From 2cc719983603f0e9d24da256b58d6abb79e3884a Mon Sep 17 00:00:00 2001 From: Biju Das Date: Thu, 25 Jul 2024 09:45:54 +0100 Subject: [PATCH 001/410] ASoC: dt-bindings: renesas,rz-ssi: Document port property Document optional port property that connects to I2S signals. Signed-off-by: Biju Das Link: https://patch.msgid.link/20240725084559.13127-2-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 8b9695f5decc..f4610eaed1e1 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -87,6 +87,10 @@ properties: '#sound-dai-cells': const: 0 + port: + $ref: audio-graph-port.yaml#/definitions/port-base + description: Connection to controller providing I2S signals + required: - compatible - reg From 3d2a69eb503d15171a7ba51cf0b562728ac396b7 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 10 Jul 2024 15:52:30 +0200 Subject: [PATCH 002/410] ASoC: codecs: wsa881x: Drop unused version readout Driver does not use the device version after reading it from the registers, so simplify by dropping unneeded code. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-1-f1c54966ccde@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0478599d0f35..103258d768eb 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -680,7 +680,6 @@ struct wsa881x_priv { * For backwards compatibility. */ unsigned int sd_n_val; - int version; int active_ports; bool port_prepared[WSA881X_MAX_SWR_PORTS]; bool port_enable[WSA881X_MAX_SWR_PORTS]; @@ -691,7 +690,6 @@ static void wsa881x_init(struct wsa881x_priv *wsa881x) struct regmap *rm = wsa881x->regmap; unsigned int val = 0; - regmap_read(rm, WSA881X_CHIP_ID1, &wsa881x->version); regmap_register_patch(wsa881x->regmap, wsa881x_rev_2_0, ARRAY_SIZE(wsa881x_rev_2_0)); From 2fbf16992e5aa14acf0441320033a01a32309ded Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 10 Jul 2024 15:52:31 +0200 Subject: [PATCH 003/410] ASoC: codecs: wsa883x: Handle reading version failure If reading version and variant from registers fails (which is unlikely but possible, because it is a read over bus), the driver will proceed and perform device configuration based on uninitialized stack variables. Handle it a bit better - bail out without doing any init and failing the update status Soundwire callback. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-2-f1c54966ccde@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index d0ab4e2290b6..89e256bfb553 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -997,15 +997,19 @@ static const struct reg_sequence reg_init[] = { {WSA883X_GMAMP_SUP1, 0xE2}, }; -static void wsa883x_init(struct wsa883x_priv *wsa883x) +static int wsa883x_init(struct wsa883x_priv *wsa883x) { struct regmap *regmap = wsa883x->regmap; - int variant, version; + int variant, version, ret; - regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + if (ret) + return ret; wsa883x->variant = variant & WSA883X_ID_MASK; - regmap_read(regmap, WSA883X_CHIP_ID0, &version); + ret = regmap_read(regmap, WSA883X_CHIP_ID0, &version); + if (ret) + return ret; wsa883x->version = version; switch (wsa883x->variant) { @@ -1040,6 +1044,8 @@ static void wsa883x_init(struct wsa883x_priv *wsa883x) WSA883X_DRE_OFFSET_MASK, wsa883x->comp_offset); } + + return 0; } static int wsa883x_update_status(struct sdw_slave *slave, @@ -1048,7 +1054,7 @@ static int wsa883x_update_status(struct sdw_slave *slave, struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) - wsa883x_init(wsa883x); + return wsa883x_init(wsa883x); return 0; } From cd15fded0e1090bf713647a5bcfd83e372152844 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 10 Jul 2024 15:52:32 +0200 Subject: [PATCH 004/410] ASoC: codecs: wsa883x: Simplify handling variant/version Driver does not use detected variant/version variables past the init function, so do not store them in the state container structure. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-3-f1c54966ccde@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 17 +++++++---------- 1 file changed, 7 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 89e256bfb553..63b27ade2bca 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -438,8 +438,6 @@ struct wsa883x_priv { struct gpio_desc *sd_n; bool port_prepared[WSA883X_MAX_SWR_PORTS]; bool port_enable[WSA883X_MAX_SWR_PORTS]; - int version; - int variant; int active_ports; int dev_mode; int comp_offset; @@ -1005,29 +1003,28 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x) ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); if (ret) return ret; - wsa883x->variant = variant & WSA883X_ID_MASK; + variant = variant & WSA883X_ID_MASK; ret = regmap_read(regmap, WSA883X_CHIP_ID0, &version); if (ret) return ret; - wsa883x->version = version; - switch (wsa883x->variant) { + switch (variant) { case WSA8830: dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n", - wsa883x->version); + version); break; case WSA8835: dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n", - wsa883x->version); + version); break; case WSA8832: dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n", - wsa883x->version); + version); break; case WSA8835_V2: dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n", - wsa883x->version); + version); break; default: break; @@ -1038,7 +1035,7 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x) /* Initial settings */ regmap_multi_reg_write(regmap, reg_init, ARRAY_SIZE(reg_init)); - if (wsa883x->variant == WSA8830 || wsa883x->variant == WSA8832) { + if (variant == WSA8830 || variant == WSA8832) { wsa883x->comp_offset = COMP_OFFSET3; regmap_update_bits(regmap, WSA883X_DRE_CTL_0, WSA883X_DRE_OFFSET_MASK, From 7eb62acd43c9299630f0e859f56981072401c5b6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Wed, 10 Jul 2024 15:52:33 +0200 Subject: [PATCH 005/410] ASoC: codecs: wsa884x: Simplify handling variant Driver does not use detected variant variable past the init function, so do not store it in the state container structure. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-4-f1c54966ccde@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index d17ae17b2938..c0f9a3e5ad14 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -703,7 +703,6 @@ struct wsa884x_priv { struct reset_control *sd_reset; bool port_prepared[WSA884X_MAX_SWR_PORTS]; bool port_enable[WSA884X_MAX_SWR_PORTS]; - unsigned int variant; int active_ports; int dev_mode; bool hw_init; @@ -1465,7 +1464,7 @@ static void wsa884x_init(struct wsa884x_priv *wsa884x) unsigned int variant = 0; if (!regmap_read(wsa884x->regmap, WSA884X_OTP_REG_0, &variant)) - wsa884x->variant = variant & WSA884X_OTP_REG_0_ID_MASK; + variant = variant & WSA884X_OTP_REG_0_ID_MASK; regmap_multi_reg_write(wsa884x->regmap, wsa884x_reg_init, ARRAY_SIZE(wsa884x_reg_init)); @@ -1474,7 +1473,7 @@ static void wsa884x_init(struct wsa884x_priv *wsa884x) wo_ctl_0 |= FIELD_PREP(WSA884X_ANA_WO_CTL_0_DAC_CM_CLAMP_EN_MASK, WSA884X_ANA_WO_CTL_0_DAC_CM_CLAMP_EN_MODE_SPEAKER); /* Assume that compander is enabled by default unless it is haptics sku */ - if (wsa884x->variant == WSA884X_OTP_ID_WSA8845H) + if (variant == WSA884X_OTP_ID_WSA8845H) wo_ctl_0 |= FIELD_PREP(WSA884X_ANA_WO_CTL_0_PA_AUX_GAIN_MASK, WSA884X_ANA_WO_CTL_0_PA_AUX_18_DB); else From 00425bf8cbc9981bd975a5475cec4964544fb297 Mon Sep 17 00:00:00 2001 From: Animesh Agarwal Date: Wed, 17 Jul 2024 19:17:21 +0530 Subject: [PATCH 006/410] ASoC: dt-bindings: ti,pcm512x: Convert to dtschema Convert the PCM512x and TAS575x audio CODECs/amplifiers bindings to DT schema format. Add missing sound-dai-cells property. Cc: Daniel Baluta Signed-off-by: Animesh Agarwal Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240717134729.51661-1-animeshagarwal28@gmail.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/pcm512x.txt | 53 --------- .../devicetree/bindings/sound/ti,pcm512x.yaml | 101 ++++++++++++++++++ 2 files changed, 101 insertions(+), 53 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm512x.yaml diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt deleted file mode 100644 index 47878a6df608..000000000000 --- a/Documentation/devicetree/bindings/sound/pcm512x.txt +++ /dev/null @@ -1,53 +0,0 @@ -PCM512x and TAS575x audio CODECs/amplifiers - -These devices support both I2C and SPI (configured with pin strapping -on the board). The TAS575x devices only support I2C. - -Required properties: - - - compatible : One of "ti,pcm5121", "ti,pcm5122", "ti,pcm5141", - "ti,pcm5142", "ti,pcm5242", "ti,tas5754" or "ti,tas5756" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - - - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the - device, as covered in bindings/regulator/regulator.txt - -Optional properties: - - - clocks : A clock specifier for the clock connected as SCLK. If this - is absent the device will be configured to clock from BCLK. If pll-in - and pll-out are specified in addition to a clock, the device is - configured to accept clock input on a specified gpio pin. - - - pll-in, pll-out : gpio pins used to connect the pll using <1> - through <6>. The device will be configured for clock input on the - given pll-in pin and PLL output on the given pll-out pin. An - external connection from the pll-out pin to the SCLK pin is assumed. - Caution: the TAS-desvices only support gpios 1,2 and 3 - -Examples: - - pcm5122: pcm5122@4c { - compatible = "ti,pcm5122"; - reg = <0x4c>; - - AVDD-supply = <®_3v3_analog>; - DVDD-supply = <®_1v8>; - CPVDD-supply = <®_3v3>; - }; - - - pcm5142: pcm5142@4c { - compatible = "ti,pcm5142"; - reg = <0x4c>; - - AVDD-supply = <®_3v3_analog>; - DVDD-supply = <®_1v8>; - CPVDD-supply = <®_3v3>; - - clocks = <&sck>; - pll-in = <3>; - pll-out = <6>; - }; diff --git a/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml new file mode 100644 index 000000000000..21ea9ff5a2bb --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,pcm512x.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,pcm512x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: PCM512x and TAS575x audio CODECs/amplifiers + +maintainers: + - Animesh Agarwal + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - ti,pcm5121 + - ti,pcm5122 + - ti,pcm5141 + - ti,pcm5142 + - ti,pcm5242 + - ti,tas5754 + - ti,tas5756 + + reg: + maxItems: 1 + + AVDD-supply: true + + DVDD-supply: true + + CPVDD-supply: true + + clocks: + maxItems: 1 + description: A clock specifier for the clock connected as SCLK. If this is + absent the device will be configured to clock from BCLK. If pll-in and + pll-out are specified in addition to a clock, the device is configured to + accept clock input on a specified gpio pin. + + '#sound-dai-cells': + const: 0 + + pll-in: + description: GPIO pin used to connect the pll using <1> through <6>. The + device will be configured for clock input on the given pll-in pin. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 6 + + pll-out: + description: GPIO pin used to connect the pll using <1> through <6>. The + device will be configured for PLL output on the given pll-out pin. An + external connection from the pll-out pin to the SCLK pin is assumed. + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 1 + maximum: 6 + +required: + - compatible + - reg + - AVDD-supply + - DVDD-supply + - CPVDD-supply + +if: + properties: + compatible: + contains: + enum: + - ti,tas5754 + - ti,tas5756 + +then: + properties: + pll-in: + maximum: 3 + + pll-out: + maximum: 3 + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@4c { + compatible = "ti,pcm5142"; + reg = <0x4c>; + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + #sound-dai-cells = <0>; + clocks = <&sck>; + pll-in = <3>; + pll-out = <6>; + }; + }; From 00645b42e3ca6a35fc4a357dd769bcef41d4a077 Mon Sep 17 00:00:00 2001 From: Animesh Agarwal Date: Mon, 22 Jul 2024 12:06:51 +0530 Subject: [PATCH 007/410] ASoC: dt-bindings: fsl,imx-audio-es8328: Convert to dtschema Convert the Freescale i.MX audio complex with ES8328 codec bindings to DT schema format. Cc: Daniel Baluta Signed-off-by: Animesh Agarwal Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240722063657.23018-1-animeshagarwal28@gmail.com Signed-off-by: Mark Brown --- .../bindings/sound/fsl,imx-audio-es8328.yaml | 111 ++++++++++++++++++ .../bindings/sound/imx-audio-es8328.txt | 60 ---------- 2 files changed, 111 insertions(+), 60 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml new file mode 100644 index 000000000000..5eb6f5812cf2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml @@ -0,0 +1,111 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,imx-audio-es8328.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale i.MX audio complex with ES8328 codec + +maintainers: + - Shawn Guo + - Sascha Hauer + +allOf: + - $ref: sound-card-common.yaml# + +properties: + compatible: + const: fsl,imx-audio-es8328 + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex + + ssi-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the i.MX SSI controller + + jack-gpio: + description: Optional GPIO for headphone jack + maxItems: 1 + + audio-amp-supply: + description: Power regulator for speaker amps + + audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle to the ES8328 audio codec + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + ES8328 pins, and the jacks on the board: + + Power supplies: + * audio-amp + + ES8328 pins: + * LOUT1 + * LOUT2 + * ROUT1 + * ROUT2 + * LINPUT1 + * LINPUT2 + * RINPUT1 + * RINPUT2 + * Mic PGA + + Board connectors: + * Headphone + * Speaker + * Mic Jack + + mux-int-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The internal port of the i.MX audio muxer (AUDMUX) + enum: [1, 2, 7] + default: 1 + + mux-ext-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: The external port of the i.MX audio muxer (AUDMIX) + enum: [3, 4, 5, 6] + default: 3 + +required: + - compatible + - model + - ssi-controller + - jack-gpio + - audio-amp-supply + - audio-codec + - audio-routing + - mux-int-port + - mux-ext-port + +unevaluatedProperties: false + +examples: + - | + sound { + compatible = "fsl,imx-audio-es8328"; + model = "imx-audio-es8328"; + ssi-controller = <&ssi1>; + audio-codec = <&codec>; + jack-gpio = <&gpio5 15 0>; + audio-amp-supply = <®_audio_amp>; + audio-routing = + "Speaker", "LOUT2", + "Speaker", "ROUT2", + "Speaker", "audio-amp", + "Headphone", "ROUT1", + "Headphone", "LOUT1", + "LINPUT1", "Mic Jack", + "RINPUT1", "Mic Jack", + "Mic Jack", "Mic Bias"; + mux-int-port = <1>; + mux-ext-port = <3>; + }; diff --git a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt b/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt deleted file mode 100644 index 07b68ab206fb..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-es8328.txt +++ /dev/null @@ -1,60 +0,0 @@ -Freescale i.MX audio complex with ES8328 codec - -Required properties: -- compatible : "fsl,imx-audio-es8328" -- model : The user-visible name of this sound complex -- ssi-controller : The phandle of the i.MX SSI controller -- jack-gpio : Optional GPIO for headphone jack -- audio-amp-supply : Power regulator for speaker amps -- audio-codec : The phandle of the ES8328 audio codec -- audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names could be power supplies, ES8328 - pins, and the jacks on the board: - - Power supplies: - * audio-amp - - ES8328 pins: - * LOUT1 - * LOUT2 - * ROUT1 - * ROUT2 - * LINPUT1 - * LINPUT2 - * RINPUT1 - * RINPUT2 - * Mic PGA - - Board connectors: - * Headphone - * Speaker - * Mic Jack -- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) -- mux-ext-port : The external port of the i.MX audio muxer (AUDMIX) - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx-audio-es8328"; - model = "imx-audio-es8328"; - ssi-controller = <&ssi1>; - audio-codec = <&codec>; - jack-gpio = <&gpio5 15 0>; - audio-amp-supply = <®_audio_amp>; - audio-routing = - "Speaker", "LOUT2", - "Speaker", "ROUT2", - "Speaker", "audio-amp", - "Headphone", "ROUT1", - "Headphone", "LOUT1", - "LINPUT1", "Mic Jack", - "RINPUT1", "Mic Jack", - "Mic Jack", "Mic Bias"; - mux-int-port = <1>; - mux-ext-port = <3>; -}; From 6024f3429fd1ac4948bac9610ecd4d0139088f0b Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Fri, 26 Jul 2024 11:10:02 +0800 Subject: [PATCH 008/410] ASoC: codecs: ES8326: suspend issue We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi Link: https://patch.msgid.link/20240726031002.35055-1-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); From 8716bd241fa120aacce5e0136125b7ecc74fe3b2 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 23 Jul 2024 10:33:00 +0200 Subject: [PATCH 009/410] ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding The APQ8016 SBC and MSM8916 QDSP6 sound cards are a bit different from others: they have additional IO muxing address space and pin control. Move them to separate schema, so the original qcom,sm8250.yaml will be easier to manage. New schema is going to grow for other platforms having more of IO muxing address spaces. Cc: Adam Skladowski Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240723083300.35605-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- .../sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++++++++++++++++++ .../bindings/sound/qcom,sm8250.yaml | 137 ------------ 2 files changed, 205 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000000..6ad451549036 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla + - Stephan Gerhold + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include + #include + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index c9076dcd44c1..1d3acdc0c733 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -27,9 +27,7 @@ properties: - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - - qcom,apq8016-sbc-sndcard - qcom,apq8096-sndcard - - qcom,msm8916-qdsp6-sndcard - qcom,qcm6490-idp-sndcard - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard @@ -58,18 +56,6 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User visible long sound card name - pin-switches: - description: List of widget names for which pin switches should be created. - $ref: /schemas/types.yaml#/definitions/string-array - - widgets: - description: User specified audio sound widgets. - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - - # Only valid for some compatibles (see allOf if below) - reg: true - reg-names: true - patternProperties: ".*-dai-link$": description: @@ -122,34 +108,6 @@ required: - compatible - model -allOf: - - if: - properties: - compatible: - contains: - enum: - - qcom,apq8016-sbc-sndcard - - qcom,msm8916-qdsp6-sndcard - then: - properties: - reg: - items: - - description: Microphone I/O mux register address - - description: Speaker I/O mux register address - reg-names: - items: - - const: mic-iomux - - const: spkr-iomux - required: - - compatible - - model - - reg - - reg-names - else: - properties: - reg: false - reg-names: false - additionalProperties: false examples: @@ -231,98 +189,3 @@ examples: }; }; }; - - - | - #include - sound@7702000 { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "DB410c"; - audio-routing = - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1"; - - pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; - pinctrl-names = "default", "sleep"; - - quaternary-dai-link { - link-name = "ADV7533"; - cpu { - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; - - primary-dai-link { - link-name = "WCD"; - cpu { - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - tertiary-dai-link { - link-name = "WCD-Capture"; - cpu { - sound-dai = <&lpass MI2S_TERTIARY>; - }; - codec { - sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; - }; - }; - }; - - - | - #include - #include - sound@7702000 { - compatible = "qcom,msm8916-qdsp6-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "msm8916"; - widgets = - "Speaker", "Speaker", - "Headphone", "Headphones"; - pin-switches = "Speaker"; - audio-routing = - "Speaker", "Speaker Amp OUT", - "Speaker Amp IN", "HPH_R", - "Headphones", "HPH_L", - "Headphones", "HPH_R", - "AMIC1", "MIC BIAS Internal1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS Internal3"; - aux-devs = <&speaker_amp>; - - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&cdc_pdm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - primary-dai-link { - link-name = "Primary MI2S"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - platform { - sound-dai = <&q6routing>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - }; From 3ff810b9bebe5578a245cfa97c252ab602e703f1 Mon Sep 17 00:00:00 2001 From: Ma Ke Date: Wed, 17 Jul 2024 19:54:36 +0800 Subject: [PATCH 010/410] ASoC: rt5682s: Return devm_of_clk_add_hw_provider to transfer the error Return devm_of_clk_add_hw_provider() in order to transfer the error, if it fails due to resource allocation failure or device tree clock provider registration failure. Fixes: bdd229ab26be ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Ma Ke Link: https://patch.msgid.link/20240717115436.3449492-1-make24@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index f50f196d700d..ce2e88e066f3 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2828,7 +2828,9 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) } if (dev->of_node) { - devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + ret = devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + if (ret) + return ret; } else { ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, init.name, dev_name(dev)); From b3f35bae68c0ff9b9339b819ec5f5f341d798bbe Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Tue, 23 Jul 2024 16:46:07 +0200 Subject: [PATCH 011/410] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240723144607.123240-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 73a588289408..76900274acf3 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2297,36 +2297,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); From 4f8cd05a43058b165b83f12f656e60415d2ff5be Mon Sep 17 00:00:00 2001 From: Biju Das Date: Mon, 15 Jul 2024 10:23:20 +0100 Subject: [PATCH 012/410] ASoC: sh: rz-ssi: Add full duplex support Add full duplex support, to support simultaneous playback/record on the same ssi channel. Signed-off-by: Biju Das Link: https://patch.msgid.link/20240715092322.119879-1-biju.das.jz@bp.renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rz-ssi.c | 289 +++++++++++++++++++++++++++++------------- 1 file changed, 198 insertions(+), 91 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 9d103646973a..d0bf0487bf1b 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -52,6 +52,7 @@ #define SSIFCR_RIE BIT(2) #define SSIFCR_TFRST BIT(1) #define SSIFCR_RFRST BIT(0) +#define SSIFCR_FIFO_RST (SSIFCR_TFRST | SSIFCR_RFRST) #define SSIFSR_TDC_MASK 0x3f #define SSIFSR_TDC_SHIFT 24 @@ -130,6 +131,14 @@ struct rz_ssi_priv { bool lrckp_fsync_fall; /* LR clock polarity (SSICR.LRCKP) */ bool bckp_rise; /* Bit clock polarity (SSICR.BCKP) */ bool dma_rt; + + /* Full duplex communication support */ + struct { + unsigned int rate; + unsigned int channels; + unsigned int sample_width; + unsigned int sample_bits; + } hw_params_cache; }; static void rz_ssi_dma_complete(void *data); @@ -208,6 +217,11 @@ static bool rz_ssi_stream_is_valid(struct rz_ssi_priv *ssi, return ret; } +static inline bool rz_ssi_is_stream_running(struct rz_ssi_stream *strm) +{ + return strm->substream && strm->running; +} + static void rz_ssi_stream_init(struct rz_ssi_stream *strm, struct snd_pcm_substream *substream) { @@ -303,59 +317,10 @@ static int rz_ssi_clk_setup(struct rz_ssi_priv *ssi, unsigned int rate, return 0; } -static int rz_ssi_start(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) -{ - bool is_play = rz_ssi_stream_is_play(ssi, strm->substream); - u32 ssicr, ssifcr; - - ssicr = rz_ssi_reg_readl(ssi, SSICR); - ssifcr = rz_ssi_reg_readl(ssi, SSIFCR) & ~0xF; - - /* FIFO interrupt thresholds */ - if (rz_ssi_is_dma_enabled(ssi)) - rz_ssi_reg_writel(ssi, SSISCR, 0); - else - rz_ssi_reg_writel(ssi, SSISCR, - SSISCR_TDES(strm->fifo_sample_size / 2 - 1) | - SSISCR_RDFS(0)); - - /* enable IRQ */ - if (is_play) { - ssicr |= SSICR_TUIEN | SSICR_TOIEN; - ssifcr |= SSIFCR_TIE | SSIFCR_RFRST; - } else { - ssicr |= SSICR_RUIEN | SSICR_ROIEN; - ssifcr |= SSIFCR_RIE | SSIFCR_TFRST; - } - - rz_ssi_reg_writel(ssi, SSICR, ssicr); - rz_ssi_reg_writel(ssi, SSIFCR, ssifcr); - - /* Clear all error flags */ - rz_ssi_reg_mask_setl(ssi, SSISR, - (SSISR_TOIRQ | SSISR_TUIRQ | SSISR_ROIRQ | - SSISR_RUIRQ), 0); - - strm->running = 1; - ssicr |= is_play ? SSICR_TEN : SSICR_REN; - rz_ssi_reg_writel(ssi, SSICR, ssicr); - - return 0; -} - -static int rz_ssi_stop(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) +static void rz_ssi_set_idle(struct rz_ssi_priv *ssi) { int timeout; - strm->running = 0; - - /* Disable TX/RX */ - rz_ssi_reg_mask_setl(ssi, SSICR, SSICR_TEN | SSICR_REN, 0); - - /* Cancel all remaining DMA transactions */ - if (rz_ssi_is_dma_enabled(ssi)) - dmaengine_terminate_async(strm->dma_ch); - /* Disable irqs */ rz_ssi_reg_mask_setl(ssi, SSICR, SSICR_TUIEN | SSICR_TOIEN | SSICR_RUIEN | SSICR_ROIEN, 0); @@ -380,6 +345,82 @@ static int rz_ssi_stop(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) /* Hold FIFOs in reset */ rz_ssi_reg_mask_setl(ssi, SSIFCR, 0, SSIFCR_TFRST | SSIFCR_RFRST); +} + +static int rz_ssi_start(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) +{ + bool is_play = rz_ssi_stream_is_play(ssi, strm->substream); + bool is_full_duplex; + u32 ssicr, ssifcr; + + is_full_duplex = rz_ssi_is_stream_running(&ssi->playback) || + rz_ssi_is_stream_running(&ssi->capture); + ssicr = rz_ssi_reg_readl(ssi, SSICR); + ssifcr = rz_ssi_reg_readl(ssi, SSIFCR); + if (!is_full_duplex) { + ssifcr &= ~0xF; + } else { + rz_ssi_reg_mask_setl(ssi, SSICR, SSICR_TEN | SSICR_REN, 0); + rz_ssi_set_idle(ssi); + ssifcr &= ~SSIFCR_FIFO_RST; + } + + /* FIFO interrupt thresholds */ + if (rz_ssi_is_dma_enabled(ssi)) + rz_ssi_reg_writel(ssi, SSISCR, 0); + else + rz_ssi_reg_writel(ssi, SSISCR, + SSISCR_TDES(strm->fifo_sample_size / 2 - 1) | + SSISCR_RDFS(0)); + + /* enable IRQ */ + if (is_play) { + ssicr |= SSICR_TUIEN | SSICR_TOIEN; + ssifcr |= SSIFCR_TIE; + if (!is_full_duplex) + ssifcr |= SSIFCR_RFRST; + } else { + ssicr |= SSICR_RUIEN | SSICR_ROIEN; + ssifcr |= SSIFCR_RIE; + if (!is_full_duplex) + ssifcr |= SSIFCR_TFRST; + } + + rz_ssi_reg_writel(ssi, SSICR, ssicr); + rz_ssi_reg_writel(ssi, SSIFCR, ssifcr); + + /* Clear all error flags */ + rz_ssi_reg_mask_setl(ssi, SSISR, + (SSISR_TOIRQ | SSISR_TUIRQ | SSISR_ROIRQ | + SSISR_RUIRQ), 0); + + strm->running = 1; + if (is_full_duplex) + ssicr |= SSICR_TEN | SSICR_REN; + else + ssicr |= is_play ? SSICR_TEN : SSICR_REN; + + rz_ssi_reg_writel(ssi, SSICR, ssicr); + + return 0; +} + +static int rz_ssi_stop(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) +{ + strm->running = 0; + + if (rz_ssi_is_stream_running(&ssi->playback) || + rz_ssi_is_stream_running(&ssi->capture)) + return 0; + + /* Disable TX/RX */ + rz_ssi_reg_mask_setl(ssi, SSICR, SSICR_TEN | SSICR_REN, 0); + + /* Cancel all remaining DMA transactions */ + if (rz_ssi_is_dma_enabled(ssi)) + dmaengine_terminate_async(strm->dma_ch); + + rz_ssi_set_idle(ssi); return 0; } @@ -512,66 +553,90 @@ static int rz_ssi_pio_send(struct rz_ssi_priv *ssi, struct rz_ssi_stream *strm) static irqreturn_t rz_ssi_interrupt(int irq, void *data) { - struct rz_ssi_stream *strm = NULL; + struct rz_ssi_stream *strm_playback = NULL; + struct rz_ssi_stream *strm_capture = NULL; struct rz_ssi_priv *ssi = data; u32 ssisr = rz_ssi_reg_readl(ssi, SSISR); if (ssi->playback.substream) - strm = &ssi->playback; - else if (ssi->capture.substream) - strm = &ssi->capture; - else + strm_playback = &ssi->playback; + if (ssi->capture.substream) + strm_capture = &ssi->capture; + + if (!strm_playback && !strm_capture) return IRQ_HANDLED; /* Left over TX/RX interrupt */ if (irq == ssi->irq_int) { /* error or idle */ - if (ssisr & SSISR_TUIRQ) - strm->uerr_num++; - if (ssisr & SSISR_TOIRQ) - strm->oerr_num++; - if (ssisr & SSISR_RUIRQ) - strm->uerr_num++; - if (ssisr & SSISR_ROIRQ) - strm->oerr_num++; + bool is_stopped = false; + int i, count; - if (ssisr & (SSISR_TUIRQ | SSISR_TOIRQ | SSISR_RUIRQ | - SSISR_ROIRQ)) { - /* Error handling */ - /* You must reset (stop/restart) after each interrupt */ - rz_ssi_stop(ssi, strm); + if (rz_ssi_is_dma_enabled(ssi)) + count = 4; + else + count = 1; - /* Clear all flags */ - rz_ssi_reg_mask_setl(ssi, SSISR, SSISR_TOIRQ | - SSISR_TUIRQ | SSISR_ROIRQ | - SSISR_RUIRQ, 0); + if (ssisr & (SSISR_RUIRQ | SSISR_ROIRQ | SSISR_TUIRQ | SSISR_TOIRQ)) + is_stopped = true; - /* Add/remove more data */ - strm->transfer(ssi, strm); + if (ssi->capture.substream && is_stopped) { + if (ssisr & SSISR_RUIRQ) + strm_capture->uerr_num++; + if (ssisr & SSISR_ROIRQ) + strm_capture->oerr_num++; - /* Resume */ - rz_ssi_start(ssi, strm); + rz_ssi_stop(ssi, strm_capture); } + + if (ssi->playback.substream && is_stopped) { + if (ssisr & SSISR_TUIRQ) + strm_playback->uerr_num++; + if (ssisr & SSISR_TOIRQ) + strm_playback->oerr_num++; + + rz_ssi_stop(ssi, strm_playback); + } + + /* Clear all flags */ + rz_ssi_reg_mask_setl(ssi, SSISR, SSISR_TOIRQ | SSISR_TUIRQ | + SSISR_ROIRQ | SSISR_RUIRQ, 0); + + /* Add/remove more data */ + if (ssi->capture.substream && is_stopped) { + for (i = 0; i < count; i++) + strm_capture->transfer(ssi, strm_capture); + } + + if (ssi->playback.substream && is_stopped) { + for (i = 0; i < count; i++) + strm_playback->transfer(ssi, strm_playback); + } + + /* Resume */ + if (ssi->playback.substream && is_stopped) + rz_ssi_start(ssi, &ssi->playback); + if (ssi->capture.substream && is_stopped) + rz_ssi_start(ssi, &ssi->capture); } - if (!strm->running) + if (!rz_ssi_is_stream_running(&ssi->playback) && + !rz_ssi_is_stream_running(&ssi->capture)) return IRQ_HANDLED; /* tx data empty */ - if (irq == ssi->irq_tx) - strm->transfer(ssi, &ssi->playback); + if (irq == ssi->irq_tx && rz_ssi_is_stream_running(&ssi->playback)) + strm_playback->transfer(ssi, &ssi->playback); /* rx data full */ - if (irq == ssi->irq_rx) { - strm->transfer(ssi, &ssi->capture); + if (irq == ssi->irq_rx && rz_ssi_is_stream_running(&ssi->capture)) { + strm_capture->transfer(ssi, &ssi->capture); rz_ssi_reg_mask_setl(ssi, SSIFSR, SSIFSR_RDF, 0); } if (irq == ssi->irq_rt) { - struct snd_pcm_substream *substream = strm->substream; - - if (rz_ssi_stream_is_play(ssi, substream)) { - strm->transfer(ssi, &ssi->playback); + if (ssi->playback.substream) { + strm_playback->transfer(ssi, &ssi->playback); } else { - strm->transfer(ssi, &ssi->capture); + strm_capture->transfer(ssi, &ssi->capture); rz_ssi_reg_mask_setl(ssi, SSIFSR, SSIFSR_RDF, 0); } } @@ -731,9 +796,12 @@ static int rz_ssi_dai_trigger(struct snd_pcm_substream *substream, int cmd, switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* Soft Reset */ - rz_ssi_reg_mask_setl(ssi, SSIFCR, 0, SSIFCR_SSIRST); - rz_ssi_reg_mask_setl(ssi, SSIFCR, SSIFCR_SSIRST, 0); - udelay(5); + if (!rz_ssi_is_stream_running(&ssi->playback) && + !rz_ssi_is_stream_running(&ssi->capture)) { + rz_ssi_reg_mask_setl(ssi, SSIFCR, 0, SSIFCR_SSIRST); + rz_ssi_reg_mask_setl(ssi, SSIFCR, SSIFCR_SSIRST, 0); + udelay(5); + } rz_ssi_stream_init(strm, substream); @@ -824,14 +892,41 @@ static int rz_ssi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } +static bool rz_ssi_is_valid_hw_params(struct rz_ssi_priv *ssi, unsigned int rate, + unsigned int channels, + unsigned int sample_width, + unsigned int sample_bits) +{ + if (ssi->hw_params_cache.rate != rate || + ssi->hw_params_cache.channels != channels || + ssi->hw_params_cache.sample_width != sample_width || + ssi->hw_params_cache.sample_bits != sample_bits) + return false; + + return true; +} + +static void rz_ssi_cache_hw_params(struct rz_ssi_priv *ssi, unsigned int rate, + unsigned int channels, + unsigned int sample_width, + unsigned int sample_bits) +{ + ssi->hw_params_cache.rate = rate; + ssi->hw_params_cache.channels = channels; + ssi->hw_params_cache.sample_width = sample_width; + ssi->hw_params_cache.sample_bits = sample_bits; +} + static int rz_ssi_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct rz_ssi_priv *ssi = snd_soc_dai_get_drvdata(dai); + struct rz_ssi_stream *strm = rz_ssi_stream_get(ssi, substream); unsigned int sample_bits = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); if (sample_bits != 16) { dev_err(ssi->dev, "Unsupported sample width: %d\n", @@ -845,8 +940,20 @@ static int rz_ssi_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return rz_ssi_clk_setup(ssi, params_rate(params), - params_channels(params)); + if (rz_ssi_is_stream_running(&ssi->playback) || + rz_ssi_is_stream_running(&ssi->capture)) { + if (rz_ssi_is_valid_hw_params(ssi, rate, channels, + strm->sample_width, sample_bits)) + return 0; + + dev_err(ssi->dev, "Full duplex needs same HW params\n"); + return -EINVAL; + } + + rz_ssi_cache_hw_params(ssi, rate, channels, strm->sample_width, + sample_bits); + + return rz_ssi_clk_setup(ssi, rate, channels); } static const struct snd_soc_dai_ops rz_ssi_dai_ops = { From 42eb47310f89eca3226e8e427bc9d571149dc866 Mon Sep 17 00:00:00 2001 From: Chen Ni Date: Tue, 9 Jul 2024 16:51:31 +0800 Subject: [PATCH 013/410] ASoC: mediatek: mt8192: remove redundant null pointer check before of_node_put of_node_put() has taken the null pointer check into account. So it is safe to remove the duplicated check before of_node_put(). Signed-off-by: Chen Ni Reviewed-by: AngeloGioacchino Del Regno Link: https://patch.msgid.link/20240709085131.1436128-1-nichen@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 8b323fb19925..db00704e206d 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1108,9 +1108,7 @@ static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) err_headset_codec: of_node_put(speaker_codec); err_speaker_codec: - if (hdmi_codec) - of_node_put(hdmi_codec); - + of_node_put(hdmi_codec); return ret; } From aaa5e1aa39074fb466f6ef3df4de6903741dfeec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 Jul 2024 17:52:36 +0200 Subject: [PATCH 014/410] ASoC: Use __counted_by() annotation for snd_soc_pcm_runtime The struct snd_soc_pcm_runtime has a flex array of snd_soc_component objects at its end, and the size is kept in num_components field. We can add __counted_by() annotation for compiler's assistance to catch array overflows. A slight additional change is the assignment of rtd->components[]; the array counter has to be incremented at first for avoiding false-positive reports from compilers. Also, the allocation size of snd_soc_pcm_runtime is cleaned up with the standard struct_size() helper, too. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240726155237.21961-1-tiwai@suse.de Signed-off-by: Mark Brown --- include/sound/soc.h | 3 ++- sound/soc/soc-core.c | 13 ++++++------- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a8e66bbf932b..e844f6afc5b5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1209,8 +1209,9 @@ struct snd_soc_pcm_runtime { bool initialized; + /* CPU/Codec/Platform */ int num_components; - struct snd_soc_component *components[]; /* CPU/Codec/Platform */ + struct snd_soc_component *components[] __counted_by(num_components); }; /* see soc_new_pcm_runtime() */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 724fe1f033b5..80bacea6bb90 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -326,8 +326,8 @@ static int snd_soc_rtd_add_component(struct snd_soc_pcm_runtime *rtd, } /* see for_each_rtd_components */ - rtd->components[rtd->num_components] = component; - rtd->num_components++; + rtd->num_components++; // increment flex array count at first + rtd->components[rtd->num_components - 1] = component; return 0; } @@ -494,7 +494,6 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct snd_soc_pcm_runtime *rtd; - struct snd_soc_component *component; struct device *dev; int ret; int stream; @@ -521,10 +520,10 @@ static struct snd_soc_pcm_runtime *soc_new_pcm_runtime( * for rtd */ rtd = devm_kzalloc(dev, - sizeof(*rtd) + - sizeof(component) * (dai_link->num_cpus + - dai_link->num_codecs + - dai_link->num_platforms), + struct_size(rtd, components, + dai_link->num_cpus + + dai_link->num_codecs + + dai_link->num_platforms), GFP_KERNEL); if (!rtd) { device_unregister(dev); From d57ef03314f529e76385a9d5108c115459b54c2b Mon Sep 17 00:00:00 2001 From: Geert Uytterhoeven Date: Mon, 22 Jul 2024 14:04:00 +0200 Subject: [PATCH 015/410] ASoC: dt-bindings: dlg,da7213: Convert to json-schema Convert the Dialog Semiconductor DA7212/DA7213 Audio Codec Device Tree binding documentation to json-schema. Add missing properties. Signed-off-by: Geert Uytterhoeven Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/7645c9024a1762d281f4067504bc32a7a3d27caa.1721649741.git.geert+renesas@glider.be Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/da7213.txt | 45 -------- .../devicetree/bindings/sound/dlg,da7213.yaml | 103 ++++++++++++++++++ MAINTAINERS | 1 + 3 files changed, 104 insertions(+), 45 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/da7213.txt create mode 100644 Documentation/devicetree/bindings/sound/dlg,da7213.yaml diff --git a/Documentation/devicetree/bindings/sound/da7213.txt b/Documentation/devicetree/bindings/sound/da7213.txt deleted file mode 100644 index 94584c96c4ae..000000000000 --- a/Documentation/devicetree/bindings/sound/da7213.txt +++ /dev/null @@ -1,45 +0,0 @@ -Dialog Semiconductor DA7212/DA7213 Audio Codec bindings - -====== - -Required properties: -- compatible : Should be "dlg,da7212" or "dlg,da7213" -- reg: Specifies the I2C slave address - -Optional properties: -- clocks : phandle and clock specifier for codec MCLK. -- clock-names : Clock name string for 'clocks' attribute, should be "mclk". - -- dlg,micbias1-lvl : Voltage (mV) for Mic Bias 1 - [<1600>, <2200>, <2500>, <3000>] -- dlg,micbias2-lvl : Voltage (mV) for Mic Bias 2 - [<1600>, <2200>, <2500>, <3000>] -- dlg,dmic-data-sel : DMIC channel select based on clock edge. - ["lrise_rfall", "lfall_rrise"] -- dlg,dmic-samplephase : When to sample audio from DMIC. - ["on_clkedge", "between_clkedge"] -- dlg,dmic-clkrate : DMIC clock frequency (Hz). - [<1500000>, <3000000>] - - - VDDA-supply : Regulator phandle for Analogue power supply - - VDDMIC-supply : Regulator phandle for Mic Bias - - VDDIO-supply : Regulator phandle for I/O power supply - -====== - -Example: - - codec_i2c: da7213@1a { - compatible = "dlg,da7213"; - reg = <0x1a>; - - clocks = <&clks 201>; - clock-names = "mclk"; - - dlg,micbias1-lvl = <2500>; - dlg,micbias2-lvl = <2500>; - - dlg,dmic-data-sel = "lrise_rfall"; - dlg,dmic-samplephase = "between_clkedge"; - dlg,dmic-clkrate = <3000000>; - }; diff --git a/Documentation/devicetree/bindings/sound/dlg,da7213.yaml b/Documentation/devicetree/bindings/sound/dlg,da7213.yaml new file mode 100644 index 000000000000..c2dede1e82ff --- /dev/null +++ b/Documentation/devicetree/bindings/sound/dlg,da7213.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/dlg,da7213.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dialog Semiconductor DA7212/DA7213 Audio Codec + +maintainers: + - Support Opensource + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - dlg,da7212 + - dlg,da7213 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + "#sound-dai-cells": + const: 0 + + dlg,micbias1-lvl: + description: Voltage (mV) for Mic Bias 1 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1600, 2200, 2500, 3000 ] + + dlg,micbias2-lvl: + description: Voltage (mV) for Mic Bias 2 + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1600, 2200, 2500, 3000 ] + + dlg,dmic-data-sel: + description: DMIC channel select based on clock edge + enum: [ lrise_rfall, lfall_rrise ] + + dlg,dmic-samplephase: + description: When to sample audio from DMIC + enum: [ on_clkedge, between_clkedge ] + + dlg,dmic-clkrate: + description: DMIC clock frequency (Hz) + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 1500000, 3000000 ] + + VDDA-supply: + description: Analogue power supply + + VDDIO-supply: + description: I/O power supply + + VDDMIC-supply: + description: Mic Bias + + VDDSP-supply: + description: Speaker supply + + ports: + $ref: audio-graph-port.yaml#/definitions/ports + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "dlg,da7213"; + reg = <0x1a>; + + clocks = <&clks 201>; + clock-names = "mclk"; + + #sound-dai-cells = <0>; + + dlg,micbias1-lvl = <2500>; + dlg,micbias2-lvl = <2500>; + + dlg,dmic-data-sel = "lrise_rfall"; + dlg,dmic-samplephase = "between_clkedge"; + dlg,dmic-clkrate = <3000000>; + }; + }; diff --git a/MAINTAINERS b/MAINTAINERS index 42decde38320..88a40557db32 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6500,6 +6500,7 @@ F: Documentation/devicetree/bindings/regulator/da92*.txt F: Documentation/devicetree/bindings/regulator/dlg,da9*.yaml F: Documentation/devicetree/bindings/regulator/dlg,slg51000.yaml F: Documentation/devicetree/bindings/sound/da[79]*.txt +F: Documentation/devicetree/bindings/sound/dlg,da7213.yaml F: Documentation/devicetree/bindings/thermal/dlg,da9062-thermal.yaml F: Documentation/devicetree/bindings/watchdog/dlg,da9062-watchdog.yaml F: Documentation/hwmon/da90??.rst From 275d57ae441f34749cbf8621441ce2148f83d5e6 Mon Sep 17 00:00:00 2001 From: Chen Ni Date: Tue, 16 Jul 2024 10:53:07 +0800 Subject: [PATCH 016/410] ASoC: cs42l42: Convert comma to semicolon Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni Reviewed-by: Richard Fitzgerald Reviewed-by: Dragan Simic Link: https://patch.msgid.link/20240716025307.400156-1-nichen@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; From 874d04fe15d12cafa09dd36e8555cea4eb0653f6 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:43 +0200 Subject: [PATCH 017/410] ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports Two arrays (with 'struct sdw_dpn_prop' and 'struct sdw_port_config') store configuration of Soundwire ports, thus each of their element is indexed according to the port number (enum wsa_port_ids, e.g. WSA881X_PORT_DAC). Except the indexing, they also store port number offset by one in member 'num'. Entire code depends on that correlation between array index and port number, thus make it explicit by using designators. The code is functionally the same, but more obvious for reading. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-1-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++------------------ 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0478599d0f35..a5e05c05fd3d 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -386,33 +386,32 @@ enum wsa_port_ids { /* 4 ports */ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { - { - /* DAC */ - .num = 1, + [WSA881X_PORT_DAC] = { + .num = WSA881X_PORT_DAC + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* COMP */ - .num = 2, + }, + [WSA881X_PORT_COMP] = { + .num = WSA881X_PORT_COMP + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* BOOST */ - .num = 3, + }, + [WSA881X_PORT_BOOST] = { + .num = WSA881X_PORT_BOOST + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* VISENSE */ - .num = 4, + }, + [WSA881X_PORT_VISENSE] = { + .num = WSA881X_PORT_VISENSE + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, @@ -422,17 +421,20 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = { }; static const struct sdw_port_config wsa881x_pconfig[WSA881X_MAX_SWR_PORTS] = { - { - .num = 1, + [WSA881X_PORT_DAC] = { + .num = WSA881X_PORT_DAC + 1, .ch_mask = 0x1, - }, { - .num = 2, + }, + [WSA881X_PORT_COMP] = { + .num = WSA881X_PORT_COMP + 1, .ch_mask = 0xf, - }, { - .num = 3, + }, + [WSA881X_PORT_BOOST] = { + .num = WSA881X_PORT_BOOST + 1, .ch_mask = 0x3, - }, { /* IV feedback */ - .num = 4, + }, + [WSA881X_PORT_VISENSE] = { + .num = WSA881X_PORT_VISENSE + 1, .ch_mask = 0x3, }, }; From add41ea55060d5e41d62268aa0bda2a27e0f5053 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:44 +0200 Subject: [PATCH 018/410] ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports Two arrays (with 'struct sdw_dpn_prop' and 'struct sdw_port_config') store configuration of Soundwire ports, thus each of their element is indexed according to the port number (enum wsa_port_ids, e.g. WSA883X_PORT_DAC). Except the indexing, they also store port number offset by one in member 'num'. Entire code depends on that correlation between array index and port number, thus make it explicit by using designators. The code is functionally the same, but more obvious for reading. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-2-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++------------------ 1 file changed, 22 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index d0ab4e2290b6..9dc2e4d96b10 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -482,33 +482,32 @@ static const struct soc_enum wsa_dev_mode_enum = /* 4 ports */ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA883X_MAX_SWR_PORTS] = { - { - /* DAC */ - .num = 1, + [WSA883X_PORT_DAC] = { + .num = WSA883X_PORT_DAC + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* COMP */ - .num = 2, + }, + [WSA883X_PORT_COMP] = { + .num = WSA883X_PORT_COMP + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* BOOST */ - .num = 3, + }, + [WSA883X_PORT_BOOST] = { + .num = WSA883X_PORT_BOOST + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { - /* VISENSE */ - .num = 4, + }, + [WSA883X_PORT_VISENSE] = { + .num = WSA883X_PORT_VISENSE + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, @@ -518,17 +517,20 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA883X_MAX_SWR_PORTS] = { }; static const struct sdw_port_config wsa883x_pconfig[WSA883X_MAX_SWR_PORTS] = { - { - .num = 1, + [WSA883X_PORT_DAC] = { + .num = WSA883X_PORT_DAC + 1, .ch_mask = 0x1, - }, { - .num = 2, + }, + [WSA883X_PORT_COMP] = { + .num = WSA883X_PORT_COMP + 1, .ch_mask = 0xf, - }, { - .num = 3, + }, + [WSA883X_PORT_BOOST] = { + .num = WSA883X_PORT_BOOST + 1, .ch_mask = 0x3, - }, { /* IV feedback */ - .num = 4, + }, + [WSA883X_PORT_VISENSE] = { + .num = WSA883X_PORT_VISENSE + 1, .ch_mask = 0x3, }, }; From 125ed86b0d669334dbc567f441d10163ff0c44bc Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:45 +0200 Subject: [PATCH 019/410] ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports Two arrays (with 'struct sdw_dpn_prop' and 'struct sdw_port_config') store configuration of Soundwire ports, thus each of their element is indexed according to the port number (enum wsa884x_port_ids, e.g. WSA884X_PORT_DAC). Except the indexing, they also store port number offset by one in member 'num'. Entire code depends on that correlation between array index and port number, thus make it explicit by using designators. The code is functionally the same, but more obvious for reading. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-3-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 1 file changed, 22 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index d17ae17b2938..d3d09c3bab2d 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -782,42 +782,47 @@ static const struct soc_enum wsa884x_dev_mode_enum = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wsa884x_dev_mode_text), wsa884x_dev_mode_text); static struct sdw_dpn_prop wsa884x_sink_dpn_prop[WSA884X_MAX_SWR_PORTS] = { - { + [WSA884X_PORT_DAC] = { .num = WSA884X_PORT_DAC + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { + }, + [WSA884X_PORT_COMP] = { .num = WSA884X_PORT_COMP + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { + }, + [WSA884X_PORT_BOOST] = { .num = WSA884X_PORT_BOOST + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { + }, + [WSA884X_PORT_PBR] = { .num = WSA884X_PORT_PBR + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { + }, + [WSA884X_PORT_VISENSE] = { .num = WSA884X_PORT_VISENSE + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, .max_ch = 1, .simple_ch_prep_sm = true, .read_only_wordlength = true, - }, { + }, + [WSA884X_PORT_CPS] = { .num = WSA884X_PORT_CPS + 1, .type = SDW_DPN_SIMPLE, .min_ch = 1, @@ -828,22 +833,27 @@ static struct sdw_dpn_prop wsa884x_sink_dpn_prop[WSA884X_MAX_SWR_PORTS] = { }; static const struct sdw_port_config wsa884x_pconfig[WSA884X_MAX_SWR_PORTS] = { - { + [WSA884X_PORT_DAC] = { .num = WSA884X_PORT_DAC + 1, .ch_mask = 0x1, - }, { + }, + [WSA884X_PORT_COMP] = { .num = WSA884X_PORT_COMP + 1, .ch_mask = 0xf, - }, { + }, + [WSA884X_PORT_BOOST] = { .num = WSA884X_PORT_BOOST + 1, .ch_mask = 0x3, - }, { + }, + [WSA884X_PORT_PBR] = { .num = WSA884X_PORT_PBR + 1, .ch_mask = 0x1, - }, { + }, + [WSA884X_PORT_VISENSE] = { .num = WSA884X_PORT_VISENSE + 1, .ch_mask = 0x3, - }, { + }, + [WSA884X_PORT_CPS] = { .num = WSA884X_PORT_CPS + 1, .ch_mask = 0x3, }, From 06fa8271273d8181cb8727e63aeec3f87a48d8c7 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:46 +0200 Subject: [PATCH 020/410] ASoC: codecs: wcd938x: Drop unused defines and enums Drop defines and enums not used in the driver. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-4-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.c | 18 ------------------ 1 file changed, 18 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index 12b32d5dc580..aa92f1bfc921 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -29,7 +29,6 @@ #define WCD938X_MAX_SUPPLY (4) #define WCD938X_MBHC_MAX_BUTTONS (8) #define TX_ADC_MAX (4) -#define WCD938X_TX_MAX_SWR_PORTS (5) #define WCD938X_RATES_MASK (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000 |\ @@ -39,8 +38,6 @@ SNDRV_PCM_RATE_176400) #define WCD938X_FORMATS_S16_S24_LE (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE) -/* Convert from vout ctl to micbias voltage in mV */ -#define WCD_VOUT_CTL_TO_MICB(v) (1000 + v * 50) #define SWR_CLK_RATE_0P6MHZ (600000) #define SWR_CLK_RATE_1P2MHZ (1200000) #define SWR_CLK_RATE_2P4MHZ (2400000) @@ -48,8 +45,6 @@ #define SWR_CLK_RATE_9P6MHZ (9600000) #define SWR_CLK_RATE_11P2896MHZ (1128960) -#define WCD938X_DRV_NAME "wcd938x_codec" -#define WCD938X_VERSION_1_0 (1) #define EAR_RX_PATH_AUX (1) #define ADC_MODE_VAL_HIFI 0x01 @@ -72,7 +67,6 @@ /* Z value compared in milliOhm */ #define WCD938X_MBHC_IS_SECOND_RAMP_REQUIRED(z) ((z > 400000) || (z < 32000)) #define WCD938X_MBHC_ZDET_CONST (86 * 16384) -#define WCD938X_MBHC_MOISTURE_RREF R_24_KOHM #define WCD_MBHC_HS_V_MAX 1600 #define WCD938X_EAR_PA_GAIN_TLV(xname, reg, shift, max, invert, tlv_array) \ @@ -89,18 +83,6 @@ enum { WCD9385 = 5, }; -enum { - TX_HDR12 = 0, - TX_HDR34, - TX_HDR_MAX, -}; - -enum { - WCD_RX1, - WCD_RX2, - WCD_RX3 -}; - enum { /* INTR_CTRL_INT_MASK_0 */ WCD938X_IRQ_MBHC_BUTTON_PRESS_DET = 0, From 42f3a2caf80910d0c251b2a407d4d220c0d3a79f Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:47 +0200 Subject: [PATCH 021/410] ASoC: codecs: wcd937x: Move max port number defines to enum Instead of having separate define to indicate number of TX and RX Soundwire ports, move it to the enums defining actual port indices/values. This makes it more obvious why such value was chosen as number of TX/RX ports. Note: the enums start from 1, thus number of ports equals to the last vaue in the enum. WCD937X_MAX_SWR_PORTS is used in one of structures in the header, so entire enum must be moved to the top of the header file. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-5-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/wcd937x.h b/sound/soc/codecs/wcd937x.h index 37bff16e88dd..35f3d48bd7dd 100644 --- a/sound/soc/codecs/wcd937x.h +++ b/sound/soc/codecs/wcd937x.h @@ -484,10 +484,25 @@ #define WCD937X_MAX_MICBIAS 3 #define WCD937X_MAX_BULK_SUPPLY 4 -#define WCD937X_MAX_TX_SWR_PORTS 4 -#define WCD937X_MAX_SWR_PORTS 5 #define WCD937X_MAX_SWR_CH_IDS 15 +enum wcd937x_tx_sdw_ports { + WCD937X_ADC_1_PORT = 1, + WCD937X_ADC_2_3_PORT, + WCD937X_DMIC_0_3_MBHC_PORT, + WCD937X_DMIC_4_6_PORT, + WCD937X_MAX_TX_SWR_PORTS = WCD937X_DMIC_4_6_PORT, +}; + +enum wcd937x_rx_sdw_ports { + WCD937X_HPH_PORT = 1, + WCD937X_CLSH_PORT, + WCD937X_COMP_PORT, + WCD937X_LO_PORT, + WCD937X_DSD_PORT, + WCD937X_MAX_SWR_PORTS = WCD937X_DSD_PORT, +}; + struct wcd937x_sdw_ch_info { int port_num; unsigned int ch_mask; @@ -581,13 +596,6 @@ enum { WCD937X_NUM_IRQS, }; -enum wcd937x_tx_sdw_ports { - WCD937X_ADC_1_PORT = 1, - WCD937X_ADC_2_3_PORT, - WCD937X_DMIC_0_3_MBHC_PORT, - WCD937X_DMIC_4_6_PORT, -}; - enum wcd937x_tx_sdw_channels { WCD937X_ADC1, WCD937X_ADC2, @@ -602,14 +610,6 @@ enum wcd937x_tx_sdw_channels { WCD937X_DMIC6, }; -enum wcd937x_rx_sdw_ports { - WCD937X_HPH_PORT = 1, - WCD937X_CLSH_PORT, - WCD937X_COMP_PORT, - WCD937X_LO_PORT, - WCD937X_DSD_PORT, -}; - enum wcd937x_rx_sdw_channels { WCD937X_HPH_L, WCD937X_HPH_R, From 5e388488f0a1dd6d340f3925e7b371e212ee3cc2 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:48 +0200 Subject: [PATCH 022/410] ASoC: codecs: wcd938x: Move max port number defines to enum Instead of having separate define to indicate number of TX and RX Soundwire ports, move it to the enums defining actual port indices/values. This makes it more obvious why such value was chosen as number of TX/RX ports. Note: the enums start from 1, thus number of ports equals to the last vaue in the enum. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-6-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd938x.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd938x.h b/sound/soc/codecs/wcd938x.h index b2ad98026ae2..fb6a0e4ef337 100644 --- a/sound/soc/codecs/wcd938x.h +++ b/sound/soc/codecs/wcd938x.h @@ -585,8 +585,6 @@ #define WCD938X_DIGITAL_DEM_BYPASS_DATA3 (0x34D8) #define WCD938X_MAX_REGISTER (WCD938X_DIGITAL_DEM_BYPASS_DATA3) -#define WCD938X_MAX_SWR_PORTS 5 -#define WCD938X_MAX_TX_SWR_PORTS 4 #define WCD938X_MAX_SWR_CH_IDS 15 struct wcd938x_sdw_ch_info { @@ -606,6 +604,7 @@ enum wcd938x_tx_sdw_ports { /* DMIC0_0, DMIC0_1, DMIC1_0, DMIC1_1 */ WCD938X_DMIC_0_3_MBHC_PORT, WCD938X_DMIC_4_7_PORT, + WCD938X_MAX_TX_SWR_PORTS = WCD938X_DMIC_4_7_PORT, }; enum wcd938x_tx_sdw_channels { @@ -630,6 +629,7 @@ enum wcd938x_rx_sdw_ports { WCD938X_COMP_PORT, WCD938X_LO_PORT, WCD938X_DSD_PORT, + WCD938X_MAX_SWR_PORTS = WCD938X_DSD_PORT, }; enum wcd938x_rx_sdw_channels { From a9d843e6b231e550f8141f27e930f90ded4edae2 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Thu, 25 Jul 2024 13:23:49 +0200 Subject: [PATCH 023/410] ASoC: codecs: wcd939x: Move max port number defines to enum Instead of having separate define to indicate number of TX and RX Soundwire ports, move it to the enums defining actual port indices/values. This makes it more obvious why such value was chosen as number of TX/RX ports. Note: the enums start from 1, thus number of ports equals to the last vaue in the enum. Signed-off-by: Krzysztof Kozlowski Reviewed-by: Dmitry Baryshkov Link: https://patch.msgid.link/20240725-asoc-wsa88xx-port-arrays-v1-7-80a03f440c72@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd939x.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd939x.h b/sound/soc/codecs/wcd939x.h index 1571c2120cfc..3204fb10b58d 100644 --- a/sound/soc/codecs/wcd939x.h +++ b/sound/soc/codecs/wcd939x.h @@ -842,9 +842,6 @@ #define WCD939X_DSD_HPHR_CFG5 (0x35a6) #define WCD939X_MAX_REGISTER (WCD939X_DSD_HPHR_CFG5) -#define WCD939X_MAX_SWR_PORTS (6) -#define WCD939X_MAX_RX_SWR_PORTS (6) -#define WCD939X_MAX_TX_SWR_PORTS (4) #define WCD939X_MAX_SWR_CH_IDS (15) struct wcd939x_sdw_ch_info { @@ -863,6 +860,7 @@ enum wcd939x_tx_sdw_ports { WCD939X_ADC_DMIC_1_2_PORT, WCD939X_DMIC_0_3_MBHC_PORT, WCD939X_DMIC_3_7_PORT, + WCD939X_MAX_TX_SWR_PORTS = WCD939X_DMIC_3_7_PORT, }; enum wcd939x_tx_sdw_channels { @@ -888,6 +886,8 @@ enum wcd939x_rx_sdw_ports { WCD939X_LO_PORT, WCD939X_DSD_PORT, WCD939X_HIFI_PCM_PORT, + WCD939X_MAX_RX_SWR_PORTS = WCD939X_HIFI_PCM_PORT, + WCD939X_MAX_SWR_PORTS = WCD939X_MAX_RX_SWR_PORTS, }; enum wcd939x_rx_sdw_channels { From e620b496c78706bb71691502e0381eb344afeaea Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Tue, 16 Jul 2024 14:41:17 +0800 Subject: [PATCH 024/410] ASoC: tas2781: Add TAS2563 into the Header Add TAS2563 into the Header in case of misunderstanding and add channel No information for error debug in tasdevice_dev_read. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240716064120.158-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } From 4e9652003bc39032c3ab79e607aa3d1913c7cf57 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 26 Jul 2024 17:28:15 +0200 Subject: [PATCH 025/410] ALSA: control: Annotate snd_kcontrol with __counted_by() struct snd_kcontrol contains a flex array of snd_kcontrol_volatile objects at its end, and the array size is stored in count field. This can be annotated gracefully with __counted_by() for catching possible array overflows. One additional change is the order of the count field initialization; The assignment of the count field is moved before assignment of vd[] elements for avoiding false-positive warnings from compilers. Link: https://patch.msgid.link/20240726152840.8629-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/control.h | 2 +- sound/core/control.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index c1659036c4a7..13511c50825f 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -81,7 +81,7 @@ struct snd_kcontrol { unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); - struct snd_kcontrol_volatile vd[]; /* volatile data */ + struct snd_kcontrol_volatile vd[] __counted_by(count); /* volatile data */ }; #define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list) diff --git a/sound/core/control.c b/sound/core/control.c index f64a555f404f..c3aad64ffbf5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -237,11 +237,11 @@ static int snd_ctl_new(struct snd_kcontrol **kctl, unsigned int count, if (!*kctl) return -ENOMEM; + (*kctl)->count = count; for (idx = 0; idx < count; idx++) { (*kctl)->vd[idx].access = access; (*kctl)->vd[idx].owner = file; } - (*kctl)->count = count; return 0; } From 0642a3c5cacc0321c755d45ae48f2c84475469a6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2024 16:13:14 +0200 Subject: [PATCH 026/410] ALSA: ump: Update substream name from assigned FB names We had a nice name scheme in ALSA sequencer UMP binding for each sequencer port referring to each assigned Function Block name, while the legacy rawmidi refers only to the UMP Endpoint name. It's better to align both. This patch moves the UMP Group attribute update functions into the core UMP code from the sequencer binding code, and improve the substream name of the legacy rawmidi. Link: https://patch.msgid.link/20240729141315.18253-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/ump.h | 10 +++++ sound/core/seq/seq_ump_client.c | 75 +++------------------------------ sound/core/ump.c | 74 ++++++++++++++++++++++++++++++-- 3 files changed, 86 insertions(+), 73 deletions(-) diff --git a/include/sound/ump.h b/include/sound/ump.h index 91238dabe307..7f68056acdff 100644 --- a/include/sound/ump.h +++ b/include/sound/ump.h @@ -13,6 +13,14 @@ struct snd_ump_ops; struct ump_cvt_to_ump; struct snd_seq_ump_ops; +struct snd_ump_group { + int group; /* group index (0-based) */ + unsigned int dir_bits; /* directions */ + bool active; /* activeness */ + bool valid; /* valid group (referred by blocks) */ + char name[64]; /* group name */ +}; + struct snd_ump_endpoint { struct snd_rawmidi core; /* raw UMP access */ @@ -41,6 +49,8 @@ struct snd_ump_endpoint { struct mutex open_mutex; + struct snd_ump_group groups[SNDRV_UMP_MAX_GROUPS]; /* table of groups */ + #if IS_ENABLED(CONFIG_SND_UMP_LEGACY_RAWMIDI) spinlock_t legacy_locks[2]; struct snd_rawmidi *legacy_rmidi; diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index 9cdfbeae3ed5..637154bd1a3f 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -23,15 +23,6 @@ enum { STR_OUT = SNDRV_RAWMIDI_STREAM_OUTPUT }; -/* object per UMP group; corresponding to a sequencer port */ -struct seq_ump_group { - int group; /* group index (0-based) */ - unsigned int dir_bits; /* directions */ - bool active; /* activeness */ - bool valid; /* valid group (referred by blocks) */ - char name[64]; /* seq port name */ -}; - /* context for UMP input parsing, per EP */ struct seq_ump_input_buffer { unsigned char len; /* total length in words */ @@ -48,7 +39,6 @@ struct seq_ump_client { int opened[2]; /* current opens for each direction */ struct snd_rawmidi_file out_rfile; /* rawmidi for output */ struct seq_ump_input_buffer input; /* input parser context */ - struct seq_ump_group groups[SNDRV_UMP_MAX_GROUPS]; /* table of groups */ void *ump_info[SNDRV_UMP_MAX_BLOCKS + 1]; /* shadow of seq client ump_info */ struct work_struct group_notify_work; /* FB change notification */ }; @@ -175,7 +165,7 @@ static int seq_ump_unuse(void *pdata, struct snd_seq_port_subscribe *info) /* fill port_info from the given UMP EP and group info */ static void fill_port_info(struct snd_seq_port_info *port, struct seq_ump_client *client, - struct seq_ump_group *group) + struct snd_ump_group *group) { unsigned int rawmidi_info = client->ump->core.info_flags; @@ -212,7 +202,7 @@ static void fill_port_info(struct snd_seq_port_info *port, } /* skip non-existing group for static blocks */ -static bool skip_group(struct seq_ump_client *client, struct seq_ump_group *group) +static bool skip_group(struct seq_ump_client *client, struct snd_ump_group *group) { return !group->valid && (client->ump->info.flags & SNDRV_UMP_EP_INFO_STATIC_BLOCKS); @@ -221,7 +211,7 @@ static bool skip_group(struct seq_ump_client *client, struct seq_ump_group *grou /* create a new sequencer port per UMP group */ static int seq_ump_group_init(struct seq_ump_client *client, int group_index) { - struct seq_ump_group *group = &client->groups[group_index]; + struct snd_ump_group *group = &client->ump->groups[group_index]; struct snd_seq_port_info *port __free(kfree) = NULL; struct snd_seq_port_callback pcallbacks; @@ -261,7 +251,7 @@ static void update_port_infos(struct seq_ump_client *client) return; for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) { - if (skip_group(client, &client->groups[i])) + if (skip_group(client, &client->ump->groups[i])) continue; old->addr.client = client->seq_client; @@ -271,7 +261,7 @@ static void update_port_infos(struct seq_ump_client *client) old); if (err < 0) return; - fill_port_info(new, client, &client->groups[i]); + fill_port_info(new, client, &client->ump->groups[i]); if (old->capability == new->capability && !strcmp(old->name, new->name)) continue; @@ -285,57 +275,6 @@ static void update_port_infos(struct seq_ump_client *client) } } -/* update dir_bits and active flag for all groups in the client */ -static void update_group_attrs(struct seq_ump_client *client) -{ - struct snd_ump_block *fb; - struct seq_ump_group *group; - int i; - - for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) { - group = &client->groups[i]; - *group->name = 0; - group->dir_bits = 0; - group->active = 0; - group->group = i; - group->valid = false; - } - - list_for_each_entry(fb, &client->ump->block_list, list) { - if (fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) - break; - group = &client->groups[fb->info.first_group]; - for (i = 0; i < fb->info.num_groups; i++, group++) { - group->valid = true; - if (fb->info.active) - group->active = 1; - switch (fb->info.direction) { - case SNDRV_UMP_DIR_INPUT: - group->dir_bits |= (1 << STR_IN); - break; - case SNDRV_UMP_DIR_OUTPUT: - group->dir_bits |= (1 << STR_OUT); - break; - case SNDRV_UMP_DIR_BIDIRECTION: - group->dir_bits |= (1 << STR_OUT) | (1 << STR_IN); - break; - } - if (!*fb->info.name) - continue; - if (!*group->name) { - /* store the first matching name */ - strscpy(group->name, fb->info.name, - sizeof(group->name)); - } else { - /* when overlapping, concat names */ - strlcat(group->name, ", ", sizeof(group->name)); - strlcat(group->name, fb->info.name, - sizeof(group->name)); - } - } - } -} - /* create a UMP Endpoint port */ static int create_ump_endpoint_port(struct seq_ump_client *client) { @@ -432,7 +371,7 @@ static void setup_client_group_filter(struct seq_ump_client *client) return; filter = ~(1U << 0); /* always allow groupless messages */ for (p = 0; p < SNDRV_UMP_MAX_GROUPS; p++) { - if (client->groups[p].active) + if (client->ump->groups[p].active) filter &= ~(1U << (p + 1)); } cptr->group_filter = filter; @@ -445,7 +384,6 @@ static void handle_group_notify(struct work_struct *work) struct seq_ump_client *client = container_of(work, struct seq_ump_client, group_notify_work); - update_group_attrs(client); update_port_infos(client); setup_client_group_filter(client); } @@ -508,7 +446,6 @@ static int snd_seq_ump_probe(struct device *_dev) client->ump_info[fb->info.block_id + 1] = &fb->info; setup_client_midi_version(client); - update_group_attrs(client); for (p = 0; p < SNDRV_UMP_MAX_GROUPS; p++) { err = seq_ump_group_init(client, p); diff --git a/sound/core/ump.c b/sound/core/ump.c index 0f0d7e895c5a..e39e9cda4912 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -524,6 +524,58 @@ static void snd_ump_proc_read(struct snd_info_entry *entry, } } +/* update dir_bits and active flag for all groups in the client */ +static void update_group_attrs(struct snd_ump_endpoint *ump) +{ + struct snd_ump_block *fb; + struct snd_ump_group *group; + int i; + + for (i = 0; i < SNDRV_UMP_MAX_GROUPS; i++) { + group = &ump->groups[i]; + *group->name = 0; + group->dir_bits = 0; + group->active = 0; + group->group = i; + group->valid = false; + } + + list_for_each_entry(fb, &ump->block_list, list) { + if (fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) + break; + group = &ump->groups[fb->info.first_group]; + for (i = 0; i < fb->info.num_groups; i++, group++) { + group->valid = true; + if (fb->info.active) + group->active = 1; + switch (fb->info.direction) { + case SNDRV_UMP_DIR_INPUT: + group->dir_bits |= (1 << SNDRV_RAWMIDI_STREAM_INPUT); + break; + case SNDRV_UMP_DIR_OUTPUT: + group->dir_bits |= (1 << SNDRV_RAWMIDI_STREAM_OUTPUT); + break; + case SNDRV_UMP_DIR_BIDIRECTION: + group->dir_bits |= (1 << SNDRV_RAWMIDI_STREAM_INPUT) | + (1 << SNDRV_RAWMIDI_STREAM_OUTPUT); + break; + } + if (!*fb->info.name) + continue; + if (!*group->name) { + /* store the first matching name */ + strscpy(group->name, fb->info.name, + sizeof(group->name)); + } else { + /* when overlapping, concat names */ + strlcat(group->name, ", ", sizeof(group->name)); + strlcat(group->name, fb->info.name, + sizeof(group->name)); + } + } + } +} + /* * UMP endpoint and function block handling */ @@ -792,8 +844,10 @@ static int ump_handle_fb_info_msg(struct snd_ump_endpoint *ump, if (fb) { fill_fb_info(ump, &fb->info, buf); - if (ump->parsed) + if (ump->parsed) { + update_group_attrs(ump); seq_notify_fb_change(ump, fb); + } } return 1; /* finished */ @@ -822,8 +876,10 @@ static int ump_handle_fb_name_msg(struct snd_ump_endpoint *ump, ret = ump_append_string(ump, fb->info.name, sizeof(fb->info.name), buf->raw, 3); /* notify the FB name update to sequencer, too */ - if (ret > 0 && ump->parsed) + if (ret > 0 && ump->parsed) { + update_group_attrs(ump); seq_notify_fb_change(ump, fb); + } return ret; } @@ -995,6 +1051,9 @@ int snd_ump_parse_endpoint(struct snd_ump_endpoint *ump) continue; } + /* initialize group attributions */ + update_group_attrs(ump); + error: ump->parsed = true; ump_request_close(ump); @@ -1172,10 +1231,17 @@ static void fill_substream_names(struct snd_ump_endpoint *ump, struct snd_rawmidi *rmidi, int dir) { struct snd_rawmidi_substream *s; + const char *name; + int idx; - list_for_each_entry(s, &rmidi->streams[dir].substreams, list) + list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { + idx = ump->legacy_mapping[s->number]; + name = ump->groups[idx].name; + if (!*name) + name = ump->info.name; snprintf(s->name, sizeof(s->name), "Group %d (%.16s)", - ump->legacy_mapping[s->number] + 1, ump->info.name); + idx + 1, name); + } } int snd_ump_attach_legacy_rawmidi(struct snd_ump_endpoint *ump, From 8abe0423ddd3ca7b9cea09c0a39249f65e646768 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2024 16:15:16 +0200 Subject: [PATCH 027/410] ALSA: hda: Keep PM disablement for deny-listed instance We have a runtime PM deny-list for the devices that show the problems (typically click noises) at runtime suspend/resume, and when it matches, the driver disables the default runtime PM. However, we still allow the runtime PM changed via power_save module option dynamically, and the desktop system often tweaks it. This ended up with a re-enablement of the runtime PM that surprises users, suddenly suffering from the noises. This patch changes the driver behavior slightly: when the device is listed in the deny-list, ignore the power_save option change and keep the original (that is, off) runtime PM state. Link: https://patch.msgid.link/20240729141519.18398-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 5 ++++- sound/pci/hda/hda_intel.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b33602e64d17..440f1b37e071 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -933,7 +933,8 @@ static int __maybe_unused param_set_xint(const char *val, const struct kernel_pa mutex_lock(&card_list_lock); list_for_each_entry(hda, &card_list, list) { chip = &hda->chip; - if (!hda->probe_continued || chip->disabled) + if (!hda->probe_continued || chip->disabled || + hda->runtime_pm_disabled) continue; snd_hda_set_power_save(&chip->bus, power_save * 1000); } @@ -2243,6 +2244,7 @@ static const struct snd_pci_quirk power_save_denylist[] = { static void set_default_power_save(struct azx *chip) { + struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int val = power_save; if (pm_blacklist) { @@ -2253,6 +2255,7 @@ static void set_default_power_save(struct azx *chip) dev_info(chip->card->dev, "device %04x:%04x is on the power_save denylist, forcing power_save to 0\n", q->subvendor, q->subdevice); val = 0; + hda->runtime_pm_disabled = 1; } } snd_hda_set_power_save(&chip->bus, val * 1000); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 0f39418f9328..2d1725f86ef1 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -22,6 +22,7 @@ struct hda_intel { /* extra flags */ unsigned int irq_pending_warned:1; unsigned int probe_continued:1; + unsigned int runtime_pm_disabled:1; /* vga_switcheroo setup */ unsigned int use_vga_switcheroo:1; From 3bb668264db5c68a02b557b3052644181bb4f4be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2024 16:15:17 +0200 Subject: [PATCH 028/410] ALSA: hda: Enhance pm_blacklist option We want sometimes to keep the runtime PM disabled persistently just like we did for the PM deny-list in the previous change, e.g. for testing some buggy device. This patch enhances the existing pm_blacklist option for achieving it easily. The default behavior doesn't change -- the driver looks up the deny list and disables the runtime PM if matches. However, when pm_blacklist=1 option is set, now the driver disables the runtime PM completely, just like the deny-list does. Update the documentation for this option, too. Link: https://patch.msgid.link/20240729141519.18398-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- Documentation/sound/alsa-configuration.rst | 3 +++ sound/pci/hda/hda_intel.c | 14 ++++++++++---- 2 files changed, 13 insertions(+), 4 deletions(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 829c672d9fe6..04254474fa04 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -1059,6 +1059,9 @@ power_save Automatic power-saving timeout (in second, 0 = disable) power_save_controller Reset HD-audio controller in power-saving mode (default = on) +pm_blacklist + Enable / disable power-management deny-list (default = look up PM + deny-list, 0 = skip PM deny-list, 1 = force to turn off runtime PM) align_buffer_size Force rounding of buffer/period sizes to multiples of 128 bytes. This is more efficient in terms of memory access but isn't diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 440f1b37e071..ca5953445636 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -175,8 +175,8 @@ module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); -static bool pm_blacklist = true; -module_param(pm_blacklist, bool, 0644); +static int pm_blacklist = -1; +module_param(pm_blacklist, bint, 0644); MODULE_PARM_DESC(pm_blacklist, "Enable power-management denylist"); /* reset the HD-audio controller in power save mode. @@ -188,7 +188,7 @@ module_param(power_save_controller, bool, 0644); MODULE_PARM_DESC(power_save_controller, "Reset controller in power save mode."); #else /* CONFIG_PM */ #define power_save 0 -#define pm_blacklist false +#define pm_blacklist 0 #define power_save_controller false #endif /* CONFIG_PM */ @@ -930,6 +930,9 @@ static int __maybe_unused param_set_xint(const char *val, const struct kernel_pa if (ret || prev == power_save) return ret; + if (pm_blacklist > 0) + return 0; + mutex_lock(&card_list_lock); list_for_each_entry(hda, &card_list, list) { chip = &hda->chip; @@ -2247,7 +2250,7 @@ static void set_default_power_save(struct azx *chip) struct hda_intel *hda = container_of(chip, struct hda_intel, chip); int val = power_save; - if (pm_blacklist) { + if (pm_blacklist < 0) { const struct snd_pci_quirk *q; q = snd_pci_quirk_lookup(chip->pci, power_save_denylist); @@ -2257,6 +2260,9 @@ static void set_default_power_save(struct azx *chip) val = 0; hda->runtime_pm_disabled = 1; } + } else if (pm_blacklist > 0) { + dev_info(chip->card->dev, "Forcing power_save to 0 via option\n"); + val = 0; } snd_hda_set_power_save(&chip->bus, val * 1000); } From fcc62b19104a67b9a2941513771e09389b75bd95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2024 18:06:58 +0200 Subject: [PATCH 029/410] ALSA: control: Take power_ref lock primarily The code path for kcontrol accesses have often nested locks of both card's controls_rwsem and power_ref, and applies in that order. However, what could take much longer is the latter, power_ref; it waits for the power state of the device, and it pretty much depends on the user's action. This patch swaps the locking order of those locks to a more natural way, namely, power_ref -> controls_rwsem, in order to shorten the time of possible nested locks. For consistency, power_ref is taken always in the top-level caller side (that is, *_user() functions and the ioctl handler itself). Link: https://patch.msgid.link/20240729160659.4516-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 54 ++++++++++++++++++++++++++++---------------- 1 file changed, 34 insertions(+), 20 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index c3aad64ffbf5..2b857757bb19 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1167,9 +1167,7 @@ static int __snd_ctl_elem_info(struct snd_card *card, #ifdef CONFIG_SND_DEBUG info->access = 0; #endif - result = snd_power_ref_and_wait(card); - if (!result) - result = kctl->info(kctl, info); + result = kctl->info(kctl, info); snd_power_unref(card); if (result >= 0) { snd_BUG_ON(info->access); @@ -1208,12 +1206,17 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, static int snd_ctl_elem_info_user(struct snd_ctl_file *ctl, struct snd_ctl_elem_info __user *_info) { + struct snd_card *card = ctl->card; struct snd_ctl_elem_info info; int result; if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; + result = snd_power_ref_and_wait(card); + if (result) + return result; result = snd_ctl_elem_info(ctl, &info); + snd_power_unref(card); if (result < 0) return result; /* drop internal access flags */ @@ -1257,10 +1260,7 @@ static int snd_ctl_elem_read(struct snd_card *card, if (!snd_ctl_skip_validation(&info)) fill_remaining_elem_value(control, &info, pattern); - ret = snd_power_ref_and_wait(card); - if (!ret) - ret = kctl->get(kctl, control); - snd_power_unref(card); + ret = kctl->get(kctl, control); if (ret < 0) return ret; if (!snd_ctl_skip_validation(&info) && @@ -1285,7 +1285,11 @@ static int snd_ctl_elem_read_user(struct snd_card *card, if (IS_ERR(control)) return PTR_ERR(no_free_ptr(control)); + result = snd_power_ref_and_wait(card); + if (result) + return result; result = snd_ctl_elem_read(card, control); + snd_power_unref(card); if (result < 0) return result; @@ -1300,7 +1304,7 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - int result; + int result = 0; down_write(&card->controls_rwsem); kctl = snd_ctl_find_id_locked(card, &control->id); @@ -1318,9 +1322,8 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, } snd_ctl_build_ioff(&control->id, kctl, index_offset); - result = snd_power_ref_and_wait(card); /* validate input values */ - if (IS_ENABLED(CONFIG_SND_CTL_INPUT_VALIDATION) && !result) { + if (IS_ENABLED(CONFIG_SND_CTL_INPUT_VALIDATION)) { struct snd_ctl_elem_info info; memset(&info, 0, sizeof(info)); @@ -1332,7 +1335,6 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, } if (!result) result = kctl->put(kctl, control); - snd_power_unref(card); if (result < 0) { up_write(&card->controls_rwsem); return result; @@ -1361,7 +1363,11 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, return PTR_ERR(no_free_ptr(control)); card = file->card; + result = snd_power_ref_and_wait(card); + if (result < 0) + return result; result = snd_ctl_elem_write(card, file, control); + snd_power_unref(card); if (result < 0) return result; @@ -1830,7 +1836,7 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, {SNDRV_CTL_TLV_OP_CMD, SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND}, }; struct snd_kcontrol_volatile *vd = &kctl->vd[snd_ctl_get_ioff(kctl, id)]; - int i, ret; + int i; /* Check support of the request for this element. */ for (i = 0; i < ARRAY_SIZE(pairs); ++i) { @@ -1848,11 +1854,7 @@ static int call_tlv_handler(struct snd_ctl_file *file, int op_flag, vd->owner != NULL && vd->owner != file) return -EPERM; - ret = snd_power_ref_and_wait(file->card); - if (!ret) - ret = kctl->tlv.c(kctl, op_flag, size, buf); - snd_power_unref(file->card); - return ret; + return kctl->tlv.c(kctl, op_flag, size, buf); } static int read_tlv_buf(struct snd_kcontrol *kctl, struct snd_ctl_elem_id *id, @@ -1965,16 +1967,28 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_SUBSCRIBE_EVENTS: return snd_ctl_subscribe_events(ctl, ip); case SNDRV_CTL_IOCTL_TLV_READ: - scoped_guard(rwsem_read, &ctl->card->controls_rwsem) + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; + scoped_guard(rwsem_read, &card->controls_rwsem) err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_READ); + snd_power_unref(card); return err; case SNDRV_CTL_IOCTL_TLV_WRITE: - scoped_guard(rwsem_write, &ctl->card->controls_rwsem) + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; + scoped_guard(rwsem_write, &card->controls_rwsem) err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_WRITE); + snd_power_unref(card); return err; case SNDRV_CTL_IOCTL_TLV_COMMAND: - scoped_guard(rwsem_write, &ctl->card->controls_rwsem) + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; + scoped_guard(rwsem_write, &card->controls_rwsem) err = snd_ctl_tlv_ioctl(ctl, argp, SNDRV_CTL_TLV_OP_CMD); + snd_power_unref(card); return err; case SNDRV_CTL_IOCTL_POWER: return -ENOPROTOOPT; From 80565764c7f53b47be266c90d635285e295684dc Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:12:10 +0000 Subject: [PATCH 030/410] ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG rsnd_mod_confirm_ssi() confirms mod sanity, it should always be confirmed, not only when DEBUG. This patch tidyup it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ed7bk4qt.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index afd69c6eb654..0f190abf00e7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -262,7 +262,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; - rsnd_mod_confirm_src(src_mod); + rsnd_mod_make_sure(src_mod, RSND_MOD_SRC); rsnd_adg_get_timesel_ratio(priv, io, in_rate, out_rate, @@ -291,7 +291,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int shift = (id % 4) * 8; u32 mask = 0xFF << shift; - rsnd_mod_confirm_ssi(ssi_mod); + rsnd_mod_make_sure(ssi_mod, RSND_MOD_SSI); val = val << shift; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d640..8cf5d9001f43 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -871,15 +871,6 @@ void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); -#ifdef DEBUG -#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) -#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) -#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -#else -#define rsnd_mod_confirm_ssi(mssi) -#define rsnd_mod_confirm_src(msrc) -#define rsnd_mod_confirm_dvc(mdvc) -#endif /* * If you don't need interrupt status debug message, From 22c406c9bf5e28d9fed0bf37ac9d544e56127fd3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:32:22 +0000 Subject: [PATCH 031/410] ASoC: rsnd: use pcm_dmaengine code rsnd is implementing own DMAEngine code, but we can replace it with pcm_dmaengine code, because these are almost same. Let's use existing and stable code. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87cymvk3t5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 17 --------- sound/soc/sh/rcar/dma.c | 75 ++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 2 +- 5 files changed, 9 insertions(+), 87 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7bddfd5e38d6..426632996a0a 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_RCAR depends on COMMON_CLK depends on OF select SND_SIMPLE_CARD_UTILS + select SND_DMAENGINE_PCM select REGMAP_MMIO help This option enables R-Car SRU/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 63b3c8bf0fde..15cb5e7008f9 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -660,23 +660,6 @@ static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) return rsnd_rdai_get(priv, dai->id); } -/* - * rsnd_soc_dai functions - */ -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io) -{ - struct snd_pcm_substream *substream = io->substream; - - /* - * this function should be called... - * - * - if rsnd_dai_pointer_update() returns true - * - without spin lock - */ - - snd_pcm_period_elapsed(substream); -} - static void rsnd_dai_stream_init(struct rsnd_dai_stream *io, struct snd_pcm_substream *substream) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7b499eee5080..2342bbb6fe92 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -7,6 +7,7 @@ #include #include +#include #include "rsnd.h" /* @@ -22,8 +23,6 @@ struct rsnd_dmaen { struct dma_chan *chan; - dma_cookie_t cookie; - unsigned int dma_len; }; struct rsnd_dmapp { @@ -66,20 +65,6 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -static void __rsnd_dmaen_complete(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - if (rsnd_io_is_working(io)) - rsnd_dai_period_elapsed(io); -} - -static void rsnd_dmaen_complete(void *data) -{ - struct rsnd_mod *mod = data; - - rsnd_mod_interrupt(mod, __rsnd_dmaen_complete); -} - static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_dai_stream *io, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) @@ -98,13 +83,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - - if (dmaen->chan) - dmaengine_terminate_async(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_STOP); } static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, @@ -120,7 +99,7 @@ static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, * Let's call it under prepare */ if (dmaen->chan) - dma_release_channel(dmaen->chan); + snd_dmaengine_pcm_close_release_chan(io->substream); dmaen->chan = NULL; @@ -153,7 +132,7 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, return -EIO; } - return 0; + return snd_dmaengine_pcm_open(io->substream, dmaen->chan); } static int rsnd_dmaen_start(struct rsnd_mod *mod, @@ -162,12 +141,9 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - int is_play = rsnd_io_is_play(io); int ret; /* @@ -195,7 +171,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, } } - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.direction = snd_pcm_substream_to_dma_direction(io->substream); cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = buswidth; @@ -209,32 +185,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return -EIO; - } - - desc->callback = rsnd_dmaen_complete; - desc->callback_param = rsnd_mod_get(dma); - - dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); - - dmaen->cookie = dmaengine_submit(desc); - if (dmaen->cookie < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return -EIO; - } - - dma_async_issue_pending(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_START); } struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, @@ -307,19 +258,7 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct dma_tx_state state; - enum dma_status status; - unsigned int pos = 0; - - status = dmaengine_tx_status(dmaen->chan, dmaen->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - if (state.residue > 0 && state.residue <= dmaen->dma_len) - pos = dmaen->dma_len - state.residue; - } - *pointer = bytes_to_frames(runtime, pos); + *pointer = snd_dmaengine_pcm_pointer(io->substream); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 8cf5d9001f43..3c164d8e3b16 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -576,7 +576,6 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, #define rsnd_rdai_width_get(rdai) \ rsnd_rdai_width_ctrl(rdai, 0) int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8d2a86383ae0..b3d4e8ae07ef 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -706,7 +706,7 @@ rsnd_ssi_interrupt_out: spin_unlock(&priv->lock); if (elapsed) - rsnd_dai_period_elapsed(io); + snd_pcm_period_elapsed(io->substream); if (stop) snd_pcm_stop_xrun(io->substream); From d5742b5d4d7b99531e352ea814506641f9fc8981 Mon Sep 17 00:00:00 2001 From: Yue Haibing Date: Wed, 31 Jul 2024 10:29:49 +0800 Subject: [PATCH 032/410] ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate' MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The variable savedbitclkrate is assigned and never used, so can be removed. sound/soc/fsl/lpc3xxx-i2s.c:42:13: warning: variable ‘savedbitclkrate’ set but not used [-Wunused-but-set-variable] Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Yue Haibing Link: https://patch.msgid.link/20240731022949.135016-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/fsl/lpc3xxx-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index af995ca081a3..62ef624d6dd4 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -39,7 +39,7 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 { u32 i2srate; u32 idxx, idyy; - u32 savedbitclkrate, diff, trate, baseclk; + u32 diff, trate, baseclk; /* Adjust rate for sample size (bits) and 2 channels and offset for * divider in clock output @@ -53,14 +53,12 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 /* Find the best divider */ *clkx = *clky = 0; - savedbitclkrate = 0; diff = ~0; for (idxx = 1; idxx < 0xFF; idxx++) { for (idyy = 1; idyy < 0xFF; idyy++) { trate = (baseclk * idxx) / idyy; if (abs(trate - i2srate) < diff) { diff = abs(trate - i2srate); - savedbitclkrate = trate; *clkx = idxx; *clky = idyy; } From fef1ac950c600ba50ef4d65ca03c8dae9be7f9ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Aug 2024 08:42:01 +0200 Subject: [PATCH 033/410] ALSA: control: Fix leftover snd_power_unref() One snd_power_unref() was forgotten and left at __snd_ctl_elem_info() in the previous change for reorganizing the locking order. Fixes: fcc62b19104a ("ALSA: control: Take power_ref lock primarily") Link: https://github.com/thesofproject/linux/pull/5127 Link: https://patch.msgid.link/20240801064203.30284-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 2b857757bb19..f668826649b3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1168,7 +1168,6 @@ static int __snd_ctl_elem_info(struct snd_card *card, info->access = 0; #endif result = kctl->info(kctl, info); - snd_power_unref(card); if (result >= 0) { snd_BUG_ON(info->access); index_offset = snd_ctl_get_ioff(kctl, &info->id); From 9c27301342a53fdf349e4c630d75c7c12bbbc580 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Aug 2024 08:48:05 +0200 Subject: [PATCH 034/410] ALSA: memalloc: Use DMA API for x86 WC page allocations, too The memalloc helper used a house-made code for allocation of WC pages on x86, since the standard DMA API doesn't cover it well. Meanwhile, the manually allocated pages won't work together with IOMMU, resulting in faults, so we should switch to the DMA API in that case, instead. This patch tries to switch back to DMA API for WC pages on x86, but with some additional tweaks that are missing. Link: https://bugzilla.kernel.org/show_bug.cgi?id=219087 Link: https://patch.msgid.link/20240801064808.31205-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 35 +++++++++++++++++------------------ 1 file changed, 17 insertions(+), 18 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f901504b5afc..428652fda926 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -492,40 +492,39 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { */ /* x86-specific allocations */ #ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - return do_alloc_pages(dmab->dev.dev, size, &dmab->addr, true); -} - -static void snd_dma_wc_free(struct snd_dma_buffer *dmab) -{ - do_free_pages(dmab->area, dmab->bytes, true); -} - -static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return snd_dma_continuous_mmap(dmab, area); -} +#define x86_fallback(dmab) (!get_dma_ops(dmab->dev.dev)) #else +#define x86_fallback(dmab) false +#endif + static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { + if (x86_fallback(dmab)) + return do_alloc_pages(dmab->dev.dev, size, &dmab->addr, true); return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_wc_free(struct snd_dma_buffer *dmab) { + if (x86_fallback(dmab)) { + do_free_pages(dmab->area, dmab->bytes, true); + return; + } dma_free_wc(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { +#ifdef CONFIG_SND_DMA_SGBUF + if (x86_fallback(dmab)) { + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return snd_dma_continuous_mmap(dmab, area); + } +#endif return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } -#endif /* CONFIG_SND_DMA_SGBUF */ static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -548,7 +547,7 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, DEFAULT_GFP, 0); #ifdef CONFIG_SND_DMA_SGBUF - if (!sgt && !get_dma_ops(dmab->dev.dev)) + if (!sgt && x86_fallback(dmab)) return snd_dma_sg_fallback_alloc(dmab, size); #endif if (!sgt) From e469e2045f1b78418f39a2461a9fdf73c2789a1a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Aug 2024 08:48:06 +0200 Subject: [PATCH 035/410] ALSA: memalloc: Let IOMMU handle S/G primarily The recent changes in IOMMU made the non-contiguous page allocations as default, hence we can simply use the standard DMA allocation for the S/G pages as well. In this patch, we simplify the code by trying the standard DMA allocation at first, instead of dma_alloc_noncontiguous(). For the case without IOMMU, we still need to manage the S/G pages manually, so we keep the same fallback routines like before. The fallback types (SNDRV_DMA_TYPE_DEV_SG_FALLBACK & co) are dropped / folded into SNDRV_DMA_TYPE_DEV_SG and co now. The allocation via the standard DMA call overrides the type accordingly, hence we don't have to have extra fallback types any longer. OTOH, SNDRV_DMA_TYPE_DEV_SG is no longer an alias but became its own type back again. Note that this patch requires another prerequisite fix for memmalloc helper to use the DMA API for WC pages on x86. Link: https://bugzilla.kernel.org/show_bug.cgi?id=219087 Link: https://patch.msgid.link/20240801064808.31205-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/memalloc.h | 7 +-- sound/core/memalloc.c | 107 ++++++++++++--------------------------- 2 files changed, 34 insertions(+), 80 deletions(-) diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h index 43d524580bd2..9dd475cf4e8c 100644 --- a/include/sound/memalloc.h +++ b/include/sound/memalloc.h @@ -42,17 +42,12 @@ struct snd_dma_device { #define SNDRV_DMA_TYPE_NONCONTIG 8 /* non-coherent SG buffer */ #define SNDRV_DMA_TYPE_NONCOHERENT 9 /* non-coherent buffer */ #ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_NONCONTIG +#define SNDRV_DMA_TYPE_DEV_SG 3 /* S/G pages */ #define SNDRV_DMA_TYPE_DEV_WC_SG 6 /* SG write-combined */ #else #define SNDRV_DMA_TYPE_DEV_SG SNDRV_DMA_TYPE_DEV /* no SG-buf support */ #define SNDRV_DMA_TYPE_DEV_WC_SG SNDRV_DMA_TYPE_DEV_WC #endif -/* fallback types, don't use those directly */ -#ifdef CONFIG_SND_DMA_SGBUF -#define SNDRV_DMA_TYPE_DEV_SG_FALLBACK 10 -#define SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK 11 -#endif /* * info for buffer allocation diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 428652fda926..f3ad9f85adf1 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -26,10 +26,6 @@ static const struct snd_malloc_ops *snd_dma_get_ops(struct snd_dma_buffer *dmab); -#ifdef CONFIG_SND_DMA_SGBUF -static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size); -#endif - static void *__snd_dma_alloc_pages(struct snd_dma_buffer *dmab, size_t size) { const struct snd_malloc_ops *ops = snd_dma_get_ops(dmab); @@ -540,16 +536,8 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) struct sg_table *sgt; void *p; -#ifdef CONFIG_SND_DMA_SGBUF - if (cpu_feature_enabled(X86_FEATURE_XENPV)) - return snd_dma_sg_fallback_alloc(dmab, size); -#endif sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, DEFAULT_GFP, 0); -#ifdef CONFIG_SND_DMA_SGBUF - if (!sgt && x86_fallback(dmab)) - return snd_dma_sg_fallback_alloc(dmab, size); -#endif if (!sgt) return NULL; @@ -666,53 +654,7 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { .get_chunk_size = snd_dma_noncontig_get_chunk_size, }; -/* x86-specific SG-buffer with WC pages */ #ifdef CONFIG_SND_DMA_SGBUF -#define sg_wc_address(it) ((unsigned long)page_address(sg_page_iter_page(it))) - -static void *snd_dma_sg_wc_alloc(struct snd_dma_buffer *dmab, size_t size) -{ - void *p = snd_dma_noncontig_alloc(dmab, size); - struct sg_table *sgt = dmab->private_data; - struct sg_page_iter iter; - - if (!p) - return NULL; - if (dmab->dev.type != SNDRV_DMA_TYPE_DEV_WC_SG) - return p; - for_each_sgtable_page(sgt, &iter, 0) - set_memory_wc(sg_wc_address(&iter), 1); - return p; -} - -static void snd_dma_sg_wc_free(struct snd_dma_buffer *dmab) -{ - struct sg_table *sgt = dmab->private_data; - struct sg_page_iter iter; - - for_each_sgtable_page(sgt, &iter, 0) - set_memory_wb(sg_wc_address(&iter), 1); - snd_dma_noncontig_free(dmab); -} - -static int snd_dma_sg_wc_mmap(struct snd_dma_buffer *dmab, - struct vm_area_struct *area) -{ - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return dma_mmap_noncontiguous(dmab->dev.dev, area, - dmab->bytes, dmab->private_data); -} - -static const struct snd_malloc_ops snd_dma_sg_wc_ops = { - .alloc = snd_dma_sg_wc_alloc, - .free = snd_dma_sg_wc_free, - .mmap = snd_dma_sg_wc_mmap, - .sync = snd_dma_noncontig_sync, - .get_addr = snd_dma_noncontig_get_addr, - .get_page = snd_dma_noncontig_get_page, - .get_chunk_size = snd_dma_noncontig_get_chunk_size, -}; - /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { bool use_dma_alloc_coherent; @@ -750,6 +692,7 @@ static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, kfree(sgbuf); } +/* fallback manual S/G buffer allocations */ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) { struct snd_dma_sg_fallback *sgbuf; @@ -759,12 +702,6 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) dma_addr_t addr; void *p; - /* correct the type */ - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) - dmab->dev.type = SNDRV_DMA_TYPE_DEV_SG_FALLBACK; - else if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; - sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) return NULL; @@ -809,7 +746,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) if (!p) goto error; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) set_pages_array_wc(sgbuf->pages, sgbuf->count); dmab->private_data = sgbuf; @@ -826,7 +763,7 @@ static void snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab) { struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) set_pages_array_wb(sgbuf->pages, sgbuf->count); vunmap(dmab->area); __snd_dma_sg_fallback_free(dmab, dmab->private_data); @@ -846,13 +783,38 @@ static int snd_dma_sg_fallback_mmap(struct snd_dma_buffer *dmab, { struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); return vm_map_pages(area, sgbuf->pages, sgbuf->count); } -static const struct snd_malloc_ops snd_dma_sg_fallback_ops = { - .alloc = snd_dma_sg_fallback_alloc, +static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + int type = dmab->dev.type; + void *p; + + if (cpu_feature_enabled(X86_FEATURE_XENPV)) + return snd_dma_sg_fallback_alloc(dmab, size); + + /* try the standard DMA API allocation at first */ + if (type == SNDRV_DMA_TYPE_DEV_WC_SG) + dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC; + else + dmab->dev.type = SNDRV_DMA_TYPE_DEV; + p = __snd_dma_alloc_pages(dmab, size); + if (p) + return p; + + dmab->dev.type = type; /* restore the type */ + /* if IOMMU is present but failed, give up */ + if (!x86_fallback(dmab)) + return NULL; + /* try fallback */ + return snd_dma_sg_fallback_alloc(dmab, size); +} + +static const struct snd_malloc_ops snd_dma_sg_ops = { + .alloc = snd_dma_sg_alloc, .free = snd_dma_sg_fallback_free, .mmap = snd_dma_sg_fallback_mmap, .get_addr = snd_dma_sg_fallback_get_addr, @@ -926,15 +888,12 @@ static const struct snd_malloc_ops *snd_dma_ops[] = { [SNDRV_DMA_TYPE_NONCONTIG] = &snd_dma_noncontig_ops, [SNDRV_DMA_TYPE_NONCOHERENT] = &snd_dma_noncoherent_ops, #ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_wc_ops, + [SNDRV_DMA_TYPE_DEV_SG] = &snd_dma_sg_ops, + [SNDRV_DMA_TYPE_DEV_WC_SG] = &snd_dma_sg_ops, #endif #ifdef CONFIG_GENERIC_ALLOCATOR [SNDRV_DMA_TYPE_DEV_IRAM] = &snd_dma_iram_ops, #endif /* CONFIG_GENERIC_ALLOCATOR */ -#ifdef CONFIG_SND_DMA_SGBUF - [SNDRV_DMA_TYPE_DEV_SG_FALLBACK] = &snd_dma_sg_fallback_ops, - [SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK] = &snd_dma_sg_fallback_ops, -#endif #endif /* CONFIG_HAS_DMA */ }; From 7ca1d0ed1ad006b874a4ec94a004d194ab0cfb5f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:01 +0000 Subject: [PATCH 036/410] ALSA: pci: pcxhr: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87v80nk52q.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/pci/pcxhr/pcxhr_mix22.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index f340458fd2e1..e1435afc4907 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -535,7 +535,7 @@ int hr222_update_analog_audio_level(struct snd_pcxhr *chip, { dev_dbg(chip->card->dev, "hr222_update_analog_audio_level(%s chan=%d)\n", - is_capture ? "capture" : "playback", channel); + snd_pcm_direction_name(is_capture), channel); if (is_capture) { int level_l, level_r, level_mic; /* we have to update all levels */ From fc5aeeabd28b18c8750e6f7ba269afa8405bcf2a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:07 +0000 Subject: [PATCH 037/410] ALSA: pci: rme9652: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ttg7k52k.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 267c7848974a..56d335f0e196 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5610,15 +5610,13 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, /* dev_dbg(hdspm->card->dev, "Allocated sample buffer for %s at 0x%08X\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "playback" : "capture", + snd_pcm_direction_name(substream->stream), snd_pcm_sgbuf_get_addr(substream, 0)); */ /* dev_dbg(hdspm->card->dev, "set_hwparams: %s %d Hz, %d channels, bs = %d\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? - "playback" : "capture", + snd_pcm_direction_name(substream->stream), params_rate(params), params_channels(params), params_buffer_size(params)); */ From 469b77e421b92ce4662f6f1a47f1e7af9dbd14bd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:12 +0000 Subject: [PATCH 038/410] ALSA: trace: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87sevrk52f.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- include/trace/events/asoc.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 202fc3680c36..6696dbcc2b96 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -8,6 +8,7 @@ #include #include #include +#include #define DAPM_DIRECT "(direct)" #define DAPM_ARROW(dir) (((dir) == SND_SOC_DAPM_DIR_OUT) ? "->" : "<-") @@ -212,7 +213,7 @@ TRACE_EVENT(snd_soc_dapm_connected, ), TP_printk("%s: found %d paths", - __entry->stream ? "capture" : "playback", __entry->paths) + snd_pcm_direction_name(__entry->stream), __entry->paths) ); TRACE_EVENT(snd_soc_jack_irq, From e1a642aba479e861e3d552b0548d1274d6a1c122 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:20 +0000 Subject: [PATCH 039/410] ALSA: aloop: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87r0bbk528.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/drivers/aloop.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index d6dd4b8c750a..439d12ad8787 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -900,8 +900,7 @@ static void loopback_snd_timer_dpcm_info(struct loopback_pcm *dpcm, cable->snd_timer.id.device, cable->snd_timer.id.subdevice); snd_iprintf(buffer, " timer open:\t\t%s\n", - (cable->snd_timer.stream == SNDRV_PCM_STREAM_CAPTURE) ? - "capture" : "playback"); + snd_pcm_direction_name(cable->snd_timer.stream)); } static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) From a48fee68a8fa35fc1a9b924c06d3e023d067ff41 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:30 +0000 Subject: [PATCH 040/410] ALSA: pcm_timer: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87plqvk51y.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Takashi Iwai --- sound/core/pcm_timer.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index c43484b22b34..ab0e5bd70f8f 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -108,8 +108,7 @@ void snd_pcm_timer_init(struct snd_pcm_substream *substream) if (snd_timer_new(substream->pcm->card, "PCM", &tid, &timer) < 0) return; sprintf(timer->name, "PCM %s %i-%i-%i", - substream->stream == SNDRV_PCM_STREAM_CAPTURE ? - "capture" : "playback", + snd_pcm_direction_name(substream->stream), tid.card, tid.device, tid.subdevice); timer->hw = snd_pcm_timer; if (snd_device_register(timer->card, timer) < 0) { From 6588fcc8833d8338b994edd97a4446bae95ff12c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:16 +0530 Subject: [PATCH 041/410] ASoC: intel: rename codec_info and dai_info structures names To make it generic, rename structure 'sof_sdw_codec_info' as 'asoc_sdw_codec_info' and 'sof_sdw_dai_info' as 'asoc_sdw_dai_info'. These structures will be moved to common header file so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bridge_cs35l56.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 26 +++++++++---------- sound/soc/intel/boards/sof_sdw_common.h | 24 ++++++++--------- sound/soc/intel/boards/sof_sdw_cs42l43.c | 2 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 2 +- sound/soc/intel/boards/sof_sdw_maxim.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 2 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 2 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 2 +- 9 files changed, 32 insertions(+), 32 deletions(-) diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/intel/boards/bridge_cs35l56.c index c3995e724aed..b289bcb3847a 100644 --- a/sound/soc/intel/boards/bridge_cs35l56.c +++ b/sound/soc/intel/boards/bridge_cs35l56.c @@ -127,7 +127,7 @@ int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, int bridge_cs35l56_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { if (sof_sdw_quirk & SOF_SIDECAR_AMPS) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e5feaef669d1..26c467bb81a3 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -731,7 +731,7 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = sdw_shutdown, }; -static struct sof_sdw_codec_info codec_info_list[] = { +static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x700, .dais = { @@ -1220,7 +1220,7 @@ static struct sof_sdw_codec_info codec_info_list[] = { }, }; -static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr) +static struct asoc_sdw_codec_info *find_codec_info_part(const u64 adr) { unsigned int part_id, sdw_version; int i; @@ -1241,7 +1241,7 @@ static struct sof_sdw_codec_info *find_codec_info_part(const u64 adr) } -static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) +static struct asoc_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) { int i; @@ -1255,8 +1255,8 @@ static struct sof_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) return NULL; } -static struct sof_sdw_codec_info *find_codec_info_dai(const char *dai_name, - int *dai_index) +static struct asoc_sdw_codec_info *find_codec_info_dai(const char *dai_name, + int *dai_index) { int i, j; @@ -1355,7 +1355,7 @@ static bool is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, } static const char *get_codec_name(struct device *dev, - const struct sof_sdw_codec_info *codec_info, + const struct asoc_sdw_codec_info *codec_info, const struct snd_soc_acpi_link_adr *adr_link, int adr_index) { @@ -1383,7 +1383,7 @@ static const char *get_codec_name(struct device *dev, static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; - struct sof_sdw_codec_info *codec_info; + struct asoc_sdw_codec_info *codec_info; struct snd_soc_dai *dai; int dai_index; int ret; @@ -1451,8 +1451,8 @@ struct sof_sdw_endpoint { const char *name_prefix; bool include_sidecar; - struct sof_sdw_codec_info *codec_info; - const struct sof_sdw_dai_info *dai_info; + struct asoc_sdw_codec_info *codec_info; + const struct asoc_sdw_dai_info *dai_info; }; struct sof_sdw_dailink { @@ -1529,7 +1529,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, for (i = 0; i < adr_link->num_adr; i++) { const struct snd_soc_acpi_adr_device *adr_dev = &adr_link->adr_d[i]; - struct sof_sdw_codec_info *codec_info; + struct asoc_sdw_codec_info *codec_info; const char *codec_name; if (!adr_dev->name_prefix) { @@ -1563,7 +1563,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, for (j = 0; j < adr_dev->num_endpoints; j++) { const struct snd_soc_acpi_endpoint *adr_end; - const struct sof_sdw_dai_info *dai_info; + const struct asoc_sdw_dai_info *dai_info; struct sof_sdw_dailink *sof_dai; int stream; @@ -1790,7 +1790,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, static int create_ssp_dailinks(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, int *be_id, - struct sof_sdw_codec_info *ssp_info, + struct asoc_sdw_codec_info *ssp_info, unsigned long ssp_mask) { struct device *dev = card->dev; @@ -1914,7 +1914,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; struct snd_soc_codec_conf *codec_conf; - struct sof_sdw_codec_info *ssp_info; + struct asoc_sdw_codec_info *ssp_info; struct sof_sdw_endpoint *sof_ends; struct sof_sdw_dailink *sof_dais; int num_devs = 0; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 2a3145d1feb6..ebacb55de318 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -79,9 +79,9 @@ enum { #define SOF_SDW_MAX_DAI_NUM 8 -struct sof_sdw_codec_info; +struct asoc_sdw_codec_info; -struct sof_sdw_dai_info { +struct asoc_sdw_dai_info { const bool direction[2]; /* playback & capture support */ const char *dai_name; const int dai_type; @@ -92,7 +92,7 @@ struct sof_sdw_dai_info { const int num_widgets; int (*init)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); int (*rtd_init)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); @@ -100,7 +100,7 @@ struct sof_sdw_dai_info { unsigned long quirk; }; -struct sof_sdw_codec_info { +struct asoc_sdw_codec_info { const int part_id; const int version_id; const char *codec_name; @@ -108,7 +108,7 @@ struct sof_sdw_codec_info { const u8 acpi_id[ACPI_ID_LEN]; const bool ignore_pch_dmic; const struct snd_soc_ops *ops; - struct sof_sdw_dai_info dais[SOF_SDW_MAX_DAI_NUM]; + struct asoc_sdw_dai_info dais[SOF_SDW_MAX_DAI_NUM]; const int dai_num; int (*codec_card_late_probe)(struct snd_soc_card *card); @@ -153,14 +153,14 @@ int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); /* RT711 support */ int sof_sdw_rt711_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT711-SDCA support */ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); @@ -170,20 +170,20 @@ extern const struct snd_soc_ops sof_sdw_rt1308_i2s_ops; /* generic amp support */ int sof_sdw_rt_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* MAXIM codec support */ int sof_sdw_maxim_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); /* CS42L43 support */ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); /* CS AMP support */ @@ -194,12 +194,12 @@ int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, struct snd_soc_codec_conf **codec_conf); int bridge_cs35l56_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); int sof_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback); /* dai_link init callbacks */ diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index b7e2750c1074..e8370c3c96fd 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -123,7 +123,7 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *da int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { /* Do init on playback link only. */ diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 10e08207619a..206fe7610a56 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -47,7 +47,7 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) int sof_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { /* Do init on playback link only. */ diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index b7f73177867f..8faa06824869 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -120,7 +120,7 @@ static int mx8373_sdw_late_probe(struct snd_soc_card *card) int sof_sdw_maxim_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { info->amp_num++; diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 60ff4d88e2dc..9ea8de609669 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -126,7 +126,7 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l int sof_sdw_rt711_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index d1c0f91ce589..da1a8cc1c63d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -256,7 +256,7 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_ int sof_sdw_rt_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 4254e30ee4c3..b63f4952e7d1 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -180,7 +180,7 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, - struct sof_sdw_codec_info *info, + struct asoc_sdw_codec_info *info, bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); From 408a454ee8886912cc8a64fcd012e0a1eb30fd0b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:17 +0530 Subject: [PATCH 042/410] ASoC: intel: rename soundwire common header macros Rename sof quirk macros, dai type and dai link macros with "SOC_SDW" tag. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bridge_cs35l56.c | 6 +- sound/soc/intel/boards/sof_sdw.c | 184 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 46 ++--- sound/soc/intel/boards/sof_sdw_cs42l43.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 6 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 8 +- 6 files changed, 126 insertions(+), 126 deletions(-) diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/intel/boards/bridge_cs35l56.c index b289bcb3847a..6b2526c781e5 100644 --- a/sound/soc/intel/boards/bridge_cs35l56.c +++ b/sound/soc/intel/boards/bridge_cs35l56.c @@ -98,7 +98,7 @@ static const struct snd_soc_dai_link bridge_dai_template = { int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, int *num_dais, int *num_devs) { - if (sof_sdw_quirk & SOF_SIDECAR_AMPS) { + if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { (*num_dais)++; (*num_devs) += ARRAY_SIZE(bridge_cs35l56_name_prefixes); } @@ -110,7 +110,7 @@ int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, struct snd_soc_codec_conf **codec_conf) { - if (sof_sdw_quirk & SOF_SIDECAR_AMPS) { + if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { **dai_links = bridge_dai_template; for (int i = 0; i < ARRAY_SIZE(bridge_cs35l56_name_prefixes); i++) { @@ -130,7 +130,7 @@ int bridge_cs35l56_spk_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback) { - if (sof_sdw_quirk & SOF_SIDECAR_AMPS) + if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) info->amp_num += ARRAY_SIZE(bridge_cs35l56_name_prefixes); return 0; diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 26c467bb81a3..64fb53772c04 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -23,24 +23,24 @@ MODULE_PARM_DESC(quirk, "Board-specific quirk override"); static void log_quirks(struct device *dev) { - if (SOF_JACK_JDSRC(sof_sdw_quirk)) + if (SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) dev_dbg(dev, "quirk realtek,jack-detect-source %ld\n", - SOF_JACK_JDSRC(sof_sdw_quirk)); - if (sof_sdw_quirk & SOF_SDW_FOUR_SPK) - dev_err(dev, "quirk SOF_SDW_FOUR_SPK enabled but no longer supported\n"); + SOC_SDW_JACK_JDSRC(sof_sdw_quirk)); + if (sof_sdw_quirk & SOC_SDW_FOUR_SPK) + dev_err(dev, "quirk SOC_SDW_FOUR_SPK enabled but no longer supported\n"); if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) dev_dbg(dev, "quirk SOF_SDW_TGL_HDMI enabled\n"); - if (sof_sdw_quirk & SOF_SDW_PCH_DMIC) - dev_dbg(dev, "quirk SOF_SDW_PCH_DMIC enabled\n"); + if (sof_sdw_quirk & SOC_SDW_PCH_DMIC) + dev_dbg(dev, "quirk SOC_SDW_PCH_DMIC enabled\n"); if (SOF_SSP_GET_PORT(sof_sdw_quirk)) dev_dbg(dev, "SSP port %ld\n", SOF_SSP_GET_PORT(sof_sdw_quirk)); - if (sof_sdw_quirk & SOF_SDW_NO_AGGREGATION) - dev_err(dev, "quirk SOF_SDW_NO_AGGREGATION enabled but no longer supported\n"); - if (sof_sdw_quirk & SOF_CODEC_SPKR) - dev_dbg(dev, "quirk SOF_CODEC_SPKR enabled\n"); - if (sof_sdw_quirk & SOF_SIDECAR_AMPS) - dev_dbg(dev, "quirk SOF_SIDECAR_AMPS enabled\n"); + if (sof_sdw_quirk & SOC_SDW_NO_AGGREGATION) + dev_err(dev, "quirk SOC_SDW_NO_AGGREGATION enabled but no longer supported\n"); + if (sof_sdw_quirk & SOC_SDW_CODEC_SPKR) + dev_dbg(dev, "quirk SOC_SDW_CODEC_SPKR enabled\n"); + if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) + dev_dbg(dev, "quirk SOC_SDW_SIDECAR_AMPS enabled\n"); } static int sof_sdw_quirk_cb(const struct dmi_system_id *id) @@ -57,7 +57,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), DMI_MATCH(DMI_PRODUCT_NAME, "CometLake Client"), }, - .driver_data = (void *)SOF_SDW_PCH_DMIC, + .driver_data = (void *)SOC_SDW_PCH_DMIC, }, { .callback = sof_sdw_quirk_cb, @@ -99,7 +99,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), DMI_MATCH(DMI_PRODUCT_NAME, "Ice Lake Client"), }, - .driver_data = (void *)SOF_SDW_PCH_DMIC, + .driver_data = (void *)SOC_SDW_PCH_DMIC, }, /* TigerLake devices */ { @@ -111,7 +111,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | RT711_JD1 | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | SOF_SSP_PORT(SOF_I2S_SSP2)), }, { @@ -159,7 +159,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Volteer"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, @@ -170,7 +170,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Ripto"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC), + SOC_SDW_PCH_DMIC), }, { /* @@ -185,7 +185,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "HP Spectre x360 Conv"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD1), }, { @@ -199,7 +199,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "8709"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD1), }, { @@ -210,7 +210,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "LAPBC"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD1), }, { @@ -221,7 +221,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "LAPBC710"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD1), }, { @@ -232,7 +232,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "LAPRC"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD2_100K), }, { @@ -243,7 +243,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | RT711_JD2_100K), }, /* TigerLake-SDCA devices */ @@ -293,7 +293,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Brya"), }, .driver_data = (void *)(SOF_SDW_TGL_HDMI | - SOF_SDW_PCH_DMIC | + SOC_SDW_PCH_DMIC | SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, @@ -501,7 +501,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Google"), DMI_MATCH(DMI_PRODUCT_NAME, "Rex"), }, - .driver_data = (void *)(SOF_SDW_PCH_DMIC | + .driver_data = (void *)(SOC_SDW_PCH_DMIC | SOF_BT_OFFLOAD_SSP(1) | SOF_SSP_BT_OFFLOAD_PRESENT), }, @@ -529,7 +529,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") }, - .driver_data = (void *)(SOF_SIDECAR_AMPS), + .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS), }, { .callback = sof_sdw_quirk_cb, @@ -537,7 +537,7 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") }, - .driver_data = (void *)(SOF_SIDECAR_AMPS), + .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS), }, {} }; @@ -738,8 +738,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt700-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .rtd_init = rt700_rtd_init, .controls = rt700_controls, .num_controls = ARRAY_SIZE(rt700_controls), @@ -756,8 +756,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt711-sdca-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, @@ -776,8 +776,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt711-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt711_init, .exit = sof_sdw_rt711_exit, .rtd_init = rt711_rtd_init, @@ -796,8 +796,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt712-sdca-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, @@ -809,8 +809,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "rt712-sdca-aif2", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt712_spk_rtd_init, @@ -829,8 +829,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt712-sdca-dmic-aif1", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -843,8 +843,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt712-sdca-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, @@ -863,8 +863,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt712-sdca-dmic-aif1", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -877,8 +877,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "rt1308-aif", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, @@ -897,8 +897,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt1316-aif", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, @@ -916,8 +916,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt1318-aif", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt_amp_spk_rtd_init, @@ -937,8 +937,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt715-sdca-aif2", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -952,8 +952,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt715-sdca-aif2", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -967,8 +967,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt715-aif2", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -982,8 +982,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt715-aif2", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -996,8 +996,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt722-sdca-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .init = sof_sdw_rt_sdca_jack_init, .exit = sof_sdw_rt_sdca_jack_exit, .rtd_init = rt_sdca_jack_rtd_init, @@ -1009,9 +1009,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "rt722-sdca-aif2", - .dai_type = SOF_SDW_DAI_TYPE_AMP, + .dai_type = SOC_SDW_DAI_TYPE_AMP, /* No feedback capability is provided by rt722-sdca codec driver*/ - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = sof_sdw_rt_amp_init, .exit = sof_sdw_rt_amp_exit, .rtd_init = rt722_spk_rtd_init, @@ -1023,8 +1023,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "rt722-sdca-aif3", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = rt_dmic_rtd_init, }, }, @@ -1036,8 +1036,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "max98373-aif1", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = sof_sdw_maxim_init, .rtd_init = maxim_spk_rtd_init, .controls = maxim_controls, @@ -1054,8 +1054,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "max98363-aif1", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = sof_sdw_maxim_init, .rtd_init = maxim_spk_rtd_init, .controls = maxim_controls, @@ -1072,8 +1072,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "rt5682-sdw", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .rtd_init = rt5682_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), @@ -1089,8 +1089,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "cs35l56-sdw1", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = sof_sdw_cs_amp_init, .rtd_init = cs_spk_rtd_init, .controls = generic_spk_controls, @@ -1107,8 +1107,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "cs42l42-sdw", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, .rtd_init = cs42l42_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), @@ -1127,8 +1127,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, false}, .dai_name = "cs42l43-dp5", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .rtd_init = cs42l43_hs_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), @@ -1138,8 +1138,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "cs42l43-dp1", - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, .rtd_init = cs42l43_dmic_rtd_init, .widgets = generic_dmic_widgets, .num_widgets = ARRAY_SIZE(generic_dmic_widgets), @@ -1147,21 +1147,21 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {false, true}, .dai_name = "cs42l43-dp2", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_UNUSED_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, }, { .direction = {true, false}, .dai_name = "cs42l43-dp6", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_UNUSED_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, .init = sof_sdw_cs42l43_spk_init, .rtd_init = cs42l43_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, .num_widgets = ARRAY_SIZE(generic_spk_widgets), - .quirk = SOF_CODEC_SPKR | SOF_SIDECAR_AMPS, + .quirk = SOC_SDW_CODEC_SPKR | SOC_SDW_SIDECAR_AMPS, }, }, .dai_num = 4, @@ -1173,8 +1173,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "sdw-mockup-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, }, }, .dai_num = 1, @@ -1186,8 +1186,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "sdw-mockup-aif1", - .dai_type = SOF_SDW_DAI_TYPE_JACK, - .dailink = {SDW_JACK_OUT_DAI_ID, SDW_JACK_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, }, }, .dai_num = 1, @@ -1199,8 +1199,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .direction = {true, true}, .dai_name = "sdw-mockup-aif1", - .dai_type = SOF_SDW_DAI_TYPE_AMP, - .dailink = {SDW_AMP_OUT_DAI_ID, SDW_AMP_IN_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, }, }, .dai_num = 1, @@ -1212,8 +1212,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .dai_name = "sdw-mockup-aif1", .direction = {false, true}, - .dai_type = SOF_SDW_DAI_TYPE_MIC, - .dailink = {SDW_UNUSED_DAI_ID, SDW_DMIC_DAI_ID}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, }, }, .dai_num = 1, @@ -1767,7 +1767,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, int ret, i; for (i = 0; i < SDW_MAX_LINKS; i++) - ctx->sdw_pin_index[i] = SDW_INTEL_BIDIR_PDI_BASE; + ctx->sdw_pin_index[i] = SOC_SDW_INTEL_BIDIR_PDI_BASE; /* generate DAI links by each sdw link */ while (sof_dais->initialised) { @@ -1971,7 +1971,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) hdmi_num = SOF_PRE_TGL_HDMI_COUNT; /* enable dmic01 & dmic16k */ - if (sof_sdw_quirk & SOF_SDW_PCH_DMIC || mach_params->dmic_num) + if (sof_sdw_quirk & SOC_SDW_PCH_DMIC || mach_params->dmic_num) dmic_num = 2; if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index ebacb55de318..6ddfb1f9639a 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -14,16 +14,16 @@ #include #include "sof_hdmi_common.h" -#define MAX_NO_PROPS 2 +#define SOC_SDW_MAX_NO_PROPS 2 #define MAX_HDMI_NUM 4 -#define SDW_UNUSED_DAI_ID -1 -#define SDW_JACK_OUT_DAI_ID 0 -#define SDW_JACK_IN_DAI_ID 1 -#define SDW_AMP_OUT_DAI_ID 2 -#define SDW_AMP_IN_DAI_ID 3 -#define SDW_DMIC_DAI_ID 4 -#define SDW_MAX_CPU_DAIS 16 -#define SDW_INTEL_BIDIR_PDI_BASE 2 +#define SOC_SDW_UNUSED_DAI_ID -1 +#define SOC_SDW_JACK_OUT_DAI_ID 0 +#define SOC_SDW_JACK_IN_DAI_ID 1 +#define SOC_SDW_AMP_OUT_DAI_ID 2 +#define SOC_SDW_AMP_IN_DAI_ID 3 +#define SOC_SDW_DMIC_DAI_ID 4 +#define SOC_SDW_MAX_CPU_DAIS 16 +#define SOC_SDW_INTEL_BIDIR_PDI_BASE 2 #define SDW_MAX_LINKS 4 @@ -44,27 +44,27 @@ enum { SOF_I2S_SSP5 = BIT(5), }; -#define SOF_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) +#define SOC_SDW_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) /* Deprecated and no longer supported by the code */ -#define SOF_SDW_FOUR_SPK BIT(4) +#define SOC_SDW_FOUR_SPK BIT(4) #define SOF_SDW_TGL_HDMI BIT(5) -#define SOF_SDW_PCH_DMIC BIT(6) +#define SOC_SDW_PCH_DMIC BIT(6) #define SOF_SSP_PORT(x) (((x) & GENMASK(5, 0)) << 7) #define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0)) /* Deprecated and no longer supported by the code */ -#define SOF_SDW_NO_AGGREGATION BIT(14) +#define SOC_SDW_NO_AGGREGATION BIT(14) /* If a CODEC has an optional speaker output, this quirk will enable it */ -#define SOF_CODEC_SPKR BIT(15) +#define SOC_SDW_CODEC_SPKR BIT(15) /* * If the CODEC has additional devices attached directly to it. * * For the cs42l43: * - 0 - No speaker output - * - SOF_CODEC_SPKR - CODEC internal speaker - * - SOF_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker - * - SOF_CODEC_SPKR | SOF_SIDECAR_AMPS - Not currently supported + * - SOC_SDW_CODEC_SPKR - CODEC internal speaker + * - SOC_SDW_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker + * - SOC_SDW_CODEC_SPKR | SOC_SDW_SIDECAR_AMPS - Not currently supported */ -#define SOF_SIDECAR_AMPS BIT(16) +#define SOC_SDW_SIDECAR_AMPS BIT(16) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 15 @@ -73,11 +73,11 @@ enum { (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(18) -#define SOF_SDW_DAI_TYPE_JACK 0 -#define SOF_SDW_DAI_TYPE_AMP 1 -#define SOF_SDW_DAI_TYPE_MIC 2 +#define SOC_SDW_DAI_TYPE_JACK 0 +#define SOC_SDW_DAI_TYPE_AMP 1 +#define SOC_SDW_DAI_TYPE_MIC 2 -#define SOF_SDW_MAX_DAI_NUM 8 +#define SOC_SDW_MAX_DAI_NUM 8 struct asoc_sdw_codec_info; @@ -108,7 +108,7 @@ struct asoc_sdw_codec_info { const u8 acpi_id[ACPI_ID_LEN]; const bool ignore_pch_dmic; const struct snd_soc_ops *ops; - struct asoc_sdw_dai_info dais[SOF_SDW_MAX_DAI_NUM]; + struct asoc_sdw_dai_info dais[SOC_SDW_MAX_DAI_NUM]; const int dai_num; int (*codec_card_late_probe)(struct snd_soc_card *card); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index e8370c3c96fd..1688f29a977a 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -104,7 +104,7 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *da struct snd_soc_card *card = rtd->card; int ret; - if (!(sof_sdw_quirk & SOF_SIDECAR_AMPS)) { + if (!(sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS)) { /* Will be set by the bridge code in this case */ card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk", diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index 9ea8de609669..fff76291fca2 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -23,13 +23,13 @@ */ static int rt711_add_codec_device_props(struct device *sdw_dev) { - struct property_entry props[MAX_NO_PROPS] = {}; + struct property_entry props[SOC_SDW_MAX_NO_PROPS] = {}; struct fwnode_handle *fwnode; int ret; - if (!SOF_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) return 0; - props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOF_JACK_JDSRC(sof_sdw_quirk)); + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(sof_sdw_quirk)); fwnode = fwnode_create_software_node(props, NULL); if (IS_ERR(fwnode)) diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index b63f4952e7d1..9c3b93f808a2 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -23,14 +23,14 @@ */ static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev) { - struct property_entry props[MAX_NO_PROPS] = {}; + struct property_entry props[SOC_SDW_MAX_NO_PROPS] = {}; struct fwnode_handle *fwnode; int ret; - if (!SOF_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) return 0; - props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOF_JACK_JDSRC(sof_sdw_quirk)); + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(sof_sdw_quirk)); fwnode = fwnode_create_software_node(props, NULL); if (IS_ERR(fwnode)) @@ -168,7 +168,7 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link if (!ctx->headset_codec_dev) return 0; - if (!SOF_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) return 0; device_remove_software_node(ctx->headset_codec_dev); From 96990cfeff61d0a1053dded6d1a4dceafb2f1562 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:18 +0530 Subject: [PATCH 043/410] ASoC: intel: rename soundwire machine driver soc ops Rename Soundwire generic machine driver soc ops with tag "asoc". Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 26 ++++++++++++------------- sound/soc/intel/boards/sof_sdw_common.h | 14 ++++++------- sound/soc/intel/boards/sof_sdw_maxim.c | 12 ++++++------ 3 files changed, 26 insertions(+), 26 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 64fb53772c04..9395daf220e9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -594,12 +594,12 @@ static const struct snd_kcontrol_new rt700_controls[] = { }; /* these wrappers are only needed to avoid typecast compilation errors */ -int sdw_startup(struct snd_pcm_substream *substream) +int asoc_sdw_startup(struct snd_pcm_substream *substream) { return sdw_startup_stream(substream); } -int sdw_prepare(struct snd_pcm_substream *substream) +int asoc_sdw_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; @@ -617,7 +617,7 @@ int sdw_prepare(struct snd_pcm_substream *substream) return sdw_prepare_stream(sdw_stream); } -int sdw_trigger(struct snd_pcm_substream *substream, int cmd) +int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; @@ -656,8 +656,8 @@ int sdw_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } -int sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +int asoc_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai_link_ch_map *ch_maps; @@ -699,7 +699,7 @@ int sdw_hw_params(struct snd_pcm_substream *substream, return 0; } -int sdw_hw_free(struct snd_pcm_substream *substream) +int asoc_sdw_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sdw_stream_runtime *sdw_stream; @@ -717,18 +717,18 @@ int sdw_hw_free(struct snd_pcm_substream *substream) return sdw_deprepare_stream(sdw_stream); } -void sdw_shutdown(struct snd_pcm_substream *substream) +void asoc_sdw_shutdown(struct snd_pcm_substream *substream) { sdw_shutdown_stream(substream); } static const struct snd_soc_ops sdw_ops = { - .startup = sdw_startup, - .prepare = sdw_prepare, - .trigger = sdw_trigger, - .hw_params = sdw_hw_params, - .hw_free = sdw_hw_free, - .shutdown = sdw_shutdown, + .startup = asoc_sdw_startup, + .prepare = asoc_sdw_prepare, + .trigger = asoc_sdw_trigger, + .hw_params = asoc_sdw_hw_params, + .hw_free = asoc_sdw_hw_free, + .shutdown = asoc_sdw_shutdown, }; static struct asoc_sdw_codec_info codec_info_list[] = { diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 6ddfb1f9639a..e88b5627560b 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -134,13 +134,13 @@ struct mc_private { extern unsigned long sof_sdw_quirk; -int sdw_startup(struct snd_pcm_substream *substream); -int sdw_prepare(struct snd_pcm_substream *substream); -int sdw_trigger(struct snd_pcm_substream *substream, int cmd); -int sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params); -int sdw_hw_free(struct snd_pcm_substream *substream); -void sdw_shutdown(struct snd_pcm_substream *substream); +int asoc_sdw_startup(struct snd_pcm_substream *substream); +int asoc_sdw_prepare(struct snd_pcm_substream *substream); +int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd); +int asoc_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +int asoc_sdw_hw_free(struct snd_pcm_substream *substream); +void asoc_sdw_shutdown(struct snd_pcm_substream *substream); /* generic HDMI support */ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index 8faa06824869..f689dc29c493 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -80,7 +80,7 @@ static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) int ret; /* according to soc_pcm_prepare dai link prepare is called first */ - ret = sdw_prepare(substream); + ret = asoc_sdw_prepare(substream); if (ret < 0) return ret; @@ -92,7 +92,7 @@ static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) int ret; /* according to soc_pcm_hw_free dai link free is called first */ - ret = sdw_hw_free(substream); + ret = asoc_sdw_hw_free(substream); if (ret < 0) return ret; @@ -100,12 +100,12 @@ static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) } static const struct snd_soc_ops max_98373_sdw_ops = { - .startup = sdw_startup, + .startup = asoc_sdw_startup, .prepare = mx8373_sdw_prepare, - .trigger = sdw_trigger, - .hw_params = sdw_hw_params, + .trigger = asoc_sdw_trigger, + .hw_params = asoc_sdw_hw_params, .hw_free = mx8373_sdw_hw_free, - .shutdown = sdw_shutdown, + .shutdown = asoc_sdw_shutdown, }; static int mx8373_sdw_late_probe(struct snd_soc_card *card) From bd5838c8999838059a25a963730bba1face1e53b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:19 +0530 Subject: [PATCH 044/410] ASoC: intel: rename soundwire codec helper functions Rename SoundWire codec helper functions with "asoc_sdw" tag. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bridge_cs35l56.c | 18 +- sound/soc/intel/boards/sof_sdw.c | 154 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 114 ++++++------- sound/soc/intel/boards/sof_sdw_cs42l42.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 16 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 10 +- sound/soc/intel/boards/sof_sdw_dmic.c | 2 +- sound/soc/intel/boards/sof_sdw_maxim.c | 22 +-- sound/soc/intel/boards/sof_sdw_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw_rt700.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 12 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 2 +- sound/soc/intel/boards/sof_sdw_rt722_sdca.c | 2 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 14 +- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 2 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 12 +- 16 files changed, 193 insertions(+), 193 deletions(-) diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/intel/boards/bridge_cs35l56.c index 6b2526c781e5..55e5cfbb2f14 100644 --- a/sound/soc/intel/boards/bridge_cs35l56.c +++ b/sound/soc/intel/boards/bridge_cs35l56.c @@ -95,8 +95,8 @@ static const struct snd_soc_dai_link bridge_dai_template = { SND_SOC_DAILINK_REG(bridge_dai), }; -int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, - int *num_dais, int *num_devs) +int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card, + int *num_dais, int *num_devs) { if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { (*num_dais)++; @@ -106,9 +106,9 @@ int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, return 0; } -int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, - struct snd_soc_codec_conf **codec_conf) +int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf) { if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { **dai_links = bridge_dai_template; @@ -125,10 +125,10 @@ int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, return 0; } -int bridge_cs35l56_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_bridge_cs35l56_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) info->amp_num += ARRAY_SIZE(bridge_cs35l56_name_prefixes); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9395daf220e9..64418976aba4 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -740,7 +740,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt700-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = rt700_rtd_init, + .rtd_init = asoc_sdw_rt700_rtd_init, .controls = rt700_controls, .num_controls = ARRAY_SIZE(rt700_controls), .widgets = rt700_widgets, @@ -758,9 +758,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt711-sdca-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt_sdca_jack_init, - .exit = sof_sdw_rt_sdca_jack_exit, - .rtd_init = rt_sdca_jack_rtd_init, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -778,9 +778,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt711-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt711_init, - .exit = sof_sdw_rt711_exit, - .rtd_init = rt711_rtd_init, + .init = asoc_sdw_rt711_init, + .exit = asoc_sdw_rt711_exit, + .rtd_init = asoc_sdw_rt711_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -798,9 +798,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt_sdca_jack_init, - .exit = sof_sdw_rt_sdca_jack_exit, - .rtd_init = rt_sdca_jack_rtd_init, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -811,9 +811,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-aif2", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = sof_sdw_rt_amp_init, - .exit = sof_sdw_rt_amp_exit, - .rtd_init = rt712_spk_rtd_init, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt712_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -831,7 +831,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -845,9 +845,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt_sdca_jack_init, - .exit = sof_sdw_rt_sdca_jack_exit, - .rtd_init = rt_sdca_jack_rtd_init, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -865,7 +865,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt712-sdca-dmic-aif1", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -879,9 +879,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt1308-aif", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = sof_sdw_rt_amp_init, - .exit = sof_sdw_rt_amp_exit, - .rtd_init = rt_amp_spk_rtd_init, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -889,7 +889,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { }, }, .dai_num = 1, - .ops = &sof_sdw_rt1308_i2s_ops, + .ops = &soc_sdw_rt1308_i2s_ops, }, { .part_id = 0x1316, @@ -899,9 +899,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt1316-aif", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = sof_sdw_rt_amp_init, - .exit = sof_sdw_rt_amp_exit, - .rtd_init = rt_amp_spk_rtd_init, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -918,9 +918,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt1318-aif", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = sof_sdw_rt_amp_init, - .exit = sof_sdw_rt_amp_exit, - .rtd_init = rt_amp_spk_rtd_init, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -939,7 +939,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-sdca-aif2", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -954,7 +954,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-sdca-aif2", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -969,7 +969,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -984,7 +984,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt715-aif2", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 1, @@ -998,9 +998,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt722-sdca-aif1", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = sof_sdw_rt_sdca_jack_init, - .exit = sof_sdw_rt_sdca_jack_exit, - .rtd_init = rt_sdca_jack_rtd_init, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -1012,9 +1012,9 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_type = SOC_SDW_DAI_TYPE_AMP, /* No feedback capability is provided by rt722-sdca codec driver*/ .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = sof_sdw_rt_amp_init, - .exit = sof_sdw_rt_amp_exit, - .rtd_init = rt722_spk_rtd_init, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt722_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -1025,7 +1025,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt722-sdca-aif3", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = rt_dmic_rtd_init, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, }, }, .dai_num = 3, @@ -1038,8 +1038,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "max98373-aif1", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = sof_sdw_maxim_init, - .rtd_init = maxim_spk_rtd_init, + .init = asoc_sdw_maxim_init, + .rtd_init = asoc_sdw_maxim_spk_rtd_init, .controls = maxim_controls, .num_controls = ARRAY_SIZE(maxim_controls), .widgets = maxim_widgets, @@ -1056,8 +1056,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "max98363-aif1", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = sof_sdw_maxim_init, - .rtd_init = maxim_spk_rtd_init, + .init = asoc_sdw_maxim_init, + .rtd_init = asoc_sdw_maxim_spk_rtd_init, .controls = maxim_controls, .num_controls = ARRAY_SIZE(maxim_controls), .widgets = maxim_widgets, @@ -1074,7 +1074,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "rt5682-sdw", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = rt5682_rtd_init, + .rtd_init = asoc_sdw_rt5682_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -1091,8 +1091,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs35l56-sdw1", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = sof_sdw_cs_amp_init, - .rtd_init = cs_spk_rtd_init, + .init = asoc_sdw_cs_amp_init, + .rtd_init = asoc_sdw_cs_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -1109,7 +1109,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l42-sdw", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = cs42l42_rtd_init, + .rtd_init = asoc_sdw_cs42l42_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -1121,15 +1121,15 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x4243, .codec_name = "cs42l43-codec", - .count_sidecar = bridge_cs35l56_count_sidecar, - .add_sidecar = bridge_cs35l56_add_sidecar, + .count_sidecar = asoc_sdw_bridge_cs35l56_count_sidecar, + .add_sidecar = asoc_sdw_bridge_cs35l56_add_sidecar, .dais = { { .direction = {true, false}, .dai_name = "cs42l43-dp5", .dai_type = SOC_SDW_DAI_TYPE_JACK, .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .rtd_init = cs42l43_hs_rtd_init, + .rtd_init = asoc_sdw_cs42l43_hs_rtd_init, .controls = generic_jack_controls, .num_controls = ARRAY_SIZE(generic_jack_controls), .widgets = generic_jack_widgets, @@ -1140,7 +1140,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l43-dp1", .dai_type = SOC_SDW_DAI_TYPE_MIC, .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = cs42l43_dmic_rtd_init, + .rtd_init = asoc_sdw_cs42l43_dmic_rtd_init, .widgets = generic_dmic_widgets, .num_widgets = ARRAY_SIZE(generic_dmic_widgets), }, @@ -1155,8 +1155,8 @@ static struct asoc_sdw_codec_info codec_info_list[] = { .dai_name = "cs42l43-dp6", .dai_type = SOC_SDW_DAI_TYPE_AMP, .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = sof_sdw_cs42l43_spk_init, - .rtd_init = cs42l43_spk_rtd_init, + .init = asoc_sdw_cs42l43_spk_init, + .rtd_init = asoc_sdw_cs42l43_spk_rtd_init, .controls = generic_spk_controls, .num_controls = ARRAY_SIZE(generic_spk_controls), .widgets = generic_spk_widgets, @@ -1220,7 +1220,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { }, }; -static struct asoc_sdw_codec_info *find_codec_info_part(const u64 adr) +static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr) { unsigned int part_id, sdw_version; int i; @@ -1241,7 +1241,7 @@ static struct asoc_sdw_codec_info *find_codec_info_part(const u64 adr) } -static struct asoc_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) +static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id) { int i; @@ -1255,8 +1255,8 @@ static struct asoc_sdw_codec_info *find_codec_info_acpi(const u8 *acpi_id) return NULL; } -static struct asoc_sdw_codec_info *find_codec_info_dai(const char *dai_name, - int *dai_index) +static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, + int *dai_index) { int i, j; @@ -1320,12 +1320,12 @@ static int init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai return 0; } -static bool is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, - unsigned int sdw_version, - unsigned int mfg_id, - unsigned int part_id, - unsigned int class_id, - int index_in_link) +static bool asoc_sdw_is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, + unsigned int sdw_version, + unsigned int mfg_id, + unsigned int part_id, + unsigned int class_id, + int index_in_link) { int i; @@ -1354,10 +1354,10 @@ static bool is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, return true; } -static const char *get_codec_name(struct device *dev, - const struct asoc_sdw_codec_info *codec_info, - const struct snd_soc_acpi_link_adr *adr_link, - int adr_index) +static const char *asoc_sdw_get_codec_name(struct device *dev, + const struct asoc_sdw_codec_info *codec_info, + const struct snd_soc_acpi_link_adr *adr_link, + int adr_index) { u64 adr = adr_link->adr_d[adr_index].adr; unsigned int sdw_version = SDW_VERSION(adr); @@ -1369,8 +1369,8 @@ static const char *get_codec_name(struct device *dev, if (codec_info->codec_name) return devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); - else if (is_unique_device(adr_link, sdw_version, mfg_id, part_id, - class_id, adr_index)) + else if (asoc_sdw_is_unique_device(adr_link, sdw_version, mfg_id, part_id, + class_id, adr_index)) return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x", link_id, mfg_id, part_id, class_id); else @@ -1380,7 +1380,7 @@ static const char *get_codec_name(struct device *dev, return NULL; } -static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) +static int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct asoc_sdw_codec_info *codec_info; @@ -1390,7 +1390,7 @@ static int sof_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) int i; for_each_rtd_codec_dais(rtd, i, dai) { - codec_info = find_codec_info_dai(dai->name, &dai_index); + codec_info = asoc_sdw_find_codec_info_dai(dai->name, &dai_index); if (!codec_info) return -EINVAL; @@ -1538,13 +1538,13 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, return -EINVAL; } - codec_info = find_codec_info_part(adr_dev->adr); + codec_info = asoc_sdw_find_codec_info_part(adr_dev->adr); if (!codec_info) return -EINVAL; ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic; - codec_name = get_codec_name(dev, codec_info, adr_link, i); + codec_name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, i); if (!codec_name) return -ENOMEM; @@ -1736,7 +1736,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, init_dai_link(dev, *dai_links, be_id, name, playback, capture, cpus, num_cpus, codecs, num_codecs, - sof_sdw_rtd_init, &sdw_ops); + asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -1831,7 +1831,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic01", 0, 1, // DMIC only supports capture "DMIC01 Pin", "dmic-codec", "dmic-hifi", - sof_sdw_dmic_init, NULL); + asoc_sdw_dmic_init, NULL); if (ret) return ret; @@ -1840,7 +1840,7 @@ static int create_dmic_dailinks(struct snd_soc_card *card, ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic16k", 0, 1, // DMIC only supports capture "DMIC16k Pin", "dmic-codec", "dmic-hifi", - /* don't call sof_sdw_dmic_init() twice */ + /* don't call asoc_sdw_dmic_init() twice */ NULL, NULL); if (ret) return ret; @@ -1956,7 +1956,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) * system only when I2S mode is supported, not sdw mode. * Here check ACPI ID to confirm I2S is supported. */ - ssp_info = find_codec_info_acpi(mach->id); + ssp_info = asoc_sdw_find_codec_info_acpi(mach->id); if (ssp_info) { ssp_mask = SOF_SSP_GET_PORT(sof_sdw_quirk); ssp_num = hweight_long(ssp_mask); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index e88b5627560b..28db2f1c6dae 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -148,75 +148,75 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); /* DMIC support */ -int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); +int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); /* RT711 support */ -int sof_sdw_rt711_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); -int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +int asoc_sdw_rt711_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT711-SDCA support */ -int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, +int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + +/* RT1308 I2S support */ +extern const struct snd_soc_ops soc_sdw_rt1308_i2s_ops; + +/* generic amp support */ +int asoc_sdw_rt_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + +/* MAXIM codec support */ +int asoc_sdw_maxim_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + +/* CS42L43 support */ +int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct asoc_sdw_codec_info *info, bool playback); -int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - -/* RT1308 I2S support */ -extern const struct snd_soc_ops sof_sdw_rt1308_i2s_ops; - -/* generic amp support */ -int sof_sdw_rt_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); -int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - -/* MAXIM codec support */ -int sof_sdw_maxim_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - -/* CS42L43 support */ -int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); /* CS AMP support */ -int bridge_cs35l56_count_sidecar(struct snd_soc_card *card, - int *num_dais, int *num_devs); -int bridge_cs35l56_add_sidecar(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, - struct snd_soc_codec_conf **codec_conf); -int bridge_cs35l56_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); +int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card, + int *num_dais, int *num_devs); +int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf); +int asoc_sdw_bridge_cs35l56_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); -int sof_sdw_cs_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); +int asoc_sdw_cs_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); /* dai_link init callbacks */ -int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index fc18e4aa3dbe..d28477c50469 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -36,7 +36,7 @@ static struct snd_soc_jack_pin cs42l42_jack_pins[] = { }, }; -int cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index 1688f29a977a..bb371b2649cf 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -48,7 +48,7 @@ static struct snd_soc_jack_pin sof_jack_pins[] = { }, }; -int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); @@ -99,7 +99,7 @@ int cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai return ret; } -int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; @@ -121,10 +121,10 @@ int cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *da return ret; } -int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { /* Do init on playback link only. */ if (!playback) @@ -132,10 +132,10 @@ int sof_sdw_cs42l43_spk_init(struct snd_soc_card *card, info->amp_num++; - return bridge_cs35l56_spk_init(card, dai_links, info, playback); + return asoc_sdw_bridge_cs35l56_spk_init(card, dai_links, info, playback); } -int cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/intel/boards/sof_sdw_cs_amp.c index 206fe7610a56..6479974bd2c3 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/intel/boards/sof_sdw_cs_amp.c @@ -14,7 +14,7 @@ #define CODEC_NAME_SIZE 8 -int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { const char *dai_name = rtd->dai_link->codecs->dai_name; struct snd_soc_card *card = rtd->card; @@ -45,10 +45,10 @@ int cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) return 0; } -int sof_sdw_cs_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_cs_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { /* Do init on playback link only. */ if (!playback) diff --git a/sound/soc/intel/boards/sof_sdw_dmic.c b/sound/soc/intel/boards/sof_sdw_dmic.c index 19df0f7a1d85..d9f2e072f401 100644 --- a/sound/soc/intel/boards/sof_sdw_dmic.c +++ b/sound/soc/intel/boards/sof_sdw_dmic.c @@ -19,7 +19,7 @@ static const struct snd_soc_dapm_route dmic_map[] = { {"DMic", NULL, "SoC DMIC"}, }; -int sof_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd) +int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index f689dc29c493..fd8347e28849 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -21,7 +21,7 @@ static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { { "Right Spk", NULL, "Right BE_OUT" }, }; -int maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; @@ -75,7 +75,7 @@ static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enabl return 0; } -static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) +static int asoc_sdw_mx8373_prepare(struct snd_pcm_substream *substream) { int ret; @@ -87,7 +87,7 @@ static int mx8373_sdw_prepare(struct snd_pcm_substream *substream) return mx8373_enable_spk_pin(substream, true); } -static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) +static int asoc_sdw_mx8373_hw_free(struct snd_pcm_substream *substream) { int ret; @@ -101,14 +101,14 @@ static int mx8373_sdw_hw_free(struct snd_pcm_substream *substream) static const struct snd_soc_ops max_98373_sdw_ops = { .startup = asoc_sdw_startup, - .prepare = mx8373_sdw_prepare, + .prepare = asoc_sdw_mx8373_prepare, .trigger = asoc_sdw_trigger, .hw_params = asoc_sdw_hw_params, - .hw_free = mx8373_sdw_hw_free, + .hw_free = asoc_sdw_mx8373_hw_free, .shutdown = asoc_sdw_shutdown, }; -static int mx8373_sdw_late_probe(struct snd_soc_card *card) +static int asoc_sdw_mx8373_sdw_late_probe(struct snd_soc_card *card) { struct snd_soc_dapm_context *dapm = &card->dapm; @@ -118,10 +118,10 @@ static int mx8373_sdw_late_probe(struct snd_soc_card *card) return snd_soc_dapm_sync(dapm); } -int sof_sdw_maxim_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_maxim_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { info->amp_num++; @@ -133,7 +133,7 @@ int sof_sdw_maxim_init(struct snd_soc_card *card, */ break; case SOF_SDW_PART_ID_MAX98373: - info->codec_card_late_probe = mx8373_sdw_late_probe; + info->codec_card_late_probe = asoc_sdw_mx8373_sdw_late_probe; dai_links->ops = &max_98373_sdw_ops; break; default: diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 67737815d016..3584638e2192 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -35,7 +35,7 @@ static struct snd_soc_jack_pin rt5682_jack_pins[] = { }, }; -int rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index 0db730071be2..a90d69573736 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -33,7 +33,7 @@ static struct snd_soc_jack_pin rt700_jack_pins[] = { }, }; -int rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index fff76291fca2..e4d300d7d5ef 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -59,7 +59,7 @@ static struct snd_soc_jack_pin rt711_jack_pins[] = { }, }; -int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -111,7 +111,7 @@ int rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) return ret; } -int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -124,10 +124,10 @@ int sof_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_l return 0; } -int sof_sdw_rt711_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_rt711_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev; diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c index 788796461885..bb09d1ddafd2 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt712_sdca.c @@ -26,7 +26,7 @@ static const struct snd_soc_dapm_route rt712_spk_map[] = { { "Speaker", NULL, "rt712 SPOR" }, }; -int rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c index 083d281bd052..2da9134ad1a3 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt722_sdca.c @@ -19,7 +19,7 @@ static const struct snd_soc_dapm_route rt722_spk_map[] = { { "Speaker", NULL, "rt722 SPK" }, }; -int rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index da1a8cc1c63d..4cf392c84cc7 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -173,7 +173,7 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_ return rt1318_map; } -int rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; const struct snd_soc_dapm_route *rt_amp_map; @@ -233,11 +233,11 @@ static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, } /* machine stream operations */ -const struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { +const struct snd_soc_ops soc_sdw_rt1308_i2s_ops = { .hw_params = rt1308_i2s_hw_params, }; -int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -254,10 +254,10 @@ int sof_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_ return 0; } -int sof_sdw_rt_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_rt_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev1, *sdw_dev2; diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/intel/boards/sof_sdw_rt_dmic.c index ea7c1a4bc566..64960b059834 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c +++ b/sound/soc/intel/boards/sof_sdw_rt_dmic.c @@ -12,7 +12,7 @@ #include "sof_board_helpers.h" #include "sof_sdw_common.h" -int rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct snd_soc_component *component; diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index 9c3b93f808a2..d84aec2b4c78 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -83,7 +83,7 @@ static const char * const need_sdca_suffix[] = { "rt711", "rt713" }; -int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) +int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -161,7 +161,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *d return ret; } -int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) +int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -178,10 +178,10 @@ int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link return 0; } -int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback) +int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback) { struct mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev; From a2b5ec0ca5fcd6f609733b37c52e23ddb7328ffd Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:20 +0530 Subject: [PATCH 045/410] ASoC: intel: rename maxim codec macros Rename maxim codec part id macros. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-6-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw_maxim.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/intel/boards/sof_sdw_maxim.c index fd8347e28849..9933224fcf68 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/intel/boards/sof_sdw_maxim.c @@ -13,8 +13,8 @@ #include "sof_sdw_common.h" static int maxim_part_id; -#define SOF_SDW_PART_ID_MAX98363 0x8363 -#define SOF_SDW_PART_ID_MAX98373 0x8373 +#define SOC_SDW_PART_ID_MAX98363 0x8363 +#define SOC_SDW_PART_ID_MAX98373 0x8373 static const struct snd_soc_dapm_route max_98373_dapm_routes[] = { { "Left Spk", NULL, "Left BE_OUT" }, @@ -127,12 +127,12 @@ int asoc_sdw_maxim_init(struct snd_soc_card *card, maxim_part_id = info->part_id; switch (maxim_part_id) { - case SOF_SDW_PART_ID_MAX98363: + case SOC_SDW_PART_ID_MAX98363: /* Default ops are set in function init_dai_link. * called as part of function create_sdw_dailink */ break; - case SOF_SDW_PART_ID_MAX98373: + case SOC_SDW_PART_ID_MAX98373: info->codec_card_late_probe = asoc_sdw_mx8373_sdw_late_probe; dai_links->ops = &max_98373_sdw_ops; break; From b1f7cbf0d5746ba270bc090f6f85a4b176410854 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:21 +0530 Subject: [PATCH 046/410] ASoC: intel: rename ignore_pch_dmic variable name Rename 'ignore_pch_dmic' variable name as 'ignore_internal_dmic'. This variable will be moved to common header file and will be used by other platform machine driver code. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-7-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 12 ++++++------ sound/soc/intel/boards/sof_sdw_common.h | 4 ++-- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 64418976aba4..28021d33fd2d 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -932,7 +932,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x714, .version_id = 3, - .ignore_pch_dmic = true, + .ignore_internal_dmic = true, .dais = { { .direction = {false, true}, @@ -947,7 +947,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x715, .version_id = 3, - .ignore_pch_dmic = true, + .ignore_internal_dmic = true, .dais = { { .direction = {false, true}, @@ -962,7 +962,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x714, .version_id = 2, - .ignore_pch_dmic = true, + .ignore_internal_dmic = true, .dais = { { .direction = {false, true}, @@ -977,7 +977,7 @@ static struct asoc_sdw_codec_info codec_info_list[] = { { .part_id = 0x715, .version_id = 2, - .ignore_pch_dmic = true, + .ignore_internal_dmic = true, .dais = { { .direction = {false, true}, @@ -1542,7 +1542,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, if (!codec_info) return -EINVAL; - ctx->ignore_pch_dmic |= codec_info->ignore_pch_dmic; + ctx->ignore_internal_dmic |= codec_info->ignore_internal_dmic; codec_name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, i); if (!codec_name) @@ -2018,7 +2018,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) /* dmic */ if (dmic_num > 0) { - if (ctx->ignore_pch_dmic) { + if (ctx->ignore_internal_dmic) { dev_warn(dev, "Ignoring PCH DMIC\n"); } else { ret = create_dmic_dailinks(card, &dai_links, &be_id); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 28db2f1c6dae..c1b58180efe5 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -106,7 +106,7 @@ struct asoc_sdw_codec_info { const char *codec_name; int amp_num; const u8 acpi_id[ACPI_ID_LEN]; - const bool ignore_pch_dmic; + const bool ignore_internal_dmic; const struct snd_soc_ops *ops; struct asoc_sdw_dai_info dais[SOC_SDW_MAX_DAI_NUM]; const int dai_num; @@ -129,7 +129,7 @@ struct mc_private { /* To store SDW Pin index for each SoundWire link */ unsigned int sdw_pin_index[SDW_MAX_LINKS]; bool append_dai_type; - bool ignore_pch_dmic; + bool ignore_internal_dmic; }; extern unsigned long sof_sdw_quirk; From d39388e6555cb805beb985a7ece1c771c611037c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:22 +0530 Subject: [PATCH 047/410] ASoC: intel/sdw-utils: move soundwire machine driver soc ops Move Intel SoundWire generic machine driver soc ops to common place so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-8-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 23 ++++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_sdw.c | 131 +-------------------- sound/soc/intel/boards/sof_sdw_common.h | 9 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 3 + sound/soc/sdw_utils/soc_sdw_utils.c | 150 ++++++++++++++++++++++++ 9 files changed, 188 insertions(+), 138 deletions(-) create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h new file mode 100644 index 000000000000..cf4cdb66b2de --- /dev/null +++ b/include/sound/soc_sdw_utils.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * This file incorporates work covered by the following copyright notice: + * Copyright (c) 2020 Intel Corporation + * Copyright(c) 2024 Advanced Micro Devices, Inc. + * + */ + +#ifndef SOC_SDW_UTILS_H +#define SOC_SDW_UTILS_H + +#include + +int asoc_sdw_startup(struct snd_pcm_substream *substream); +int asoc_sdw_prepare(struct snd_pcm_substream *substream); +int asoc_sdw_prepare(struct snd_pcm_substream *substream); +int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd); +int asoc_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params); +int asoc_sdw_hw_free(struct snd_pcm_substream *substream); +void asoc_sdw_shutdown(struct snd_pcm_substream *substream); + +#endif diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index a52afb423b46..e87bd15a8b43 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -126,6 +126,8 @@ source "sound/soc/xtensa/Kconfig" # Supported codecs source "sound/soc/codecs/Kconfig" +source "sound/soc/sdw_utils/Kconfig" + # generic frame-work source "sound/soc/generic/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index fd61847dd1eb..775bb38c2ed4 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -75,3 +75,4 @@ obj-$(CONFIG_SND_SOC) += uniphier/ obj-$(CONFIG_SND_SOC) += ux500/ obj-$(CONFIG_SND_SOC) += xilinx/ obj-$(CONFIG_SND_SOC) += xtensa/ +obj-$(CONFIG_SND_SOC) += sdw_utils/ diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f1faa5dd2a4f..3d59d6982763 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -656,6 +656,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE + select SND_SOC_SDW_UTILS select SND_SOC_MAX98363 select SND_SOC_MAX98373_I2C select SND_SOC_MAX98373_SDW diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 28021d33fd2d..fc73db4af186 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -11,7 +11,6 @@ #include #include #include -#include #include #include "sof_sdw_common.h" #include "../../codecs/rt711.h" @@ -593,135 +592,6 @@ static const struct snd_kcontrol_new rt700_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; -/* these wrappers are only needed to avoid typecast compilation errors */ -int asoc_sdw_startup(struct snd_pcm_substream *substream) -{ - return sdw_startup_stream(substream); -} - -int asoc_sdw_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct sdw_stream_runtime *sdw_stream; - struct snd_soc_dai *dai; - - /* Find stream from first CPU DAI */ - dai = snd_soc_rtd_to_cpu(rtd, 0); - - sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); - if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); - return PTR_ERR(sdw_stream); - } - - return sdw_prepare_stream(sdw_stream); -} - -int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct sdw_stream_runtime *sdw_stream; - struct snd_soc_dai *dai; - int ret; - - /* Find stream from first CPU DAI */ - dai = snd_soc_rtd_to_cpu(rtd, 0); - - sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); - if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); - return PTR_ERR(sdw_stream); - } - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - ret = sdw_enable_stream(sdw_stream); - break; - - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - ret = sdw_disable_stream(sdw_stream); - break; - default: - ret = -EINVAL; - break; - } - - if (ret) - dev_err(rtd->dev, "%s trigger %d failed: %d\n", __func__, cmd, ret); - - return ret; -} - -int asoc_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai_link_ch_map *ch_maps; - int ch = params_channels(params); - unsigned int ch_mask; - int num_codecs; - int step; - int i; - - if (!rtd->dai_link->ch_maps) - return 0; - - /* Identical data will be sent to all codecs in playback */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ch_mask = GENMASK(ch - 1, 0); - step = 0; - } else { - num_codecs = rtd->dai_link->num_codecs; - - if (ch < num_codecs || ch % num_codecs != 0) { - dev_err(rtd->dev, "Channels number %d is invalid when codec number = %d\n", - ch, num_codecs); - return -EINVAL; - } - - ch_mask = GENMASK(ch / num_codecs - 1, 0); - step = hweight_long(ch_mask); - - } - - /* - * The captured data will be combined from each cpu DAI if the dai - * link has more than one codec DAIs. Set codec channel mask and - * ASoC will set the corresponding channel numbers for each cpu dai. - */ - for_each_link_ch_maps(rtd->dai_link, i, ch_maps) - ch_maps->ch_mask = ch_mask << (i * step); - - return 0; -} - -int asoc_sdw_hw_free(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct sdw_stream_runtime *sdw_stream; - struct snd_soc_dai *dai; - - /* Find stream from first CPU DAI */ - dai = snd_soc_rtd_to_cpu(rtd, 0); - - sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); - if (IS_ERR(sdw_stream)) { - dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); - return PTR_ERR(sdw_stream); - } - - return sdw_deprepare_stream(sdw_stream); -} - -void asoc_sdw_shutdown(struct snd_pcm_substream *substream) -{ - sdw_shutdown_stream(substream); -} - static const struct snd_soc_ops sdw_ops = { .startup = asoc_sdw_startup, .prepare = asoc_sdw_prepare, @@ -2232,3 +2102,4 @@ MODULE_AUTHOR("Rander Wang "); MODULE_AUTHOR("Pierre-Louis Bossart "); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index c1b58180efe5..d97aedeef9e8 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -12,6 +12,7 @@ #include #include #include +#include #include "sof_hdmi_common.h" #define SOC_SDW_MAX_NO_PROPS 2 @@ -134,14 +135,6 @@ struct mc_private { extern unsigned long sof_sdw_quirk; -int asoc_sdw_startup(struct snd_pcm_substream *substream); -int asoc_sdw_prepare(struct snd_pcm_substream *substream); -int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd); -int asoc_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params); -int asoc_sdw_hw_free(struct snd_pcm_substream *substream); -void asoc_sdw_shutdown(struct snd_pcm_substream *substream); - /* generic HDMI support */ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/sdw_utils/Kconfig b/sound/soc/sdw_utils/Kconfig new file mode 100644 index 000000000000..d915083c3889 --- /dev/null +++ b/sound/soc/sdw_utils/Kconfig @@ -0,0 +1,6 @@ +# SPDX-License-Identifier: GPL-2.0-only +config SND_SOC_SDW_UTILS + tristate + help + This option enables to use SoundWire common helper functions and + SoundWire codec helper functions in machine driver. diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile new file mode 100644 index 000000000000..29b2852be287 --- /dev/null +++ b/sound/soc/sdw_utils/Makefile @@ -0,0 +1,3 @@ +# SPDX-License-Identifier: GPL-2.0-only +snd-soc-sdw-utils-y := soc_sdw_utils.o +obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c new file mode 100644 index 000000000000..cccab173fd17 --- /dev/null +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -0,0 +1,150 @@ +// SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: +// Copyright (c) 2020 Intel Corporation +// Copyright(c) 2024 Advanced Micro Devices, Inc. +/* + * soc-sdw-utils.c - common SoundWire machine driver helper functions + */ + +#include +#include +#include +#include +#include + +/* these wrappers are only needed to avoid typecast compilation errors */ +int asoc_sdw_startup(struct snd_pcm_substream *substream) +{ + return sdw_startup_stream(substream); +} +EXPORT_SYMBOL_NS(asoc_sdw_startup, SND_SOC_SDW_UTILS); + +int asoc_sdw_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sdw_stream_runtime *sdw_stream; + struct snd_soc_dai *dai; + + /* Find stream from first CPU DAI */ + dai = snd_soc_rtd_to_cpu(rtd, 0); + + sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); + if (IS_ERR(sdw_stream)) { + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); + return PTR_ERR(sdw_stream); + } + + return sdw_prepare_stream(sdw_stream); +} +EXPORT_SYMBOL_NS(asoc_sdw_prepare, SND_SOC_SDW_UTILS); + +int asoc_sdw_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sdw_stream_runtime *sdw_stream; + struct snd_soc_dai *dai; + int ret; + + /* Find stream from first CPU DAI */ + dai = snd_soc_rtd_to_cpu(rtd, 0); + + sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); + if (IS_ERR(sdw_stream)) { + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); + return PTR_ERR(sdw_stream); + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + ret = sdw_enable_stream(sdw_stream); + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + ret = sdw_disable_stream(sdw_stream); + break; + default: + ret = -EINVAL; + break; + } + + if (ret) + dev_err(rtd->dev, "%s trigger %d failed: %d\n", __func__, cmd, ret); + + return ret; +} +EXPORT_SYMBOL_NS(asoc_sdw_trigger, SND_SOC_SDW_UTILS); + +int asoc_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct snd_soc_dai_link_ch_map *ch_maps; + int ch = params_channels(params); + unsigned int ch_mask; + int num_codecs; + int step; + int i; + + if (!rtd->dai_link->ch_maps) + return 0; + + /* Identical data will be sent to all codecs in playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ch_mask = GENMASK(ch - 1, 0); + step = 0; + } else { + num_codecs = rtd->dai_link->num_codecs; + + if (ch < num_codecs || ch % num_codecs != 0) { + dev_err(rtd->dev, "Channels number %d is invalid when codec number = %d\n", + ch, num_codecs); + return -EINVAL; + } + + ch_mask = GENMASK(ch / num_codecs - 1, 0); + step = hweight_long(ch_mask); + } + + /* + * The captured data will be combined from each cpu DAI if the dai + * link has more than one codec DAIs. Set codec channel mask and + * ASoC will set the corresponding channel numbers for each cpu dai. + */ + for_each_link_ch_maps(rtd->dai_link, i, ch_maps) + ch_maps->ch_mask = ch_mask << (i * step); + + return 0; +} +EXPORT_SYMBOL_NS(asoc_sdw_hw_params, SND_SOC_SDW_UTILS); + +int asoc_sdw_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct sdw_stream_runtime *sdw_stream; + struct snd_soc_dai *dai; + + /* Find stream from first CPU DAI */ + dai = snd_soc_rtd_to_cpu(rtd, 0); + + sdw_stream = snd_soc_dai_get_stream(dai, substream->stream); + if (IS_ERR(sdw_stream)) { + dev_err(rtd->dev, "no stream found for DAI %s\n", dai->name); + return PTR_ERR(sdw_stream); + } + + return sdw_deprepare_stream(sdw_stream); +} +EXPORT_SYMBOL_NS(asoc_sdw_hw_free, SND_SOC_SDW_UTILS); + +void asoc_sdw_shutdown(struct snd_pcm_substream *substream) +{ + sdw_shutdown_stream(substream); +} +EXPORT_SYMBOL_NS(asoc_sdw_shutdown, SND_SOC_SDW_UTILS); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 73619137c633a89a90f8c8c5fa59171aaff6e913 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:23 +0530 Subject: [PATCH 048/410] ASoC: intel: move soundwire machine driver common structures Move intel generic SoundWire machine driver common structures to soc_sdw_utils.h file. These structures will be used in other platform SoundWire machine driver code. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-9-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 43 +++++++++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 43 ------------------------- 2 files changed, 43 insertions(+), 43 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index cf4cdb66b2de..1ae5523bbcf8 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -11,6 +11,49 @@ #include +#define SOC_SDW_MAX_DAI_NUM 8 + +struct asoc_sdw_codec_info; + +struct asoc_sdw_dai_info { + const bool direction[2]; /* playback & capture support */ + const char *dai_name; + const int dai_type; + const int dailink[2]; /* dailink id for each direction */ + const struct snd_kcontrol_new *controls; + const int num_controls; + const struct snd_soc_dapm_widget *widgets; + const int num_widgets; + int (*init)(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + int (*rtd_init)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); + bool rtd_init_done; /* Indicate that the rtd_init callback is done */ + unsigned long quirk; +}; + +struct asoc_sdw_codec_info { + const int part_id; + const int version_id; + const char *codec_name; + int amp_num; + const u8 acpi_id[ACPI_ID_LEN]; + const bool ignore_internal_dmic; + const struct snd_soc_ops *ops; + struct asoc_sdw_dai_info dais[SOC_SDW_MAX_DAI_NUM]; + const int dai_num; + + int (*codec_card_late_probe)(struct snd_soc_card *card); + + int (*count_sidecar)(struct snd_soc_card *card, + int *num_dais, int *num_devs); + int (*add_sidecar)(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf); +}; + int asoc_sdw_startup(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index d97aedeef9e8..688cbc3afb29 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -78,49 +78,6 @@ enum { #define SOC_SDW_DAI_TYPE_AMP 1 #define SOC_SDW_DAI_TYPE_MIC 2 -#define SOC_SDW_MAX_DAI_NUM 8 - -struct asoc_sdw_codec_info; - -struct asoc_sdw_dai_info { - const bool direction[2]; /* playback & capture support */ - const char *dai_name; - const int dai_type; - const int dailink[2]; /* dailink id for each direction */ - const struct snd_kcontrol_new *controls; - const int num_controls; - const struct snd_soc_dapm_widget *widgets; - const int num_widgets; - int (*init)(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - int (*exit)(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - int (*rtd_init)(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); - bool rtd_init_done; /* Indicate that the rtd_init callback is done */ - unsigned long quirk; -}; - -struct asoc_sdw_codec_info { - const int part_id; - const int version_id; - const char *codec_name; - int amp_num; - const u8 acpi_id[ACPI_ID_LEN]; - const bool ignore_internal_dmic; - const struct snd_soc_ops *ops; - struct asoc_sdw_dai_info dais[SOC_SDW_MAX_DAI_NUM]; - const int dai_num; - - int (*codec_card_late_probe)(struct snd_soc_card *card); - - int (*count_sidecar)(struct snd_soc_card *card, - int *num_dais, int *num_devs); - int (*add_sidecar)(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, - struct snd_soc_codec_conf **codec_conf); -}; - struct mc_private { struct snd_soc_card card; struct snd_soc_jack sdw_headset; From 941d6933eb31b13d09a7293bc92d9d494513a372 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:24 +0530 Subject: [PATCH 049/410] ASoC: intel/sdw_utils: move soundwire machine driver helper functions Move below Intel SoundWire machine driver helper functions to soc_sdw_utils.c file so that it can be used by other platform machine driver. - asoc_sdw_is_unique_device() - asoc_sdw_get_codec_name() Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-10-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 5 +++ sound/soc/intel/boards/sof_sdw.c | 60 ---------------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 61 +++++++++++++++++++++++++++++ 3 files changed, 66 insertions(+), 60 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 1ae5523bbcf8..7ca3a6afdfb1 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -10,6 +10,7 @@ #define SOC_SDW_UTILS_H #include +#include #define SOC_SDW_MAX_DAI_NUM 8 @@ -63,4 +64,8 @@ int asoc_sdw_hw_params(struct snd_pcm_substream *substream, int asoc_sdw_hw_free(struct snd_pcm_substream *substream); void asoc_sdw_shutdown(struct snd_pcm_substream *substream); +const char *asoc_sdw_get_codec_name(struct device *dev, + const struct asoc_sdw_codec_info *codec_info, + const struct snd_soc_acpi_link_adr *adr_link, + int adr_index); #endif diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index fc73db4af186..e1d2b744987f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1190,66 +1190,6 @@ static int init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai return 0; } -static bool asoc_sdw_is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, - unsigned int sdw_version, - unsigned int mfg_id, - unsigned int part_id, - unsigned int class_id, - int index_in_link) -{ - int i; - - for (i = 0; i < adr_link->num_adr; i++) { - unsigned int sdw1_version, mfg1_id, part1_id, class1_id; - u64 adr; - - /* skip itself */ - if (i == index_in_link) - continue; - - adr = adr_link->adr_d[i].adr; - - sdw1_version = SDW_VERSION(adr); - mfg1_id = SDW_MFG_ID(adr); - part1_id = SDW_PART_ID(adr); - class1_id = SDW_CLASS_ID(adr); - - if (sdw_version == sdw1_version && - mfg_id == mfg1_id && - part_id == part1_id && - class_id == class1_id) - return false; - } - - return true; -} - -static const char *asoc_sdw_get_codec_name(struct device *dev, - const struct asoc_sdw_codec_info *codec_info, - const struct snd_soc_acpi_link_adr *adr_link, - int adr_index) -{ - u64 adr = adr_link->adr_d[adr_index].adr; - unsigned int sdw_version = SDW_VERSION(adr); - unsigned int link_id = SDW_DISCO_LINK_ID(adr); - unsigned int unique_id = SDW_UNIQUE_ID(adr); - unsigned int mfg_id = SDW_MFG_ID(adr); - unsigned int part_id = SDW_PART_ID(adr); - unsigned int class_id = SDW_CLASS_ID(adr); - - if (codec_info->codec_name) - return devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); - else if (asoc_sdw_is_unique_device(adr_link, sdw_version, mfg_id, part_id, - class_id, adr_index)) - return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x", - link_id, mfg_id, part_id, class_id); - else - return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x:%01x", - link_id, mfg_id, part_id, class_id, unique_id); - - return NULL; -} - static int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index cccab173fd17..2b59ddc61078 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -146,5 +146,66 @@ void asoc_sdw_shutdown(struct snd_pcm_substream *substream) } EXPORT_SYMBOL_NS(asoc_sdw_shutdown, SND_SOC_SDW_UTILS); +static bool asoc_sdw_is_unique_device(const struct snd_soc_acpi_link_adr *adr_link, + unsigned int sdw_version, + unsigned int mfg_id, + unsigned int part_id, + unsigned int class_id, + int index_in_link) +{ + int i; + + for (i = 0; i < adr_link->num_adr; i++) { + unsigned int sdw1_version, mfg1_id, part1_id, class1_id; + u64 adr; + + /* skip itself */ + if (i == index_in_link) + continue; + + adr = adr_link->adr_d[i].adr; + + sdw1_version = SDW_VERSION(adr); + mfg1_id = SDW_MFG_ID(adr); + part1_id = SDW_PART_ID(adr); + class1_id = SDW_CLASS_ID(adr); + + if (sdw_version == sdw1_version && + mfg_id == mfg1_id && + part_id == part1_id && + class_id == class1_id) + return false; + } + + return true; +} + +const char *asoc_sdw_get_codec_name(struct device *dev, + const struct asoc_sdw_codec_info *codec_info, + const struct snd_soc_acpi_link_adr *adr_link, + int adr_index) +{ + u64 adr = adr_link->adr_d[adr_index].adr; + unsigned int sdw_version = SDW_VERSION(adr); + unsigned int link_id = SDW_DISCO_LINK_ID(adr); + unsigned int unique_id = SDW_UNIQUE_ID(adr); + unsigned int mfg_id = SDW_MFG_ID(adr); + unsigned int part_id = SDW_PART_ID(adr); + unsigned int class_id = SDW_CLASS_ID(adr); + + if (codec_info->codec_name) + return devm_kstrdup(dev, codec_info->codec_name, GFP_KERNEL); + else if (asoc_sdw_is_unique_device(adr_link, sdw_version, mfg_id, part_id, + class_id, adr_index)) + return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x", + link_id, mfg_id, part_id, class_id); + else + return devm_kasprintf(dev, GFP_KERNEL, "sdw:0:%01x:%04x:%04x:%02x:%01x", + link_id, mfg_id, part_id, class_id, unique_id); + + return NULL; +} +EXPORT_SYMBOL_NS(asoc_sdw_get_codec_name, SND_SOC_SDW_UTILS); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 4776d0c9088634677749e42ae8b36b4da46226db Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:25 +0530 Subject: [PATCH 050/410] ASoC: intel/sdw_utils: move dmic codec helper function Move generic dmic codec helper function implementation to sdw_utils folder so that this function can be used by other platform machine drivers. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-11-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 4 ++++ sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw_common.h | 3 --- sound/soc/sdw_utils/Makefile | 2 +- .../boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} | 8 +++++--- 5 files changed, 10 insertions(+), 8 deletions(-) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (76%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 7ca3a6afdfb1..0ffbd9847532 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -68,4 +68,8 @@ const char *asoc_sdw_get_codec_name(struct device *dev, const struct asoc_sdw_codec_info *codec_info, const struct snd_soc_acpi_link_adr *adr_link, int adr_index); + +/* DMIC support */ +int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); + #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index dc6fe110f279..8ac6f7b5fbee 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -43,7 +43,6 @@ snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_rt_dmic.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ - sof_sdw_dmic.o \ sof_sdw_hdmi.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 688cbc3afb29..81b654407651 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -97,9 +97,6 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); -/* DMIC support */ -int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); - /* RT711 support */ int asoc_sdw_rt711_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 29b2852be287..de8aff8744d8 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sdw-utils-y := soc_sdw_utils.o +snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_dmic.c b/sound/soc/sdw_utils/soc_sdw_dmic.c similarity index 76% rename from sound/soc/intel/boards/sof_sdw_dmic.c rename to sound/soc/sdw_utils/soc_sdw_dmic.c index d9f2e072f401..fc2aae985084 100644 --- a/sound/soc/intel/boards/sof_sdw_dmic.c +++ b/sound/soc/sdw_utils/soc_sdw_dmic.c @@ -1,14 +1,16 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_dmic - Helpers to handle dmic from generic machine driver + * soc_sdw_dmic - Helpers to handle dmic from generic machine driver */ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_widget dmic_widgets[] = { SND_SOC_DAPM_MIC("SoC DMIC", NULL), @@ -40,4 +42,4 @@ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd) return ret; } - +EXPORT_SYMBOL_NS(asoc_sdw_dmic_init, SND_SOC_SDW_UTILS); From a9831fd1c0e6353366bbde6e1dd7b15a2dd6d825 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:26 +0530 Subject: [PATCH 051/410] ASoC: intel/sdw_utils: move rtk dmic helper functions Move rtk SoundWire dmic helper functions implementation to sdw_utils folder to make it generic. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-12-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 3 +++ sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/sdw_utils/Makefile | 2 +- .../sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} | 9 +++++---- 5 files changed, 9 insertions(+), 7 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (77%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 0ffbd9847532..9fa102fc03c3 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -72,4 +72,7 @@ const char *asoc_sdw_get_codec_name(struct device *dev, /* DMIC support */ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); +/* dai_link init callbacks */ +int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); + #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 8ac6f7b5fbee..dca8eecfa820 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -40,7 +40,6 @@ snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \ - sof_sdw_rt_dmic.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 81b654407651..73227ebf8e7b 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -162,7 +162,6 @@ int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index de8aff8744d8..2c8f70465a12 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,3 +1,3 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o +snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt_dmic.c b/sound/soc/sdw_utils/soc_sdw_rt_dmic.c similarity index 77% rename from sound/soc/intel/boards/sof_sdw_rt_dmic.c rename to sound/soc/sdw_utils/soc_sdw_rt_dmic.c index 64960b059834..7f24806d809d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_dmic.c +++ b/sound/soc/sdw_utils/soc_sdw_rt_dmic.c @@ -1,16 +1,17 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2024 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt_dmic - Helpers to handle Realtek SDW DMIC from generic machine driver + * soc_sdw_rt_dmic - Helpers to handle Realtek SDW DMIC from generic machine driver */ #include #include #include #include -#include "sof_board_helpers.h" -#include "sof_sdw_common.h" +#include int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { @@ -39,4 +40,4 @@ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_da return 0; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_rt_dmic_rtd_init, SND_SOC_SDW_UTILS); From 09c60bc9da91f188e2aedb0bac94d3e129fa20f9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:27 +0530 Subject: [PATCH 052/410] ASoC: intel/sdw_utils: move rt712 sdca helper functions Move RT712 SDCA codec helper file to sdw_utils folder so that these helper functions can be used by other platform machine drivers. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-13-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/sdw_utils/Makefile | 3 ++- .../soc_sdw_rt712_sdca.c} | 8 +++++--- 5 files changed, 9 insertions(+), 6 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (80%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 9fa102fc03c3..6fd305253e2a 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -74,5 +74,6 @@ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index dca8eecfa820..3bc9d25fc9bb 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -39,7 +39,7 @@ snd-soc-sof-sdw-y += sof_sdw.o \ bridge_cs35l56.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ - sof_sdw_rt712_sdca.o sof_sdw_rt722_sdca.o \ + sof_sdw_rt722_sdca.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 73227ebf8e7b..b190aae1e093 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -160,7 +160,6 @@ int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 2c8f70465a12..f9a2baa49617 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,3 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only -snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o +snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ + soc_sdw_rt712_sdca.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c b/sound/soc/sdw_utils/soc_sdw_rt712_sdca.c similarity index 80% rename from sound/soc/intel/boards/sof_sdw_rt712_sdca.c rename to sound/soc/sdw_utils/soc_sdw_rt712_sdca.c index bb09d1ddafd2..5127210b9a03 100644 --- a/sound/soc/intel/boards/sof_sdw_rt712_sdca.c +++ b/sound/soc/sdw_utils/soc_sdw_rt712_sdca.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2023 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt712_sdca - Helpers to handle RT712-SDCA from generic machine driver + * soc_sdw_rt712_sdca - Helpers to handle RT712-SDCA from generic machine driver */ #include @@ -13,7 +15,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include /* * dapm routes for rt712 spk will be registered dynamically according @@ -43,4 +45,4 @@ int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ return ret; } - +EXPORT_SYMBOL_NS(asoc_sdw_rt712_spk_rtd_init, SND_SOC_SDW_UTILS); From 89b3456e9afa4c61879f6d744f6d2c6fadc2b2fc Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:28 +0530 Subject: [PATCH 053/410] ASoC: intel/sdw_utils: move rt722 sdca helper functions Move RT722 SDCA codec helper file to sdw_utils folder to make it generic. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-14-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 1 + sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/sdw_utils/Makefile | 2 +- .../soc_sdw_rt722_sdca.c} | 8 +++++--- 5 files changed, 7 insertions(+), 6 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (75%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 6fd305253e2a..5bc2a89bced4 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -75,5 +75,6 @@ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3bc9d25fc9bb..f3baf9ecfbb7 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -39,7 +39,6 @@ snd-soc-sof-sdw-y += sof_sdw.o \ bridge_cs35l56.o \ sof_sdw_rt5682.o sof_sdw_rt700.o \ sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ - sof_sdw_rt722_sdca.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index b190aae1e093..c7672dc1320c 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -160,7 +160,6 @@ int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index f9a2baa49617..261c60098e88 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,4 +1,4 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ - soc_sdw_rt712_sdca.o + soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c b/sound/soc/sdw_utils/soc_sdw_rt722_sdca.c similarity index 75% rename from sound/soc/intel/boards/sof_sdw_rt722_sdca.c rename to sound/soc/sdw_utils/soc_sdw_rt722_sdca.c index 2da9134ad1a3..6a402172289f 100644 --- a/sound/soc/intel/boards/sof_sdw_rt722_sdca.c +++ b/sound/soc/sdw_utils/soc_sdw_rt722_sdca.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2023 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt722_sdca - Helpers to handle RT722-SDCA from generic machine driver + * soc_sdw_rt722_sdca - Helpers to handle RT722-SDCA from generic machine driver */ #include @@ -13,7 +15,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_route rt722_spk_map[] = { { "Speaker", NULL, "rt722 SPK" }, @@ -36,4 +38,4 @@ int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ return ret; } - +EXPORT_SYMBOL_NS(asoc_sdw_rt722_spk_rtd_init, SND_SOC_SDW_UTILS); From 4f54856b4ea460e9048c11e3f698c38fb072529d Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:29 +0530 Subject: [PATCH 054/410] ASoC: intel: split soundwire machine driver private data Split intel generic SoundWire machine driver private data into two structures. One structure is generic one which can be used by other platform machine driver and the other one is intel specific one. Move generic machine driver private data to soc_sdw_utils.h. Define a void pointer in generic machine driver private data structure and assign the vendor specific structure in mc_probe() call. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-15-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 10 +++++ sound/soc/intel/boards/sof_sdw.c | 37 ++++++++++++------- sound/soc/intel/boards/sof_sdw_common.h | 8 +--- sound/soc/intel/boards/sof_sdw_cs42l42.c | 2 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 2 +- sound/soc/intel/boards/sof_sdw_hdmi.c | 14 ++++--- sound/soc/intel/boards/sof_sdw_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw_rt700.c | 2 +- sound/soc/intel/boards/sof_sdw_rt711.c | 6 +-- sound/soc/intel/boards/sof_sdw_rt_amp.c | 4 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 6 +-- 11 files changed, 55 insertions(+), 38 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 5bc2a89bced4..eb713cdf4079 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -55,6 +55,16 @@ struct asoc_sdw_codec_info { struct snd_soc_codec_conf **codec_conf); }; +struct asoc_sdw_mc_private { + struct snd_soc_card card; + struct snd_soc_jack sdw_headset; + struct device *headset_codec_dev; /* only one headset per card */ + struct device *amp_dev1, *amp_dev2; + bool append_dai_type; + bool ignore_internal_dmic; + void *private; +}; + int asoc_sdw_startup(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e1d2b744987f..236e3fab66b9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1319,7 +1319,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, int *num_devs) { struct device *dev = card->dev; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link; @@ -1440,7 +1440,8 @@ static int create_sdw_dailink(struct snd_soc_card *card, int *be_id, struct snd_soc_codec_conf **codec_conf) { struct device *dev = card->dev; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; struct sof_sdw_endpoint *sof_end; int stream; int ret; @@ -1519,7 +1520,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, if (cur_link != sof_end->link_mask) { int link_num = ffs(sof_end->link_mask) - 1; - int pin_num = ctx->sdw_pin_index[link_num]++; + int pin_num = intel_ctx->sdw_pin_index[link_num]++; cur_link = sof_end->link_mask; @@ -1573,11 +1574,12 @@ static int create_sdw_dailinks(struct snd_soc_card *card, struct sof_sdw_dailink *sof_dais, struct snd_soc_codec_conf **codec_conf) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; int ret, i; for (i = 0; i < SDW_MAX_LINKS; i++) - ctx->sdw_pin_index[i] = SOC_SDW_INTEL_BIDIR_PDI_BASE; + intel_ctx->sdw_pin_index[i] = SOC_SDW_INTEL_BIDIR_PDI_BASE; /* generate DAI links by each sdw link */ while (sof_dais->initialised) { @@ -1665,7 +1667,8 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, int hdmi_num) { struct device *dev = card->dev; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; int i, ret; for (i = 0; i < hdmi_num; i++) { @@ -1673,7 +1676,7 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); char *codec_name, *codec_dai_name; - if (ctx->hdmi.idisp_codec) { + if (intel_ctx->hdmi.idisp_codec) { codec_name = "ehdaudio0D2"; codec_dai_name = devm_kasprintf(dev, GFP_KERNEL, "intel-hdmi-hifi%d", i + 1); @@ -1721,7 +1724,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); int sdw_be_num = 0, ssp_num = 0, dmic_num = 0, bt_num = 0; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; struct snd_soc_codec_conf *codec_conf; struct asoc_sdw_codec_info *ssp_info; @@ -1773,7 +1777,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) } if (mach_params->codec_mask & IDISP_CODEC_MASK) - ctx->hdmi.idisp_codec = true; + intel_ctx->hdmi.idisp_codec = true; if (sof_sdw_quirk & SOF_SDW_TGL_HDMI) hdmi_num = SOF_TGL_HDMI_COUNT; @@ -1789,7 +1793,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) dev_dbg(dev, "sdw %d, ssp %d, dmic %d, hdmi %d, bt: %d\n", sdw_be_num, ssp_num, dmic_num, - ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); + intel_ctx->hdmi.idisp_codec ? hdmi_num : 0, bt_num); codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); if (!codec_conf) { @@ -1862,7 +1866,8 @@ err_dai: static int sof_sdw_card_late_probe(struct snd_soc_card *card) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; int ret = 0; int i; @@ -1875,7 +1880,7 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) } } - if (ctx->hdmi.idisp_codec) + if (intel_ctx->hdmi.idisp_codec) ret = sof_sdw_hdmi_card_late_probe(card); return ret; @@ -1934,16 +1939,22 @@ static int mc_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); struct snd_soc_card *card; - struct mc_private *ctx; + struct asoc_sdw_mc_private *ctx; + struct intel_mc_ctx *intel_ctx; int amp_num = 0, i; int ret; dev_dbg(&pdev->dev, "Entry\n"); + intel_ctx = devm_kzalloc(&pdev->dev, sizeof(*intel_ctx), GFP_KERNEL); + if (!intel_ctx) + return -ENOMEM; + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) return -ENOMEM; + ctx->private = intel_ctx; card = &ctx->card; card->dev = &pdev->dev; card->name = "soundwire"; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index c7672dc1320c..7954472c11bb 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -78,16 +78,10 @@ enum { #define SOC_SDW_DAI_TYPE_AMP 1 #define SOC_SDW_DAI_TYPE_MIC 2 -struct mc_private { - struct snd_soc_card card; - struct snd_soc_jack sdw_headset; +struct intel_mc_ctx { struct sof_hdmi_private hdmi; - struct device *headset_codec_dev; /* only one headset per card */ - struct device *amp_dev1, *amp_dev2; /* To store SDW Pin index for each SoundWire link */ unsigned int sdw_pin_index[SDW_MAX_LINKS]; - bool append_dai_type; - bool ignore_internal_dmic; }; extern unsigned long sof_sdw_quirk; diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/intel/boards/sof_sdw_cs42l42.c index d28477c50469..3ce2f65f994a 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l42.c @@ -39,7 +39,7 @@ static struct snd_soc_jack_pin cs42l42_jack_pins[] = { int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/intel/boards/sof_sdw_cs42l43.c index bb371b2649cf..47d05fe7de53 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/intel/boards/sof_sdw_cs42l43.c @@ -51,7 +51,7 @@ static struct snd_soc_jack_pin sof_jack_pins[] = { int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &ctx->sdw_headset; struct snd_soc_card *card = rtd->card; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_hdmi.c b/sound/soc/intel/boards/sof_sdw_hdmi.c index f34fabdf9d93..4084119a9a5f 100644 --- a/sound/soc/intel/boards/sof_sdw_hdmi.c +++ b/sound/soc/intel/boards/sof_sdw_hdmi.c @@ -17,23 +17,25 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd) { - struct mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - ctx->hdmi.hdmi_comp = dai->component; + intel_ctx->hdmi.hdmi_comp = dai->component; return 0; } int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; - if (!ctx->hdmi.idisp_codec) + if (!intel_ctx->hdmi.idisp_codec) return 0; - if (!ctx->hdmi.hdmi_comp) + if (!intel_ctx->hdmi.hdmi_comp) return -EINVAL; - return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp); + return hda_dsp_hdmi_build_controls(card, intel_ctx->hdmi.hdmi_comp); } diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/intel/boards/sof_sdw_rt5682.c index 3584638e2192..7e52720e0195 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/intel/boards/sof_sdw_rt5682.c @@ -38,7 +38,7 @@ static struct snd_soc_jack_pin rt5682_jack_pins[] = { int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/intel/boards/sof_sdw_rt700.c index a90d69573736..0abaaeacfd90 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/intel/boards/sof_sdw_rt700.c @@ -36,7 +36,7 @@ static struct snd_soc_jack_pin rt700_jack_pins[] = { int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/intel/boards/sof_sdw_rt711.c index e4d300d7d5ef..fb5bc611b5e7 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/intel/boards/sof_sdw_rt711.c @@ -62,7 +62,7 @@ static struct snd_soc_jack_pin rt711_jack_pins[] = { int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; @@ -113,7 +113,7 @@ int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); if (!ctx->headset_codec_dev) return 0; @@ -129,7 +129,7 @@ int asoc_sdw_rt711_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 4cf392c84cc7..fff746671c1d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -239,7 +239,7 @@ const struct snd_soc_ops soc_sdw_rt1308_i2s_ops = { int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); if (ctx->amp_dev1) { device_remove_software_node(ctx->amp_dev1); @@ -259,7 +259,7 @@ int asoc_sdw_rt_amp_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev1, *sdw_dev2; int ret; diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c index d84aec2b4c78..8059e483835d 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c @@ -86,7 +86,7 @@ static const char * const need_sdca_suffix[] = { int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component; struct snd_soc_jack *jack; int ret; @@ -163,7 +163,7 @@ int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_s int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); if (!ctx->headset_codec_dev) return 0; @@ -183,7 +183,7 @@ int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback) { - struct mc_private *ctx = snd_soc_card_get_drvdata(card); + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct device *sdw_dev; int ret; From 139e17740200d8e92677942bfee6c8cfe4da3009 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:30 +0530 Subject: [PATCH 055/410] ASoC: intel/sdw_utils: move rt5682 codec helper function Move rt5682 sdw codec helper function to common place holder to make it generic. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-16-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 1 + sound/soc/intel/boards/Makefile | 4 ++-- sound/soc/intel/boards/sof_sdw_common.h | 1 - sound/soc/sdw_utils/Makefile | 3 ++- .../sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} | 8 +++++--- 5 files changed, 10 insertions(+), 7 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (88%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index eb713cdf4079..ed97d78336da 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -86,5 +86,6 @@ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index f3baf9ecfbb7..80c33e4b4cfe 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -37,8 +37,8 @@ snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ bridge_cs35l56.o \ - sof_sdw_rt5682.o sof_sdw_rt700.o \ - sof_sdw_rt711.o sof_sdw_rt_sdca_jack_common.o \ + sof_sdw_rt700.o sof_sdw_rt711.o \ + sof_sdw_rt_sdca_jack_common.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 7954472c11bb..bbd09698c69d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -151,7 +151,6 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 261c60098e88..fea2b6ae6975 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,4 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ - soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o + soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o \ + soc_sdw_rt5682.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt5682.c b/sound/soc/sdw_utils/soc_sdw_rt5682.c similarity index 88% rename from sound/soc/intel/boards/sof_sdw_rt5682.c rename to sound/soc/sdw_utils/soc_sdw_rt5682.c index 7e52720e0195..80b4caa92667 100644 --- a/sound/soc/intel/boards/sof_sdw_rt5682.c +++ b/sound/soc/sdw_utils/soc_sdw_rt5682.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver + * soc_sdw_rt5682 - Helpers to handle RT5682 from generic machine driver */ #include @@ -15,7 +17,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_route rt5682_map[] = { /*Headphones*/ @@ -86,4 +88,4 @@ int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai return ret; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_rt5682_rtd_init, SND_SOC_SDW_UTILS); From da5b1831673208e235cadc9107207adb092c2eb9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:31 +0530 Subject: [PATCH 056/410] ASoC: intel/sdw_utils: move rtk jack common helper functions Move RTK codec jack common helper functions to common place holder (sdw_utils folder) to make it generic so that it will be used by other platform machine driver code. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-17-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 11 ++++++++++ sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw.c | 1 + sound/soc/intel/boards/sof_sdw_common.h | 10 ---------- sound/soc/sdw_utils/Makefile | 2 +- .../soc_sdw_rt_sdca_jack_common.c} | 20 +++++++++++-------- 6 files changed, 25 insertions(+), 20 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (90%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index ed97d78336da..6b6bab8d3310 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -13,6 +13,8 @@ #include #define SOC_SDW_MAX_DAI_NUM 8 +#define SOC_SDW_MAX_NO_PROPS 2 +#define SOC_SDW_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) struct asoc_sdw_codec_info; @@ -63,6 +65,7 @@ struct asoc_sdw_mc_private { bool append_dai_type; bool ignore_internal_dmic; void *private; + unsigned long mc_quirk; }; int asoc_sdw_startup(struct snd_pcm_substream *substream); @@ -82,8 +85,16 @@ const char *asoc_sdw_get_codec_name(struct device *dev, /* DMIC support */ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); +/* RT711-SDCA support */ +int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 80c33e4b4cfe..0f1b2c288162 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -38,7 +38,6 @@ snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ bridge_cs35l56.o \ sof_sdw_rt700.o sof_sdw_rt711.o \ - sof_sdw_rt_sdca_jack_common.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 236e3fab66b9..e310843974a7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1973,6 +1973,7 @@ static int mc_probe(struct platform_device *pdev) log_quirks(card->dev); + ctx->mc_quirk = sof_sdw_quirk; /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) codec_info_list[i].amp_num = 0; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index bbd09698c69d..af656716c9d2 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -15,7 +15,6 @@ #include #include "sof_hdmi_common.h" -#define SOC_SDW_MAX_NO_PROPS 2 #define MAX_HDMI_NUM 4 #define SOC_SDW_UNUSED_DAI_ID -1 #define SOC_SDW_JACK_OUT_DAI_ID 0 @@ -45,7 +44,6 @@ enum { SOF_I2S_SSP5 = BIT(5), }; -#define SOC_SDW_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) /* Deprecated and no longer supported by the code */ #define SOC_SDW_FOUR_SPK BIT(4) #define SOF_SDW_TGL_HDMI BIT(5) @@ -98,13 +96,6 @@ int asoc_sdw_rt711_init(struct snd_soc_card *card, bool playback); int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); -/* RT711-SDCA support */ -int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); -int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - /* RT1308 I2S support */ extern const struct snd_soc_ops soc_sdw_rt1308_i2s_ops; @@ -154,6 +145,5 @@ int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index fea2b6ae6975..68b8fddeb77e 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o \ - soc_sdw_rt5682.o + soc_sdw_rt5682.o soc_sdw_rt_sdca_jack_common.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c similarity index 90% rename from sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c rename to sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c index 8059e483835d..3e6211dc1599 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c +++ b/sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt711_sdca - Helpers to handle RT711-SDCA from generic machine driver + * soc_sdw_rt711_sdca - Helpers to handle RT711-SDCA from generic machine driver */ #include @@ -15,22 +17,22 @@ #include #include #include -#include "sof_sdw_common.h" +#include /* * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev) +static int rt_sdca_jack_add_codec_device_props(struct device *sdw_dev, unsigned long quirk) { struct property_entry props[SOC_SDW_MAX_NO_PROPS] = {}; struct fwnode_handle *fwnode; int ret; - if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(quirk)) return 0; - props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(sof_sdw_quirk)); + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(quirk)); fwnode = fwnode_create_software_node(props, NULL); if (IS_ERR(fwnode)) @@ -160,6 +162,7 @@ int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_s return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_rt_sdca_jack_rtd_init, SND_SOC_SDW_UTILS); int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -168,7 +171,7 @@ int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_lin if (!ctx->headset_codec_dev) return 0; - if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(ctx->mc_quirk)) return 0; device_remove_software_node(ctx->headset_codec_dev); @@ -177,6 +180,7 @@ int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_lin return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_rt_sdca_jack_exit, SND_SOC_SDW_UTILS); int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -198,7 +202,7 @@ int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, if (!sdw_dev) return -EPROBE_DEFER; - ret = rt_sdca_jack_add_codec_device_props(sdw_dev); + ret = rt_sdca_jack_add_codec_device_props(sdw_dev, ctx->mc_quirk); if (ret < 0) { put_device(sdw_dev); return ret; @@ -207,4 +211,4 @@ int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, return 0; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_rt_sdca_jack_init, SND_SOC_SDW_UTILS); From 8e84fd22dc425fd0b005a20156eeb67f019ae39f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:32 +0530 Subject: [PATCH 057/410] ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions Move RT700 and RT711 Soundwire codec helper functions to common place holder so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-18-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 9 +++++++++ sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw_common.h | 9 --------- sound/soc/sdw_utils/Makefile | 1 + .../soc_sdw_rt700.c} | 8 +++++--- .../soc_sdw_rt711.c} | 18 +++++++++++------- 6 files changed, 26 insertions(+), 20 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (88%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (85%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 6b6bab8d3310..3e55cac33176 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -85,6 +85,13 @@ const char *asoc_sdw_get_codec_name(struct device *dev, /* DMIC support */ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); +/* RT711 support */ +int asoc_sdw_rt711_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + /* RT711-SDCA support */ int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -95,6 +102,8 @@ int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_lin /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 0f1b2c288162..9f92f49cc517 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -37,7 +37,6 @@ snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_maxim.o sof_sdw_rt_amp.o \ bridge_cs35l56.o \ - sof_sdw_rt700.o sof_sdw_rt711.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ sof_sdw_hdmi.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index af656716c9d2..1d7e6df02247 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -89,13 +89,6 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); -/* RT711 support */ -int asoc_sdw_rt711_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); -int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - /* RT1308 I2S support */ extern const struct snd_soc_ops soc_sdw_rt1308_i2s_ops; @@ -142,8 +135,6 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 68b8fddeb77e..20516094f453 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -1,5 +1,6 @@ # SPDX-License-Identifier: GPL-2.0-only snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ + soc_sdw_rt700.o soc_sdw_rt711.o \ soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o \ soc_sdw_rt5682.o soc_sdw_rt_sdca_jack_common.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt700.c b/sound/soc/sdw_utils/soc_sdw_rt700.c similarity index 88% rename from sound/soc/intel/boards/sof_sdw_rt700.c rename to sound/soc/sdw_utils/soc_sdw_rt700.c index 0abaaeacfd90..4dbeeeca3434 100644 --- a/sound/soc/intel/boards/sof_sdw_rt700.c +++ b/sound/soc/sdw_utils/soc_sdw_rt700.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt700 - Helpers to handle RT700 from generic machine driver + * soc_sdw_rt700 - Helpers to handle RT700 from generic machine driver */ #include @@ -13,7 +15,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_route rt700_map[] = { /* Headphones */ @@ -83,4 +85,4 @@ int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai return ret; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_rt700_rtd_init, SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_rt711.c b/sound/soc/sdw_utils/soc_sdw_rt711.c similarity index 85% rename from sound/soc/intel/boards/sof_sdw_rt711.c rename to sound/soc/sdw_utils/soc_sdw_rt711.c index fb5bc611b5e7..38b4126dd45f 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711.c +++ b/sound/soc/sdw_utils/soc_sdw_rt711.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt711 - Helpers to handle RT711 from generic machine driver + * soc_sdw_rt711 - Helpers to handle RT711 from generic machine driver */ #include @@ -15,21 +17,21 @@ #include #include #include -#include "sof_sdw_common.h" +#include /* * Note this MUST be called before snd_soc_register_card(), so that the props * are in place before the codec component driver's probe function parses them. */ -static int rt711_add_codec_device_props(struct device *sdw_dev) +static int rt711_add_codec_device_props(struct device *sdw_dev, unsigned long quirk) { struct property_entry props[SOC_SDW_MAX_NO_PROPS] = {}; struct fwnode_handle *fwnode; int ret; - if (!SOC_SDW_JACK_JDSRC(sof_sdw_quirk)) + if (!SOC_SDW_JACK_JDSRC(quirk)) return 0; - props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(sof_sdw_quirk)); + props[0] = PROPERTY_ENTRY_U32("realtek,jd-src", SOC_SDW_JACK_JDSRC(quirk)); fwnode = fwnode_create_software_node(props, NULL); if (IS_ERR(fwnode)) @@ -110,6 +112,7 @@ int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_rt711_rtd_init, SND_SOC_SDW_UTILS); int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -123,6 +126,7 @@ int asoc_sdw_rt711_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_ return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_rt711_exit, SND_SOC_SDW_UTILS); int asoc_sdw_rt711_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -144,7 +148,7 @@ int asoc_sdw_rt711_init(struct snd_soc_card *card, if (!sdw_dev) return -EPROBE_DEFER; - ret = rt711_add_codec_device_props(sdw_dev); + ret = rt711_add_codec_device_props(sdw_dev, ctx->mc_quirk); if (ret < 0) { put_device(sdw_dev); return ret; @@ -153,4 +157,4 @@ int asoc_sdw_rt711_init(struct snd_soc_card *card, return 0; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_rt711_init, SND_SOC_SDW_UTILS); From ccc96ae2814a7faad591af68bd0115e4d2b256b4 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:33 +0530 Subject: [PATCH 058/410] ASoC: intel/sdw_utils: move rtk amp codec helper functions Move RTK amp codec helper functions related implementation to common place holder to make it generic so that these helper functions will be used by other platform machine driver modules. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-19-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 11 +++++++++++ sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 11 ----------- sound/soc/sdw_utils/Makefile | 3 ++- .../soc_sdw_rt_amp.c} | 14 ++++++++++---- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +++--- 6 files changed, 27 insertions(+), 20 deletions(-) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (93%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 3e55cac33176..450542eb3ea0 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -99,9 +99,20 @@ int asoc_sdw_rt_sdca_jack_init(struct snd_soc_card *card, bool playback); int asoc_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +/* RT1308 I2S support */ +extern const struct snd_soc_ops soc_sdw_rt1308_i2s_ops; + +/* generic amp support */ +int asoc_sdw_rt_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); +int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); + /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt700_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9f92f49cc517..70c56bdc180c 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -35,7 +35,7 @@ snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ - sof_sdw_maxim.o sof_sdw_rt_amp.o \ + sof_sdw_maxim.o \ bridge_cs35l56.o \ sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ sof_sdw_cs_amp.o \ diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 1d7e6df02247..7f856c6018aa 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -89,16 +89,6 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); -/* RT1308 I2S support */ -extern const struct snd_soc_ops soc_sdw_rt1308_i2s_ops; - -/* generic amp support */ -int asoc_sdw_rt_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); -int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); - /* MAXIM codec support */ int asoc_sdw_maxim_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -135,6 +125,5 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 20516094f453..7851af9b8593 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -2,5 +2,6 @@ snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ soc_sdw_rt700.o soc_sdw_rt711.o \ soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o \ - soc_sdw_rt5682.o soc_sdw_rt_sdca_jack_common.o + soc_sdw_rt5682.o soc_sdw_rt_sdca_jack_common.o \ + soc_sdw_rt_amp.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/sdw_utils/soc_sdw_rt_amp.c similarity index 93% rename from sound/soc/intel/boards/sof_sdw_rt_amp.c rename to sound/soc/sdw_utils/soc_sdw_rt_amp.c index fff746671c1d..42be01405ab4 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_rt_amp.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2022 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_rt_amp - Helpers to handle RT1308/RT1316/RT1318 from generic machine driver + * soc_sdw_rt_amp - Helpers to handle RT1308/RT1316/RT1318 from generic machine driver */ #include @@ -14,9 +16,9 @@ #include #include #include -#include "sof_sdw_common.h" -#include "sof_sdw_amp_coeff_tables.h" -#include "../../codecs/rt1308.h" +#include +#include "soc_sdw_rt_amp_coeff_tables.h" +#include "../codecs/rt1308.h" #define CODEC_NAME_SIZE 7 @@ -199,6 +201,7 @@ int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_rt_amp_spk_rtd_init, SND_SOC_SDW_UTILS); static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) @@ -236,6 +239,7 @@ static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, const struct snd_soc_ops soc_sdw_rt1308_i2s_ops = { .hw_params = rt1308_i2s_hw_params, }; +EXPORT_SYMBOL_NS(soc_sdw_rt1308_i2s_ops, SND_SOC_SDW_UTILS); int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link) { @@ -253,6 +257,7 @@ int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_rt_amp_exit, SND_SOC_SDW_UTILS); int asoc_sdw_rt_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -295,3 +300,4 @@ int asoc_sdw_rt_amp_init(struct snd_soc_card *card, return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_rt_amp_init, SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h b/sound/soc/sdw_utils/soc_sdw_rt_amp_coeff_tables.h similarity index 97% rename from sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h rename to sound/soc/sdw_utils/soc_sdw_rt_amp_coeff_tables.h index 4a3e6fdbd623..4803d134d071 100644 --- a/sound/soc/intel/boards/sof_sdw_amp_coeff_tables.h +++ b/sound/soc/sdw_utils/soc_sdw_rt_amp_coeff_tables.h @@ -2,11 +2,11 @@ */ /* - * sof_sdw_amp_coeff_tables.h - related coefficients for amplifier parameters + * soc_sdw_rt_amp_coeff_tables.h - related coefficients for RTK amplifier parameters */ -#ifndef SND_SOC_SOF_SDW_AMP_COEFF_H -#define SND_SOC_SOF_SDW_AMP_COEFF_H +#ifndef SND_SOC_SDW_RT_SDW_AMP_COEFF_H +#define SND_SOC_SDW_RT_SDW_AMP_COEFF_H #define RT1308_MAX_BQ_REG 480 #define RT1316_MAX_BQ_REG 580 From 5fa46627d5118bf58b80df45489f49e1e316473a Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:34 +0530 Subject: [PATCH 059/410] ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions To make it generic, move Cirrus Soundwire codec helper functions to common place holder so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-20-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 40 +++++++++++++++++++ sound/soc/intel/boards/Makefile | 3 -- sound/soc/intel/boards/sof_sdw_common.h | 39 ------------------ sound/soc/sdw_utils/Makefile | 5 ++- .../soc_sdw_bridge_cs35l56.c} | 38 ++++++++++++------ .../soc_sdw_cs42l42.c} | 9 +++-- .../soc_sdw_cs42l43.c} | 20 ++++++---- .../soc_sdw_cs_amp.c} | 8 +++- 8 files changed, 94 insertions(+), 68 deletions(-) rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (88%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (85%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (80%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 450542eb3ea0..d5dd887b9d15 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -16,6 +16,19 @@ #define SOC_SDW_MAX_NO_PROPS 2 #define SOC_SDW_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) +/* If a CODEC has an optional speaker output, this quirk will enable it */ +#define SOC_SDW_CODEC_SPKR BIT(15) +/* + * If the CODEC has additional devices attached directly to it. + * + * For the cs42l43: + * - 0 - No speaker output + * - SOC_SDW_CODEC_SPKR - CODEC internal speaker + * - SOC_SDW_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker + * - SOC_SDW_CODEC_SPKR | SOF_SIDECAR_AMPS - Not currently supported + */ +#define SOC_SDW_SIDECAR_AMPS BIT(16) + struct asoc_sdw_codec_info; struct asoc_sdw_dai_info { @@ -109,6 +122,28 @@ int asoc_sdw_rt_amp_init(struct snd_soc_card *card, bool playback); int asoc_sdw_rt_amp_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); +/* CS42L43 support */ +int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + +/* CS AMP support */ +int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card, + int *num_dais, int *num_devs); +int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + struct snd_soc_codec_conf **codec_conf); +int asoc_sdw_bridge_cs35l56_spk_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + +int asoc_sdw_cs_amp_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); @@ -118,5 +153,10 @@ int asoc_sdw_rt711_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai int asoc_sdw_rt712_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt722_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt5682_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 70c56bdc180c..1ee903e12249 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -36,9 +36,6 @@ snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ sof_sdw_maxim.o \ - bridge_cs35l56.o \ - sof_sdw_cs42l42.o sof_sdw_cs42l43.o \ - sof_sdw_cs_amp.o \ sof_sdw_hdmi.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 7f856c6018aa..b95daa32e343 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -52,18 +52,6 @@ enum { #define SOF_SSP_GET_PORT(quirk) (((quirk) >> 7) & GENMASK(5, 0)) /* Deprecated and no longer supported by the code */ #define SOC_SDW_NO_AGGREGATION BIT(14) -/* If a CODEC has an optional speaker output, this quirk will enable it */ -#define SOC_SDW_CODEC_SPKR BIT(15) -/* - * If the CODEC has additional devices attached directly to it. - * - * For the cs42l43: - * - 0 - No speaker output - * - SOC_SDW_CODEC_SPKR - CODEC internal speaker - * - SOC_SDW_SIDECAR_AMPS - 2x Sidecar amplifiers + CODEC internal speaker - * - SOC_SDW_CODEC_SPKR | SOC_SDW_SIDECAR_AMPS - Not currently supported - */ -#define SOC_SDW_SIDECAR_AMPS BIT(16) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 15 @@ -95,35 +83,8 @@ int asoc_sdw_maxim_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback); -/* CS42L43 support */ -int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - -/* CS AMP support */ -int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card, - int *num_dais, int *num_devs); -int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, - struct snd_soc_dai_link **dai_links, - struct snd_soc_codec_conf **codec_conf); -int asoc_sdw_bridge_cs35l56_spk_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - -int asoc_sdw_cs_amp_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - /* dai_link init callbacks */ -int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); -int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index 7851af9b8593..c15b08f3ab0b 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -3,5 +3,8 @@ snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ soc_sdw_rt700.o soc_sdw_rt711.o \ soc_sdw_rt712_sdca.o soc_sdw_rt722_sdca.o \ soc_sdw_rt5682.o soc_sdw_rt_sdca_jack_common.o \ - soc_sdw_rt_amp.o + soc_sdw_rt_amp.o \ + soc_sdw_bridge_cs35l56.o \ + soc_sdw_cs42l42.o soc_sdw_cs42l43.o \ + soc_sdw_cs_amp.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/bridge_cs35l56.c b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c similarity index 73% rename from sound/soc/intel/boards/bridge_cs35l56.c rename to sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c index 55e5cfbb2f14..fcc3ef685af7 100644 --- a/sound/soc/intel/boards/bridge_cs35l56.c +++ b/sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c @@ -1,6 +1,11 @@ // SPDX-License-Identifier: GPL-2.0-only -// -// Intel SOF Machine Driver with Cirrus Logic CS35L56 Smart Amp +// This file incorporates work covered by the following copyright notice: +// Copyright (c) 2024 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. + +/* + * soc_sdw_bridge_cs35l56 - codec helper functions for handling CS35L56 Smart AMP + */ #include #include @@ -9,7 +14,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_widget bridge_widgets[] = { SND_SOC_DAPM_SPK("Bridge Speaker", NULL), @@ -25,7 +30,7 @@ static const char * const bridge_cs35l56_name_prefixes[] = { "AMPR", }; -static int bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd) +static int asoc_sdw_bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; int i, ret; @@ -73,7 +78,7 @@ static int bridge_cs35l56_asp_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static const struct snd_soc_pcm_stream bridge_params = { +static const struct snd_soc_pcm_stream asoc_sdw_bridge_params = { .formats = SNDRV_PCM_FMTBIT_S16_LE, .rate_min = 48000, .rate_max = 48000, @@ -81,7 +86,7 @@ static const struct snd_soc_pcm_stream bridge_params = { .channels_max = 2, }; -SND_SOC_DAILINK_DEFS(bridge_dai, +SND_SOC_DAILINK_DEFS(asoc_sdw_bridge_dai, DAILINK_COMP_ARRAY(COMP_CODEC("cs42l43-codec", "cs42l43-asp")), DAILINK_COMP_ARRAY(COMP_CODEC("spi-cs35l56-left", "cs35l56-asp1"), COMP_CODEC("spi-cs35l56-right", "cs35l56-asp1")), @@ -89,28 +94,33 @@ SND_SOC_DAILINK_DEFS(bridge_dai, static const struct snd_soc_dai_link bridge_dai_template = { .name = "cs42l43-cs35l56", - .init = bridge_cs35l56_asp_init, - .c2c_params = &bridge_params, + .init = asoc_sdw_bridge_cs35l56_asp_init, + .c2c_params = &asoc_sdw_bridge_params, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBC_CFC, - SND_SOC_DAILINK_REG(bridge_dai), + SND_SOC_DAILINK_REG(asoc_sdw_bridge_dai), }; int asoc_sdw_bridge_cs35l56_count_sidecar(struct snd_soc_card *card, int *num_dais, int *num_devs) { - if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS) { (*num_dais)++; (*num_devs) += ARRAY_SIZE(bridge_cs35l56_name_prefixes); } return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_bridge_cs35l56_count_sidecar, SND_SOC_SDW_UTILS); int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, struct snd_soc_codec_conf **codec_conf) { - if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) { + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS) { **dai_links = bridge_dai_template; for (int i = 0; i < ARRAY_SIZE(bridge_cs35l56_name_prefixes); i++) { @@ -124,14 +134,18 @@ int asoc_sdw_bridge_cs35l56_add_sidecar(struct snd_soc_card *card, return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_bridge_cs35l56_add_sidecar, SND_SOC_SDW_UTILS); int asoc_sdw_bridge_cs35l56_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct asoc_sdw_codec_info *info, bool playback) { - if (sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS) + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + + if (ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS) info->amp_num += ARRAY_SIZE(bridge_cs35l56_name_prefixes); return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_bridge_cs35l56_spk_init, SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l42.c b/sound/soc/sdw_utils/soc_sdw_cs42l42.c similarity index 88% rename from sound/soc/intel/boards/sof_sdw_cs42l42.c rename to sound/soc/sdw_utils/soc_sdw_cs42l42.c index 3ce2f65f994a..78a6cb059ac0 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l42.c +++ b/sound/soc/sdw_utils/soc_sdw_cs42l42.c @@ -1,8 +1,9 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2023 Intel Corporation - +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_cs42l42 - Helpers to handle CS42L42 from generic machine driver + * soc_sdw_cs42l42 - Helpers to handle CS42L42 from generic machine driver */ #include @@ -15,7 +16,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_route cs42l42_map[] = { /* HP jack connectors - unknown if we have jack detection */ @@ -87,4 +88,4 @@ int asoc_sdw_cs42l42_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_da return ret; } -MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); +EXPORT_SYMBOL_NS(asoc_sdw_cs42l42_rtd_init, SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_cs42l43.c b/sound/soc/sdw_utils/soc_sdw_cs42l43.c similarity index 85% rename from sound/soc/intel/boards/sof_sdw_cs42l43.c rename to sound/soc/sdw_utils/soc_sdw_cs42l43.c index 47d05fe7de53..adb1c008e871 100644 --- a/sound/soc/intel/boards/sof_sdw_cs42l43.c +++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c @@ -1,9 +1,11 @@ // SPDX-License-Identifier: GPL-2.0-only // Based on sof_sdw_rt5682.c +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2023 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_cs42l43 - Helpers to handle CS42L43 from generic machine driver + * soc_sdw_cs42l43 - Helpers to handle CS42L43 from generic machine driver */ #include #include @@ -16,7 +18,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static const struct snd_soc_dapm_route cs42l43_hs_map[] = { { "Headphone", NULL, "cs42l43 AMP3_OUT" }, @@ -37,7 +39,7 @@ static const struct snd_soc_dapm_route cs42l43_dmic_map[] = { { "cs42l43 PDM2_DIN", NULL, "DMIC" }, }; -static struct snd_soc_jack_pin sof_jack_pins[] = { +static struct snd_soc_jack_pin soc_jack_pins[] = { { .pin = "Headphone", .mask = SND_JACK_HEADPHONE, @@ -73,8 +75,8 @@ int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc SND_JACK_HEADSET | SND_JACK_LINEOUT | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, - jack, sof_jack_pins, - ARRAY_SIZE(sof_jack_pins)); + jack, soc_jack_pins, + ARRAY_SIZE(soc_jack_pins)); if (ret) { dev_err(card->dev, "Failed to create jack: %d\n", ret); return ret; @@ -98,13 +100,15 @@ int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_hs_rtd_init, SND_SOC_SDW_UTILS); int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { struct snd_soc_card *card = rtd->card; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); int ret; - if (!(sof_sdw_quirk & SOC_SDW_SIDECAR_AMPS)) { + if (!(ctx->mc_quirk & SOC_SDW_SIDECAR_AMPS)) { /* Will be set by the bridge code in this case */ card->components = devm_kasprintf(card->dev, GFP_KERNEL, "%s spk:cs42l43-spk", @@ -120,6 +124,7 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_spk_rtd_init, SND_SOC_SDW_UTILS); int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -134,6 +139,7 @@ int asoc_sdw_cs42l43_spk_init(struct snd_soc_card *card, return asoc_sdw_bridge_cs35l56_spk_init(card, dai_links, info, playback); } +EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_spk_init, SND_SOC_SDW_UTILS); int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { @@ -152,4 +158,4 @@ int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_s return ret; } - +EXPORT_SYMBOL_NS(asoc_sdw_cs42l43_dmic_rtd_init, SND_SOC_SDW_UTILS); diff --git a/sound/soc/intel/boards/sof_sdw_cs_amp.c b/sound/soc/sdw_utils/soc_sdw_cs_amp.c similarity index 80% rename from sound/soc/intel/boards/sof_sdw_cs_amp.c rename to sound/soc/sdw_utils/soc_sdw_cs_amp.c index 6479974bd2c3..58b059b68016 100644 --- a/sound/soc/intel/boards/sof_sdw_cs_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_cs_amp.c @@ -1,8 +1,10 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2023 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. /* - * sof_sdw_cs_amp - Helpers to handle CS35L56 from generic machine driver + * soc_sdw_cs_amp - Helpers to handle CS35L56 from generic machine driver */ #include @@ -10,7 +12,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include #define CODEC_NAME_SIZE 8 @@ -44,6 +46,7 @@ int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_cs_spk_rtd_init, SND_SOC_SDW_UTILS); int asoc_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, @@ -58,3 +61,4 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card, return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_cs_amp_init, SND_SOC_SDW_UTILS); From 051b7cb3fde16fd8788078d697d84069b0e50e03 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 14:44:35 +0530 Subject: [PATCH 060/410] ASoC: intel/sdw_utils: move maxim codec helper functions Move maxim codec helper functions to common place holder so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801091446.10457-21-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 7 +++++++ sound/soc/intel/boards/Makefile | 1 - sound/soc/intel/boards/sof_sdw_common.h | 10 ---------- sound/soc/sdw_utils/Makefile | 3 ++- .../sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} | 14 +++++++++----- 5 files changed, 18 insertions(+), 17 deletions(-) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (86%) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index d5dd887b9d15..9e84d1ab6cd0 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -144,6 +144,12 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card, struct asoc_sdw_codec_info *info, bool playback); +/* MAXIM codec support */ +int asoc_sdw_maxim_init(struct snd_soc_card *card, + struct snd_soc_dai_link *dai_links, + struct asoc_sdw_codec_info *info, + bool playback); + /* dai_link init callbacks */ int asoc_sdw_rt_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); @@ -158,5 +164,6 @@ int asoc_sdw_cs42l43_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); +int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 1ee903e12249..5bd8dc2d166a 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -35,7 +35,6 @@ snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ - sof_sdw_maxim.o \ sof_sdw_hdmi.o obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o obj-$(CONFIG_SND_SOC_INTEL_SOF_CS42L42_MACH) += snd-soc-sof_cs42l42.o diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index b95daa32e343..664c3404eb81 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -77,14 +77,4 @@ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); int sof_sdw_hdmi_card_late_probe(struct snd_soc_card *card); -/* MAXIM codec support */ -int asoc_sdw_maxim_init(struct snd_soc_card *card, - struct snd_soc_dai_link *dai_links, - struct asoc_sdw_codec_info *info, - bool playback); - -/* dai_link init callbacks */ - -int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai); - #endif diff --git a/sound/soc/sdw_utils/Makefile b/sound/soc/sdw_utils/Makefile index c15b08f3ab0b..28229ed96ffb 100644 --- a/sound/soc/sdw_utils/Makefile +++ b/sound/soc/sdw_utils/Makefile @@ -6,5 +6,6 @@ snd-soc-sdw-utils-y := soc_sdw_utils.o soc_sdw_dmic.o soc_sdw_rt_dmic.o \ soc_sdw_rt_amp.o \ soc_sdw_bridge_cs35l56.o \ soc_sdw_cs42l42.o soc_sdw_cs42l43.o \ - soc_sdw_cs_amp.o + soc_sdw_cs_amp.o \ + soc_sdw_maxim.o obj-$(CONFIG_SND_SOC_SDW_UTILS) += snd-soc-sdw-utils.o diff --git a/sound/soc/intel/boards/sof_sdw_maxim.c b/sound/soc/sdw_utils/soc_sdw_maxim.c similarity index 86% rename from sound/soc/intel/boards/sof_sdw_maxim.c rename to sound/soc/sdw_utils/soc_sdw_maxim.c index 9933224fcf68..cdcd8df37e1d 100644 --- a/sound/soc/intel/boards/sof_sdw_maxim.c +++ b/sound/soc/sdw_utils/soc_sdw_maxim.c @@ -1,7 +1,9 @@ // SPDX-License-Identifier: GPL-2.0-only +// This file incorporates work covered by the following copyright notice: // Copyright (c) 2020 Intel Corporation +// Copyright (c) 2024 Advanced Micro Devices, Inc. // -// sof_sdw_maxim - Helpers to handle maxim codecs +// soc_sdw_maxim - Helpers to handle maxim codecs // codec devices from generic machine driver #include @@ -10,7 +12,7 @@ #include #include #include -#include "sof_sdw_common.h" +#include static int maxim_part_id; #define SOC_SDW_PART_ID_MAX98363 0x8363 @@ -41,8 +43,9 @@ int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_ return ret; } +EXPORT_SYMBOL_NS(asoc_sdw_maxim_spk_rtd_init, SND_SOC_SDW_UTILS); -static int mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable) +static int asoc_sdw_mx8373_enable_spk_pin(struct snd_pcm_substream *substream, bool enable) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *codec_dai; @@ -84,7 +87,7 @@ static int asoc_sdw_mx8373_prepare(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return mx8373_enable_spk_pin(substream, true); + return asoc_sdw_mx8373_enable_spk_pin(substream, true); } static int asoc_sdw_mx8373_hw_free(struct snd_pcm_substream *substream) @@ -96,7 +99,7 @@ static int asoc_sdw_mx8373_hw_free(struct snd_pcm_substream *substream) if (ret < 0) return ret; - return mx8373_enable_spk_pin(substream, false); + return asoc_sdw_mx8373_enable_spk_pin(substream, false); } static const struct snd_soc_ops max_98373_sdw_ops = { @@ -142,3 +145,4 @@ int asoc_sdw_maxim_init(struct snd_soc_card *card, } return 0; } +EXPORT_SYMBOL_NS(asoc_sdw_maxim_init, SND_SOC_SDW_UTILS); From 8f87e292a34813e4551e1513c62b7222900481cb Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:07 +0530 Subject: [PATCH 061/410] ASoC: intel/sdw_utils: move dai id common macros Move dai id common macros from intel SoundWire generic driver to soc_sdw_utils.h file so that it can be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 7 +++++++ sound/soc/intel/boards/sof_sdw_common.h | 6 ------ 2 files changed, 7 insertions(+), 6 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 9e84d1ab6cd0..13941ddd24c8 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -29,6 +29,13 @@ */ #define SOC_SDW_SIDECAR_AMPS BIT(16) +#define SOC_SDW_UNUSED_DAI_ID -1 +#define SOC_SDW_JACK_OUT_DAI_ID 0 +#define SOC_SDW_JACK_IN_DAI_ID 1 +#define SOC_SDW_AMP_OUT_DAI_ID 2 +#define SOC_SDW_AMP_IN_DAI_ID 3 +#define SOC_SDW_DMIC_DAI_ID 4 + struct asoc_sdw_codec_info; struct asoc_sdw_dai_info { diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 664c3404eb81..8bfdffde16a3 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -16,12 +16,6 @@ #include "sof_hdmi_common.h" #define MAX_HDMI_NUM 4 -#define SOC_SDW_UNUSED_DAI_ID -1 -#define SOC_SDW_JACK_OUT_DAI_ID 0 -#define SOC_SDW_JACK_IN_DAI_ID 1 -#define SOC_SDW_AMP_OUT_DAI_ID 2 -#define SOC_SDW_AMP_IN_DAI_ID 3 -#define SOC_SDW_DMIC_DAI_ID 4 #define SOC_SDW_MAX_CPU_DAIS 16 #define SOC_SDW_INTEL_BIDIR_PDI_BASE 2 From 6e7af1fdf7da7f0b79475f573d3c1f49a826bd68 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:08 +0530 Subject: [PATCH 062/410] ASoC: intel/sdw_utils: move soundwire dai type macros Move SoundWire dai type macros to common header file(soc_sdw_util.h). So that these macros will be used by other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 4 ++++ sound/soc/intel/boards/sof_sdw_common.h | 4 ---- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 13941ddd24c8..7912ff7d2bb9 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -36,6 +36,10 @@ #define SOC_SDW_AMP_IN_DAI_ID 3 #define SOC_SDW_DMIC_DAI_ID 4 +#define SOC_SDW_DAI_TYPE_JACK 0 +#define SOC_SDW_DAI_TYPE_AMP 1 +#define SOC_SDW_DAI_TYPE_MIC 2 + struct asoc_sdw_codec_info; struct asoc_sdw_dai_info { diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 8bfdffde16a3..02f3eebd019d 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -54,10 +54,6 @@ enum { (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(18) -#define SOC_SDW_DAI_TYPE_JACK 0 -#define SOC_SDW_DAI_TYPE_AMP 1 -#define SOC_SDW_DAI_TYPE_MIC 2 - struct intel_mc_ctx { struct sof_hdmi_private hdmi; /* To store SDW Pin index for each SoundWire link */ From e377c94773171e8a97e716e57ec67d49b02ded0b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:09 +0530 Subject: [PATCH 063/410] ASoC: intel/sdw_utils: move soundwire codec_info_list structure SoundWire 'codec_info_list' structure is not a platform specific one. Move codec_info_list structure to common file soc_sdw_utils.c. Move codec helper functions which uses codec_info_list structure to common place holder and rename the function by adding _sdw tag. This will allow to use 'codec_info_list' structure and it's helper functions in other platform machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 13 + sound/soc/intel/boards/sof_sdw.c | 658 +--------------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 657 +++++++++++++++++++++++++++ 3 files changed, 676 insertions(+), 652 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 7912ff7d2bb9..9d99a460ba27 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -90,8 +90,12 @@ struct asoc_sdw_mc_private { bool ignore_internal_dmic; void *private; unsigned long mc_quirk; + int codec_info_list_count; }; +extern struct asoc_sdw_codec_info codec_info_list[]; +int asoc_sdw_get_codec_info_list_count(void); + int asoc_sdw_startup(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); int asoc_sdw_prepare(struct snd_pcm_substream *substream); @@ -106,6 +110,15 @@ const char *asoc_sdw_get_codec_name(struct device *dev, const struct snd_soc_acpi_link_adr *adr_link, int adr_index); +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr); + +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id); + +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, + int *dai_index); + +int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd); + /* DMIC support */ int asoc_sdw_dmic_init(struct snd_soc_pcm_runtime *rtd); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e310843974a7..87f3e5aa1477 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -548,50 +548,6 @@ static struct snd_soc_dai_link_component platform_component[] = { } }; -static const struct snd_soc_dapm_widget generic_dmic_widgets[] = { - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static const struct snd_soc_dapm_widget generic_jack_widgets[] = { - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), -}; - -static const struct snd_kcontrol_new generic_jack_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - -static const struct snd_soc_dapm_widget generic_spk_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_kcontrol_new generic_spk_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - -static const struct snd_soc_dapm_widget maxim_widgets[] = { - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - -static const struct snd_kcontrol_new maxim_controls[] = { - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - -static const struct snd_soc_dapm_widget rt700_widgets[] = { - SND_SOC_DAPM_HP("Headphones", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_kcontrol_new rt700_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphones"), - SOC_DAPM_PIN_SWITCH("AMIC"), - SOC_DAPM_PIN_SWITCH("Speaker"), -}; - static const struct snd_soc_ops sdw_ops = { .startup = asoc_sdw_startup, .prepare = asoc_sdw_prepare, @@ -601,547 +557,6 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -static struct asoc_sdw_codec_info codec_info_list[] = { - { - .part_id = 0x700, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt700-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = asoc_sdw_rt700_rtd_init, - .controls = rt700_controls, - .num_controls = ARRAY_SIZE(rt700_controls), - .widgets = rt700_widgets, - .num_widgets = ARRAY_SIZE(rt700_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x711, - .version_id = 3, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt711-sdca-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = asoc_sdw_rt_sdca_jack_init, - .exit = asoc_sdw_rt_sdca_jack_exit, - .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x711, - .version_id = 2, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt711-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = asoc_sdw_rt711_init, - .exit = asoc_sdw_rt711_exit, - .rtd_init = asoc_sdw_rt711_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x712, - .version_id = 3, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt712-sdca-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = asoc_sdw_rt_sdca_jack_init, - .exit = asoc_sdw_rt_sdca_jack_exit, - .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - { - .direction = {true, false}, - .dai_name = "rt712-sdca-aif2", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = asoc_sdw_rt_amp_init, - .exit = asoc_sdw_rt_amp_exit, - .rtd_init = asoc_sdw_rt712_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - }, - .dai_num = 2, - }, - { - .part_id = 0x1712, - .version_id = 3, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt712-sdca-dmic-aif1", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x713, - .version_id = 3, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt712-sdca-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = asoc_sdw_rt_sdca_jack_init, - .exit = asoc_sdw_rt_sdca_jack_exit, - .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x1713, - .version_id = 3, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt712-sdca-dmic-aif1", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x1308, - .acpi_id = "10EC1308", - .dais = { - { - .direction = {true, false}, - .dai_name = "rt1308-aif", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = asoc_sdw_rt_amp_init, - .exit = asoc_sdw_rt_amp_exit, - .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - }, - .dai_num = 1, - .ops = &soc_sdw_rt1308_i2s_ops, - }, - { - .part_id = 0x1316, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt1316-aif", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = asoc_sdw_rt_amp_init, - .exit = asoc_sdw_rt_amp_exit, - .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x1318, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt1318-aif", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = asoc_sdw_rt_amp_init, - .exit = asoc_sdw_rt_amp_exit, - .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x714, - .version_id = 3, - .ignore_internal_dmic = true, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt715-sdca-aif2", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x715, - .version_id = 3, - .ignore_internal_dmic = true, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt715-sdca-aif2", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x714, - .version_id = 2, - .ignore_internal_dmic = true, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt715-aif2", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x715, - .version_id = 2, - .ignore_internal_dmic = true, - .dais = { - { - .direction = {false, true}, - .dai_name = "rt715-aif2", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x722, - .version_id = 3, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt722-sdca-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .init = asoc_sdw_rt_sdca_jack_init, - .exit = asoc_sdw_rt_sdca_jack_exit, - .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - { - .direction = {true, false}, - .dai_name = "rt722-sdca-aif2", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - /* No feedback capability is provided by rt722-sdca codec driver*/ - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = asoc_sdw_rt_amp_init, - .exit = asoc_sdw_rt_amp_exit, - .rtd_init = asoc_sdw_rt722_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - { - .direction = {false, true}, - .dai_name = "rt722-sdca-aif3", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_rt_dmic_rtd_init, - }, - }, - .dai_num = 3, - }, - { - .part_id = 0x8373, - .dais = { - { - .direction = {true, true}, - .dai_name = "max98373-aif1", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = asoc_sdw_maxim_init, - .rtd_init = asoc_sdw_maxim_spk_rtd_init, - .controls = maxim_controls, - .num_controls = ARRAY_SIZE(maxim_controls), - .widgets = maxim_widgets, - .num_widgets = ARRAY_SIZE(maxim_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x8363, - .dais = { - { - .direction = {true, false}, - .dai_name = "max98363-aif1", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = asoc_sdw_maxim_init, - .rtd_init = asoc_sdw_maxim_spk_rtd_init, - .controls = maxim_controls, - .num_controls = ARRAY_SIZE(maxim_controls), - .widgets = maxim_widgets, - .num_widgets = ARRAY_SIZE(maxim_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x5682, - .dais = { - { - .direction = {true, true}, - .dai_name = "rt5682-sdw", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = asoc_sdw_rt5682_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x3556, - .dais = { - { - .direction = {true, true}, - .dai_name = "cs35l56-sdw1", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - .init = asoc_sdw_cs_amp_init, - .rtd_init = asoc_sdw_cs_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x4242, - .dais = { - { - .direction = {true, true}, - .dai_name = "cs42l42-sdw", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - .rtd_init = asoc_sdw_cs42l42_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x4243, - .codec_name = "cs42l43-codec", - .count_sidecar = asoc_sdw_bridge_cs35l56_count_sidecar, - .add_sidecar = asoc_sdw_bridge_cs35l56_add_sidecar, - .dais = { - { - .direction = {true, false}, - .dai_name = "cs42l43-dp5", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .rtd_init = asoc_sdw_cs42l43_hs_rtd_init, - .controls = generic_jack_controls, - .num_controls = ARRAY_SIZE(generic_jack_controls), - .widgets = generic_jack_widgets, - .num_widgets = ARRAY_SIZE(generic_jack_widgets), - }, - { - .direction = {false, true}, - .dai_name = "cs42l43-dp1", - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - .rtd_init = asoc_sdw_cs42l43_dmic_rtd_init, - .widgets = generic_dmic_widgets, - .num_widgets = ARRAY_SIZE(generic_dmic_widgets), - }, - { - .direction = {false, true}, - .dai_name = "cs42l43-dp2", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - }, - { - .direction = {true, false}, - .dai_name = "cs42l43-dp6", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, - .init = asoc_sdw_cs42l43_spk_init, - .rtd_init = asoc_sdw_cs42l43_spk_rtd_init, - .controls = generic_spk_controls, - .num_controls = ARRAY_SIZE(generic_spk_controls), - .widgets = generic_spk_widgets, - .num_widgets = ARRAY_SIZE(generic_spk_widgets), - .quirk = SOC_SDW_CODEC_SPKR | SOC_SDW_SIDECAR_AMPS, - }, - }, - .dai_num = 4, - }, - { - .part_id = 0xaaaa, /* generic codec mockup */ - .version_id = 0, - .dais = { - { - .direction = {true, true}, - .dai_name = "sdw-mockup-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0xaa55, /* headset codec mockup */ - .version_id = 0, - .dais = { - { - .direction = {true, true}, - .dai_name = "sdw-mockup-aif1", - .dai_type = SOC_SDW_DAI_TYPE_JACK, - .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x55aa, /* amplifier mockup */ - .version_id = 0, - .dais = { - { - .direction = {true, true}, - .dai_name = "sdw-mockup-aif1", - .dai_type = SOC_SDW_DAI_TYPE_AMP, - .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, - }, - }, - .dai_num = 1, - }, - { - .part_id = 0x5555, - .version_id = 0, - .dais = { - { - .dai_name = "sdw-mockup-aif1", - .direction = {false, true}, - .dai_type = SOC_SDW_DAI_TYPE_MIC, - .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, - }, - }, - .dai_num = 1, - }, -}; - -static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr) -{ - unsigned int part_id, sdw_version; - int i; - - part_id = SDW_PART_ID(adr); - sdw_version = SDW_VERSION(adr); - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) - /* - * A codec info is for all sdw version with the part id if - * version_id is not specified in the codec info. - */ - if (part_id == codec_info_list[i].part_id && - (!codec_info_list[i].version_id || - sdw_version == codec_info_list[i].version_id)) - return &codec_info_list[i]; - - return NULL; - -} - -static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id) -{ - int i; - - if (!acpi_id[0]) - return NULL; - - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) - if (!memcmp(codec_info_list[i].acpi_id, acpi_id, ACPI_ID_LEN)) - return &codec_info_list[i]; - - return NULL; -} - -static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, - int *dai_index) -{ - int i, j; - - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { - for (j = 0; j < codec_info_list[i].dai_num; j++) { - if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) { - *dai_index = j; - return &codec_info_list[i]; - } - } - } - - return NULL; -} - static void init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, int *be_id, char *name, int playback, int capture, struct snd_soc_dai_link_component *cpus, int cpus_num, @@ -1190,69 +605,6 @@ static int init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai return 0; } -static int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_card *card = rtd->card; - struct asoc_sdw_codec_info *codec_info; - struct snd_soc_dai *dai; - int dai_index; - int ret; - int i; - - for_each_rtd_codec_dais(rtd, i, dai) { - codec_info = asoc_sdw_find_codec_info_dai(dai->name, &dai_index); - if (!codec_info) - return -EINVAL; - - /* - * A codec dai can be connected to different dai links for capture and playback, - * but we only need to call the rtd_init function once. - * The rtd_init for each codec dai is independent. So, the order of rtd_init - * doesn't matter. - */ - if (codec_info->dais[dai_index].rtd_init_done) - continue; - - /* - * Add card controls and dapm widgets for the first codec dai. - * The controls and widgets will be used for all codec dais. - */ - - if (i > 0) - goto skip_add_controls_widgets; - - if (codec_info->dais[dai_index].controls) { - ret = snd_soc_add_card_controls(card, codec_info->dais[dai_index].controls, - codec_info->dais[dai_index].num_controls); - if (ret) { - dev_err(card->dev, "%#x controls addition failed: %d\n", - codec_info->part_id, ret); - return ret; - } - } - if (codec_info->dais[dai_index].widgets) { - ret = snd_soc_dapm_new_controls(&card->dapm, - codec_info->dais[dai_index].widgets, - codec_info->dais[dai_index].num_widgets); - if (ret) { - dev_err(card->dev, "%#x widgets addition failed: %d\n", - codec_info->part_id, ret); - return ret; - } - } - -skip_add_controls_widgets: - if (codec_info->dais[dai_index].rtd_init) { - ret = codec_info->dais[dai_index].rtd_init(rtd, dai); - if (ret) - return ret; - } - codec_info->dais[dai_index].rtd_init_done = true; - } - - return 0; -} - struct sof_sdw_endpoint { struct list_head list; @@ -1871,7 +1223,7 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) int ret = 0; int i; - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + for (i = 0; i < ctx->codec_info_list_count; i++) { if (codec_info_list[i].codec_card_late_probe) { ret = codec_info_list[i].codec_card_late_probe(card); @@ -1907,10 +1259,11 @@ static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card static void mc_dailink_exit_loop(struct snd_soc_card *card) { struct snd_soc_dai_link *dai_link; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); int ret; int i, j; - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + for (i = 0; i < ctx->codec_info_list_count; i++) { for (j = 0; j < codec_info_list[i].dai_num; j++) { codec_info_list[i].dais[j].rtd_init_done = false; /* Check each dai in codec_info_lis to see if it is used in the link */ @@ -1955,6 +1308,7 @@ static int mc_probe(struct platform_device *pdev) return -ENOMEM; ctx->private = intel_ctx; + ctx->codec_info_list_count = asoc_sdw_get_codec_info_list_count(); card = &ctx->card; card->dev = &pdev->dev; card->name = "soundwire"; @@ -1975,7 +1329,7 @@ static int mc_probe(struct platform_device *pdev) ctx->mc_quirk = sof_sdw_quirk; /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + for (i = 0; i < ctx->codec_info_list_count; i++) codec_info_list[i].amp_num = 0; if (mach->mach_params.subsystem_id_set) { @@ -1993,7 +1347,7 @@ static int mc_probe(struct platform_device *pdev) * amp_num will only be increased for active amp * codecs on used platform */ - for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + for (i = 0; i < ctx->codec_info_list_count; i++) amp_num += codec_info_list[i].amp_num; card->components = devm_kasprintf(card->dev, GFP_KERNEL, diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 2b59ddc61078..a496b4eff6e3 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -12,6 +12,663 @@ #include #include +static const struct snd_soc_dapm_widget generic_dmic_widgets[] = { + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static const struct snd_soc_dapm_widget generic_jack_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_kcontrol_new generic_jack_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static const struct snd_soc_dapm_widget generic_spk_widgets[] = { + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_kcontrol_new generic_spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static const struct snd_soc_dapm_widget maxim_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + +static const struct snd_kcontrol_new maxim_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget rt700_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_kcontrol_new rt700_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("AMIC"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +struct asoc_sdw_codec_info codec_info_list[] = { + { + .part_id = 0x700, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt700-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .rtd_init = asoc_sdw_rt700_rtd_init, + .controls = rt700_controls, + .num_controls = ARRAY_SIZE(rt700_controls), + .widgets = rt700_widgets, + .num_widgets = ARRAY_SIZE(rt700_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x711, + .version_id = 3, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt711-sdca-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x711, + .version_id = 2, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt711-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .init = asoc_sdw_rt711_init, + .exit = asoc_sdw_rt711_exit, + .rtd_init = asoc_sdw_rt711_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x712, + .version_id = 3, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt712-sdca-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + { + .direction = {true, false}, + .dai_name = "rt712-sdca-aif2", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt712_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 2, + }, + { + .part_id = 0x1712, + .version_id = 3, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt712-sdca-dmic-aif1", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x713, + .version_id = 3, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt712-sdca-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x1713, + .version_id = 3, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt712-sdca-dmic-aif1", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x1308, + .acpi_id = "10EC1308", + .dais = { + { + .direction = {true, false}, + .dai_name = "rt1308-aif", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 1, + .ops = &soc_sdw_rt1308_i2s_ops, + }, + { + .part_id = 0x1316, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt1316-aif", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x1318, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt1318-aif", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x714, + .version_id = 3, + .ignore_internal_dmic = true, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt715-sdca-aif2", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x715, + .version_id = 3, + .ignore_internal_dmic = true, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt715-sdca-aif2", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x714, + .version_id = 2, + .ignore_internal_dmic = true, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt715-aif2", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x715, + .version_id = 2, + .ignore_internal_dmic = true, + .dais = { + { + .direction = {false, true}, + .dai_name = "rt715-aif2", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x722, + .version_id = 3, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt722-sdca-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .init = asoc_sdw_rt_sdca_jack_init, + .exit = asoc_sdw_rt_sdca_jack_exit, + .rtd_init = asoc_sdw_rt_sdca_jack_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + { + .direction = {true, false}, + .dai_name = "rt722-sdca-aif2", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + /* No feedback capability is provided by rt722-sdca codec driver*/ + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt722_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + { + .direction = {false, true}, + .dai_name = "rt722-sdca-aif3", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_rt_dmic_rtd_init, + }, + }, + .dai_num = 3, + }, + { + .part_id = 0x8373, + .dais = { + { + .direction = {true, true}, + .dai_name = "max98373-aif1", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, + .init = asoc_sdw_maxim_init, + .rtd_init = asoc_sdw_maxim_spk_rtd_init, + .controls = maxim_controls, + .num_controls = ARRAY_SIZE(maxim_controls), + .widgets = maxim_widgets, + .num_widgets = ARRAY_SIZE(maxim_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x8363, + .dais = { + { + .direction = {true, false}, + .dai_name = "max98363-aif1", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_maxim_init, + .rtd_init = asoc_sdw_maxim_spk_rtd_init, + .controls = maxim_controls, + .num_controls = ARRAY_SIZE(maxim_controls), + .widgets = maxim_widgets, + .num_widgets = ARRAY_SIZE(maxim_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x5682, + .dais = { + { + .direction = {true, true}, + .dai_name = "rt5682-sdw", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .rtd_init = asoc_sdw_rt5682_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x3556, + .dais = { + { + .direction = {true, true}, + .dai_name = "cs35l56-sdw1", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, + .init = asoc_sdw_cs_amp_init, + .rtd_init = asoc_sdw_cs_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x4242, + .dais = { + { + .direction = {true, true}, + .dai_name = "cs42l42-sdw", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + .rtd_init = asoc_sdw_cs42l42_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x4243, + .codec_name = "cs42l43-codec", + .count_sidecar = asoc_sdw_bridge_cs35l56_count_sidecar, + .add_sidecar = asoc_sdw_bridge_cs35l56_add_sidecar, + .dais = { + { + .direction = {true, false}, + .dai_name = "cs42l43-dp5", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .rtd_init = asoc_sdw_cs42l43_hs_rtd_init, + .controls = generic_jack_controls, + .num_controls = ARRAY_SIZE(generic_jack_controls), + .widgets = generic_jack_widgets, + .num_widgets = ARRAY_SIZE(generic_jack_widgets), + }, + { + .direction = {false, true}, + .dai_name = "cs42l43-dp1", + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + .rtd_init = asoc_sdw_cs42l43_dmic_rtd_init, + .widgets = generic_dmic_widgets, + .num_widgets = ARRAY_SIZE(generic_dmic_widgets), + }, + { + .direction = {false, true}, + .dai_name = "cs42l43-dp2", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + }, + { + .direction = {true, false}, + .dai_name = "cs42l43-dp6", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_cs42l43_spk_init, + .rtd_init = asoc_sdw_cs42l43_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + .quirk = SOC_SDW_CODEC_SPKR | SOC_SDW_SIDECAR_AMPS, + }, + }, + .dai_num = 4, + }, + { + .part_id = 0xaaaa, /* generic codec mockup */ + .version_id = 0, + .dais = { + { + .direction = {true, true}, + .dai_name = "sdw-mockup-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0xaa55, /* headset codec mockup */ + .version_id = 0, + .dais = { + { + .direction = {true, true}, + .dai_name = "sdw-mockup-aif1", + .dai_type = SOC_SDW_DAI_TYPE_JACK, + .dailink = {SOC_SDW_JACK_OUT_DAI_ID, SOC_SDW_JACK_IN_DAI_ID}, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x55aa, /* amplifier mockup */ + .version_id = 0, + .dais = { + { + .direction = {true, true}, + .dai_name = "sdw-mockup-aif1", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, + }, + }, + .dai_num = 1, + }, + { + .part_id = 0x5555, + .version_id = 0, + .dais = { + { + .dai_name = "sdw-mockup-aif1", + .direction = {false, true}, + .dai_type = SOC_SDW_DAI_TYPE_MIC, + .dailink = {SOC_SDW_UNUSED_DAI_ID, SOC_SDW_DMIC_DAI_ID}, + }, + }, + .dai_num = 1, + }, +}; +EXPORT_SYMBOL_NS(codec_info_list, SND_SOC_SDW_UTILS); + +int asoc_sdw_get_codec_info_list_count(void) +{ + return ARRAY_SIZE(codec_info_list); +}; +EXPORT_SYMBOL_NS(asoc_sdw_get_codec_info_list_count, SND_SOC_SDW_UTILS); + +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr) +{ + unsigned int part_id, sdw_version; + int i; + + part_id = SDW_PART_ID(adr); + sdw_version = SDW_VERSION(adr); + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + /* + * A codec info is for all sdw version with the part id if + * version_id is not specified in the codec info. + */ + if (part_id == codec_info_list[i].part_id && + (!codec_info_list[i].version_id || + sdw_version == codec_info_list[i].version_id)) + return &codec_info_list[i]; + + return NULL; +} +EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_part, SND_SOC_SDW_UTILS); + +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id) +{ + int i; + + if (!acpi_id[0]) + return NULL; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) + if (!memcmp(codec_info_list[i].acpi_id, acpi_id, ACPI_ID_LEN)) + return &codec_info_list[i]; + + return NULL; +} +EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_acpi, SND_SOC_SDW_UTILS); + +struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, int *dai_index) +{ + int i, j; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + for (j = 0; j < codec_info_list[i].dai_num; j++) { + if (!strcmp(codec_info_list[i].dais[j].dai_name, dai_name)) { + *dai_index = j; + return &codec_info_list[i]; + } + } + } + + return NULL; +} +EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_dai, SND_SOC_SDW_UTILS); + +int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct asoc_sdw_codec_info *codec_info; + struct snd_soc_dai *dai; + int dai_index; + int ret; + int i; + + for_each_rtd_codec_dais(rtd, i, dai) { + codec_info = asoc_sdw_find_codec_info_dai(dai->name, &dai_index); + if (!codec_info) + return -EINVAL; + + /* + * A codec dai can be connected to different dai links for capture and playback, + * but we only need to call the rtd_init function once. + * The rtd_init for each codec dai is independent. So, the order of rtd_init + * doesn't matter. + */ + if (codec_info->dais[dai_index].rtd_init_done) + continue; + + /* + * Add card controls and dapm widgets for the first codec dai. + * The controls and widgets will be used for all codec dais. + */ + + if (i > 0) + goto skip_add_controls_widgets; + + if (codec_info->dais[dai_index].controls) { + ret = snd_soc_add_card_controls(card, codec_info->dais[dai_index].controls, + codec_info->dais[dai_index].num_controls); + if (ret) { + dev_err(card->dev, "%#x controls addition failed: %d\n", + codec_info->part_id, ret); + return ret; + } + } + if (codec_info->dais[dai_index].widgets) { + ret = snd_soc_dapm_new_controls(&card->dapm, + codec_info->dais[dai_index].widgets, + codec_info->dais[dai_index].num_widgets); + if (ret) { + dev_err(card->dev, "%#x widgets addition failed: %d\n", + codec_info->part_id, ret); + return ret; + } + } + +skip_add_controls_widgets: + if (codec_info->dais[dai_index].rtd_init) { + ret = codec_info->dais[dai_index].rtd_init(rtd, dai); + if (ret) + return ret; + } + codec_info->dais[dai_index].rtd_init_done = true; + } + + return 0; +} +EXPORT_SYMBOL_NS(asoc_sdw_rtd_init, SND_SOC_SDW_UTILS); + /* these wrappers are only needed to avoid typecast compilation errors */ int asoc_sdw_startup(struct snd_pcm_substream *substream) { From 778dcb08832a5e526e447971f7ca72cb6dd2307b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:10 +0530 Subject: [PATCH 064/410] ASoC: intel/sdw_utils: move machine driver dai link helper functions Move machine driver dai link helper functions to common place holder, So that it can be used by other platform machine driver. Rename these functions with "asoc_sdw" tag as a prefix. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 5 +++ sound/soc/intel/boards/sof_sdw.c | 54 ++--------------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 52 +++++++++++++++++++++++++++ 3 files changed, 59 insertions(+), 52 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 9d99a460ba27..b3b6d6b7ce2f 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -117,6 +117,11 @@ struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id); struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, int *dai_index); +struct snd_soc_dai_link *asoc_sdw_mc_find_codec_dai_used(struct snd_soc_card *card, + const char *dai_name); + +void asoc_sdw_mc_dailink_exit_loop(struct snd_soc_card *card); + int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd); /* DMIC support */ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 87f3e5aa1477..07b1d6994304 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1238,56 +1238,6 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) return ret; } -/* helper to get the link that the codec DAI is used */ -static struct snd_soc_dai_link *mc_find_codec_dai_used(struct snd_soc_card *card, - const char *dai_name) -{ - struct snd_soc_dai_link *dai_link; - int i; - int j; - - for_each_card_prelinks(card, i, dai_link) { - for (j = 0; j < dai_link->num_codecs; j++) { - /* Check each codec in a link */ - if (!strcmp(dai_link->codecs[j].dai_name, dai_name)) - return dai_link; - } - } - return NULL; -} - -static void mc_dailink_exit_loop(struct snd_soc_card *card) -{ - struct snd_soc_dai_link *dai_link; - struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); - int ret; - int i, j; - - for (i = 0; i < ctx->codec_info_list_count; i++) { - for (j = 0; j < codec_info_list[i].dai_num; j++) { - codec_info_list[i].dais[j].rtd_init_done = false; - /* Check each dai in codec_info_lis to see if it is used in the link */ - if (!codec_info_list[i].dais[j].exit) - continue; - /* - * We don't need to call .exit function if there is no matched - * dai link found. - */ - dai_link = mc_find_codec_dai_used(card, - codec_info_list[i].dais[j].dai_name); - if (dai_link) { - /* Do the .exit function if the codec dai is used in the link */ - ret = codec_info_list[i].dais[j].exit(card, dai_link); - if (ret) - dev_warn(card->dev, - "codec exit failed %d\n", - ret); - break; - } - } - } -} - static int mc_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); @@ -1368,7 +1318,7 @@ static int mc_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(card->dev, card); if (ret) { dev_err_probe(card->dev, ret, "snd_soc_register_card failed %d\n", ret); - mc_dailink_exit_loop(card); + asoc_sdw_mc_dailink_exit_loop(card); return ret; } @@ -1381,7 +1331,7 @@ static void mc_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - mc_dailink_exit_loop(card); + asoc_sdw_mc_dailink_exit_loop(card); } static const struct platform_device_id mc_id_table[] = { diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index a496b4eff6e3..409a50147349 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -864,5 +864,57 @@ const char *asoc_sdw_get_codec_name(struct device *dev, } EXPORT_SYMBOL_NS(asoc_sdw_get_codec_name, SND_SOC_SDW_UTILS); +/* helper to get the link that the codec DAI is used */ +struct snd_soc_dai_link *asoc_sdw_mc_find_codec_dai_used(struct snd_soc_card *card, + const char *dai_name) +{ + struct snd_soc_dai_link *dai_link; + int i; + int j; + + for_each_card_prelinks(card, i, dai_link) { + for (j = 0; j < dai_link->num_codecs; j++) { + /* Check each codec in a link */ + if (!strcmp(dai_link->codecs[j].dai_name, dai_name)) + return dai_link; + } + } + return NULL; +} +EXPORT_SYMBOL_NS(asoc_sdw_mc_find_codec_dai_used, SND_SOC_SDW_UTILS); + +void asoc_sdw_mc_dailink_exit_loop(struct snd_soc_card *card) +{ + struct snd_soc_dai_link *dai_link; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + int ret; + int i, j; + + for (i = 0; i < ctx->codec_info_list_count; i++) { + for (j = 0; j < codec_info_list[i].dai_num; j++) { + codec_info_list[i].dais[j].rtd_init_done = false; + /* Check each dai in codec_info_lis to see if it is used in the link */ + if (!codec_info_list[i].dais[j].exit) + continue; + /* + * We don't need to call .exit function if there is no matched + * dai link found. + */ + dai_link = asoc_sdw_mc_find_codec_dai_used(card, + codec_info_list[i].dais[j].dai_name); + if (dai_link) { + /* Do the .exit function if the codec dai is used in the link */ + ret = codec_info_list[i].dais[j].exit(card, dai_link); + if (ret) + dev_warn(card->dev, + "codec exit failed %d\n", + ret); + break; + } + } + } +} +EXPORT_SYMBOL_NS(asoc_sdw_mc_dailink_exit_loop, SND_SOC_SDW_UTILS); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 5bd414c7b80e39ef11bd86db76e726962c1bfc92 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:11 +0530 Subject: [PATCH 065/410] ASoC: sdw_utils: refactor sof_sdw_card_late_probe function Refactor sof_sdw_card_late_probe() function and derive a generic function soc_sdw_card_late_probe() function which can be used by SoundWire generic machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 2 ++ sound/soc/intel/boards/sof_sdw.c | 12 +++--------- sound/soc/sdw_utils/soc_sdw_utils.c | 16 ++++++++++++++++ 3 files changed, 21 insertions(+), 9 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index b3b6d6b7ce2f..14e21a992f56 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -122,6 +122,8 @@ struct snd_soc_dai_link *asoc_sdw_mc_find_codec_dai_used(struct snd_soc_card *ca void asoc_sdw_mc_dailink_exit_loop(struct snd_soc_card *card); +int asoc_sdw_card_late_probe(struct snd_soc_card *card); + int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd); /* DMIC support */ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 07b1d6994304..65b15f594aed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1221,16 +1221,10 @@ static int sof_sdw_card_late_probe(struct snd_soc_card *card) struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; int ret = 0; - int i; - for (i = 0; i < ctx->codec_info_list_count; i++) { - if (codec_info_list[i].codec_card_late_probe) { - ret = codec_info_list[i].codec_card_late_probe(card); - - if (ret < 0) - return ret; - } - } + ret = asoc_sdw_card_late_probe(card); + if (ret < 0) + return ret; if (intel_ctx->hdmi.idisp_codec) ret = sof_sdw_hdmi_card_late_probe(card); diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 409a50147349..613ecc3bed92 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -916,5 +916,21 @@ void asoc_sdw_mc_dailink_exit_loop(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(asoc_sdw_mc_dailink_exit_loop, SND_SOC_SDW_UTILS); +int asoc_sdw_card_late_probe(struct snd_soc_card *card) +{ + int ret = 0; + int i; + + for (i = 0; i < ARRAY_SIZE(codec_info_list); i++) { + if (codec_info_list[i].codec_card_late_probe) { + ret = codec_info_list[i].codec_card_late_probe(card); + if (ret < 0) + return ret; + } + } + return ret; +} +EXPORT_SYMBOL_NS(asoc_sdw_card_late_probe, SND_SOC_SDW_UTILS); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 59f8b622d52e30c5be7f14212c26ca37e810ece9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:12 +0530 Subject: [PATCH 066/410] ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() To make it generic, refactor existing implementation for init_dai_link() and init_simple_dai_link() as mentioned below. - Move init_dai_link() and init_simple_dai_link() to common place holder - Rename the functions with "asoc_sdw" as prefix. - Pass the platform specific 'platform_component' structure and its size as arguments for init_simple_dai_link() function and allocate one more extra dlc for platform component. - Pass the 'platform_component' and 'num_platforms' as arguments for init_dai_link(). Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-6-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 16 +++++ sound/soc/intel/boards/sof_sdw.c | 105 +++++++++------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 54 ++++++++++++++ 3 files changed, 104 insertions(+), 71 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 14e21a992f56..e366b7968c2d 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -124,6 +124,22 @@ void asoc_sdw_mc_dailink_exit_loop(struct snd_soc_card *card); int asoc_sdw_card_late_probe(struct snd_soc_card *card); +void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, + int *be_id, char *name, int playback, int capture, + struct snd_soc_dai_link_component *cpus, int cpus_num, + struct snd_soc_dai_link_component *platform_component, + int num_platforms, struct snd_soc_dai_link_component *codecs, + int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops); + +int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, + int *be_id, char *name, int playback, int capture, + const char *cpu_dai_name, const char *platform_comp_name, + int num_platforms, const char *codec_name, + const char *codec_dai_name, + int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops); + int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd); /* DMIC support */ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 65b15f594aed..d258728d64cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -557,54 +557,6 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -static void init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, - int *be_id, char *name, int playback, int capture, - struct snd_soc_dai_link_component *cpus, int cpus_num, - struct snd_soc_dai_link_component *codecs, int codecs_num, - int (*init)(struct snd_soc_pcm_runtime *rtd), - const struct snd_soc_ops *ops) -{ - dev_dbg(dev, "create dai link %s, id %d\n", name, *be_id); - dai_links->id = (*be_id)++; - dai_links->name = name; - dai_links->platforms = platform_component; - dai_links->num_platforms = ARRAY_SIZE(platform_component); - dai_links->no_pcm = 1; - dai_links->cpus = cpus; - dai_links->num_cpus = cpus_num; - dai_links->codecs = codecs; - dai_links->num_codecs = codecs_num; - dai_links->dpcm_playback = playback; - dai_links->dpcm_capture = capture; - dai_links->init = init; - dai_links->ops = ops; -} - -static int init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, - int *be_id, char *name, int playback, int capture, - const char *cpu_dai_name, - const char *codec_name, const char *codec_dai_name, - int (*init)(struct snd_soc_pcm_runtime *rtd), - const struct snd_soc_ops *ops) -{ - struct snd_soc_dai_link_component *dlc; - - /* Allocate two DLCs one for the CPU, one for the CODEC */ - dlc = devm_kcalloc(dev, 2, sizeof(*dlc), GFP_KERNEL); - if (!dlc || !name || !cpu_dai_name || !codec_name || !codec_dai_name) - return -ENOMEM; - - dlc[0].dai_name = cpu_dai_name; - - dlc[1].name = codec_name; - dlc[1].dai_name = codec_dai_name; - - init_dai_link(dev, dai_links, be_id, name, playback, capture, - &dlc[0], 1, &dlc[1], 1, init, ops); - - return 0; -} - struct sof_sdw_endpoint { struct list_head list; @@ -897,9 +849,10 @@ static int create_sdw_dailink(struct snd_soc_card *card, playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); capture = (stream == SNDRV_PCM_STREAM_CAPTURE); - init_dai_link(dev, *dai_links, be_id, name, playback, capture, - cpus, num_cpus, codecs, num_codecs, - asoc_sdw_rtd_init, &sdw_ops); + asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, + cpus, num_cpus, platform_component, + ARRAY_SIZE(platform_component), codecs, num_codecs, + asoc_sdw_rtd_init, &sdw_ops); /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations @@ -969,10 +922,12 @@ static int create_ssp_dailinks(struct snd_soc_card *card, int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; - ret = init_simple_dai_link(dev, *dai_links, be_id, name, - playback, capture, cpu_dai_name, - codec_name, ssp_info->dais[0].dai_name, - NULL, ssp_info->ops); + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, + playback, capture, cpu_dai_name, + platform_component->name, + ARRAY_SIZE(platform_component), codec_name, + ssp_info->dais[0].dai_name, NULL, + ssp_info->ops); if (ret) return ret; @@ -992,20 +947,24 @@ static int create_dmic_dailinks(struct snd_soc_card *card, struct device *dev = card->dev; int ret; - ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic01", - 0, 1, // DMIC only supports capture - "DMIC01 Pin", "dmic-codec", "dmic-hifi", - asoc_sdw_dmic_init, NULL); + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, "dmic01", + 0, 1, // DMIC only supports capture + "DMIC01 Pin", platform_component->name, + ARRAY_SIZE(platform_component), + "dmic-codec", "dmic-hifi", + asoc_sdw_dmic_init, NULL); if (ret) return ret; (*dai_links)++; - ret = init_simple_dai_link(dev, *dai_links, be_id, "dmic16k", - 0, 1, // DMIC only supports capture - "DMIC16k Pin", "dmic-codec", "dmic-hifi", - /* don't call asoc_sdw_dmic_init() twice */ - NULL, NULL); + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, "dmic16k", + 0, 1, // DMIC only supports capture + "DMIC16k Pin", platform_component->name, + ARRAY_SIZE(platform_component), + "dmic-codec", "dmic-hifi", + /* don't call asoc_sdw_dmic_init() twice */ + NULL, NULL); if (ret) return ret; @@ -1037,10 +996,12 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, codec_dai_name = "snd-soc-dummy-dai"; } - ret = init_simple_dai_link(dev, *dai_links, be_id, name, - 1, 0, // HDMI only supports playback - cpu_dai_name, codec_name, codec_dai_name, - i == 0 ? sof_sdw_hdmi_init : NULL, NULL); + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, + 1, 0, // HDMI only supports playback + cpu_dai_name, platform_component->name, + ARRAY_SIZE(platform_component), + codec_name, codec_dai_name, + i == 0 ? sof_sdw_hdmi_init : NULL, NULL); if (ret) return ret; @@ -1060,9 +1021,11 @@ static int create_bt_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); int ret; - ret = init_simple_dai_link(dev, *dai_links, be_id, name, - 1, 1, cpu_dai_name, snd_soc_dummy_dlc.name, - snd_soc_dummy_dlc.dai_name, NULL, NULL); + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, + 1, 1, cpu_dai_name, platform_component->name, + ARRAY_SIZE(platform_component), + snd_soc_dummy_dlc.name, snd_soc_dummy_dlc.dai_name, + NULL, NULL); if (ret) return ret; diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 613ecc3bed92..6183629d1754 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -932,5 +932,59 @@ int asoc_sdw_card_late_probe(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(asoc_sdw_card_late_probe, SND_SOC_SDW_UTILS); +void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, + int *be_id, char *name, int playback, int capture, + struct snd_soc_dai_link_component *cpus, int cpus_num, + struct snd_soc_dai_link_component *platform_component, + int num_platforms, struct snd_soc_dai_link_component *codecs, + int codecs_num, int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops) +{ + dev_dbg(dev, "create dai link %s, id %d\n", name, *be_id); + dai_links->id = (*be_id)++; + dai_links->name = name; + dai_links->platforms = platform_component; + dai_links->num_platforms = num_platforms; + dai_links->no_pcm = 1; + dai_links->cpus = cpus; + dai_links->num_cpus = cpus_num; + dai_links->codecs = codecs; + dai_links->num_codecs = codecs_num; + dai_links->dpcm_playback = playback; + dai_links->dpcm_capture = capture; + dai_links->init = init; + dai_links->ops = ops; +} +EXPORT_SYMBOL_NS(asoc_sdw_init_dai_link, SND_SOC_SDW_UTILS); + +int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *dai_links, + int *be_id, char *name, int playback, int capture, + const char *cpu_dai_name, const char *platform_comp_name, + int num_platforms, const char *codec_name, + const char *codec_dai_name, + int (*init)(struct snd_soc_pcm_runtime *rtd), + const struct snd_soc_ops *ops) +{ + struct snd_soc_dai_link_component *dlc; + + /* Allocate three DLCs one for the CPU, one for platform and one for the CODEC */ + dlc = devm_kcalloc(dev, 3, sizeof(*dlc), GFP_KERNEL); + if (!dlc || !name || !cpu_dai_name || !platform_comp_name || !codec_name || !codec_dai_name) + return -ENOMEM; + + dlc[0].dai_name = cpu_dai_name; + dlc[1].name = platform_comp_name; + + dlc[2].name = codec_name; + dlc[2].dai_name = codec_dai_name; + + asoc_sdw_init_dai_link(dev, dai_links, be_id, name, playback, capture, + &dlc[0], 1, &dlc[1], num_platforms, + &dlc[2], 1, init, ops); + + return 0; +} +EXPORT_SYMBOL_NS(asoc_sdw_init_simple_dai_link, SND_SOC_SDW_UTILS); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 0b8f009ae92fa94ab3c0ca881fce02c336056a73 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:13 +0530 Subject: [PATCH 067/410] ASoC: soc-acpi: add pci revision id field in mach params structure Few IP's may have same PCI vendor id and device id and different revision id. Add revision id field to the 'snd_soc_acpi_mach_params' so that using this field platform specific implementation can be added in machine driver. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-7-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 38ccec4e3fcd..b6d301946244 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -70,6 +70,7 @@ static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) * @dai_drivers: pointer to dai_drivers, used e.g. in nocodec mode * @subsystem_vendor: optional PCI SSID vendor value * @subsystem_device: optional PCI SSID device value + * @subsystem_rev: optional PCI SSID revision value * @subsystem_id_set: true if a value has been written to * subsystem_vendor and subsystem_device. */ @@ -86,6 +87,7 @@ struct snd_soc_acpi_mach_params { struct snd_soc_dai_driver *dai_drivers; unsigned short subsystem_vendor; unsigned short subsystem_device; + unsigned short subsystem_rev; bool subsystem_id_set; }; From 57677ccde7522170c50eb44258d54c6584356f47 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:14 +0530 Subject: [PATCH 068/410] ASoC: amd: acp: add soundwire machines for acp6.3 based platform Add Soundwire machines for acp6.3 based platform. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-8-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 4 ++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 ++++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + sound/soc/sof/amd/Kconfig | 1 + 5 files changed, 98 insertions(+) create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 30590a23ad63..19859b1b73c5 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -13,6 +13,10 @@ config SND_SOC_AMD_ACP_COMMON This option enables common modules for Audio-Coprocessor i.e. ACP IP block on AMD platforms. +config SND_SOC_ACPI_AMD_MATCH + tristate + select SND_SOC_ACPI if ACPI + if SND_SOC_AMD_ACP_COMMON config SND_SOC_AMD_ACP_PDM diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index b068bf1f920e..516a44f3ffb6 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -22,6 +22,7 @@ snd-acp70-y := acp70.o snd-acp-mach-y := acp-mach-common.o snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o snd-acp-sof-mach-y := acp-sof-mach.o +snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -38,3 +39,4 @@ obj-$(CONFIG_SND_AMD_SOUNDWIRE_ACPI) += snd-amd-sdw-acpi.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_MACH) += snd-acp-sof-mach.o +obj-$(CONFIG_SND_SOC_ACPI_AMD_MATCH) += snd-soc-acpi-amd-match.o diff --git a/sound/soc/amd/acp/amd-acp63-acpi-match.c b/sound/soc/amd/acp/amd-acp63-acpi-match.c new file mode 100644 index 000000000000..be9367913073 --- /dev/null +++ b/sound/soc/amd/acp/amd-acp63-acpi-match.c @@ -0,0 +1,90 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * amd-acp63-acpi-match.c - tables and support for ACP 6.3 platform + * ACPI enumeration. + * + * Copyright 2024 Advanced Micro Devices, Inc. + */ + +#include +#include "../mach-config.h" + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0 +}; + +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1 +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1 +}; + +static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + }, + { + .adr = 0x000030025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "rt1316-1" + }, + { + .adr = 0x000032025D131601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "rt1316-2" + }, +}; + +static const struct snd_soc_acpi_adr_device rt714_adr[] = { + { + .adr = 0x130025d071401ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt714" + } +}; + +static const struct snd_soc_acpi_link_adr acp63_4_in_1_sdca[] = { + { .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_rt1316_group_adr), + .adr_d = rt711_rt1316_group_adr, + }, + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt714_adr), + .adr_d = rt714_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_sdw_machines[] = { + { + .link_mask = BIT(0) | BIT(1), + .links = acp63_4_in_1_sdca, + .drv_name = "amd_sof_sdw", + .sof_tplg_filename = "sof-acp_6_3-rt711-l0-rt1316-l0-rt714-l1.tplg", + .fw_filename = "sof-acp_6_3.ri", + }, + {}, +}; +EXPORT_SYMBOL(snd_soc_acpi_amd_acp63_sof_sdw_machines); + +MODULE_DESCRIPTION("AMD ACP6.3 tables and support for ACPI enumeration"); +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); diff --git a/sound/soc/amd/mach-config.h b/sound/soc/amd/mach-config.h index 7af0f9cf3921..32aa8a6931f4 100644 --- a/sound/soc/amd/mach-config.h +++ b/sound/soc/amd/mach-config.h @@ -23,6 +23,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_amd_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_vangogh_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_sdw_machines[]; struct config_entry { u32 flags; diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 2729c6eb3feb..848c031ed5fb 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -23,6 +23,7 @@ config SND_SOC_SOF_AMD_COMMON select SND_AMD_ACP_CONFIG select SND_SOC_SOF_XTENSA select SND_SOC_SOF_ACP_PROBES + select SND_SOC_ACPI_AMD_MATCH select SND_SOC_ACPI if ACPI help This option is not user-selectable but automatically handled by From 15049b6a6c19d799095e5dbe9a4772862b11ee23 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:15 +0530 Subject: [PATCH 069/410] ASoC: SOF: amd: add alternate machines for acp6.3 based platform Add SoundWire machines as alternate machines for acp6.3 based platform. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-9-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/pci-acp63.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index fc8984447365..54d42f83ce9e 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -48,6 +48,7 @@ static const struct sof_amd_acp_desc acp63_chip_info = { static const struct sof_dev_desc acp63_desc = { .machines = snd_soc_acpi_amd_acp63_sof_machines, + .alt_machines = snd_soc_acpi_amd_acp63_sof_sdw_machines, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, From b7cdb4a89cc864ead6470a57947815c49870b09a Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:16 +0530 Subject: [PATCH 070/410] ASoC: SOF: amd: update mach params subsystem_rev variable Add pci_rev variable in acp sof driver private data structure and assign this value to mach_params structure subsystem_rev variable. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-10-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-common.c | 3 +++ sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + 3 files changed, 5 insertions(+) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index 81bb93e98358..dbcaac84cb73 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -153,6 +153,7 @@ static struct snd_soc_acpi_mach *amd_sof_sdw_machine_select(struct snd_sof_dev * break; } if (mach && mach->link_mask) { + mach->mach_params.subsystem_rev = acp_data->pci_rev; mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; mach->mach_params.platform = dev_name(sdev->dev); @@ -173,6 +174,7 @@ static struct snd_soc_acpi_mach *amd_sof_sdw_machine_select(struct snd_sof_dev * struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev) { struct snd_sof_pdata *sof_pdata = sdev->pdata; + struct acp_dev_data *acp_data = sdev->pdata->hw_pdata; const struct sof_dev_desc *desc = sof_pdata->desc; struct snd_soc_acpi_mach *mach = NULL; @@ -186,6 +188,7 @@ struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev) } } + mach->mach_params.subsystem_rev = acp_data->pci_rev; sof_pdata->tplg_filename = mach->sof_tplg_filename; sof_pdata->fw_filename = mach->fw_filename; diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 74fd5f2b148b..7b122656efd1 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -695,6 +695,7 @@ int amd_sof_acp_probe(struct snd_sof_dev *sdev) pci_set_master(pci); adata->addr = addr; adata->reg_range = chip->reg_end_addr - chip->reg_start_addr; + adata->pci_rev = pci->revision; mutex_init(&adata->acp_lock); sdev->pdata->hw_pdata = adata; adata->smn_dev = pci_get_device(PCI_VENDOR_ID_AMD, chip->host_bridge_id, NULL); diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 87e79d500865..ec9170b3f068 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -251,6 +251,7 @@ struct acp_dev_data { bool is_dram_in_use; bool is_sram_in_use; bool sdw_en_stat; + unsigned int pci_rev; }; void memcpy_to_scratch(struct snd_sof_dev *sdev, u32 offset, unsigned int *src, size_t bytes); From cb8ea62e6402067ba092d4c1d66a9440513a572b Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 1 Aug 2024 16:48:17 +0530 Subject: [PATCH 071/410] ASoC: amd/sdw_utils: add sof based soundwire generic machine driver Add sof based Soundwire generic driver for amd platforms. Currently support added for ACP6.3 based platforms. Link: https://github.com/thesofproject/linux/pull/5068 Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240801111821.18076-11-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/Kconfig | 18 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 742 +++++++++++++++++++++++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 44 ++ 4 files changed, 806 insertions(+) create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 19859b1b73c5..88391e4c17e3 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -119,6 +119,24 @@ config SND_SOC_AMD_SOF_MACH help This option enables SOF sound card support for ACP audio. +config SND_SOC_AMD_SOF_SDW_MACH + tristate "AMD SOF Soundwire Machine Driver Support" + depends on X86 && PCI && ACPI + depends on SOUNDWIRE + select SND_SOC_SDW_UTILS + select SND_SOC_DMIC + select SND_SOC_RT711_SDW + select SND_SOC_RT711_SDCA_SDW + select SND_SOC_RT1316_SDW + select SND_SOC_RT715_SDW + select SND_SOC_RT715_SDCA_SDW + help + This option enables SOF sound card support for SoundWire enabled + AMD platforms along with ACP PDM controller. + Say Y if you want to enable SoundWire based machine driver support + on AMD platform. + If unsure select "N". + endif # SND_SOC_AMD_ACP_COMMON config SND_AMD_SOUNDWIRE_ACPI diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 516a44f3ffb6..82cf5d180b3a 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -23,6 +23,7 @@ snd-acp-mach-y := acp-mach-common.o snd-acp-legacy-mach-y := acp-legacy-mach.o acp3x-es83xx/acp3x-es83xx.o snd-acp-sof-mach-y := acp-sof-mach.o snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o +snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -40,3 +41,4 @@ obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o obj-$(CONFIG_SND_SOC_AMD_SOF_MACH) += snd-acp-sof-mach.o obj-$(CONFIG_SND_SOC_ACPI_AMD_MATCH) += snd-soc-acpi-amd-match.o +obj-$(CONFIG_SND_SOC_AMD_SOF_SDW_MACH) += snd-acp-sdw-sof-mach.o diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c new file mode 100644 index 000000000000..3419675e45a9 --- /dev/null +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -0,0 +1,742 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright(c) 2024 Advanced Micro Devices, Inc. + +/* + * acp-sdw-sof-mach - ASoC Machine driver for AMD SoundWire platforms + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "soc_amd_sdw_common.h" +#include "../../codecs/rt711.h" + +static unsigned long sof_sdw_quirk = RT711_JD1; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, int, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + +static void log_quirks(struct device *dev) +{ + if (SOC_JACK_JDSRC(sof_sdw_quirk)) + dev_dbg(dev, "quirk realtek,jack-detect-source %ld\n", + SOC_JACK_JDSRC(sof_sdw_quirk)); + if (sof_sdw_quirk & ASOC_SDW_ACP_DMIC) + dev_dbg(dev, "quirk SOC_SDW_ACP_DMIC enabled\n"); +} + +static int sof_sdw_quirk_cb(const struct dmi_system_id *id) +{ + sof_sdw_quirk = (unsigned long)id->driver_data; + return 1; +} + +static const struct dmi_system_id sof_sdw_quirk_table[] = { + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AMD"), + DMI_MATCH(DMI_PRODUCT_NAME, "Birman-PHX"), + }, + .driver_data = (void *)RT711_JD2, + }, + {} +}; + +static struct snd_soc_dai_link_component platform_component[] = { + { + /* name might be overridden during probe */ + .name = "0000:04:00.5", + } +}; + +static const struct snd_soc_ops sdw_ops = { + .startup = asoc_sdw_startup, + .prepare = asoc_sdw_prepare, + .trigger = asoc_sdw_trigger, + .hw_params = asoc_sdw_hw_params, + .hw_free = asoc_sdw_hw_free, + .shutdown = asoc_sdw_shutdown, +}; + +/* + * get BE dailink number and CPU DAI number based on sdw link adr. + * Since some sdw slaves may be aggregated, the CPU DAI number + * may be larger than the number of BE dailinks. + */ +static int get_dailink_info(struct device *dev, + const struct snd_soc_acpi_link_adr *adr_link, + int *sdw_be_num, int *codecs_num) +{ + bool group_visited[AMD_SDW_MAX_GROUPS]; + int i; + int j; + + *sdw_be_num = 0; + + if (!adr_link) + return -EINVAL; + + for (i = 0; i < AMD_SDW_MAX_GROUPS; i++) + group_visited[i] = false; + + for (; adr_link->num_adr; adr_link++) { + const struct snd_soc_acpi_endpoint *endpoint; + struct asoc_sdw_codec_info *codec_info; + int stream; + u64 adr; + + /* make sure the link mask has a single bit set */ + if (!is_power_of_2(adr_link->mask)) + return -EINVAL; + + for (i = 0; i < adr_link->num_adr; i++) { + adr = adr_link->adr_d[i].adr; + codec_info = asoc_sdw_find_codec_info_part(adr); + if (!codec_info) + return -EINVAL; + + *codecs_num += codec_info->dai_num; + + if (!adr_link->adr_d[i].name_prefix) { + dev_err(dev, "codec 0x%llx does not have a name prefix\n", + adr_link->adr_d[i].adr); + return -EINVAL; + } + + endpoint = adr_link->adr_d[i].endpoints; + if (endpoint->aggregated && !endpoint->group_id) { + dev_err(dev, "invalid group id on link %x\n", + adr_link->mask); + return -EINVAL; + } + + for (j = 0; j < codec_info->dai_num; j++) { + /* count DAI number for playback and capture */ + for_each_pcm_streams(stream) { + if (!codec_info->dais[j].direction[stream]) + continue; + + /* count BE for each non-aggregated slave or group */ + if (!endpoint->aggregated || + !group_visited[endpoint->group_id]) + (*sdw_be_num)++; + } + } + + if (endpoint->aggregated) + group_visited[endpoint->group_id] = true; + } + } + return 0; +} + +static int fill_sdw_codec_dlc(struct device *dev, + const struct snd_soc_acpi_link_adr *adr_link, + struct snd_soc_dai_link_component *codec, + int adr_index, int dai_index) +{ + u64 adr = adr_link->adr_d[adr_index].adr; + struct asoc_sdw_codec_info *codec_info; + + codec_info = asoc_sdw_find_codec_info_part(adr); + if (!codec_info) + return -EINVAL; + + codec->name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, adr_index); + if (!codec->name) + return -ENOMEM; + + codec->dai_name = codec_info->dais[dai_index].dai_name; + dev_err(dev, "codec->dai_name:%s\n", codec->dai_name); + return 0; +} + +static int set_codec_init_func(struct snd_soc_card *card, + const struct snd_soc_acpi_link_adr *adr_link, + struct snd_soc_dai_link *dai_links, + bool playback, int group_id, int adr_index, int dai_index) +{ + int i = adr_index; + + do { + /* + * Initialize the codec. If codec is part of an aggregated + * group (group_id>0), initialize all codecs belonging to + * same group. + * The first link should start with adr_link->adr_d[adr_index] + * because that is the device that we want to initialize and + * we should end immediately if it is not aggregated (group_id=0) + */ + for ( ; i < adr_link->num_adr; i++) { + struct asoc_sdw_codec_info *codec_info; + + codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; + + /* The group_id is > 0 iff the codec is aggregated */ + if (adr_link->adr_d[i].endpoints->group_id != group_id) + continue; + if (codec_info->dais[dai_index].init) + codec_info->dais[dai_index].init(card, + dai_links, + codec_info, + playback); + + if (!group_id) + return 0; + } + + i = 0; + adr_link++; + } while (adr_link->mask); + + return 0; +} + +/* + * check endpoint status in slaves and gather link ID for all slaves in + * the same group to generate different CPU DAI. Now only support + * one sdw link with all slaves set with only single group id. + * + * one slave on one sdw link with aggregated = 0 + * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI + * + * two or more slaves on one sdw link with aggregated = 1 + * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs + */ +static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, + struct device *dev, int *cpu_dai_id, int *cpu_dai_num, + int *codec_num, unsigned int *group_id, + int adr_index) +{ + int i; + + if (!adr_link->adr_d[adr_index].endpoints->aggregated) { + cpu_dai_id[0] = ffs(adr_link->mask) - 1; + *cpu_dai_num = 1; + *codec_num = 1; + *group_id = 0; + return 0; + } + + *codec_num = 0; + *cpu_dai_num = 0; + *group_id = adr_link->adr_d[adr_index].endpoints->group_id; + + /* Count endpoints with the same group_id in the adr_link */ + for (; adr_link && adr_link->num_adr; adr_link++) { + unsigned int link_codecs = 0; + + for (i = 0; i < adr_link->num_adr; i++) { + if (adr_link->adr_d[i].endpoints->aggregated && + adr_link->adr_d[i].endpoints->group_id == *group_id) + link_codecs++; + } + + if (link_codecs) { + *codec_num += link_codecs; + + if (*cpu_dai_num >= ACP63_SDW_MAX_CPU_DAIS) { + dev_err(dev, "cpu_dai_id array overflowed\n"); + return -EINVAL; + } + + cpu_dai_id[(*cpu_dai_num)++] = ffs(adr_link->mask) - 1; + } + } + + return 0; +} + +static int get_acp63_cpu_pin_id(u32 sdw_link_id, int be_id, int *cpu_pin_id, struct device *dev) +{ + switch (sdw_link_id) { + case AMD_SDW0: + switch (be_id) { + case SOC_SDW_JACK_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO0_TX; + break; + case SOC_SDW_JACK_IN_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO0_RX; + break; + case SOC_SDW_AMP_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO1_TX; + break; + case SOC_SDW_AMP_IN_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO1_RX; + break; + case SOC_SDW_DMIC_DAI_ID: + *cpu_pin_id = ACP63_SW0_AUDIO2_RX; + break; + default: + dev_err(dev, "Invalid be id:%d\n", be_id); + return -EINVAL; + } + break; + case AMD_SDW1: + switch (be_id) { + case SOC_SDW_JACK_OUT_DAI_ID: + case SOC_SDW_AMP_OUT_DAI_ID: + *cpu_pin_id = ACP63_SW1_AUDIO0_TX; + break; + case SOC_SDW_JACK_IN_DAI_ID: + case SOC_SDW_AMP_IN_DAI_ID: + case SOC_SDW_DMIC_DAI_ID: + *cpu_pin_id = ACP63_SW1_AUDIO0_RX; + break; + default: + dev_err(dev, "invalid be_id:%d\n", be_id); + return -EINVAL; + } + break; + default: + dev_err(dev, "Invalid link id:%d\n", sdw_link_id); + return -EINVAL; + } + return 0; +} + +static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; + +static int create_sdw_dailink(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, + const struct snd_soc_acpi_link_adr *adr_link, + struct snd_soc_codec_conf **codec_conf, + int *be_id, int adr_index, int dai_index) +{ + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct amd_mc_ctx *amd_ctx = (struct amd_mc_ctx *)ctx->private; + struct device *dev = card->dev; + const struct snd_soc_acpi_link_adr *adr_link_next; + struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; + struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *cpus; + struct asoc_sdw_codec_info *codec_info; + int cpu_dai_id[ACP63_SDW_MAX_CPU_DAIS]; + int cpu_dai_num; + unsigned int group_id; + unsigned int sdw_link_id; + int codec_dlc_index = 0; + int codec_num; + int stream; + int i = 0; + int j, k; + int ret; + int cpu_pin_id; + + ret = get_slave_info(adr_link, dev, cpu_dai_id, &cpu_dai_num, &codec_num, + &group_id, adr_index); + if (ret) + return ret; + codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + sdw_codec_ch_maps = devm_kcalloc(dev, codec_num, + sizeof(*sdw_codec_ch_maps), GFP_KERNEL); + if (!sdw_codec_ch_maps) + return -ENOMEM; + + /* generate codec name on different links in the same group */ + j = adr_index; + for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr && + i < cpu_dai_num; adr_link_next++) { + /* skip the link excluded by this processed group */ + if (cpu_dai_id[i] != ffs(adr_link_next->mask) - 1) + continue; + + /* j reset after loop, adr_index only applies to first link */ + for (k = 0 ; (j < adr_link_next->num_adr) && (k < codec_num) ; j++, k++) { + const struct snd_soc_acpi_endpoint *endpoints; + + endpoints = adr_link_next->adr_d[j].endpoints; + if (group_id && (!endpoints->aggregated || + endpoints->group_id != group_id)) + continue; + + /* sanity check */ + if (*codec_conf >= card->codec_conf + card->num_configs) { + dev_err(dev, "codec_conf array overflowed\n"); + return -EINVAL; + } + + ret = fill_sdw_codec_dlc(dev, adr_link_next, + &codecs[codec_dlc_index], + j, dai_index); + if (ret) + return ret; + (*codec_conf)->dlc = codecs[codec_dlc_index]; + (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix; + + sdw_codec_ch_maps[codec_dlc_index].cpu = i; + sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index; + + codec_dlc_index++; + (*codec_conf)++; + } + j = 0; + + /* check next link to create codec dai in the processed group */ + i++; + } + + /* find codec info to create BE DAI */ + codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[adr_index].adr); + if (!codec_info) + return -EINVAL; + + ctx->ignore_internal_dmic |= codec_info->ignore_internal_dmic; + + sdw_link_id = (adr_link->adr_d[adr_index].adr) >> 48; + for_each_pcm_streams(stream) { + char *name, *cpu_name; + int playback, capture; + static const char * const sdw_stream_name[] = { + "SDW%d-PIN%d-PLAYBACK", + "SDW%d-PIN%d-CAPTURE", + "SDW%d-PIN%d-PLAYBACK-%s", + "SDW%d-PIN%d-CAPTURE-%s", + }; + + if (!codec_info->dais[dai_index].direction[stream]) + continue; + + *be_id = codec_info->dais[dai_index].dailink[stream]; + if (*be_id < 0) { + dev_err(dev, "Invalid dailink id %d\n", *be_id); + return -EINVAL; + } + switch (amd_ctx->acp_rev) { + case ACP63_PCI_REV: + ret = get_acp63_cpu_pin_id(sdw_link_id, *be_id, &cpu_pin_id, dev); + if (ret) + return ret; + break; + default: + return -EINVAL; + } + /* create stream name according to first link id */ + if (ctx->append_dai_type) { + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream + 2], sdw_link_id, cpu_pin_id, + type_strings[codec_info->dais[dai_index].dai_type]); + } else { + name = devm_kasprintf(dev, GFP_KERNEL, + sdw_stream_name[stream], sdw_link_id, cpu_pin_id); + } + if (!name) + return -ENOMEM; + + cpus = devm_kcalloc(dev, cpu_dai_num, sizeof(*cpus), GFP_KERNEL); + if (!cpus) + return -ENOMEM; + /* + * generate CPU DAI name base on the sdw link ID and + * cpu pin id according to sdw dai driver. + */ + for (k = 0; k < cpu_dai_num; k++) { + cpu_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", sdw_link_id, cpu_pin_id); + if (!cpu_name) + return -ENOMEM; + + cpus[k].dai_name = cpu_name; + } + + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); + capture = (stream == SNDRV_PCM_STREAM_CAPTURE); + asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, + playback, capture, + cpus, cpu_dai_num, + platform_component, ARRAY_SIZE(platform_component), + codecs, codec_num, + asoc_sdw_rtd_init, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + (*dai_links)->nonatomic = true; + (*dai_links)->ch_maps = sdw_codec_ch_maps; + + ret = set_codec_init_func(card, adr_link, *dai_links, + playback, group_id, adr_index, dai_index); + if (ret < 0) { + dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id); + return ret; + } + + (*dai_links)++; + } + return 0; +} + +static int create_dmic_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id) +{ + struct device *dev = card->dev; + int ret; + + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, "acp-dmic-codec", + 0, 1, // DMIC only supports capture + "acp-sof-dmic", platform_component->name, + ARRAY_SIZE(platform_component), + "dmic-codec", "dmic-hifi", + asoc_sdw_dmic_init, NULL); + if (ret) + return ret; + + (*dai_links)++; + + return 0; +} + +static int sof_card_dai_links_create(struct snd_soc_card *card) +{ + struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(card->dev); + int sdw_be_num = 0, dmic_num = 0; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; + struct snd_soc_codec_conf *codec_conf; + int codec_conf_num = 0; + bool group_generated[AMD_SDW_MAX_GROUPS] = { }; + struct snd_soc_dai_link *dai_links; + struct asoc_sdw_codec_info *codec_info; + int num_links; + int i, j, be_id = 0; + int ret; + + ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); + if (ret < 0) { + dev_err(dev, "failed to get sdw link info %d\n", ret); + return ret; + } + + /* enable dmic */ + if (sof_sdw_quirk & ASOC_SDW_ACP_DMIC || mach_params->dmic_num) + dmic_num = 1; + + dev_dbg(dev, "sdw %d, dmic %d", sdw_be_num, dmic_num); + + /* allocate BE dailinks */ + num_links = sdw_be_num + dmic_num; + dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL); + if (!dai_links) + return -ENOMEM; + + /* allocate codec conf, will be populated when dailinks are created */ + codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf), + GFP_KERNEL); + if (!codec_conf) + return -ENOMEM; + + card->dai_link = dai_links; + card->num_links = num_links; + card->codec_conf = codec_conf; + card->num_configs = codec_conf_num; + + /* SDW */ + if (!sdw_be_num) + goto DMIC; + + for (; adr_link->num_adr; adr_link++) { + /* + * If there are two or more different devices on the same sdw link, we have to + * append the codec type to the dai link name to prevent duplicated dai link name. + * The same type devices on the same sdw link will be in the same + * snd_soc_acpi_adr_device array. They won't be described in different adr_links. + */ + for (i = 0; i < adr_link->num_adr; i++) { + /* find codec info to get dai_num */ + codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; + if (codec_info->dai_num > 1) { + ctx->append_dai_type = true; + goto out; + } + for (j = 0; j < i; j++) { + if ((SDW_PART_ID(adr_link->adr_d[i].adr) != + SDW_PART_ID(adr_link->adr_d[j].adr)) || + (SDW_MFG_ID(adr_link->adr_d[i].adr) != + SDW_MFG_ID(adr_link->adr_d[j].adr))) { + ctx->append_dai_type = true; + goto out; + } + } + } + } +out: + + /* generate DAI links by each sdw link */ + for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) { + for (i = 0; i < adr_link->num_adr; i++) { + const struct snd_soc_acpi_endpoint *endpoint; + + endpoint = adr_link->adr_d[i].endpoints; + + /* this group has been generated */ + if (endpoint->aggregated && + group_generated[endpoint->group_id]) + continue; + + /* find codec info to get dai_num */ + codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); + if (!codec_info) + return -EINVAL; + + for (j = 0; j < codec_info->dai_num ; j++) { + int current_be_id; + + ret = create_sdw_dailink(card, &dai_links, adr_link, + &codec_conf, ¤t_be_id, + i, j); + if (ret < 0) { + dev_err(dev, + "failed to create dai link %d on 0x%x\n", + j, codec_info->part_id); + return ret; + } + /* Update the be_id to match the highest ID used for SDW link */ + if (be_id < current_be_id) + be_id = current_be_id; + } + + if (endpoint->aggregated) + group_generated[endpoint->group_id] = true; + } + } + +DMIC: + /* dmic */ + if (dmic_num > 0) { + if (ctx->ignore_internal_dmic) { + dev_warn(dev, "Ignoring ACP DMIC\n"); + } else { + be_id = SOC_SDW_DMIC_DAI_ID; + ret = create_dmic_dailinks(card, &dai_links, &be_id); + if (ret) + return ret; + } + } + + WARN_ON(dai_links != card->dai_link + card->num_links); + return 0; +} + +/* SoC card */ +static const char sdw_card_long_name[] = "AMD Soundwire SOF"; + +static int mc_probe(struct platform_device *pdev) +{ + struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); + struct snd_soc_card *card; + struct amd_mc_ctx *amd_ctx; + struct asoc_sdw_mc_private *ctx; + int amp_num = 0, i; + int ret; + + amd_ctx = devm_kzalloc(&pdev->dev, sizeof(*amd_ctx), GFP_KERNEL); + if (!amd_ctx) + return -ENOMEM; + + amd_ctx->acp_rev = mach->mach_params.subsystem_rev; + amd_ctx->max_sdw_links = ACP63_SDW_MAX_LINKS; + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); + if (!ctx) + return -ENOMEM; + ctx->codec_info_list_count = asoc_sdw_get_codec_info_list_count(); + ctx->private = amd_ctx; + card = &ctx->card; + card->dev = &pdev->dev; + card->name = "amd-soundwire", + card->owner = THIS_MODULE, + card->late_probe = asoc_sdw_card_late_probe, + + snd_soc_card_set_drvdata(card, ctx); + + dmi_check_system(sof_sdw_quirk_table); + + if (quirk_override != -1) { + dev_info(card->dev, "Overriding quirk 0x%lx => 0x%x\n", + sof_sdw_quirk, quirk_override); + sof_sdw_quirk = quirk_override; + } + + log_quirks(card->dev); + + ctx->mc_quirk = sof_sdw_quirk; + /* reset amp_num to ensure amp_num++ starts from 0 in each probe */ + for (i = 0; i < ctx->codec_info_list_count; i++) + codec_info_list[i].amp_num = 0; + + ret = sof_card_dai_links_create(card); + if (ret < 0) + return ret; + + /* + * the default amp_num is zero for each codec and + * amp_num will only be increased for active amp + * codecs on used platform + */ + for (i = 0; i < ctx->codec_info_list_count; i++) + amp_num += codec_info_list[i].amp_num; + + card->components = devm_kasprintf(card->dev, GFP_KERNEL, + " cfg-amp:%d", amp_num); + if (!card->components) + return -ENOMEM; + + card->long_name = sdw_card_long_name; + + /* Register the card */ + ret = devm_snd_soc_register_card(card->dev, card); + if (ret) { + dev_err_probe(card->dev, ret, "snd_soc_register_card failed %d\n", ret); + asoc_sdw_mc_dailink_exit_loop(card); + return ret; + } + + platform_set_drvdata(pdev, card); + + return ret; +} + +static void mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + asoc_sdw_mc_dailink_exit_loop(card); +} + +static const struct platform_device_id mc_id_table[] = { + { "amd_sof_sdw", }, + {} +}; +MODULE_DEVICE_TABLE(platform, mc_id_table); + +static struct platform_driver sof_sdw_driver = { + .driver = { + .name = "amd_sof_sdw", + .pm = &snd_soc_pm_ops, + }, + .probe = mc_probe, + .remove_new = mc_remove, + .id_table = mc_id_table, +}; + +module_platform_driver(sof_sdw_driver); + +MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); +MODULE_AUTHOR("Vijendar Mukunda +#include +#include +#include + +#define ACP63_SDW_MAX_CPU_DAIS 8 +#define ACP63_SDW_MAX_LINKS 2 + +#define AMD_SDW_MAX_GROUPS 9 +#define ACP63_PCI_REV 0x63 +#define SOC_JACK_JDSRC(quirk) ((quirk) & GENMASK(3, 0)) +#define ASOC_SDW_FOUR_SPK BIT(4) +#define ASOC_SDW_ACP_DMIC BIT(5) + +#define AMD_SDW0 0 +#define AMD_SDW1 1 +#define ACP63_SW0_AUDIO0_TX 0 +#define ACP63_SW0_AUDIO1_TX 1 +#define ACP63_SW0_AUDIO2_TX 2 + +#define ACP63_SW0_AUDIO0_RX 3 +#define ACP63_SW0_AUDIO1_RX 4 +#define ACP63_SW0_AUDIO2_RX 5 + +#define ACP63_SW1_AUDIO0_TX 0 +#define ACP63_SW1_AUDIO0_RX 1 + +struct amd_mc_ctx { + unsigned int acp_rev; + unsigned int max_sdw_links; +}; + +#endif From a1c8929b0ebbfd7598f038ac74fb0a28f94ade8c Mon Sep 17 00:00:00 2001 From: "Rob Herring (Arm)" Date: Wed, 31 Jul 2024 13:12:57 -0600 Subject: [PATCH 072/410] ASoC: Use of_property_present() Use of_property_present() to test for property presence rather than of_get_property(). This is part of a larger effort to remove callers of of_get_property() and similar functions. of_get_property() leaks the DT property data pointer which is a problem for dynamically allocated nodes which may be freed. Signed-off-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240731191312.1710417-19-robh@kernel.org Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index af8b9d098d2d..931b430fa983 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -184,7 +184,7 @@ static int psc_i2s_of_probe(struct platform_device *op) /* Check for the codec handle. If it is not present then we * are done */ - if (!of_get_property(op->dev.of_node, "codec-handle", NULL)) + if (!of_property_present(op->dev.of_node, "codec-handle")) return 0; /* Due to errata in the dma mode; need to line up enabling diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 4bdbcd2635ef..05d59e03b1c5 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -213,7 +213,7 @@ int tegra_pcm_construct(struct snd_soc_component *component, * Fallback for backwards-compatibility with older device trees that * have the iommus property in the virtual, top-level "sound" node. */ - if (!of_get_property(dev->of_node, "iommus", NULL)) + if (!of_property_present(dev->of_node, "iommus")) dev = rtd->card->snd_card->dev; return tegra_pcm_dma_allocate(dev, rtd, tegra_pcm_hardware.buffer_bytes_max); From 69dd15a8ef0ae494179fd15023aa8172188db6b7 Mon Sep 17 00:00:00 2001 From: "Rob Herring (Arm)" Date: Wed, 31 Jul 2024 13:12:58 -0600 Subject: [PATCH 073/410] ASoC: Use of_property_read_bool() Use of_property_read_bool() to read boolean properties rather than of_get_property(). This is part of a larger effort to remove callers of of_get_property() and similar functions. of_get_property() leaks the DT property data pointer which is a problem for dynamically allocated nodes which may be freed. Signed-off-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240731191312.1710417-20-robh@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/ak4613.c | 4 ++-- sound/soc/soc-core.c | 8 ++++---- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index 551738abd1a5..de9e43185555 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -840,14 +840,14 @@ static void ak4613_parse_of(struct ak4613_priv *priv, /* Input 1 - 2 */ for (i = 0; i < 2; i++) { snprintf(prop, sizeof(prop), "asahi-kasei,in%d-single-end", i + 1); - if (!of_get_property(np, prop, NULL)) + if (!of_property_read_bool(np, prop)) priv->ic |= 1 << i; } /* Output 1 - 6 */ for (i = 0; i < 6; i++) { snprintf(prop, sizeof(prop), "asahi-kasei,out%d-single-end", i + 1); - if (!of_get_property(np, prop, NULL)) + if (!of_property_read_bool(np, prop)) priv->oc |= 1 << i; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 80bacea6bb90..20248a29d167 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3371,10 +3371,10 @@ unsigned int snd_soc_daifmt_parse_format(struct device_node *np, * SND_SOC_DAIFMT_INV_MASK area */ snprintf(prop, sizeof(prop), "%sbitclock-inversion", prefix); - bit = !!of_get_property(np, prop, NULL); + bit = of_property_read_bool(np, prop); snprintf(prop, sizeof(prop), "%sframe-inversion", prefix); - frame = !!of_get_property(np, prop, NULL); + frame = of_property_read_bool(np, prop); switch ((bit << 4) + frame) { case 0x11: @@ -3411,12 +3411,12 @@ unsigned int snd_soc_daifmt_parse_clock_provider_raw(struct device_node *np, * check "[prefix]frame-master" */ snprintf(prop, sizeof(prop), "%sbitclock-master", prefix); - bit = !!of_get_property(np, prop, NULL); + bit = of_property_read_bool(np, prop); if (bit && bitclkmaster) *bitclkmaster = of_parse_phandle(np, prop, 0); snprintf(prop, sizeof(prop), "%sframe-master", prefix); - frame = !!of_get_property(np, prop, NULL); + frame = of_property_read_bool(np, prop); if (frame && framemaster) *framemaster = of_parse_phandle(np, prop, 0); From 7dfdcde20179383d4acfee61692a22955d5a8217 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:39 +0000 Subject: [PATCH 074/410] ASoC: stm: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87o76fk51p.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/stm/stm32_i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index 46098e111142..a96aa308681a 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -823,7 +823,7 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* Enable i2s */ dev_dbg(cpu_dai->dev, "start I2S %s\n", - playback_flg ? "playback" : "capture"); + snd_pcm_direction_name(substream->stream)); cfg1_mask = I2S_CFG1_RXDMAEN | I2S_CFG1_TXDMAEN; regmap_update_bits(i2s->regmap, STM32_I2S_CFG1_REG, @@ -869,7 +869,7 @@ static int stm32_i2s_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop I2S %s\n", - playback_flg ? "playback" : "capture"); + snd_pcm_direction_name(substream->stream)); if (playback_flg) regmap_update_bits(i2s->regmap, STM32_I2S_IER_REG, From cda4aa0069b73e3404ee61864b4814ccc7b7a6ad Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:46 +0000 Subject: [PATCH 075/410] ASoC: sof: pcm: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87mslzk51h.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sof/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index baad4c1445aa..35a7462d8b69 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -100,7 +100,7 @@ sof_pcm_setup_connected_widgets(struct snd_sof_dev *sdev, struct snd_soc_pcm_run dpcm_end_walk_at_be); if (ret < 0) { dev_err(sdev->dev, "error: dai %s has no valid %s path\n", dai->name, - dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + snd_pcm_direction_name(dir)); return ret; } From baa779902020d8921cf652944309d1284a63811c Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:53 +0000 Subject: [PATCH 076/410] ASoC: sof: intel: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87le1jk51b.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda-stream.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 8213debde497..3ac63ce67ab1 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -216,9 +216,7 @@ hda_dsp_stream_get(struct snd_sof_dev *sdev, int direction, u32 flags) /* stream found ? */ if (!hext_stream) { - dev_err(sdev->dev, "error: no free %s streams\n", - direction == SNDRV_PCM_STREAM_PLAYBACK ? - "playback" : "capture"); + dev_err(sdev->dev, "error: no free %s streams\n", snd_pcm_direction_name(direction)); return hext_stream; } From 8156921e620827319460e8daa9831d731b0d75fd Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:05:58 +0000 Subject: [PATCH 077/410] ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87jzh3k515.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/fsl/lpc3xxx-i2s.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 62ef624d6dd4..8ae33d91a8a8 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -188,8 +188,7 @@ static int lpc3xxx_i2s_hw_params(struct snd_pcm_substream *substream, __lpc3xxx_find_clkdiv(&clkx, &clky, i2s_info_p->freq, xfersize, i2s_info_p->clkrate); - dev_dbg(dev, "Stream : %s\n", - substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + dev_dbg(dev, "Stream : %s\n", snd_pcm_direction_name(substream->stream)); dev_dbg(dev, "Desired clock rate : %d\n", i2s_info_p->freq); dev_dbg(dev, "Base clock rate : %d\n", i2s_info_p->clkrate); dev_dbg(dev, "Transfer size (bytes) : %d\n", xfersize); From d6db65bc62fd19915a9926e08b9cbcfbadd64291 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:06:03 +0000 Subject: [PATCH 078/410] ASoC: tegra: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ikwnk510.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/tegra/tegra210_i2s.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index fe4fde844d86..e93ceb7afb4c 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -85,7 +85,7 @@ static int tegra210_i2s_set_clock_rate(struct device *dev, } static int tegra210_i2s_sw_reset(struct snd_soc_component *compnt, - bool is_playback) + int stream) { struct device *dev = compnt->dev; struct tegra210_i2s *i2s = dev_get_drvdata(dev); @@ -95,7 +95,7 @@ static int tegra210_i2s_sw_reset(struct snd_soc_component *compnt, unsigned int cif_ctrl, stream_ctrl, i2s_ctrl, val; int err; - if (is_playback) { + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { reset_reg = TEGRA210_I2S_RX_SOFT_RESET; cif_reg = TEGRA210_I2S_RX_CIF_CTRL; stream_reg = TEGRA210_I2S_RX_CTRL; @@ -118,7 +118,7 @@ static int tegra210_i2s_sw_reset(struct snd_soc_component *compnt, 10, 10000); if (err) { dev_err(dev, "timeout: failed to reset I2S for %s\n", - is_playback ? "playback" : "capture"); + snd_pcm_direction_name(stream)); return err; } @@ -137,16 +137,16 @@ static int tegra210_i2s_init(struct snd_soc_dapm_widget *w, struct device *dev = compnt->dev; struct tegra210_i2s *i2s = dev_get_drvdata(dev); unsigned int val, status_reg; - bool is_playback; + int stream; int err; switch (w->reg) { case TEGRA210_I2S_RX_ENABLE: - is_playback = true; + stream = SNDRV_PCM_STREAM_PLAYBACK; status_reg = TEGRA210_I2S_RX_STATUS; break; case TEGRA210_I2S_TX_ENABLE: - is_playback = false; + stream = SNDRV_PCM_STREAM_CAPTURE; status_reg = TEGRA210_I2S_TX_STATUS; break; default: @@ -159,11 +159,11 @@ static int tegra210_i2s_init(struct snd_soc_dapm_widget *w, 10, 10000); if (err) { dev_err(dev, "timeout: previous I2S %s is still active\n", - is_playback ? "playback" : "capture"); + snd_pcm_direction_name(stream)); return err; } - return tegra210_i2s_sw_reset(compnt, is_playback); + return tegra210_i2s_sw_reset(compnt, stream); } static int __maybe_unused tegra210_i2s_runtime_suspend(struct device *dev) From ebbd6703d4635d36b35f12a1365dfd7894c0ff12 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:06:10 +0000 Subject: [PATCH 079/410] ASoC: soc-pcm: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87h6c7k50t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index bad823718ae4..5520944ac9dd 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -222,7 +222,7 @@ static void dpcm_create_debugfs_state(struct snd_soc_dpcm *dpcm, int stream) char *name; name = kasprintf(GFP_KERNEL, "%s:%s", dpcm->be->dai_link->name, - stream ? "capture" : "playback"); + snd_pcm_direction_name(stream)); if (name) { dpcm->debugfs_state = debugfs_create_dir( name, dpcm->fe->debugfs_dpcm_root); @@ -1278,7 +1278,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, snd_soc_dpcm_stream_unlock_irq(fe, stream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", - stream ? "capture" : "playback", fe->dai_link->name, + snd_pcm_direction_name(stream), fe->dai_link->name, stream ? "<-" : "->", be->dai_link->name); dpcm_create_debugfs_state(dpcm, stream); @@ -1306,7 +1306,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, continue; dev_dbg(fe->dev, "reparent %s path %s %s %s\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), dpcm->fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); @@ -1327,14 +1327,14 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) snd_soc_dpcm_stream_lock_irq(fe, stream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), dpcm->be->dai_link->name); if (dpcm->state != SND_SOC_DPCM_LINK_STATE_FREE) continue; dev_dbg(fe->dev, "freed DSP %s path %s %s %s\n", - stream ? "capture" : "playback", fe->dai_link->name, + snd_pcm_direction_name(stream), fe->dai_link->name, stream ? "<-" : "->", dpcm->be->dai_link->name); /* BEs still alive need new FE */ @@ -1441,10 +1441,10 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, if (paths > 0) dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, - stream ? "capture" : "playback"); + snd_pcm_direction_name(stream)); else if (paths == 0) dev_dbg(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, - stream ? "capture" : "playback"); + snd_pcm_direction_name(stream)); return paths; } @@ -1487,7 +1487,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), dpcm->be->dai_link->name, fe->dai_link->name); dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; dpcm_set_be_update_state(dpcm->be, stream, SND_SOC_DPCM_UPDATE_BE); @@ -1605,7 +1605,7 @@ void dpcm_be_dai_stop(struct snd_soc_pcm_runtime *fe, int stream, if (be->dpcm[stream].users == 0) { dev_err(be->dev, "ASoC: no users %s at close - state %d\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), be->dpcm[stream].state); continue; } @@ -1645,7 +1645,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) if (!be_substream) { dev_err(be->dev, "ASoC: no backend %s stream\n", - stream ? "capture" : "playback"); + snd_pcm_direction_name(stream)); continue; } @@ -1656,7 +1656,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) /* first time the dpcm is open ? */ if (be->dpcm[stream].users == DPCM_MAX_BE_USERS) { dev_err(be->dev, "ASoC: too many users %s at open %d\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), be->dpcm[stream].state); continue; } @@ -1669,7 +1669,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) continue; dev_dbg(be->dev, "ASoC: open %s BE %s\n", - stream ? "capture" : "playback", be->dai_link->name); + snd_pcm_direction_name(stream), be->dai_link->name); be_substream->runtime = fe_substream->runtime; err = __soc_pcm_open(be, be_substream); @@ -1677,7 +1677,7 @@ int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) be->dpcm[stream].users--; if (be->dpcm[stream].users < 0) dev_err(be->dev, "ASoC: no users %s at unwind %d\n", - stream ? "capture" : "playback", + snd_pcm_direction_name(stream), be->dpcm[stream].state); be->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; @@ -2531,7 +2531,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", - stream ? "capture" : "playback", fe->dai_link->name); + snd_pcm_direction_name(stream), fe->dai_link->name); if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { /* call bespoke trigger - FE takes care of all BE triggers */ @@ -2565,7 +2565,7 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", - stream ? "capture" : "playback", fe->dai_link->name); + snd_pcm_direction_name(stream), fe->dai_link->name); /* Only start the BE if the FE is ready */ if (fe->dpcm[stream].state == SND_SOC_DPCM_STATE_HW_FREE || From bb660132868b5208d6a5f2bd184425cf788f4ef9 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 02:06:16 +0000 Subject: [PATCH 080/410] ASoC: soc-dapm: use snd_pcm_direction_name() We already have snd_pcm_direction_name(). Let's use it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87frrrk50n.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 37dccd9c1ba0..d7d6dbb9d9ea 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2751,8 +2751,7 @@ static int dapm_update_dai_unlocked(struct snd_pcm_substream *substream, if (!w) return 0; - dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, - dir == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + dev_dbg(dai->dev, "Update DAI routes for %s %s\n", dai->name, snd_pcm_direction_name(dir)); snd_soc_dapm_widget_for_each_sink_path(w, p) { ret = dapm_update_dai_chan(p, p->sink, channels); From 6a965fbaac461564ae74dbfe6d9c9e9de65ea67a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:40:07 +0200 Subject: [PATCH 081/410] ASoC: Intel: soc-acpi: add PTL match tables MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For now the tables are basic for mockup devices and headset codec support Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240802124011.173820-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 41 +++++++++++++++++++ 3 files changed, 44 insertions(+) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index 4843b57798f6..daed7123df9d 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -33,6 +33,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[]; @@ -44,6 +45,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[]; /* * generic table used for HDA codec-based platforms, possibly with diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 40a74a19c508..91e146e2487d 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -12,6 +12,7 @@ snd-soc-acpi-intel-match-y := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-matc soc-acpi-intel-rpl-match.o soc-acpi-intel-mtl-match.o \ soc-acpi-intel-arl-match.o \ soc-acpi-intel-lnl-match.o \ + soc-acpi-intel-ptl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c new file mode 100644 index 000000000000..dba45fa7a453 --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -0,0 +1,41 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-acpi-intel-ptl-match.c - tables and support for PTL ACPI enumeration. + * + * Copyright (c) 2024, Intel Corporation. + * + */ + +#include +#include +#include "soc-acpi-intel-sdw-mockup-match.h" + +struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { + /* mockup tests need to be first */ + { + .link_mask = GENMASK(3, 0), + .links = sdw_mockup_headset_2amps_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt711-rt1308-rt715.tplg", + }, + { + .link_mask = BIT(0) | BIT(1) | BIT(3), + .links = sdw_mockup_headset_1amp_mic, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt711-rt1308-mono-rt715.tplg", + }, + { + .link_mask = GENMASK(2, 0), + .links = sdw_mockup_mic_headset_1amp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt715-rt711-rt1308-mono.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_sdw_machines); From 42b4763ab301c5604343aa49774426d5005711a3 Mon Sep 17 00:00:00 2001 From: Fred Oh Date: Fri, 2 Aug 2024 14:40:08 +0200 Subject: [PATCH 082/410] ASoC: SOF: Intel: add PTL specific power control register MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit PTL has some differences from MTL/LNL. Need to use different register to power up. Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Reviewed-by: Péter Ujfalusi Signed-off-by: Fred Oh Signed-off-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240802124011.173820-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/mtl.c | 16 ++++++++++++++-- sound/soc/sof/intel/mtl.h | 2 ++ 2 files changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/mtl.c b/sound/soc/sof/intel/mtl.c index 1bf274509ee6..2b9d22ccf345 100644 --- a/sound/soc/sof/intel/mtl.c +++ b/sound/soc/sof/intel/mtl.c @@ -245,6 +245,18 @@ int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) u32 cpa; u32 pgs; int ret; + u32 dsppwrctl; + u32 dsppwrsts; + const struct sof_intel_dsp_desc *chip; + + chip = get_chip_info(sdev->pdata); + if (chip->hw_ip_version > SOF_INTEL_ACE_2_0) { + dsppwrctl = PTL_HFPWRCTL2; + dsppwrsts = PTL_HFPWRSTS2; + } else { + dsppwrctl = MTL_HFPWRCTL; + dsppwrsts = MTL_HFPWRSTS; + } /* Set the DSP subsystem power on */ snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFDSSCS, @@ -264,14 +276,14 @@ int mtl_dsp_pre_fw_run(struct snd_sof_dev *sdev) } /* Power up gated-DSP-0 domain in order to access the DSP shim register block. */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, MTL_HFPWRCTL, + snd_sof_dsp_update_bits(sdev, HDA_DSP_BAR, dsppwrctl, MTL_HFPWRCTL_WPDSPHPXPG, MTL_HFPWRCTL_WPDSPHPXPG); usleep_range(1000, 1010); /* poll with timeout to check if operation successful */ pgs = MTL_HFPWRSTS_DSPHPXPGS_MASK; - ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, MTL_HFPWRSTS, dsphfpwrsts, + ret = snd_sof_dsp_read_poll_timeout(sdev, HDA_DSP_BAR, dsppwrsts, dsphfpwrsts, (dsphfpwrsts & pgs) == pgs, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_RESET_TIMEOUT_US); diff --git a/sound/soc/sof/intel/mtl.h b/sound/soc/sof/intel/mtl.h index 7acaa7e724f4..9ab4b21c960e 100644 --- a/sound/soc/sof/intel/mtl.h +++ b/sound/soc/sof/intel/mtl.h @@ -12,9 +12,11 @@ #define MTL_HFDSSCS_CPA_MASK BIT(24) #define MTL_HFSNDWIE 0x114C #define MTL_HFPWRCTL 0x1D18 +#define PTL_HFPWRCTL2 0x1D20 #define MTL_HfPWRCTL_WPIOXPG(x) BIT((x) + 8) #define MTL_HFPWRCTL_WPDSPHPXPG BIT(0) #define MTL_HFPWRSTS 0x1D1C +#define PTL_HFPWRSTS2 0x1D24 #define MTL_HFPWRSTS_DSPHPXPGS_MASK BIT(0) #define MTL_HFINTIPPTR 0x1108 #define MTL_IRQ_INTEN_L_HOST_IPC_MASK BIT(0) From 3f8c8027775901c13d1289b4c54e024d3d5d982a Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:40:09 +0200 Subject: [PATCH 083/410] ASoC: SOF: Intel: add initial support for PTL MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Clone LNL for now. Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240802124011.173820-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/Kconfig | 17 ++++++++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++++++++++ sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/shim.h | 1 + 7 files changed, 126 insertions(+) create mode 100644 sound/soc/sof/intel/pci-ptl.c diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 3396bd46b778..2c43558d96b9 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -281,6 +281,23 @@ config SND_SOC_SOF_LUNARLAKE Say Y if you have such a device. If unsure select "N". +config SND_SOC_SOF_INTEL_PTL + tristate + select SND_SOC_SOF_HDA_COMMON + select SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE + select SND_SOC_SOF_IPC4 + select SND_SOC_SOF_INTEL_LNL + +config SND_SOC_SOF_PANTHERLAKE + tristate "SOF support for Pantherlake" + default SND_SOC_SOF_PCI + select SND_SOC_SOF_INTEL_PTL + help + This adds support for Sound Open Firmware for Intel(R) platforms + using the Pantherlake processors. + Say Y if you have such a device. + If unsure select "N". + config SND_SOC_SOF_HDA_COMMON tristate diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index b56fa5530b8b..f40daa616803 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -34,6 +34,7 @@ snd-sof-pci-intel-icl-y := pci-icl.o icl.o snd-sof-pci-intel-tgl-y := pci-tgl.o tgl.o snd-sof-pci-intel-mtl-y := pci-mtl.o mtl.o snd-sof-pci-intel-lnl-y := pci-lnl.o lnl.o +snd-sof-pci-intel-ptl-y := pci-ptl.o obj-$(CONFIG_SND_SOC_SOF_MERRIFIELD) += snd-sof-pci-intel-tng.o obj-$(CONFIG_SND_SOC_SOF_INTEL_SKL) += snd-sof-pci-intel-skl.o @@ -43,3 +44,4 @@ obj-$(CONFIG_SND_SOC_SOF_INTEL_ICL) += snd-sof-pci-intel-icl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_TGL) += snd-sof-pci-intel-tgl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_MTL) += snd-sof-pci-intel-mtl.o obj-$(CONFIG_SND_SOC_SOF_INTEL_LNL) += snd-sof-pci-intel-lnl.o +obj-$(CONFIG_SND_SOC_SOF_INTEL_PTL) += snd-sof-pci-intel-ptl.o diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 1315f5bc3e31..4c88522d4048 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -69,6 +69,7 @@ static void hda_get_interfaces(struct snd_sof_dev *sdev, u32 *interface_mask) interface_mask[SOF_DAI_HOST_ACCESS] = BIT(SOF_DAI_INTEL_HDA); break; case SOF_INTEL_ACE_2_0: + case SOF_INTEL_ACE_3_0: interface_mask[SOF_DAI_DSP_ACCESS] = BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC) | BIT(SOF_DAI_INTEL_HDA) | BIT(SOF_DAI_INTEL_ALH); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 3c9e1d59e1ab..b74a472435b5 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -920,6 +920,7 @@ extern const struct sof_intel_dsp_desc adls_chip_info; extern const struct sof_intel_dsp_desc mtl_chip_info; extern const struct sof_intel_dsp_desc arl_s_chip_info; extern const struct sof_intel_dsp_desc lnl_chip_info; +extern const struct sof_intel_dsp_desc ptl_chip_info; /* Probes support */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) diff --git a/sound/soc/sof/intel/lnl.c b/sound/soc/sof/intel/lnl.c index 4b5665f82170..3d5a1f8b17e5 100644 --- a/sound/soc/sof/intel/lnl.c +++ b/sound/soc/sof/intel/lnl.c @@ -22,6 +22,7 @@ /* LunarLake ops */ struct snd_sof_dsp_ops sof_lnl_ops; +EXPORT_SYMBOL_NS(sof_lnl_ops, SND_SOC_SOF_INTEL_LNL); static const struct snd_sof_debugfs_map lnl_dsp_debugfs[] = { {"hda", HDA_DSP_HDA_BAR, 0, 0x4000, SOF_DEBUGFS_ACCESS_ALWAYS}, @@ -181,6 +182,7 @@ int sof_lnl_ops_init(struct snd_sof_dev *sdev) return 0; }; +EXPORT_SYMBOL_NS(sof_lnl_ops_init, SND_SOC_SOF_INTEL_LNL); /* Check if an SDW IRQ occurred */ static bool lnl_dsp_check_sdw_irq(struct snd_sof_dev *sdev) @@ -245,3 +247,28 @@ const struct sof_intel_dsp_desc lnl_chip_info = { .disable_interrupts = lnl_dsp_disable_interrupts, .hw_ip_version = SOF_INTEL_ACE_2_0, }; + +const struct sof_intel_dsp_desc ptl_chip_info = { + .cores_num = 5, + .init_core_mask = BIT(0), + .host_managed_cores_mask = BIT(0), + .ipc_req = MTL_DSP_REG_HFIPCXIDR, + .ipc_req_mask = MTL_DSP_REG_HFIPCXIDR_BUSY, + .ipc_ack = MTL_DSP_REG_HFIPCXIDA, + .ipc_ack_mask = MTL_DSP_REG_HFIPCXIDA_DONE, + .ipc_ctl = MTL_DSP_REG_HFIPCXCTL, + .rom_status_reg = LNL_DSP_REG_HFDSC, + .rom_init_timeout = 300, + .ssp_count = MTL_SSP_COUNT, + .d0i3_offset = MTL_HDA_VS_D0I3C, + .read_sdw_lcount = hda_sdw_check_lcount_ext, + .enable_sdw_irq = lnl_enable_sdw_irq, + .check_sdw_irq = lnl_dsp_check_sdw_irq, + .check_sdw_wakeen_irq = lnl_sdw_check_wakeen_irq, + .check_ipc_irq = mtl_dsp_check_ipc_irq, + .cl_init = mtl_dsp_cl_init, + .power_down_dsp = mtl_power_down_dsp, + .disable_interrupts = lnl_dsp_disable_interrupts, + .hw_ip_version = SOF_INTEL_ACE_3_0, +}; +EXPORT_SYMBOL_NS(ptl_chip_info, SND_SOC_SOF_INTEL_LNL); diff --git a/sound/soc/sof/intel/pci-ptl.c b/sound/soc/sof/intel/pci-ptl.c new file mode 100644 index 000000000000..69195b5e7b1a --- /dev/null +++ b/sound/soc/sof/intel/pci-ptl.c @@ -0,0 +1,77 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2024 Intel Corporation. +// + +#include +#include +#include +#include +#include +#include "../ops.h" +#include "../sof-pci-dev.h" + +/* platform specific devices */ +#include "hda.h" +#include "mtl.h" + +static const struct sof_dev_desc ptl_desc = { + .use_acpi_target_states = true, + .machines = snd_soc_acpi_intel_ptl_machines, + .alt_machines = snd_soc_acpi_intel_ptl_sdw_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &ptl_chip_info, + .ipc_supported_mask = BIT(SOF_IPC_TYPE_4), + .ipc_default = SOF_IPC_TYPE_4, + .dspless_mode_supported = true, + .default_fw_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4/ptl", + }, + .default_lib_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-lib/ptl", + }, + .default_tplg_path = { + [SOF_IPC_TYPE_4] = "intel/sof-ipc4-tplg", + }, + .default_fw_filename = { + [SOF_IPC_TYPE_4] = "sof-ptl.ri", + }, + .nocodec_tplg_filename = "sof-ptl-nocodec.tplg", + .ops = &sof_lnl_ops, + .ops_init = sof_lnl_ops_init, +}; + +/* PCI IDs */ +static const struct pci_device_id sof_pci_ids[] = { + { PCI_DEVICE_DATA(INTEL, HDA_PTL, &ptl_desc) }, /* PTL */ + { 0, } +}; +MODULE_DEVICE_TABLE(pci, sof_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_intel_ptl_driver = { + .name = "sof-audio-pci-intel-ptl", + .id_table = sof_pci_ids, + .probe = hda_pci_intel_probe, + .remove = sof_pci_remove, + .shutdown = sof_pci_shutdown, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_intel_ptl_driver); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for PantherLake platforms"); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_LNL); +MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL); +MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); +MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index 9328d2bbfd03..8709b750a11e 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -22,6 +22,7 @@ enum sof_intel_hw_ip_version { SOF_INTEL_CAVS_2_5, /* TigerLake, AlderLake */ SOF_INTEL_ACE_1_0, /* MeteorLake */ SOF_INTEL_ACE_2_0, /* LunarLake */ + SOF_INTEL_ACE_3_0, /* PantherLake */ }; /* From 77a6869afbbfad0db297e9e4b9233aac209d5385 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 2 Aug 2024 14:40:10 +0200 Subject: [PATCH 084/410] ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add rt711-sdca on sdw link0. Reviewed-by: Péter Ujfalusi Reviewed-by: Ranjani Sridharan Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240802124011.173820-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-ptl-match.c | 31 +++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index dba45fa7a453..ce4234fd3895 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -15,6 +15,31 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_machines[] = { }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_machines); +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { + { + .adr = 0x000030025D071101ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr ptl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_sdca_0_adr), + .adr_d = rt711_sdca_0_adr, + }, + {} +}; + /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { /* mockup tests need to be first */ @@ -36,6 +61,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-ptl-rt715-rt711-rt1308-mono.tplg", }, + { + .link_mask = BIT(0), + .links = ptl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt711.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_sdw_machines); From 2786d3f4943c472c10dd630ec3e0a1a892181562 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 2 Aug 2024 14:40:11 +0200 Subject: [PATCH 085/410] ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This patch adds match table for rt722 multiple function codec on link 0 and link3. The topology does not internally refer to link0 or link3, so we can simplify and use the same topology file name. We do need different tables though. Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Signed-off-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240802124011.173820-6-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-ptl-match.c | 73 +++++++++++++++++++ 1 file changed, 73 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c index ce4234fd3895..90f97a44b607 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ptl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ptl-match.c @@ -22,6 +22,31 @@ static const struct snd_soc_acpi_endpoint single_endpoint = { .group_id = 0, }; +/* + * RT722 is a multi-function codec, three endpoints are created for + * its headset, amp and dmic functions. + */ +static const struct snd_soc_acpi_endpoint rt722_endpoints[] = { + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { { .adr = 0x000030025D071101ull, @@ -31,6 +56,42 @@ static const struct snd_soc_acpi_adr_device rt711_sdca_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device rt722_0_single_adr[] = { + { + .adr = 0x000030025d072201ull, + .num_endpoints = ARRAY_SIZE(rt722_endpoints), + .endpoints = rt722_endpoints, + .name_prefix = "rt722" + } +}; + +static const struct snd_soc_acpi_adr_device rt722_3_single_adr[] = { + { + .adr = 0x000330025d072201ull, + .num_endpoints = ARRAY_SIZE(rt722_endpoints), + .endpoints = rt722_endpoints, + .name_prefix = "rt722" + } +}; + +static const struct snd_soc_acpi_link_adr ptl_rt722_only[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt722_0_single_adr), + .adr_d = rt722_0_single_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr ptl_rt722_l3[] = { + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(rt722_3_single_adr), + .adr_d = rt722_3_single_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr ptl_rvp[] = { { .mask = BIT(0), @@ -67,6 +128,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_ptl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-ptl-rt711.tplg", }, + { + .link_mask = BIT(0), + .links = ptl_rt722_only, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt722.tplg", + }, + { + .link_mask = BIT(3), + .links = ptl_rt722_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-ptl-rt722.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_ptl_sdw_machines); From cac88e96ba0961921b8068326579b26094f37ba4 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:46:06 +0200 Subject: [PATCH 086/410] ASoC: SOF: sof-priv.h: optimize snd_sof_platform_stream_params MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit reshuffle members to remove hole. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20240802124609.188954-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-priv.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 4d6a1517f9b3..28179a573762 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -132,16 +132,17 @@ struct snd_sof_pdata; /** * struct snd_sof_platform_stream_params - platform dependent stream parameters - * @stream_tag: Stream tag to use - * @use_phy_addr: Use the provided @phy_addr for configuration * @phy_addr: Platform dependent address to be used, if @use_phy_addr * is true + * @stream_tag: Stream tag to use + * @use_phy_addr: Use the provided @phy_addr for configuration * @no_ipc_position: Disable position update IPC from firmware + * @cont_update_posn: Continuous position update. */ struct snd_sof_platform_stream_params { + u32 phy_addr; u16 stream_tag; bool use_phy_address; - u32 phy_addr; bool no_ipc_position; bool cont_update_posn; }; From e9e7eeaf199c7961d634235dbedeb7b682b1dd32 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:46:07 +0200 Subject: [PATCH 087/410] ASoC: SOF: sof-priv.h: optimize snd_sof_mailbox MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Reverse the two members to remove a hole. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20240802124609.188954-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-priv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 28179a573762..6ecc58e11592 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -412,8 +412,8 @@ struct snd_sof_debugfs_map { /* mailbox descriptor, used for host <-> DSP IPC */ struct snd_sof_mailbox { - u32 offset; size_t size; + u32 offset; }; /* IPC message descriptor for host <-> DSP IO */ From 5a4413d0fa8d0a438246acaf81637d71aab1b6a0 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:46:08 +0200 Subject: [PATCH 088/410] ASoC: SOF: sof-priv.h: optimize snd_sof_ipc_msg MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Move waitq to make sure it's entirely in the same cache line, and move ipc_complete to reduce padding. struct snd_sof_ipc_msg { void * msg_data; /* 0 8 */ void * reply_data; /* 8 8 */ size_t msg_size; /* 16 8 */ size_t reply_size; /* 24 8 */ int reply_error; /* 32 4 */ bool ipc_complete; /* 36 1 */ /* XXX 3 bytes hole, try to pack */ wait_queue_head_t waitq; /* 40 88 */ /* --- cacheline 2 boundary (128 bytes) --- */ void * rx_data; /* 128 8 */ /* size: 136, cachelines: 3, members: 8 */ /* sum members: 133, holes: 1, sum holes: 3 */ /* last cacheline: 8 bytes */ }; Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20240802124609.188954-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-priv.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6ecc58e11592..843be3b6415d 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -425,11 +425,12 @@ struct snd_sof_ipc_msg { size_t reply_size; int reply_error; - /* notification, firmware initiated messages */ - void *rx_data; + bool ipc_complete; wait_queue_head_t waitq; - bool ipc_complete; + + /* notification, firmware initiated messages */ + void *rx_data; }; /** From 5821d7b4981f4915ab353f538be38defe9656f81 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Fri, 2 Aug 2024 14:46:09 +0200 Subject: [PATCH 089/410] ASoC: SOF: sof-audio.h: optimize snd_sof_pcm_stream_pipeline_list MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Invert members to remove hole. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Link: https://patch.msgid.link/20240802124609.188954-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 49be02234fc3..b9b8b64768b9 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -314,12 +314,12 @@ struct sof_token_info { /** * struct snd_sof_pcm_stream_pipeline_list - List of pipelines associated with a PCM stream - * @count: number of pipeline widgets in the @pipe_widgets array * @pipelines: array of pipelines + * @count: number of pipeline widgets in the @pipe_widgets array */ struct snd_sof_pcm_stream_pipeline_list { - u32 count; struct snd_sof_pipeline **pipelines; + u32 count; }; /* PCM stream, mapped to FW component */ From 92b796845a4a8789c2d9434c6a77baa88a99121e Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Fri, 2 Aug 2024 15:20:52 +0800 Subject: [PATCH 090/410] ASoC: tas2781: Fix a compiling warning reported by robot kernel test due to adding tas2563_dvc_table Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Fixes: 75ed63a5ab5d ("ASoC: tas2781: Add new Kontrol to set tas2563 digital Volume") Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240802072055.1462-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..faa3e194f73b --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0X7F, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 00fd4d449ff3..d87263e43fdb 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include From 11c2d223713b7a7fee848595e9f582d34adc552b Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Thu, 1 Aug 2024 22:30:05 +0200 Subject: [PATCH 091/410] ASoC: sti-sas: Constify snd_soc_component_driver struct In order to constify `snd_soc_component_driver` struct, simplify the logic and the `sti_sas_dev_data` struct. Since commit 165a57a3df02 ("ASoC: sti-sas: clean legacy in sti-sas") only only chip is supported and `sti_sas_driver` can be fully defined at compilation time. Before: ====== text data bss dec hex filename 8033 1547 16 9596 257c sound/soc/codecs/sti-sas.o After: ===== text data bss dec hex filename 8257 1163 16 9436 24dc sound/soc/codecs/sti-sas.o Signed-off-by: Christophe JAILLET Link: https://patch.msgid.link/2c08558813e3bbfae0a5302199cf6ca226e7cde1.1722544073.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/sti-sas.c | 21 +++++---------------- 1 file changed, 5 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index c421906a0694..4ab15be69f3a 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -63,10 +63,6 @@ struct sti_spdif_audio { struct sti_sas_dev_data { const struct regmap_config *regmap; const struct snd_soc_dai_ops *dac_ops; /* DAC function callbacks */ - const struct snd_soc_dapm_widget *dapm_widgets; /* dapms declaration */ - const int num_dapm_widgets; /* dapms declaration */ - const struct snd_soc_dapm_route *dapm_routes; /* route declaration */ - const int num_dapm_routes; /* route declaration */ }; /* driver data structure */ @@ -324,10 +320,6 @@ static const struct regmap_config stih407_sas_regmap = { static const struct sti_sas_dev_data stih407_data = { .regmap = &stih407_sas_regmap, .dac_ops = &stih407_dac_ops, - .dapm_widgets = stih407_sas_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(stih407_sas_dapm_widgets), - .dapm_routes = stih407_sas_route, - .num_dapm_routes = ARRAY_SIZE(stih407_sas_route), }; static struct snd_soc_dai_driver sti_sas_dai[] = { @@ -386,12 +378,16 @@ static int sti_sas_component_probe(struct snd_soc_component *component) return sti_sas_init_sas_registers(component, drvdata); } -static struct snd_soc_component_driver sti_sas_driver = { +static const struct snd_soc_component_driver sti_sas_driver = { .probe = sti_sas_component_probe, .resume = sti_sas_resume, .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, + .dapm_widgets = stih407_sas_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(stih407_sas_dapm_widgets), + .dapm_routes = stih407_sas_route, + .num_dapm_routes = ARRAY_SIZE(stih407_sas_route), }; static const struct of_device_id sti_sas_dev_match[] = { @@ -446,13 +442,6 @@ static int sti_sas_driver_probe(struct platform_device *pdev) sti_sas_dai[STI_SAS_DAI_ANALOG_OUT].ops = drvdata->dev_data->dac_ops; - /* Set dapms*/ - sti_sas_driver.dapm_widgets = drvdata->dev_data->dapm_widgets; - sti_sas_driver.num_dapm_widgets = drvdata->dev_data->num_dapm_widgets; - - sti_sas_driver.dapm_routes = drvdata->dev_data->dapm_routes; - sti_sas_driver.num_dapm_routes = drvdata->dev_data->num_dapm_routes; - /* Store context */ dev_set_drvdata(&pdev->dev, drvdata); From a1c2716738b7ba9e912e04872639dd39c72baa35 Mon Sep 17 00:00:00 2001 From: Yue Haibing Date: Fri, 2 Aug 2024 18:10:44 +0800 Subject: [PATCH 092/410] ASoC: fsl: lpc3xxx: Make some symbols static These symbols are not used outside of the files, make them static to fix sparse warnings: sound/soc/fsl/lpc3xxx-i2s.c:261:30: warning: symbol 'lpc3xxx_i2s_dai_ops' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-i2s.c:271:27: warning: symbol 'lpc3xxx_i2s_dai_driver' was not declared. Should it be static? sound/soc/fsl/lpc3xxx-pcm.c:55:39: warning: symbol 'lpc3xxx_soc_platform_driver' was not declared. Should it be static? Signed-off-by: Yue Haibing Link: https://patch.msgid.link/20240802101044.3302251-1-yuehaibing@huawei.com Signed-off-by: Mark Brown --- sound/soc/fsl/lpc3xxx-i2s.c | 4 ++-- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 8ae33d91a8a8..c65c17dfa174 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -257,7 +257,7 @@ static int lpc3xxx_i2s_dai_probe(struct snd_soc_dai *dai) return 0; } -const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { +static const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .probe = lpc3xxx_i2s_dai_probe, .startup = lpc3xxx_i2s_startup, .shutdown = lpc3xxx_i2s_shutdown, @@ -267,7 +267,7 @@ const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { .set_fmt = lpc3xxx_i2s_set_dai_fmt, }; -struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { +static struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { .playback = { .channels_min = 1, .channels_max = 2, diff --git a/sound/soc/fsl/lpc3xxx-pcm.c b/sound/soc/fsl/lpc3xxx-pcm.c index c0d499b9b8ba..e6abaf63895a 100644 --- a/sound/soc/fsl/lpc3xxx-pcm.c +++ b/sound/soc/fsl/lpc3xxx-pcm.c @@ -52,7 +52,7 @@ static const struct snd_dmaengine_pcm_config lpc3xxx_dmaengine_pcm_config = { .prealloc_buffer_size = 128 * 1024, }; -const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { +static const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { .name = "lpc32xx-pcm", }; From 839e231a53b824a62bc3696ad3ba1dcedc4f4167 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Mon, 29 Jul 2024 19:36:05 +0200 Subject: [PATCH 093/410] ASoC: cs43130: Constify snd_soc_component_driver struct In order to constify `snd_soc_component_driver` struct, duplicate `soc_component_dev_cs43130` into a `soc_component_dev_cs43130_digital` and `soc_component_dev_cs43130_analog`. These 2 new structures share the same .dapm_widgets and .dapm_routes arrays but differ for .num_dapm_widgets and .num_dapm_routes. In the digital case, the last entries are not taken into account. Doing so has several advantages: - `snd_soc_component_driver` can be declared as const to move their declarations to read-only sections. - code in the probe is simpler. There is no need to concatenate some arrays to handle the "analog" case - this saves some memory because all_hp_widgets and analog_hp_routes can be removed. Before : ====== text data bss dec hex filename 53965 8265 4512 66742 104b6 sound/soc/codecs/cs43130.o After : ===== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET Link: https://patch.msgid.link/1f04bb0366d9640d7ee361dae114ff79e4b381c1.1722274212.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 73 +++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index be4037890fdb..cb4ca80f36d2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1415,7 +1415,7 @@ static const char * const bypass_mux_text[] = { static SOC_ENUM_SINGLE_DECL(bypass_enum, SND_SOC_NOPM, 0, bypass_mux_text); static const struct snd_kcontrol_new bypass_ctrl = SOC_DAPM_ENUM("Switch", bypass_enum); -static const struct snd_soc_dapm_widget digital_hp_widgets[] = { +static const struct snd_soc_dapm_widget hp_widgets[] = { SND_SOC_DAPM_MUX("Bypass Switch", SND_SOC_NOPM, 0, 0, &bypass_ctrl), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -1447,19 +1447,16 @@ static const struct snd_soc_dapm_widget digital_hp_widgets[] = { CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD)), -}; -static const struct snd_soc_dapm_widget analog_hp_widgets[] = { +/* Some devices have some extra analog widgets */ +#define NUM_ANALOG_WIDGETS 1 + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), }; -static struct snd_soc_dapm_widget all_hp_widgets[ - ARRAY_SIZE(digital_hp_widgets) + - ARRAY_SIZE(analog_hp_widgets)]; - -static const struct snd_soc_dapm_route digital_hp_routes[] = { +static const struct snd_soc_dapm_route hp_routes[] = { {"ASPIN PCM", NULL, "ASP PCM Playback"}, {"ASPIN DoP", NULL, "ASP DoP Playback"}, {"XSPIN DoP", NULL, "XSP DoP Playback"}, @@ -1472,15 +1469,12 @@ static const struct snd_soc_dapm_route digital_hp_routes[] = { {"Bypass Switch", "Internal", "HiFi DAC"}, {"HPOUTA", NULL, "Bypass Switch"}, {"HPOUTB", NULL, "Bypass Switch"}, -}; -static const struct snd_soc_dapm_route analog_hp_routes[] = { +/* Some devices have some extra analog routes */ +#define NUM_ANALOG_ROUTES 1 {"Bypass Switch", "Alternative", "Analog Playback"}, }; -static struct snd_soc_dapm_route all_hp_routes[ - ARRAY_SIZE(digital_hp_routes) + - ARRAY_SIZE(analog_hp_routes)]; static const unsigned int cs43130_asp_src_rates[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 @@ -2398,7 +2392,7 @@ static int cs43130_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_cs43130 = { +static const struct snd_soc_component_driver soc_component_dev_cs43130_digital = { .probe = cs43130_probe, .controls = cs43130_snd_controls, .num_controls = ARRAY_SIZE(cs43130_snd_controls), @@ -2407,6 +2401,26 @@ static struct snd_soc_component_driver soc_component_dev_cs43130 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, + /* Don't take into account the ending analog widgets and routes */ + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets) - NUM_ANALOG_WIDGETS, + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes) - NUM_ANALOG_ROUTES, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs43130_analog = { + .probe = cs43130_probe, + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .set_sysclk = cs43130_component_set_sysclk, + .set_pll = cs43130_set_pll, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets), + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes), }; static const struct regmap_config cs43130_regmap = { @@ -2479,6 +2493,7 @@ static int cs43130_handle_device_data(struct cs43130_private *cs43130) static int cs43130_i2c_probe(struct i2c_client *client) { + const struct snd_soc_component_driver *component_driver; struct cs43130_private *cs43130; int ret; unsigned int reg; @@ -2596,39 +2611,15 @@ static int cs43130_i2c_probe(struct i2c_client *client) switch (cs43130->dev_id) { case CS43130_CHIP_ID: case CS43131_CHIP_ID: - memcpy(all_hp_widgets, digital_hp_widgets, - sizeof(digital_hp_widgets)); - memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), - analog_hp_widgets, sizeof(analog_hp_widgets)); - memcpy(all_hp_routes, digital_hp_routes, - sizeof(digital_hp_routes)); - memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), - analog_hp_routes, sizeof(analog_hp_routes)); - - soc_component_dev_cs43130.dapm_widgets = - all_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(all_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - all_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(all_hp_routes); + component_driver = &soc_component_dev_cs43130_analog; break; case CS43198_CHIP_ID: case CS4399_CHIP_ID: - soc_component_dev_cs43130.dapm_widgets = - digital_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(digital_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - digital_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(digital_hp_routes); + component_driver = &soc_component_dev_cs43130_digital; break; } - ret = devm_snd_soc_register_component(cs43130->dev, - &soc_component_dev_cs43130, + ret = devm_snd_soc_register_component(cs43130->dev, component_driver, cs43130_dai, ARRAY_SIZE(cs43130_dai)); if (ret < 0) { dev_err(cs43130->dev, From 0079c9d1e58a39148e6ce13bda55307ea6aa3a9e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Aug 2024 09:00:23 +0200 Subject: [PATCH 094/410] ALSA: ump: Handle MIDI 1.0 Function Block in MIDI 2.0 protocol The UMP v1.1 spec says in the section 6.2.1: "If a UMP Endpoint declares MIDI 2.0 Protocol but a Function Block represents a MIDI 1.0 connection, then may optionally be used for messages to/from that Function Block." It implies that the driver can (and should) keep MIDI 1.0 CVM exceptionally for those FBs even if UMP Endpoint is running in MIDI 2.0 protocol, and the current driver lacks of it. This patch extends the sequencer port info to indicate a MIDI 1.0 port, and tries to send/receive MIDI 1.0 CVM as is when this port is the source or sink. The sequencer port flag is set by the driver at parsing FBs and GTBs although application can set it to its own user-space clients, too. Link: https://patch.msgid.link/20240806070024.14301-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/ump.h | 1 + include/uapi/sound/asequencer.h | 2 ++ sound/core/seq/seq_ports.c | 2 ++ sound/core/seq/seq_ports.h | 2 ++ sound/core/seq/seq_ump_client.c | 4 +++- sound/core/seq/seq_ump_convert.c | 11 ++++++----- sound/core/ump.c | 10 +++++++++- 7 files changed, 25 insertions(+), 7 deletions(-) diff --git a/include/sound/ump.h b/include/sound/ump.h index 7f68056acdff..7484f62fb234 100644 --- a/include/sound/ump.h +++ b/include/sound/ump.h @@ -18,6 +18,7 @@ struct snd_ump_group { unsigned int dir_bits; /* directions */ bool active; /* activeness */ bool valid; /* valid group (referred by blocks) */ + bool is_midi1; /* belongs to a MIDI1 FB */ char name[64]; /* group name */ }; diff --git a/include/uapi/sound/asequencer.h b/include/uapi/sound/asequencer.h index 39b37edcf813..bc30c1f2a109 100644 --- a/include/uapi/sound/asequencer.h +++ b/include/uapi/sound/asequencer.h @@ -461,6 +461,8 @@ struct snd_seq_remove_events { #define SNDRV_SEQ_PORT_FLG_TIMESTAMP (1<<1) #define SNDRV_SEQ_PORT_FLG_TIME_REAL (1<<2) +#define SNDRV_SEQ_PORT_FLG_IS_MIDI1 (1<<3) /* Keep MIDI 1.0 protocol */ + /* port direction */ #define SNDRV_SEQ_PORT_DIR_UNKNOWN 0 #define SNDRV_SEQ_PORT_DIR_INPUT 1 diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index ca631ca4f2c6..535290f24eed 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -362,6 +362,8 @@ int snd_seq_set_port_info(struct snd_seq_client_port * port, port->direction |= SNDRV_SEQ_PORT_DIR_OUTPUT; } + port->is_midi1 = !!(info->flags & SNDRV_SEQ_PORT_FLG_IS_MIDI1); + return 0; } diff --git a/sound/core/seq/seq_ports.h b/sound/core/seq/seq_ports.h index b111382f697a..20a26bffecb7 100644 --- a/sound/core/seq/seq_ports.h +++ b/sound/core/seq/seq_ports.h @@ -87,6 +87,8 @@ struct snd_seq_client_port { unsigned char direction; unsigned char ump_group; + bool is_midi1; /* keep MIDI 1.0 protocol */ + #if IS_ENABLED(CONFIG_SND_SEQ_UMP) struct snd_seq_ump_midi2_bank midi2_bank[16]; /* per channel */ #endif diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index 637154bd1a3f..e5d3f4d206bf 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -189,6 +189,8 @@ static void fill_port_info(struct snd_seq_port_info *port, port->ump_group = group->group + 1; if (!group->active) port->capability |= SNDRV_SEQ_PORT_CAP_INACTIVE; + if (group->is_midi1) + port->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; port->type = SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | SNDRV_SEQ_PORT_TYPE_MIDI_UMP | SNDRV_SEQ_PORT_TYPE_HARDWARE | @@ -223,7 +225,7 @@ static int seq_ump_group_init(struct seq_ump_client *client, int group_index) return -ENOMEM; fill_port_info(port, client, group); - port->flags = SNDRV_SEQ_PORT_FLG_GIVEN_PORT; + port->flags |= SNDRV_SEQ_PORT_FLG_GIVEN_PORT; memset(&pcallbacks, 0, sizeof(pcallbacks)); pcallbacks.owner = THIS_MODULE; pcallbacks.private_data = client; diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index d9dacfbe4a9a..90656980ae26 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -595,12 +595,13 @@ int snd_seq_deliver_from_ump(struct snd_seq_client *source, type = ump_message_type(ump_ev->ump[0]); if (snd_seq_client_is_ump(dest)) { - if (snd_seq_client_is_midi2(dest) && - type == UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE) + bool is_midi2 = snd_seq_client_is_midi2(dest) && + !dest_port->is_midi1; + + if (is_midi2 && type == UMP_MSG_TYPE_MIDI1_CHANNEL_VOICE) return cvt_ump_midi1_to_midi2(dest, dest_port, event, atomic, hop); - else if (!snd_seq_client_is_midi2(dest) && - type == UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE) + else if (!is_midi2 && type == UMP_MSG_TYPE_MIDI2_CHANNEL_VOICE) return cvt_ump_midi2_to_midi1(dest, dest_port, event, atomic, hop); /* non-EP port and different group is set? */ @@ -1254,7 +1255,7 @@ int snd_seq_deliver_to_ump(struct snd_seq_client *source, return 0; /* group filtered - skip the event */ if (event->type == SNDRV_SEQ_EVENT_SYSEX) return cvt_sysex_to_ump(dest, dest_port, event, atomic, hop); - else if (snd_seq_client_is_midi2(dest)) + else if (snd_seq_client_is_midi2(dest) && !dest_port->is_midi1) return cvt_to_ump_midi2(dest, dest_port, event, atomic, hop); else return cvt_to_ump_midi1(dest, dest_port, event, atomic, hop); diff --git a/sound/core/ump.c b/sound/core/ump.c index e39e9cda4912..c7c3581bbbbc 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -538,6 +538,7 @@ static void update_group_attrs(struct snd_ump_endpoint *ump) group->active = 0; group->group = i; group->valid = false; + group->is_midi1 = false; } list_for_each_entry(fb, &ump->block_list, list) { @@ -548,6 +549,8 @@ static void update_group_attrs(struct snd_ump_endpoint *ump) group->valid = true; if (fb->info.active) group->active = 1; + if (fb->info.flags & SNDRV_UMP_BLOCK_IS_MIDI1) + group->is_midi1 = true; switch (fb->info.direction) { case SNDRV_UMP_DIR_INPUT: group->dir_bits |= (1 << SNDRV_RAWMIDI_STREAM_INPUT); @@ -1156,6 +1159,7 @@ static int process_legacy_output(struct snd_ump_endpoint *ump, struct snd_rawmidi_substream *substream; struct ump_cvt_to_ump *ctx; const int dir = SNDRV_RAWMIDI_STREAM_OUTPUT; + unsigned int protocol; unsigned char c; int group, size = 0; @@ -1168,9 +1172,13 @@ static int process_legacy_output(struct snd_ump_endpoint *ump, if (!substream) continue; ctx = &ump->out_cvts[group]; + protocol = ump->info.protocol; + if ((protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI2) && + ump->groups[group].is_midi1) + protocol = SNDRV_UMP_EP_INFO_PROTO_MIDI1; while (!ctx->ump_bytes && snd_rawmidi_transmit(substream, &c, 1) > 0) - snd_ump_convert_to_ump(ctx, group, ump->info.protocol, c); + snd_ump_convert_to_ump(ctx, group, protocol, c); if (ctx->ump_bytes && ctx->ump_bytes <= count) { size = ctx->ump_bytes; memcpy(buffer, ctx->ump, size); From 7d2fb3812acde0a76e0d361877e8295db065f9f4 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 6 Aug 2024 00:00:57 +0000 Subject: [PATCH 095/410] ASoC: remove bespoke trigger support Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87v80ewmdi.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c951..ab4e109fe71d 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd00..773f2db8c31c 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a6..df2487b700cc 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c1..302ca753a1f3 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5520944ac9dd..1edcb8d6f6ee 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2388,14 +2388,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2525,26 +2517,12 @@ out: static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", snd_pcm_direction_name(stream), fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2558,10 +2536,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2605,23 +2580,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; From 901e85677ec0bb9a69fb9eab1feafe0c4eb7d07e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Aug 2024 14:46:50 +0200 Subject: [PATCH 096/410] ALSA: usb-audio: Add input value sanity checks for standard types For an invalid input value that is out of the given range, currently USB-audio driver corrects the value silently and accepts without errors. This is no wrong behavior, per se, but the recent kselftest rather wants to have an error in such a case, hence a different behavior is expected now. This patch adds a sanity check at each control put for the standard mixer types and returns an error if an invalid value is given. Note that this covers only the standard mixer types. The mixer quirks that have own control callbacks would need different coverage. Link: https://patch.msgid.link/20240806124651.28203-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 35 +++++++++++++++++++++++++++-------- sound/usb/mixer.h | 1 + 2 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f7ce8e8c3c3e..2d27d729c3be 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1377,6 +1377,19 @@ no_res_check: #define get_min_max(cval, def) get_min_max_with_quirks(cval, def, NULL) +/* get the max value advertised via control API */ +static int get_max_exposed(struct usb_mixer_elem_info *cval) +{ + if (!cval->max_exposed) { + if (cval->res) + cval->max_exposed = + DIV_ROUND_UP(cval->max - cval->min, cval->res); + else + cval->max_exposed = cval->max - cval->min; + } + return cval->max_exposed; +} + /* get a feature/mixer unit info */ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1389,11 +1402,8 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = cval->channels; - if (cval->val_type == USB_MIXER_BOOLEAN || - cval->val_type == USB_MIXER_INV_BOOLEAN) { - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 1; - } else { + if (cval->val_type != USB_MIXER_BOOLEAN && + cval->val_type != USB_MIXER_INV_BOOLEAN) { if (!cval->initialized) { get_min_max_with_quirks(cval, 0, kcontrol); if (cval->initialized && cval->dBmin >= cval->dBmax) { @@ -1405,10 +1415,10 @@ static int mixer_ctl_feature_info(struct snd_kcontrol *kcontrol, &kcontrol->id); } } - uinfo->value.integer.min = 0; - uinfo->value.integer.max = - DIV_ROUND_UP(cval->max - cval->min, cval->res); } + + uinfo->value.integer.min = 0; + uinfo->value.integer.max = get_max_exposed(cval); return 0; } @@ -1449,6 +1459,7 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_info *cval = kcontrol->private_data; + int max_val = get_max_exposed(cval); int c, cnt, val, oval, err; int changed = 0; @@ -1461,6 +1472,8 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, if (err < 0) return filter_error(cval, err); val = ucontrol->value.integer.value[cnt]; + if (val < 0 || val > max_val) + return -EINVAL; val = get_abs_value(cval, val); if (oval != val) { snd_usb_set_cur_mix_value(cval, c + 1, cnt, val); @@ -1474,6 +1487,8 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, if (err < 0) return filter_error(cval, err); val = ucontrol->value.integer.value[0]; + if (val < 0 || val > max_val) + return -EINVAL; val = get_abs_value(cval, val); if (val != oval) { snd_usb_set_cur_mix_value(cval, 0, 0, val); @@ -2337,6 +2352,8 @@ static int mixer_ctl_procunit_put(struct snd_kcontrol *kcontrol, if (err < 0) return filter_error(cval, err); val = ucontrol->value.integer.value[0]; + if (val < 0 || val > get_max_exposed(cval)) + return -EINVAL; val = get_abs_value(cval, val); if (val != oval) { set_cur_ctl_value(cval, cval->control << 8, val); @@ -2699,6 +2716,8 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, if (err < 0) return filter_error(cval, err); val = ucontrol->value.enumerated.item[0]; + if (val < 0 || val >= cval->max) /* here cval->max = # elements */ + return -EINVAL; val = get_abs_value(cval, val); if (val != oval) { set_cur_ctl_value(cval, cval->control << 8, val); diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index d43895c1ae5c..167fbfcf01ac 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -88,6 +88,7 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int max_exposed; /* control API exposes the value in 0..max_exposed */ int dBmin, dBmax; int cached; int cache_val[MAX_CHANNELS]; From 08713dcc49060f2d0870483932c6d68ef8430acf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:22:56 +0200 Subject: [PATCH 097/410] ALSA: ump: Choose the protocol when protocol caps are changed When the protocol capability bits are changed via Endpoint Info update notification, we should check the validity of the current protocol and reset it if needed, too. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-2-tiwai@suse.de --- sound/core/ump.c | 22 ++++++++++++++++------ 1 file changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index c7c3581bbbbc..4502de891adf 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -657,6 +657,17 @@ static int ump_append_string(struct snd_ump_endpoint *ump, char *dest, format == UMP_STREAM_MSG_FORMAT_END); } +/* Choose the default protocol */ +static void choose_default_protocol(struct snd_ump_endpoint *ump) +{ + if (ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK) + return; + if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI2) + ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI2; + else + ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI1; +} + /* handle EP info stream message; update the UMP attributes */ static int ump_handle_ep_info_msg(struct snd_ump_endpoint *ump, const union snd_ump_stream_msg *buf) @@ -678,6 +689,10 @@ static int ump_handle_ep_info_msg(struct snd_ump_endpoint *ump, ump_dbg(ump, "EP info: version=%x, num_blocks=%x, proto_caps=%x\n", ump->info.version, ump->info.num_blocks, ump->info.protocol_caps); + + ump->info.protocol &= ump->info.protocol_caps; + choose_default_protocol(ump); + return 1; /* finished */ } @@ -1040,12 +1055,7 @@ int snd_ump_parse_endpoint(struct snd_ump_endpoint *ump) ump_dbg(ump, "Unable to get UMP EP stream config\n"); /* If no protocol is set by some reason, assume the valid one */ - if (!(ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK)) { - if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI2) - ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI2; - else if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI1) - ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI1; - } + choose_default_protocol(ump); /* Query and create blocks from Function Blocks */ for (blk = 0; blk < ump->info.num_blocks; blk++) { From b28654233f654cc16134b72751bf371c7bf0ce3a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:22:57 +0200 Subject: [PATCH 098/410] ALSA: usb-audio: Accept multiple protocols in GTBs It's valid to give different protocols via multiple GTBs; e.g. a MIDI 1.0 port is embedded in a MIDI 2.0 device that talks with MIDI 2.0 protocol. However, the current driver implementation assumes only a single protocol over the whole Endpoint, and it can't handle such a scenario. This patch changes the driver's behavior to parse GTBs to accept multiple protocols. Instead of switching to the last given protocol, it adds the protocol capability bits now. Meanwhile, the default protocol is chosen by the first given protocol in GTBs. Practically seen, this should be a minor issue, as new devices should specify the protocols properly via UMP Endpoint Info messages, so this is rather just covering a corner case. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-3-tiwai@suse.de --- sound/usb/midi2.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index 820d3e4b672a..fa655aa4a56f 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -607,12 +607,8 @@ static int parse_group_terminal_block(struct snd_usb_midi2_ump *rmidi, return 0; } - if (ump->info.protocol && ump->info.protocol != protocol) - usb_audio_info(rmidi->umidi->chip, - "Overriding preferred MIDI protocol in GTB %d: %x -> %x\n", - rmidi->usb_block_id, ump->info.protocol, - protocol); - ump->info.protocol = protocol; + if (!ump->info.protocol) + ump->info.protocol = protocol; protocol_caps = protocol; switch (desc->bMIDIProtocol) { @@ -624,13 +620,7 @@ static int parse_group_terminal_block(struct snd_usb_midi2_ump *rmidi, break; } - if (ump->info.protocol_caps && ump->info.protocol_caps != protocol_caps) - usb_audio_info(rmidi->umidi->chip, - "Overriding MIDI protocol caps in GTB %d: %x -> %x\n", - rmidi->usb_block_id, ump->info.protocol_caps, - protocol_caps); - ump->info.protocol_caps = protocol_caps; - + ump->info.protocol_caps |= protocol_caps; return 0; } From ac3a9185bd5f547cb16ef1388e8786ad5a6e8858 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:22:58 +0200 Subject: [PATCH 099/410] ALSA: usb-audio: Set MIDI1 flag appropriately for GTB MIDI 1.0 entry When a MIDI 1.0 protocol is specified in a GTB entry while others are set in MIDI 2.0, it should be seen as a legacy MIDI 1.0 port. Since recently we allow drivers to set a flag SNDRV_UMP_BLOCK_IS_MIDI1 to a FB for that purpose. This patch tries to set that flag when the device shows such a configuration. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-4-tiwai@suse.de --- sound/usb/midi2.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index fa655aa4a56f..4fb43d9743d7 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -863,9 +863,23 @@ static int create_gtb_block(struct snd_usb_midi2_ump *rmidi, int dir, int blk) fb->info.flags |= SNDRV_UMP_BLOCK_IS_MIDI1 | SNDRV_UMP_BLOCK_IS_LOWSPEED; + /* if MIDI 2.0 protocol is supported and yet the GTB shows MIDI 1.0, + * treat it as a MIDI 1.0-specific block + */ + if (rmidi->ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI2) { + switch (desc->bMIDIProtocol) { + case USB_MS_MIDI_PROTO_1_0_64: + case USB_MS_MIDI_PROTO_1_0_64_JRTS: + case USB_MS_MIDI_PROTO_1_0_128: + case USB_MS_MIDI_PROTO_1_0_128_JRTS: + fb->info.flags |= SNDRV_UMP_BLOCK_IS_MIDI1; + break; + } + } + usb_audio_dbg(umidi->chip, - "Created a UMP block %d from GTB, name=%s\n", - blk, fb->info.name); + "Created a UMP block %d from GTB, name=%s, flags=0x%x\n", + blk, fb->info.name, fb->info.flags); return 0; } From ebaa86c0bddd2c47c516bf2096b17c0bed71d914 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:22:59 +0200 Subject: [PATCH 100/410] ALSA: usb-audio: Update UMP group attributes for GTB blocks, too When a FB is created from a GTB instead of UMP FB Info inquiry, we missed the update of the corresponding UMP Group attributes. Export the call of updater and let it be called from the USB driver. Fixes: 0642a3c5cacc ("ALSA: ump: Update substream name from assigned FB names") Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-5-tiwai@suse.de --- include/sound/ump.h | 1 + sound/core/ump.c | 9 +++++---- sound/usb/midi2.c | 2 ++ 3 files changed, 8 insertions(+), 4 deletions(-) diff --git a/include/sound/ump.h b/include/sound/ump.h index 7484f62fb234..532c2c3ea28e 100644 --- a/include/sound/ump.h +++ b/include/sound/ump.h @@ -123,6 +123,7 @@ static inline int snd_ump_attach_legacy_rawmidi(struct snd_ump_endpoint *ump, int snd_ump_receive_ump_val(struct snd_ump_endpoint *ump, u32 val); int snd_ump_switch_protocol(struct snd_ump_endpoint *ump, unsigned int protocol); +void snd_ump_update_group_attrs(struct snd_ump_endpoint *ump); /* * Some definitions for UMP diff --git a/sound/core/ump.c b/sound/core/ump.c index 4502de891adf..243ecdbb2a6e 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -525,7 +525,7 @@ static void snd_ump_proc_read(struct snd_info_entry *entry, } /* update dir_bits and active flag for all groups in the client */ -static void update_group_attrs(struct snd_ump_endpoint *ump) +void snd_ump_update_group_attrs(struct snd_ump_endpoint *ump) { struct snd_ump_block *fb; struct snd_ump_group *group; @@ -578,6 +578,7 @@ static void update_group_attrs(struct snd_ump_endpoint *ump) } } } +EXPORT_SYMBOL_GPL(snd_ump_update_group_attrs); /* * UMP endpoint and function block handling @@ -863,7 +864,7 @@ static int ump_handle_fb_info_msg(struct snd_ump_endpoint *ump, if (fb) { fill_fb_info(ump, &fb->info, buf); if (ump->parsed) { - update_group_attrs(ump); + snd_ump_update_group_attrs(ump); seq_notify_fb_change(ump, fb); } } @@ -895,7 +896,7 @@ static int ump_handle_fb_name_msg(struct snd_ump_endpoint *ump, buf->raw, 3); /* notify the FB name update to sequencer, too */ if (ret > 0 && ump->parsed) { - update_group_attrs(ump); + snd_ump_update_group_attrs(ump); seq_notify_fb_change(ump, fb); } return ret; @@ -1065,7 +1066,7 @@ int snd_ump_parse_endpoint(struct snd_ump_endpoint *ump) } /* initialize group attributions */ - update_group_attrs(ump); + snd_ump_update_group_attrs(ump); error: ump->parsed = true; diff --git a/sound/usb/midi2.c b/sound/usb/midi2.c index 4fb43d9743d7..692dfc3c182f 100644 --- a/sound/usb/midi2.c +++ b/sound/usb/midi2.c @@ -877,6 +877,8 @@ static int create_gtb_block(struct snd_usb_midi2_ump *rmidi, int dir, int blk) } } + snd_ump_update_group_attrs(rmidi->ump); + usb_audio_dbg(umidi->chip, "Created a UMP block %d from GTB, name=%s, flags=0x%x\n", blk, fb->info.name, fb->info.flags); From 8e3f30b8dc063ef323e535b21f0c6ac7b487ee5a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:23:00 +0200 Subject: [PATCH 101/410] ALSA: seq: Print MIDI 1.0 specific port in proc output When a sequencer port assigned to a UMP Group that is specific to MIDI 1.0 among MIDI 2.0 client, mark it explicitly in the proc output, so that user can see it easily. This is an exceptional case where the message isn't converted to MIDI 1.0 even if the client is running in MIDI 2.0 mode. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-6-tiwai@suse.de --- sound/core/seq/seq_clientmgr.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8c4ee5066afe..276b5b3f68dc 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2633,13 +2633,18 @@ static void snd_seq_info_dump_ports(struct snd_info_buffer *buffer, list_for_each_entry(p, &client->ports_list_head, list) { if (p->capability & SNDRV_SEQ_PORT_CAP_INACTIVE) continue; - snd_iprintf(buffer, " Port %3d : \"%s\" (%c%c%c%c) [%s]\n", + snd_iprintf(buffer, " Port %3d : \"%s\" (%c%c%c%c) [%s]", p->addr.port, p->name, FLAG_PERM_RD(p->capability), FLAG_PERM_WR(p->capability), FLAG_PERM_EX(p->capability), FLAG_PERM_DUPLEX(p->capability), port_direction_name(p->direction)); +#if IS_ENABLED(CONFIG_SND_SEQ_UMP) + if (snd_seq_client_is_midi2(client) && p->is_midi1) + snd_iprintf(buffer, " [MIDI1]"); +#endif + snd_iprintf(buffer, "\n"); snd_seq_info_dump_subscribers(buffer, &p->c_src, 1, " Connecting To: "); snd_seq_info_dump_subscribers(buffer, &p->c_dest, 0, " Connected From: "); } From b977831342ec131aea17831295db48450ada7202 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 11:23:01 +0200 Subject: [PATCH 102/410] ALSA: seq: Fix missing seq port info bit return for MIDI 1.0 block The recent extension added a new ALSA sequencer port info flag bit SNDRV_SEQ_PORT_FLG_IS_MIDI1, but it's not reported back when inquired. Fix it to report properly. Fixes: 0079c9d1e58a ("ALSA: ump: Handle MIDI 1.0 Function Block in MIDI 2.0 protocol") Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807092303.1935-7-tiwai@suse.de --- sound/core/seq/seq_ports.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index 535290f24eed..cc2f8e846584 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -401,6 +401,9 @@ int snd_seq_get_port_info(struct snd_seq_client_port * port, info->time_queue = port->time_queue; } + if (port->is_midi1) + info->flags |= SNDRV_SEQ_PORT_FLG_IS_MIDI1; + /* UMP direction and group */ info->direction = port->direction; info->ump_group = port->ump_group; From c2c0b67dca3cb3b3cea0dd60075a1c5ba77e2fcd Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Wed, 7 Aug 2024 17:02:32 +0200 Subject: [PATCH 103/410] ASoC: tas2781-i2c: Drop weird GPIO code The tas2781-i2c driver gets an IRQ from either ACPI or device tree, then proceeds to check if the IRQ has a corresponding GPIO and in case it does enforce the GPIO as input and set a label on it. This is abuse of the API: - First we cannot guarantee that the numberspaces of the GPIOs and the IRQs are the same, i.e that an IRQ number corresponds to a GPIO number like that. - Second, GPIO chips and IRQ chips should be treated as orthogonal APIs, the irqchip needs to ascertain that the backing GPIO line is set to input etc just using the irqchip. - Third it is using the legacy API which should not be used in new code yet this was added just a year ago. Delete the offending code. If this creates problems the GPIO and irqchip maintainers can help to fix the issues. It *should* not create any problems, because the irq isn't used anywhere in the driver, it's just obtained and then left unused. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Linus Walleij Link: https://patch.msgid.link/20240807-asoc-tas-gpios-v2-1-bd0f2705d58b@linaro.org Signed-off-by: Mark Brown --- include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 24 +++--------------------- 5 files changed, 5 insertions(+), 32 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 18161d02a96f..dbda552398b5 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -81,11 +81,6 @@ struct tasdevice { bool is_loaderr; }; -struct tasdevice_irqinfo { - int irq_gpio; - int irq; -}; - struct calidata { unsigned char *data; unsigned long total_sz; @@ -93,7 +88,6 @@ struct calidata { struct tasdevice_priv { struct tasdevice tasdevice[TASDEVICE_MAX_CHANNELS]; - struct tasdevice_irqinfo irq_info; struct tasdevice_rca rcabin; struct calidata cali_data; struct tasdevice_fw *fmw; @@ -115,6 +109,7 @@ struct tasdevice_priv { unsigned int chip_id; unsigned int sysclk; + int irq; int cur_prog; int cur_conf; int fw_state; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 49bd7097d892..8a7fe48043d2 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -814,7 +814,7 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) } else return -ENODEV; - tas_hda->priv->irq_info.irq = clt->irq; + tas_hda->priv->irq = clt->irq; ret = tas2781_read_acpi(tas_hda->priv, device_name); if (ret) return dev_err_probe(tas_hda->dev, ret, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 58abbc098a91..664c371796d6 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include @@ -411,8 +410,6 @@ EXPORT_SYMBOL_GPL(tasdevice_dsp_remove); void tasdevice_remove(struct tasdevice_priv *tas_priv) { - if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) - gpio_free(tas_priv->irq_info.irq_gpio); mutex_destroy(&tas_priv->codec_lock); } EXPORT_SYMBOL_GPL(tasdevice_remove); diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 8f9a3ae7153e..f3a7605f0710 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -13,7 +13,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a57c045afdd..46cc568cea76 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include @@ -758,7 +757,7 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) { struct i2c_client *client = (struct i2c_client *)tas_priv->client; unsigned int dev_addrs[TASDEVICE_MAX_CHANNELS]; - int rc, i, ndev = 0; + int i, ndev = 0; if (tas_priv->isacpi) { ndev = device_property_read_u32_array(&client->dev, @@ -773,7 +772,7 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) "ti,audio-slots", dev_addrs, ndev); } - tas_priv->irq_info.irq_gpio = + tas_priv->irq = acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); } else if (IS_ENABLED(CONFIG_OF)) { struct device_node *np = tas_priv->dev->of_node; @@ -785,7 +784,7 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) dev_addrs[ndev++] = addr; } - tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + tas_priv->irq = of_irq_get(np, 0); } else { ndev = 1; dev_addrs[0] = client->addr; @@ -801,23 +800,6 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) __func__); strcpy(tas_priv->dev_name, tasdevice_id[tas_priv->chip_id].name); - - if (gpio_is_valid(tas_priv->irq_info.irq_gpio)) { - rc = gpio_request(tas_priv->irq_info.irq_gpio, - "AUDEV-IRQ"); - if (!rc) { - gpio_direction_input( - tas_priv->irq_info.irq_gpio); - - tas_priv->irq_info.irq = - gpio_to_irq(tas_priv->irq_info.irq_gpio); - } else - dev_err(tas_priv->dev, "%s: GPIO %d request error\n", - __func__, tas_priv->irq_info.irq_gpio); - } else - dev_err(tas_priv->dev, - "Looking up irq-gpio property failed %d\n", - tas_priv->irq_info.irq_gpio); } static int tasdevice_i2c_probe(struct i2c_client *i2c) From 1c4b509edad15192bfb64c81d3c305bbae8070db Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Wed, 7 Aug 2024 17:02:33 +0200 Subject: [PATCH 104/410] ASoC: tas2781-i2c: Get the right GPIO line The code is obtaining a GPIO reset using the reset GPIO name "reset-gpios", but the gpiolib is already adding the suffix "-gpios" to anything passed to this function and will be looking for "reset-gpios-gpios" which is most certainly not what the author desired. Fix it up. Fixes: ef3bcde75d06 ("ASoC: tas2781: Add tas2781 driver") Signed-off-by: Linus Walleij Link: https://patch.msgid.link/20240807-asoc-tas-gpios-v2-2-bd0f2705d58b@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 46cc568cea76..313c17f76ca4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -794,7 +794,7 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) tas_priv->tasdevice[i].dev_addr = dev_addrs[i]; tas_priv->reset = devm_gpiod_get_optional(&client->dev, - "reset-gpios", GPIOD_OUT_HIGH); + "reset", GPIOD_OUT_HIGH); if (IS_ERR(tas_priv->reset)) dev_err(tas_priv->dev, "%s Can't get reset GPIO\n", __func__); From caab9a1cbb9a9fca24ceabeef57b3764d861ad32 Mon Sep 17 00:00:00 2001 From: Linus Walleij Date: Wed, 7 Aug 2024 17:02:34 +0200 Subject: [PATCH 105/410] ASoC: tas*: Drop unused GPIO includes These drivers all use and has no business including the legacy headers or . Drop the surplus includes. Signed-off-by: Linus Walleij Link: https://patch.msgid.link/20240807-asoc-tas-gpios-v2-3-bd0f2705d58b@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - 4 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 9e68afc09897..684d52ec6600 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -12,7 +12,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tas2764.c b/sound/soc/codecs/tas2764.c index 5eaddf07aadc..d482cd194c08 100644 --- a/sound/soc/codecs/tas2764.c +++ b/sound/soc/codecs/tas2764.c @@ -10,7 +10,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 5601fba17c96..9f93b230652a 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/codecs/tas2780.c b/sound/soc/codecs/tas2780.c index 6902bfef185b..a1963415c931 100644 --- a/sound/soc/codecs/tas2780.c +++ b/sound/soc/codecs/tas2780.c @@ -7,7 +7,6 @@ #include #include #include -#include #include #include #include From 94cd66f8dcad8faba7a3425c324c83464ac60887 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:51 +0200 Subject: [PATCH 106/410] ALSA: portman2x4: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-2-tiwai@suse.de --- sound/drivers/portman2x4.c | 19 +++++++++---------- 1 file changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 6fd9e8870021..54d818d2f53d 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -385,9 +385,8 @@ static void portman_flush_input(struct portman *pm, unsigned char port) command = RXDATA1; break; default: - snd_printk(KERN_WARNING - "portman_flush_input() Won't flush port %i\n", - port); + dev_warn(pm->card->dev, "%s Won't flush port %i\n", + __func__, port); return; } @@ -712,7 +711,7 @@ static int snd_portman_probe(struct platform_device *pdev) err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printd("Cannot create card\n"); + dev_dbg(&pdev->dev, "Cannot create card\n"); return err; } strcpy(card->driver, DRIVER_NAME); @@ -726,21 +725,21 @@ static int snd_portman_probe(struct platform_device *pdev) &portman_cb, /* callbacks */ pdev->id); /* device number */ if (pardev == NULL) { - snd_printd("Cannot register pardevice\n"); + dev_dbg(card->dev, "Cannot register pardevice\n"); err = -EIO; goto __err; } /* claim parport */ if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + dev_dbg(card->dev, "Cannot claim parport 0x%lx\n", pardev->port->base); err = -EIO; goto free_pardev; } err = portman_create(card, pardev, &pm); if (err < 0) { - snd_printd("Cannot create main component\n"); + dev_dbg(card->dev, "Cannot create main component\n"); goto release_pardev; } card->private_data = pm; @@ -754,7 +753,7 @@ static int snd_portman_probe(struct platform_device *pdev) err = snd_portman_rawmidi_create(card); if (err < 0) { - snd_printd("Creating Rawmidi component failed\n"); + dev_dbg(card->dev, "Creating Rawmidi component failed\n"); goto __err; } @@ -768,11 +767,11 @@ static int snd_portman_probe(struct platform_device *pdev) /* At this point card will be usable */ err = snd_card_register(card); if (err < 0) { - snd_printd("Cannot register card\n"); + dev_dbg(card->dev, "Cannot register card\n"); goto __err; } - snd_printk(KERN_INFO "Portman 2x4 on 0x%lx\n", p->base); + dev_info(card->dev, "Portman 2x4 on 0x%lx\n", p->base); return 0; release_pardev: From f7d4adacc588d995e76a85f42064af2c7dc0fa83 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:52 +0200 Subject: [PATCH 107/410] ALSA: mts64: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-3-tiwai@suse.de --- sound/drivers/mts64.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index b1b333d1cf39..6fc255a6754d 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -652,8 +652,8 @@ static int snd_mts64_ctl_create(struct snd_card *card, for (i = 0; control[i]; ++i) { err = snd_ctl_add(card, snd_ctl_new1(control[i], mts)); if (err < 0) { - snd_printd("Cannot create control: %s\n", - control[i]->name); + dev_dbg(card->dev, "Cannot create control: %s\n", + control[i]->name); return err; } } @@ -926,7 +926,7 @@ static int snd_mts64_probe(struct platform_device *pdev) err = snd_card_new(&pdev->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { - snd_printd("Cannot create card\n"); + dev_dbg(&pdev->dev, "Cannot create card\n"); return err; } strcpy(card->driver, DRIVER_NAME); @@ -940,21 +940,21 @@ static int snd_mts64_probe(struct platform_device *pdev) &mts64_cb, /* callbacks */ pdev->id); /* device number */ if (!pardev) { - snd_printd("Cannot register pardevice\n"); + dev_dbg(card->dev, "Cannot register pardevice\n"); err = -EIO; goto __err; } /* claim parport */ if (parport_claim(pardev)) { - snd_printd("Cannot claim parport 0x%lx\n", pardev->port->base); + dev_dbg(card->dev, "Cannot claim parport 0x%lx\n", pardev->port->base); err = -EIO; goto free_pardev; } err = snd_mts64_create(card, pardev, &mts); if (err < 0) { - snd_printd("Cannot create main component\n"); + dev_dbg(card->dev, "Cannot create main component\n"); goto release_pardev; } card->private_data = mts; @@ -968,7 +968,7 @@ static int snd_mts64_probe(struct platform_device *pdev) err = snd_mts64_rawmidi_create(card); if (err < 0) { - snd_printd("Creating Rawmidi component failed\n"); + dev_dbg(card->dev, "Creating Rawmidi component failed\n"); goto __err; } @@ -982,11 +982,11 @@ static int snd_mts64_probe(struct platform_device *pdev) /* At this point card will be usable */ err = snd_card_register(card); if (err < 0) { - snd_printd("Cannot register card\n"); + dev_dbg(card->dev, "Cannot register card\n"); goto __err; } - snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); + dev_info(card->dev, "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; release_pardev: From 2bddeda8ac8d5f773edcb63fd60fb2db332b029d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:53 +0200 Subject: [PATCH 108/410] ALSA: mpu401: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-4-tiwai@suse.de --- sound/drivers/mpu401/mpu401.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 3398aee33baa..cd01af5fa4ed 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -56,7 +56,7 @@ static int snd_mpu401_create(struct device *devptr, int dev, int err; if (!uart_enter[dev]) - snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n"); + dev_err(devptr, "the uart_enter option is obsolete; remove it\n"); *rcard = NULL; err = snd_devm_card_new(devptr, index[dev], id[dev], THIS_MODULE, @@ -75,7 +75,7 @@ static int snd_mpu401_create(struct device *devptr, int dev, err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0, irq[dev], NULL); if (err < 0) { - printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]); + dev_err(devptr, "MPU401 not detected at 0x%lx\n", port[dev]); return err; } @@ -90,11 +90,11 @@ static int snd_mpu401_probe(struct platform_device *devptr) struct snd_card *card; if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "specify port\n"); + dev_err(&devptr->dev, "specify port\n"); return -EINVAL; } if (irq[dev] == SNDRV_AUTO_IRQ) { - snd_printk(KERN_ERR "specify or disable IRQ\n"); + dev_err(&devptr->dev, "specify or disable IRQ\n"); return -EINVAL; } err = snd_mpu401_create(&devptr->dev, dev, &card); @@ -133,11 +133,11 @@ static int snd_mpu401_pnp(int dev, struct pnp_dev *device, { if (!pnp_port_valid(device, 0) || pnp_port_flags(device, 0) & IORESOURCE_DISABLED) { - snd_printk(KERN_ERR "no PnP port\n"); + dev_err(&device->dev, "no PnP port\n"); return -ENODEV; } if (pnp_port_len(device, 0) < IO_EXTENT) { - snd_printk(KERN_ERR "PnP port length is %llu, expected %d\n", + dev_err(&device->dev, "PnP port length is %llu, expected %d\n", (unsigned long long)pnp_port_len(device, 0), IO_EXTENT); return -ENODEV; @@ -146,7 +146,7 @@ static int snd_mpu401_pnp(int dev, struct pnp_dev *device, if (!pnp_irq_valid(device, 0) || pnp_irq_flags(device, 0) & IORESOURCE_DISABLED) { - snd_printk(KERN_WARNING "no PnP irq, using polling\n"); + dev_warn(&device->dev, "no PnP irq, using polling\n"); irq[dev] = -1; } else { irq[dev] = pnp_irq(device, 0); @@ -234,7 +234,7 @@ static int __init alsa_card_mpu401_init(void) if (!snd_mpu401_devices) { #ifdef MODULE - printk(KERN_ERR "MPU-401 device not found or device busy\n"); + pr_err("MPU-401 device not found or device busy\n"); #endif snd_mpu401_unregister_all(); return -ENODEV; From 1fa884ebeb73cfe5ff078606b8888aec38103e2b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:54 +0200 Subject: [PATCH 109/410] ALSA: mpu401_uart: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The assignment of mpu->rmidi was moved to an earlier place, so that dev_*() can access to the proper device pointer. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-5-tiwai@suse.de --- sound/drivers/mpu401/mpu401_uart.c | 31 ++++++++++++++++-------------- 1 file changed, 17 insertions(+), 14 deletions(-) diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index f435b9b4ae24..8e3318e17717 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -73,8 +73,9 @@ static void snd_mpu401_uart_clear_rx(struct snd_mpu401 *mpu) mpu->read(mpu, MPU401D(mpu)); #ifdef CONFIG_SND_DEBUG if (timeout <= 0) - snd_printk(KERN_ERR "cmd: clear rx timeout (status = 0x%x)\n", - mpu->read(mpu, MPU401C(mpu))); + dev_err(mpu->rmidi->dev, + "cmd: clear rx timeout (status = 0x%x)\n", + mpu->read(mpu, MPU401C(mpu))); #endif } @@ -224,8 +225,9 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, udelay(10); #ifdef CONFIG_SND_DEBUG if (!timeout) - snd_printk(KERN_ERR "cmd: tx timeout (status = 0x%x)\n", - mpu->read(mpu, MPU401C(mpu))); + dev_err(mpu->rmidi->dev, + "cmd: tx timeout (status = 0x%x)\n", + mpu->read(mpu, MPU401C(mpu))); #endif } mpu->write(mpu, cmd, MPU401C(mpu)); @@ -244,10 +246,11 @@ static int snd_mpu401_uart_cmd(struct snd_mpu401 * mpu, unsigned char cmd, ok = 1; spin_unlock_irqrestore(&mpu->input_lock, flags); if (!ok) { - snd_printk(KERN_ERR "cmd: 0x%x failed at 0x%lx " - "(status = 0x%x, data = 0x%x)\n", cmd, mpu->port, - mpu->read(mpu, MPU401C(mpu)), - mpu->read(mpu, MPU401D(mpu))); + dev_err(mpu->rmidi->dev, + "cmd: 0x%x failed at 0x%lx (status = 0x%x, data = 0x%x)\n", + cmd, mpu->port, + mpu->read(mpu, MPU401C(mpu)), + mpu->read(mpu, MPU401D(mpu))); return 1; } return 0; @@ -546,13 +549,14 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, spin_lock_init(&mpu->timer_lock); mpu->hardware = hardware; mpu->irq = -1; + mpu->rmidi = rmidi; if (! (info_flags & MPU401_INFO_INTEGRATED)) { int res_size = hardware == MPU401_HW_PC98II ? 4 : 2; mpu->res = request_region(port, res_size, "MPU401 UART"); if (!mpu->res) { - snd_printk(KERN_ERR "mpu401_uart: " - "unable to grab port 0x%lx size %d\n", - port, res_size); + dev_err(rmidi->dev, + "mpu401_uart: unable to grab port 0x%lx size %d\n", + port, res_size); err = -EBUSY; goto free_device; } @@ -572,8 +576,8 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, if (irq >= 0) { if (request_irq(irq, snd_mpu401_uart_interrupt, 0, "MPU401 UART", (void *) mpu)) { - snd_printk(KERN_ERR "mpu401_uart: " - "unable to grab IRQ %d\n", irq); + dev_err(rmidi->dev, + "mpu401_uart: unable to grab IRQ %d\n", irq); err = -EBUSY; goto free_device; } @@ -599,7 +603,6 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, if (out_enable) rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX; } - mpu->rmidi = rmidi; if (rrawmidi) *rrawmidi = rmidi; return 0; From 1e594f9a7ba8bac7dd8e5bdd74cd786e0e00246c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:55 +0200 Subject: [PATCH 110/410] ALSA: mtpav: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The commented-out debug prints got removed, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-6-tiwai@suse.de --- sound/drivers/mtpav.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index f212f233ea61..946184a2a758 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -285,10 +285,6 @@ static void snd_mtpav_output_port_write(struct mtpav *mtp_card, snd_mtpav_send_byte(mtp_card, 0xf5); snd_mtpav_send_byte(mtp_card, portp->hwport); - /* - snd_printk(KERN_DEBUG "new outport: 0x%x\n", - (unsigned int) portp->hwport); - */ if (!(outbyte & 0x80) && portp->running_status) snd_mtpav_send_byte(mtp_card, portp->running_status); } @@ -522,8 +518,6 @@ static void snd_mtpav_read_bytes(struct mtpav *mcrd) u8 sbyt = snd_mtpav_getreg(mcrd, SREG); - /* printk(KERN_DEBUG "snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); */ - if (!(sbyt & SIGS_BYTE)) return; @@ -569,13 +563,13 @@ static int snd_mtpav_get_ISA(struct mtpav *mcard) mcard->res_port = devm_request_region(mcard->card->dev, port, 3, "MotuMTPAV MIDI"); if (!mcard->res_port) { - snd_printk(KERN_ERR "MTVAP port 0x%lx is busy\n", port); + dev_err(mcard->card->dev, "MTVAP port 0x%lx is busy\n", port); return -EBUSY; } mcard->port = port; if (devm_request_irq(mcard->card->dev, irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) { - snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); + dev_err(mcard->card->dev, "MTVAP IRQ %d busy\n", irq); return -EBUSY; } mcard->irq = irq; @@ -717,7 +711,9 @@ static int snd_mtpav_probe(struct platform_device *dev) card->private_free = snd_mtpav_free; platform_set_drvdata(dev, card); - printk(KERN_INFO "Motu MidiTimePiece on parallel port irq: %d ioport: 0x%lx\n", irq, port); + dev_info(card->dev, + "Motu MidiTimePiece on parallel port irq: %d ioport: 0x%lx\n", + irq, port); return 0; } From a2fa882d6dea19b4488b10f40334e04c77503ffc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:56 +0200 Subject: [PATCH 111/410] ALSA: opl3: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some debug prints are cleaned up with a macro, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-7-tiwai@suse.de --- sound/drivers/opl3/opl3_lib.c | 18 +++---- sound/drivers/opl3/opl3_midi.c | 95 ++++++++++++++------------------- sound/drivers/opl3/opl3_oss.c | 12 +++-- sound/drivers/opl3/opl3_synth.c | 4 +- 4 files changed, 56 insertions(+), 73 deletions(-) diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 6c1f1cc092d8..4e57e3b2f118 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -92,7 +92,7 @@ static int snd_opl3_detect(struct snd_opl3 * opl3) opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_IRQ_RESET); signature = stat1 = inb(opl3->l_port); /* Status register */ if ((stat1 & 0xe0) != 0x00) { /* Should be 0x00 */ - snd_printd("OPL3: stat1 = 0x%x\n", stat1); + dev_dbg(opl3->card->dev, "OPL3: stat1 = 0x%x\n", stat1); return -ENODEV; } /* Set timer1 to 0xff */ @@ -108,7 +108,7 @@ static int snd_opl3_detect(struct snd_opl3 * opl3) /* Reset the IRQ of the FM chip */ opl3->command(opl3, OPL3_LEFT | OPL3_REG_TIMER_CONTROL, OPL3_IRQ_RESET); if ((stat2 & 0xe0) != 0xc0) { /* There is no YM3812 */ - snd_printd("OPL3: stat2 = 0x%x\n", stat2); + dev_dbg(opl3->card->dev, "OPL3: stat2 = 0x%x\n", stat2); return -ENODEV; } @@ -289,9 +289,6 @@ void snd_opl3_interrupt(struct snd_hwdep * hw) opl3 = hw->private_data; status = inb(opl3->l_port); -#if 0 - snd_printk(KERN_DEBUG "AdLib IRQ status = 0x%x\n", status); -#endif if (!(status & 0x80)) return; @@ -365,7 +362,8 @@ EXPORT_SYMBOL(snd_opl3_new); int snd_opl3_init(struct snd_opl3 *opl3) { if (! opl3->command) { - printk(KERN_ERR "snd_opl3_init: command not defined!\n"); + dev_err(opl3->card->dev, + "snd_opl3_init: command not defined!\n"); return -EINVAL; } @@ -405,14 +403,14 @@ int snd_opl3_create(struct snd_card *card, if (! integrated) { opl3->res_l_port = request_region(l_port, 2, "OPL2/3 (left)"); if (!opl3->res_l_port) { - snd_printk(KERN_ERR "opl3: can't grab left port 0x%lx\n", l_port); + dev_err(card->dev, "opl3: can't grab left port 0x%lx\n", l_port); snd_device_free(card, opl3); return -EBUSY; } if (r_port != 0) { opl3->res_r_port = request_region(r_port, 2, "OPL2/3 (right)"); if (!opl3->res_r_port) { - snd_printk(KERN_ERR "opl3: can't grab right port 0x%lx\n", r_port); + dev_err(card->dev, "opl3: can't grab right port 0x%lx\n", r_port); snd_device_free(card, opl3); return -EBUSY; } @@ -432,8 +430,8 @@ int snd_opl3_create(struct snd_card *card, opl3->command = &snd_opl2_command; err = snd_opl3_detect(opl3); if (err < 0) { - snd_printd("OPL2/3 chip not detected at 0x%lx/0x%lx\n", - opl3->l_port, opl3->r_port); + dev_dbg(card->dev, "OPL2/3 chip not detected at 0x%lx/0x%lx\n", + opl3->l_port, opl3->r_port); snd_device_free(card, opl3); return err; } diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index e2b7be67f0e3..9bee454441b0 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -11,6 +11,13 @@ #include "opl3_voice.h" #include +#ifdef DEBUG_MIDI +#define opl3_dbg(opl3, fmt, ...) \ + dev_dbg(((struct snd_opl3 *)(opl3))->card->dev, fmt, ##__VA_ARGS__) +#else +#define opl3_dbg(opl3, fmt, ...) do {} while (0) +#endif + static void snd_opl3_note_off_unsafe(void *p, int note, int vel, struct snd_midi_channel *chan); /* @@ -107,14 +114,17 @@ static void snd_opl3_calc_pitch(unsigned char *fnum, unsigned char *blocknum, #ifdef DEBUG_ALLOC -static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) { +static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) +{ int i; - char *str = "x.24"; + const char *str = "x.24"; + char buf[MAX_OPL3_VOICES + 1]; - printk(KERN_DEBUG "time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); for (i = 0; i < opl3->max_voices; i++) - printk(KERN_CONT "%c", *(str + opl3->voices[i].state + 1)); - printk(KERN_CONT "\n"); + buf[i] = str[opl3->voices[i].state + 1]; + buf[i] = 0; + dev_dbg(opl3->card->dev, "time %.5i: %s [%.2i]: %s\n", + opl3->use_time, s, voice, buf); } #endif @@ -203,9 +213,10 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, for (i = 0; i < END; i++) { if (best[i].voice >= 0) { #ifdef DEBUG_ALLOC - printk(KERN_DEBUG "%s %iop allocation on voice %i\n", - alloc_type[i], instr_4op ? 4 : 2, - best[i].voice); + dev_dbg(opl3->card->dev, + "%s %iop allocation on voice %i\n", + alloc_type[i], instr_4op ? 4 : 2, + best[i].voice); #endif return best[i].voice; } @@ -302,10 +313,8 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) opl3 = p; -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "Note on, ch %i, inst %i, note %i, vel %i\n", - chan->number, chan->midi_program, note, vel); -#endif + opl3_dbg(opl3, "Note on, ch %i, inst %i, note %i, vel %i\n", + chan->number, chan->midi_program, note, vel); /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, drum voice numbers are notes on drum channel */ @@ -358,10 +367,8 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " --> OPL%i instrument: %s\n", - instr_4op ? 3 : 2, patch->name); -#endif + opl3_dbg(opl3, " --> OPL%i instrument: %s\n", + instr_4op ? 3 : 2, patch->name); /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, allocate voice on free OPL3 channel */ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { @@ -422,10 +429,8 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } } -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " --> setting OPL3 connection: 0x%x\n", - opl3->connection_reg); -#endif + opl3_dbg(opl3, " --> setting OPL3 connection: 0x%x\n", + opl3->connection_reg); /* * calculate volume depending on connection * between FM operators (see include/opl3.h) @@ -457,9 +462,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* Program the FM voice characteristics */ for (i = 0; i < (instr_4op ? 4 : 2); i++) { -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " --> programming operator %i\n", i); -#endif + opl3_dbg(opl3, " --> programming operator %i\n", i); op_offset = snd_opl3_regmap[voice_offset][i]; /* Set OPL3 AM_VIB register of requested voice/operator */ @@ -537,9 +540,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* Set output sound flag */ blocknum |= OPL3_KEYON_BIT; -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " --> trigger voice %i\n", voice); -#endif + opl3_dbg(opl3, " --> trigger voice %i\n", voice); /* Set OPL3 KEYON_BLOCK register of requested voice */ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); opl3->command(opl3, opl3_reg, blocknum); @@ -593,9 +594,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) bank = 0; prg = extra_prg - 1; } -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " *** allocating extra program\n"); -#endif + opl3_dbg(opl3, " *** allocating extra program\n"); goto __extra_prg; } spin_unlock_irqrestore(&opl3->voice_lock, flags); @@ -624,9 +623,7 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) } /* kill voice */ -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG " --> kill voice %i\n", voice); -#endif + opl3_dbg(opl3, " --> kill voice %i\n", voice); opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); /* clear Key ON bit */ opl3->command(opl3, opl3_reg, vp->keyon_reg); @@ -660,10 +657,8 @@ static void snd_opl3_note_off_unsafe(void *p, int note, int vel, opl3 = p; -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n", - chan->number, chan->midi_program, note); -#endif + opl3_dbg(opl3, "Note off, ch %i, inst %i, note %i\n", + chan->number, chan->midi_program, note); if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel && use_internal_drums) { @@ -703,10 +698,8 @@ void snd_opl3_note_off(void *p, int note, int vel, */ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *chan) { -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", - chan->number, chan->midi_program); -#endif + opl3_dbg(p, "Key pressure, ch#: %i, inst#: %i\n", + chan->number, chan->midi_program); } /* @@ -714,10 +707,8 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha */ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) { -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", - chan->number, chan->midi_program); -#endif + opl3_dbg(p, "Terminate note, ch#: %i, inst#: %i\n", + chan->number, chan->midi_program); } static void snd_opl3_update_pitch(struct snd_opl3 *opl3, int voice) @@ -803,10 +794,8 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) struct snd_opl3 *opl3; opl3 = p; -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "Controller, TYPE = %i, ch#: %i, inst#: %i\n", - type, chan->number, chan->midi_program); -#endif + opl3_dbg(opl3, "Controller, TYPE = %i, ch#: %i, inst#: %i\n", + type, chan->number, chan->midi_program); switch (type) { case MIDI_CTL_MSB_MODWHEEL: @@ -837,10 +826,8 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_channel_set *chset) { -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", - chan->number, chan->midi_program); -#endif + opl3_dbg(p, "NRPN, ch#: %i, inst#: %i\n", + chan->number, chan->midi_program); } /* @@ -849,7 +836,5 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset) { -#ifdef DEBUG_MIDI - snd_printk(KERN_DEBUG "SYSEX\n"); -#endif + opl3_dbg(p, "SYSEX\n"); } diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 7645365eec89..6d39b2b77b80 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -193,14 +193,14 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -EINVAL; if (count < (int)sizeof(sbi)) { - snd_printk(KERN_ERR "FM Error: Patch record too short\n"); + dev_err(opl3->card->dev, "FM Error: Patch record too short\n"); return -EINVAL; } if (copy_from_user(&sbi, buf, sizeof(sbi))) return -EFAULT; if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk(KERN_ERR "FM Error: Invalid instrument number %d\n", + dev_err(opl3->card->dev, "FM Error: Invalid instrument number %d\n", sbi.channel); return -EINVAL; } @@ -220,13 +220,15 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, unsigned long ioarg) { + struct snd_opl3 *opl3; + if (snd_BUG_ON(!arg)) return -ENXIO; + opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: - snd_printk(KERN_ERR "OPL3: " - "Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. " - "Fix the program.\n"); + dev_err(opl3->card->dev, + "OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n"); return -EINVAL; case SNDCTL_SYNTH_MEMAVL: diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 97d30a833ac8..10f622b439a0 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -158,10 +158,8 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, return 0; #endif -#ifdef CONFIG_SND_DEBUG default: - snd_printk(KERN_WARNING "unknown IOCTL: 0x%x\n", cmd); -#endif + dev_dbg(opl3->card->dev, "unknown IOCTL: 0x%x\n", cmd); } return -ENOTTY; } From 7debf0350e264c431f8c029a15d799317dd5d052 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:57 +0200 Subject: [PATCH 112/410] ALSA: opl4: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-8-tiwai@suse.de --- sound/drivers/opl4/opl4_lib.c | 8 ++++---- sound/drivers/opl4/yrw801.c | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 035645eb5e8d..8fa61596875a 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -114,7 +114,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) snd_opl4_enable_opl4(opl4); id1 = snd_opl4_read(opl4, OPL4_REG_MEMORY_CONFIGURATION); - snd_printdd("OPL4[02]=%02x\n", id1); + dev_dbg(opl4->card->dev, "OPL4[02]=%02x\n", id1); switch (id1 & OPL4_DEVICE_ID_MASK) { case 0x20: opl4->hardware = OPL3_HW_OPL4; @@ -130,7 +130,7 @@ static int snd_opl4_detect(struct snd_opl4 *opl4) snd_opl4_write(opl4, OPL4_REG_MIX_CONTROL_PCM, 0xff); id1 = snd_opl4_read(opl4, OPL4_REG_MIX_CONTROL_FM); id2 = snd_opl4_read(opl4, OPL4_REG_MIX_CONTROL_PCM); - snd_printdd("OPL4 id1=%02x id2=%02x\n", id1, id2); + dev_dbg(opl4->card->dev, "OPL4 id1=%02x id2=%02x\n", id1, id2); if (id1 != 0x00 || id2 != 0xff) return -ENODEV; @@ -200,7 +200,7 @@ int snd_opl4_create(struct snd_card *card, opl4->res_fm_port = request_region(fm_port, 8, "OPL4 FM"); opl4->res_pcm_port = request_region(pcm_port, 8, "OPL4 PCM/MIX"); if (!opl4->res_fm_port || !opl4->res_pcm_port) { - snd_printk(KERN_ERR "opl4: can't grab ports 0x%lx, 0x%lx\n", fm_port, pcm_port); + dev_err(card->dev, "opl4: can't grab ports 0x%lx, 0x%lx\n", fm_port, pcm_port); snd_opl4_free(opl4); return -EBUSY; } @@ -214,7 +214,7 @@ int snd_opl4_create(struct snd_card *card, err = snd_opl4_detect(opl4); if (err < 0) { snd_opl4_free(opl4); - snd_printd("OPL4 chip not detected at %#lx/%#lx\n", fm_port, pcm_port); + dev_dbg(card->dev, "OPL4 chip not detected at %#lx/%#lx\n", fm_port, pcm_port); return err; } diff --git a/sound/drivers/opl4/yrw801.c b/sound/drivers/opl4/yrw801.c index 6c335492d082..9e464b84b905 100644 --- a/sound/drivers/opl4/yrw801.c +++ b/sound/drivers/opl4/yrw801.c @@ -43,7 +43,7 @@ int snd_yrw801_detect(struct snd_opl4 *opl4) snd_opl4_read_memory(opl4, buf, 0x1ffffe, 2); if (buf[0] != 0x01) return -ENODEV; - snd_printdd("YRW801 ROM version %02x.%02x\n", buf[0], buf[1]); + dev_dbg(opl4->card->dev, "YRW801 ROM version %02x.%02x\n", buf[0], buf[1]); return 0; } From 4d82bf10d1625d4b19443af1aaeca2b4202e0334 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:58 +0200 Subject: [PATCH 113/410] ALSA: serial-u16550: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-9-tiwai@suse.de --- sound/drivers/serial-u16550.c | 41 ++++++++++++++++++----------------- 1 file changed, 21 insertions(+), 20 deletions(-) diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 3cbc7a4adcb4..f66e01624c68 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -224,9 +224,9 @@ static void snd_uart16550_io_loop(struct snd_uart16550 * uart) snd_rawmidi_receive(uart->midi_input[substream], &c, 1); if (status & UART_LSR_OE) - snd_printk(KERN_WARNING - "%s: Overrun on device at 0x%lx\n", - uart->rmidi->name, uart->base); + dev_warn(uart->card->dev, + "%s: Overrun on device at 0x%lx\n", + uart->rmidi->name, uart->base); } /* remember the last stream */ @@ -323,7 +323,8 @@ static int snd_uart16550_detect(struct snd_uart16550 *uart) } if (!devm_request_region(uart->card->dev, io_base, 8, "Serial MIDI")) { - snd_printk(KERN_ERR "u16550: can't grab port 0x%lx\n", io_base); + dev_err(uart->card->dev, + "u16550: can't grab port 0x%lx\n", io_base); return -EBUSY; } @@ -619,9 +620,9 @@ static int snd_uart16550_output_byte(struct snd_uart16550 *uart, } } else { if (!snd_uart16550_write_buffer(uart, midi_byte)) { - snd_printk(KERN_WARNING - "%s: Buffer overrun on device at 0x%lx\n", - uart->rmidi->name, uart->base); + dev_warn(uart->card->dev, + "%s: Buffer overrun on device at 0x%lx\n", + uart->rmidi->name, uart->base); return 0; } } @@ -775,15 +776,15 @@ static int snd_uart16550_create(struct snd_card *card, err = snd_uart16550_detect(uart); if (err <= 0) { - printk(KERN_ERR "no UART detected at 0x%lx\n", iobase); + dev_err(card->dev, "no UART detected at 0x%lx\n", iobase); return -ENODEV; } if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (devm_request_irq(card->dev, irq, snd_uart16550_interrupt, 0, "Serial MIDI", uart)) { - snd_printk(KERN_WARNING - "irq %d busy. Using Polling.\n", irq); + dev_warn(card->dev, + "irq %d busy. Using Polling.\n", irq); } else { uart->irq = irq; } @@ -879,23 +880,23 @@ static int snd_serial_probe(struct platform_device *devptr) case SNDRV_SERIAL_GENERIC: break; default: - snd_printk(KERN_ERR - "Adaptor type is out of range 0-%d (%d)\n", - SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]); + dev_err(&devptr->dev, + "Adaptor type is out of range 0-%d (%d)\n", + SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]); return -ENODEV; } if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) { - snd_printk(KERN_ERR - "Count of outputs is out of range 1-%d (%d)\n", - SNDRV_SERIAL_MAX_OUTS, outs[dev]); + dev_err(&devptr->dev, + "Count of outputs is out of range 1-%d (%d)\n", + SNDRV_SERIAL_MAX_OUTS, outs[dev]); return -ENODEV; } if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) { - snd_printk(KERN_ERR - "Count of inputs is out of range 1-%d (%d)\n", - SNDRV_SERIAL_MAX_INS, ins[dev]); + dev_err(&devptr->dev, + "Count of inputs is out of range 1-%d (%d)\n", + SNDRV_SERIAL_MAX_INS, ins[dev]); return -ENODEV; } @@ -975,7 +976,7 @@ static int __init alsa_card_serial_init(void) } if (! cards) { #ifdef MODULE - printk(KERN_ERR "serial midi soundcard not found or device busy\n"); + pr_err("serial midi soundcard not found or device busy\n"); #endif snd_serial_unregister_all(); return -ENODEV; From b5557ef985301d7ca2ba08f8e9c22ba4f8c30d5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:33:59 +0200 Subject: [PATCH 114/410] ALSA: virmidi: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-10-tiwai@suse.de --- sound/drivers/virmidi.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 58012de90c38..5f7b65ad63e3 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -83,9 +83,9 @@ static int snd_virmidi_probe(struct platform_device *devptr) vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { - snd_printk(KERN_WARNING - "too much midi devices for virmidi %d: force to use %d\n", - dev, MAX_MIDI_DEVICES); + dev_warn(&devptr->dev, + "too much midi devices for virmidi %d: force to use %d\n", + dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { @@ -155,7 +155,7 @@ static int __init alsa_card_virmidi_init(void) } if (!cards) { #ifdef MODULE - printk(KERN_ERR "Card-VirMIDI soundcard not found or device busy\n"); + pr_err("Card-VirMIDI soundcard not found or device busy\n"); #endif snd_virmidi_unregister_all(); return -ENODEV; From b426b3ba9f6fe17990893c5324727dd217d420b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:00 +0200 Subject: [PATCH 115/410] ALSA: vx_core: Drop unused dev field The vx_core.dev field has never been set but referred incorrectly at firmware loading. Pass the proper device pointer from card->dev at request_firmware(), and drop the unused dev field from vx_core, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-11-tiwai@suse.de --- include/sound/vx_core.h | 1 - sound/drivers/vx/vx_hwdep.c | 2 +- 2 files changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h index 1ddd3036bdfc..ca87fa6a8135 100644 --- a/include/sound/vx_core.h +++ b/include/sound/vx_core.h @@ -155,7 +155,6 @@ struct vx_core { unsigned int chip_status; unsigned int pcm_running; - struct device *dev; struct snd_hwdep *hwdep; struct vx_rmh irq_rmh; /* RMH used in interrupts */ diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index efbb644edba1..74f7ab330c7f 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -58,7 +58,7 @@ int snd_vx_setup_firmware(struct vx_core *chip) if (! fw_files[chip->type][i]) continue; sprintf(path, "vx/%s", fw_files[chip->type][i]); - if (request_firmware(&fw, path, chip->dev)) { + if (request_firmware(&fw, path, chip->card->dev)) { snd_printk(KERN_ERR "vx: can't load firmware %s\n", path); return -ENOENT; } From 41abc8056dd3125d4a9b7ac0f4087dfb67580d95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:01 +0200 Subject: [PATCH 116/410] ALSA: vx_core: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The commented old debug prints are dropped, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-12-tiwai@suse.de --- sound/drivers/vx/vx_core.c | 64 ++++++++++++++++--------------------- sound/drivers/vx/vx_hwdep.c | 2 +- sound/drivers/vx/vx_pcm.c | 23 +++++++------ sound/drivers/vx/vx_uer.c | 3 +- 4 files changed, 44 insertions(+), 48 deletions(-) diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 18901e5bcfcf..e2def8719ed2 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -52,7 +52,9 @@ int snd_vx_check_reg_bit(struct vx_core *chip, int reg, int mask, int bit, int t return 0; //msleep(10); } while (time_after_eq(end_time, jiffies)); - snd_printd(KERN_DEBUG "vx_check_reg_bit: timeout, reg=%s, mask=0x%x, val=0x%x\n", reg_names[reg], mask, snd_vx_inb(chip, reg)); + dev_dbg(chip->card->dev, + "vx_check_reg_bit: timeout, reg=%s, mask=0x%x, val=0x%x\n", + reg_names[reg], mask, snd_vx_inb(chip, reg)); return -EIO; } @@ -129,13 +131,14 @@ static int vx_transfer_end(struct vx_core *chip, int cmd) if (err & ISR_ERR) { err = vx_wait_for_rx_full(chip); if (err < 0) { - snd_printd(KERN_DEBUG "transfer_end: error in rx_full\n"); + dev_dbg(chip->card->dev, + "transfer_end: error in rx_full\n"); return err; } err = vx_inb(chip, RXH) << 16; err |= vx_inb(chip, RXM) << 8; err |= vx_inb(chip, RXL); - snd_printd(KERN_DEBUG "transfer_end: error = 0x%x\n", err); + dev_dbg(chip->card->dev, "transfer_end: error = 0x%x\n", err); return -(VX_ERR_MASK | err); } return 0; @@ -239,20 +242,10 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) err = vx_reset_chk(chip); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: vx_reset_chk error\n"); + dev_dbg(chip->card->dev, "vx_send_msg: vx_reset_chk error\n"); return err; } -#if 0 - printk(KERN_DEBUG "rmh: cmd = 0x%06x, length = %d, stype = %d\n", - rmh->Cmd[0], rmh->LgCmd, rmh->DspStat); - if (rmh->LgCmd > 1) { - printk(KERN_DEBUG " "); - for (i = 1; i < rmh->LgCmd; i++) - printk(KERN_CONT "0x%06x ", rmh->Cmd[i]); - printk(KERN_CONT "\n"); - } -#endif /* Check bit M is set according to length of the command */ if (rmh->LgCmd > 1) rmh->Cmd[0] |= MASK_MORE_THAN_1_WORD_COMMAND; @@ -262,7 +255,7 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) /* Wait for TX empty */ err = vx_wait_isr_bit(chip, ISR_TX_EMPTY); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: wait tx empty error\n"); + dev_dbg(chip->card->dev, "vx_send_msg: wait tx empty error\n"); return err; } @@ -274,7 +267,8 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) /* Trigger irq MESSAGE */ err = vx_send_irq_dsp(chip, IRQ_MESSAGE); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: send IRQ_MESSAGE error\n"); + dev_dbg(chip->card->dev, + "vx_send_msg: send IRQ_MESSAGE error\n"); return err; } @@ -287,13 +281,15 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) if (vx_inb(chip, ISR) & ISR_ERR) { err = vx_wait_for_rx_full(chip); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: rx_full read error\n"); + dev_dbg(chip->card->dev, + "vx_send_msg: rx_full read error\n"); return err; } err = vx_inb(chip, RXH) << 16; err |= vx_inb(chip, RXM) << 8; err |= vx_inb(chip, RXL); - snd_printd(KERN_DEBUG "msg got error = 0x%x at cmd[0]\n", err); + dev_dbg(chip->card->dev, + "msg got error = 0x%x at cmd[0]\n", err); err = -(VX_ERR_MASK | err); return err; } @@ -304,7 +300,8 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) /* Wait for TX ready */ err = vx_wait_isr_bit(chip, ISR_TX_READY); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: tx_ready error\n"); + dev_dbg(chip->card->dev, + "vx_send_msg: tx_ready error\n"); return err; } @@ -316,14 +313,16 @@ int vx_send_msg_nolock(struct vx_core *chip, struct vx_rmh *rmh) /* Trigger irq MESS_READ_NEXT */ err = vx_send_irq_dsp(chip, IRQ_MESS_READ_NEXT); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: IRQ_READ_NEXT error\n"); + dev_dbg(chip->card->dev, + "vx_send_msg: IRQ_READ_NEXT error\n"); return err; } } /* Wait for TX empty */ err = vx_wait_isr_bit(chip, ISR_TX_READY); if (err < 0) { - snd_printd(KERN_DEBUG "vx_send_msg: TX_READY error\n"); + dev_dbg(chip->card->dev, + "vx_send_msg: TX_READY error\n"); return err; } /* End of transfer */ @@ -372,9 +371,6 @@ int vx_send_rih_nolock(struct vx_core *chip, int cmd) if (chip->chip_status & VX_STAT_IS_STALE) return -EBUSY; -#if 0 - printk(KERN_DEBUG "send_rih: cmd = 0x%x\n", cmd); -#endif err = vx_reset_chk(chip); if (err < 0) return err; @@ -453,7 +449,7 @@ int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot) if (no_fillup) break; if (vx_wait_isr_bit(chip, ISR_TX_EMPTY) < 0) { - snd_printk(KERN_ERR "dsp boot failed at %d\n", i); + dev_err(chip->card->dev, "dsp boot failed at %d\n", i); return -EIO; } vx_outb(chip, TXH, 0); @@ -462,7 +458,7 @@ int snd_vx_load_boot_image(struct vx_core *chip, const struct firmware *boot) } else { const unsigned char *image = boot->data + i; if (vx_wait_isr_bit(chip, ISR_TX_EMPTY) < 0) { - snd_printk(KERN_ERR "dsp boot failed at %d\n", i); + dev_err(chip->card->dev, "dsp boot failed at %d\n", i); return -EIO; } vx_outb(chip, TXH, image[0]); @@ -510,18 +506,12 @@ irqreturn_t snd_vx_threaded_irq_handler(int irq, void *dev) if (vx_test_irq_src(chip, &events) < 0) return IRQ_HANDLED; -#if 0 - if (events & 0x000800) - printk(KERN_ERR "DSP Stream underrun ! IRQ events = 0x%x\n", events); -#endif - // printk(KERN_DEBUG "IRQ events = 0x%x\n", events); - /* We must prevent any application using this DSP * and block any further request until the application * either unregisters or reloads the DSP */ if (events & FATAL_DSP_ERROR) { - snd_printk(KERN_ERR "vx_core: fatal DSP error!!\n"); + dev_err(chip->card->dev, "vx_core: fatal DSP error!!\n"); return IRQ_HANDLED; } @@ -698,8 +688,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) /* Wait DSP ready for a new read */ err = vx_wait_isr_bit(chip, ISR_TX_EMPTY); if (err < 0) { - printk(KERN_ERR - "dsp loading error at position %d\n", i); + dev_err(chip->card->dev, + "dsp loading error at position %d\n", i); return err; } cptr = image; @@ -713,7 +703,6 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) csum = (csum >> 24) | (csum << 8); vx_outb(chip, TXL, *cptr++); } - snd_printdd(KERN_DEBUG "checksum = 0x%08x\n", csum); msleep(200); @@ -759,7 +748,8 @@ int snd_vx_resume(struct vx_core *chip) continue; err = chip->ops->load_dsp(chip, i, chip->firmware[i]); if (err < 0) { - snd_printk(KERN_ERR "vx: firmware resume error at DSP %d\n", i); + dev_err(chip->card->dev, + "vx: firmware resume error at DSP %d\n", i); return -EIO; } } diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 74f7ab330c7f..a7f8ddf4df5a 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -59,7 +59,7 @@ int snd_vx_setup_firmware(struct vx_core *chip) continue; sprintf(path, "vx/%s", fw_files[chip->type][i]); if (request_firmware(&fw, path, chip->card->dev)) { - snd_printk(KERN_ERR "vx: can't load firmware %s\n", path); + dev_err(chip->card->dev, "vx: can't load firmware %s\n", path); return -ENOENT; } err = chip->ops->load_dsp(chip, i, fw); diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index ceaeb257003b..cbc77ca1ebdd 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -190,8 +190,10 @@ static int vx_set_ibl(struct vx_core *chip, struct vx_ibl_info *info) info->max_size = rmh.Stat[1]; info->min_size = rmh.Stat[2]; info->granularity = rmh.Stat[3]; - snd_printdd(KERN_DEBUG "vx_set_ibl: size = %d, max = %d, min = %d, gran = %d\n", - info->size, info->max_size, info->min_size, info->granularity); + dev_dbg(chip->card->dev, + "%s: size = %d, max = %d, min = %d, gran = %d\n", + __func__, info->size, info->max_size, info->min_size, + info->granularity); return 0; } @@ -616,12 +618,12 @@ static int vx_pcm_playback_transfer_chunk(struct vx_core *chip, if (space < 0) { /* disconnect the host, SIZE_HBUF command always switches to the stream mode */ vx_send_rih(chip, IRQ_CONNECT_STREAM_NEXT); - snd_printd("error hbuffer\n"); + dev_dbg(chip->card->dev, "error hbuffer\n"); return space; } if (space < size) { vx_send_rih(chip, IRQ_CONNECT_STREAM_NEXT); - snd_printd("no enough hbuffer space %d\n", space); + dev_dbg(chip->card->dev, "no enough hbuffer space %d\n", space); return -EIO; /* XRUN */ } @@ -795,7 +797,8 @@ static int vx_pcm_prepare(struct snd_pcm_substream *subs) /* IEC958 status (raw-mode) was changed */ /* we reopen the pipe */ struct vx_rmh rmh; - snd_printdd(KERN_DEBUG "reopen the pipe with data_mode = %d\n", data_mode); + dev_dbg(chip->card->dev, + "reopen the pipe with data_mode = %d\n", data_mode); vx_init_rmh(&rmh, CMD_FREE_PIPE); vx_set_pipe_cmd_params(&rmh, 0, pipe->number, 0); err = vx_send_msg(chip, &rmh); @@ -812,8 +815,9 @@ static int vx_pcm_prepare(struct snd_pcm_substream *subs) } if (chip->pcm_running && chip->freq != runtime->rate) { - snd_printk(KERN_ERR "vx: cannot set different clock %d " - "from the current %d\n", runtime->rate, chip->freq); + dev_err(chip->card->dev, + "vx: cannot set different clock %d from the current %d\n", + runtime->rate, chip->freq); return -EINVAL; } vx_set_clock(chip, runtime->rate); @@ -1091,7 +1095,7 @@ void vx_pcm_update_intr(struct vx_core *chip, unsigned int events) chip->irq_rmh.Cmd[0] |= 0x00000002; /* SEL_END_OF_BUF_EVENTS */ if (vx_send_msg(chip, &chip->irq_rmh) < 0) { - snd_printdd(KERN_ERR "msg send error!!\n"); + dev_dbg(chip->card->dev, "msg send error!!\n"); return; } @@ -1141,7 +1145,8 @@ static int vx_init_audio_io(struct vx_core *chip) vx_init_rmh(&rmh, CMD_SUPPORTED); if (vx_send_msg(chip, &rmh) < 0) { - snd_printk(KERN_ERR "vx: cannot get the supported audio data\n"); + dev_err(chip->card->dev, + "vx: cannot get the supported audio data\n"); return -ENXIO; } diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 884c40be19dc..3eca22151225 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -196,7 +196,8 @@ void vx_set_internal_clock(struct vx_core *chip, unsigned int freq) /* Get real clock value */ clock = vx_calc_clock_from_freq(chip, freq); - snd_printdd(KERN_DEBUG "set internal clock to 0x%x from freq %d\n", clock, freq); + dev_dbg(chip->card->dev, + "set internal clock to 0x%x from freq %d\n", clock, freq); mutex_lock(&chip->lock); if (vx_is_pcmcia(chip)) { vx_outb(chip, HIFREQ, (clock >> 8) & 0x0f); From ca2f73ffaadacb14a7232aee5c2b5854483f5c40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:02 +0200 Subject: [PATCH 117/410] ALSA: aloop: Use standard print API Use pr_err() instead of open-coded printk() just for code simplification. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-13-tiwai@suse.de --- sound/drivers/aloop.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 439d12ad8787..3650d47b7d58 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1896,7 +1896,7 @@ static int __init alsa_card_loopback_init(void) } if (!cards) { #ifdef MODULE - printk(KERN_ERR "aloop: No loopback enabled\n"); + pr_err("aloop: No loopback enabled\n"); #endif loopback_unregister_all(); return -ENODEV; From 650dcf25e181ff1dc4f9f107f1fa71e802c14d1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:03 +0200 Subject: [PATCH 118/410] ALSA: dummy: Use standard print API Use pr_*() macro instead of open-coded printk() just for code simplification. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-14-tiwai@suse.de --- sound/drivers/dummy.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 52ff6ac3f743..8f5df9b3aaaa 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1033,8 +1033,7 @@ static int snd_dummy_probe(struct platform_device *devptr) dummy->card = card; for (mdl = dummy_models; *mdl && model[dev]; mdl++) { if (strcmp(model[dev], (*mdl)->name) == 0) { - printk(KERN_INFO - "snd-dummy: Using model '%s' for card %i\n", + pr_info("snd-dummy: Using model '%s' for card %i\n", (*mdl)->name, card->number); m = dummy->model = *mdl; break; @@ -1168,7 +1167,7 @@ static int __init alsa_card_dummy_init(void) } if (!cards) { #ifdef MODULE - printk(KERN_ERR "Dummy soundcard not found or device busy\n"); + pr_err("Dummy soundcard not found or device busy\n"); #endif snd_dummy_unregister_all(); return -ENODEV; From 9cbe416b9356e84863dff02112d3d3c095f881f8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:04 +0200 Subject: [PATCH 119/410] ALSA: pcsp: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-15-tiwai@suse.de --- sound/drivers/pcsp/pcsp.c | 21 +++++++++--------- sound/drivers/pcsp/pcsp_lib.c | 38 ++++++++++++++++----------------- sound/drivers/pcsp/pcsp_mixer.c | 2 +- 3 files changed, 31 insertions(+), 30 deletions(-) diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index 7195cb49e00f..78c9b1c7590f 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -47,11 +47,12 @@ static int snd_pcsp_create(struct snd_card *card) if (!nopcm) { if (resolution > PCSP_MAX_PERIOD_NS) { - printk(KERN_ERR "PCSP: Timer resolution is not sufficient " - "(%unS)\n", resolution); - printk(KERN_ERR "PCSP: Make sure you have HPET and ACPI " - "enabled.\n"); - printk(KERN_ERR "PCSP: Turned into nopcm mode.\n"); + dev_err(card->dev, + "PCSP: Timer resolution is not sufficient (%unS)\n", + resolution); + dev_err(card->dev, + "PCSP: Make sure you have HPET and ACPI enabled.\n"); + dev_err(card->dev, "PCSP: Turned into nopcm mode.\n"); nopcm = 1; } } @@ -61,8 +62,8 @@ static int snd_pcsp_create(struct snd_card *card) else min_div = MAX_DIV; #if PCSP_DEBUG - printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%u\n", - loops_per_jiffy, min_div, resolution); + dev_dbg(card->dev, "PCSP: lpj=%li, min_div=%i, res=%u\n", + loops_per_jiffy, min_div, resolution); #endif div = MAX_DIV / min_div; @@ -141,14 +142,14 @@ static int alsa_card_pcsp_init(struct device *dev) err = snd_card_pcsp_probe(0, dev); if (err) { - printk(KERN_ERR "PC-Speaker initialization failed.\n"); + dev_err(dev, "PC-Speaker initialization failed.\n"); return err; } /* Well, CONFIG_DEBUG_PAGEALLOC makes the sound horrible. Lets alert */ if (debug_pagealloc_enabled()) { - printk(KERN_WARNING "PCSP: CONFIG_DEBUG_PAGEALLOC is enabled, " - "which may make the sound noisy.\n"); + dev_warn(dev, + "PCSP: CONFIG_DEBUG_PAGEALLOC is enabled, which may make the sound noisy.\n"); } return 0; diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 773db4bf0876..d9bc1ea1b53c 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include "pcsp.h" @@ -105,8 +106,8 @@ static void pcsp_pointer_update(struct snd_pcsp *chip) periods_elapsed = chip->playback_ptr - chip->period_ptr; if (periods_elapsed < 0) { #if PCSP_DEBUG - printk(KERN_INFO "PCSP: buffer_bytes mod period_bytes != 0 ? " - "(%zi %zi %zi)\n", + dev_dbg(chip->card->dev, + "PCSP: buffer_bytes mod period_bytes != 0 ? (%zi %zi %zi)\n", chip->playback_ptr, period_bytes, buffer_bytes); #endif periods_elapsed += buffer_bytes; @@ -136,7 +137,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) pointer_update = !chip->thalf; ns = pcsp_timer_update(chip); if (!ns) { - printk(KERN_WARNING "PCSP: unexpected stop\n"); + dev_warn(chip->card->dev, "PCSP: unexpected stop\n"); return HRTIMER_NORESTART; } @@ -151,10 +152,10 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) static int pcsp_start_playing(struct snd_pcsp *chip) { #if PCSP_DEBUG - printk(KERN_INFO "PCSP: start_playing called\n"); + dev_dbg(chip->card->dev, "PCSP: start_playing called\n"); #endif if (atomic_read(&chip->timer_active)) { - printk(KERN_ERR "PCSP: Timer already active\n"); + dev_err(chip->card->dev, "PCSP: Timer already active\n"); return -EIO; } @@ -172,7 +173,7 @@ static int pcsp_start_playing(struct snd_pcsp *chip) static void pcsp_stop_playing(struct snd_pcsp *chip) { #if PCSP_DEBUG - printk(KERN_INFO "PCSP: stop_playing called\n"); + dev_dbg(chip->card->dev, "PCSP: stop_playing called\n"); #endif if (!atomic_read(&chip->timer_active)) return; @@ -201,7 +202,7 @@ static int snd_pcsp_playback_close(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG - printk(KERN_INFO "PCSP: close called\n"); + dev_dbg(chip->card->dev, "PCSP: close called\n"); #endif pcsp_sync_stop(chip); chip->playback_substream = NULL; @@ -220,7 +221,7 @@ static int snd_pcsp_playback_hw_free(struct snd_pcm_substream *substream) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG - printk(KERN_INFO "PCSP: hw_free called\n"); + dev_dbg(chip->card->dev, "PCSP: hw_free called\n"); #endif pcsp_sync_stop(chip); return 0; @@ -236,14 +237,13 @@ static int snd_pcsp_playback_prepare(struct snd_pcm_substream *substream) snd_pcm_format_physical_width(substream->runtime->format) >> 3; chip->is_signed = snd_pcm_format_signed(substream->runtime->format); #if PCSP_DEBUG - printk(KERN_INFO "PCSP: prepare called, " - "size=%zi psize=%zi f=%zi f1=%i fsize=%i\n", - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - snd_pcm_lib_buffer_bytes(substream) / - snd_pcm_lib_period_bytes(substream), - substream->runtime->periods, - chip->fmt_size); + dev_dbg(chip->card->dev, "PCSP: prepare called, size=%zi psize=%zi f=%zi f1=%i fsize=%i\n", + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream), + snd_pcm_lib_buffer_bytes(substream) / + snd_pcm_lib_period_bytes(substream), + substream->runtime->periods, + chip->fmt_size); #endif return 0; } @@ -252,7 +252,7 @@ static int snd_pcsp_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcsp *chip = snd_pcm_substream_chip(substream); #if PCSP_DEBUG - printk(KERN_INFO "PCSP: trigger called\n"); + dev_dbg(chip->card->dev, "PCSP: trigger called\n"); #endif switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -306,10 +306,10 @@ static int snd_pcsp_playback_open(struct snd_pcm_substream *substream) struct snd_pcsp *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; #if PCSP_DEBUG - printk(KERN_INFO "PCSP: open called\n"); + dev_dbg(chip->card->dev, "PCSP: open called\n"); #endif if (atomic_read(&chip->timer_active)) { - printk(KERN_ERR "PCSP: still active!!\n"); + dev_err(chip->card->dev, "PCSP: still active!!\n"); return -EBUSY; } runtime->hw = snd_pcsp_playback; diff --git a/sound/drivers/pcsp/pcsp_mixer.c b/sound/drivers/pcsp/pcsp_mixer.c index da33e5b620a7..c0ae942358b9 100644 --- a/sound/drivers/pcsp/pcsp_mixer.c +++ b/sound/drivers/pcsp/pcsp_mixer.c @@ -73,7 +73,7 @@ static int pcsp_treble_put(struct snd_kcontrol *kcontrol, if (treble != chip->treble) { chip->treble = treble; #if PCSP_DEBUG - printk(KERN_INFO "PCSP: rate set to %li\n", PCSP_RATE()); + dev_dbg(chip->card->dev, "PCSP: rate set to %li\n", PCSP_RATE()); #endif changed = 1; } From e71391ba9434823621886538c9db9e4c3946f1ba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:05 +0200 Subject: [PATCH 120/410] ALSA: i2c: cs8427: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-16-tiwai@suse.de --- sound/i2c/cs8427.c | 37 ++++++++++++++++++++----------------- 1 file changed, 20 insertions(+), 17 deletions(-) diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index f58b14b49045..29a1a7a0d050 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -52,8 +52,9 @@ int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, buf[1] = val; err = snd_i2c_sendbytes(device, buf, 2); if (err != 2) { - snd_printk(KERN_ERR "unable to send bytes 0x%02x:0x%02x " - "to CS8427 (%i)\n", buf[0], buf[1], err); + dev_err(device->bus->card->dev, + "unable to send bytes 0x%02x:0x%02x to CS8427 (%i)\n", + buf[0], buf[1], err); return err < 0 ? err : -EIO; } return 0; @@ -68,14 +69,14 @@ static int snd_cs8427_reg_read(struct snd_i2c_device *device, unsigned char reg) err = snd_i2c_sendbytes(device, ®, 1); if (err != 1) { - snd_printk(KERN_ERR "unable to send register 0x%x byte " - "to CS8427\n", reg); + dev_err(device->bus->card->dev, + "unable to send register 0x%x byte to CS8427\n", reg); return err < 0 ? err : -EIO; } err = snd_i2c_readbytes(device, &buf, 1); if (err != 1) { - snd_printk(KERN_ERR "unable to read register 0x%x byte " - "from CS8427\n", reg); + dev_err(device->bus->card->dev, + "unable to read register 0x%x byte from CS8427\n", reg); return err < 0 ? err : -EIO; } return buf; @@ -195,16 +196,18 @@ int snd_cs8427_init(struct snd_i2c_bus *bus, err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); if (err != CS8427_VER8427A) { /* give second chance */ - snd_printk(KERN_WARNING "invalid CS8427 signature 0x%x: " - "let me try again...\n", err); + dev_warn(device->bus->card->dev, + "invalid CS8427 signature 0x%x: let me try again...\n", + err); err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); } if (err != CS8427_VER8427A) { snd_i2c_unlock(bus); - snd_printk(KERN_ERR "unable to find CS8427 signature " - "(expected 0x%x, read 0x%x),\n", - CS8427_VER8427A, err); - snd_printk(KERN_ERR " initialization is not completed\n"); + dev_err(device->bus->card->dev, + "unable to find CS8427 signature (expected 0x%x, read 0x%x),\n", + CS8427_VER8427A, err); + dev_err(device->bus->card->dev, + " initialization is not completed\n"); return -EFAULT; } /* turn off run bit while making changes to configuration */ @@ -289,7 +292,7 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, snd_i2c_sendbytes(device, buf, 1); snd_i2c_readbytes(device, buf, 127); for (xx = 0; xx < 127; xx++) - printk(KERN_DEBUG "reg[0x%x] = 0x%x\n", xx+1, buf[xx]); + dev_dbg(device->bus->card->dev, "reg[0x%x] = 0x%x\n", xx+1, buf[xx]); } #endif @@ -392,15 +395,15 @@ static int snd_cs8427_qsubcode_get(struct snd_kcontrol *kcontrol, snd_i2c_lock(device->bus); err = snd_i2c_sendbytes(device, ®, 1); if (err != 1) { - snd_printk(KERN_ERR "unable to send register 0x%x byte " - "to CS8427\n", reg); + dev_err(device->bus->card->dev, + "unable to send register 0x%x byte to CS8427\n", reg); snd_i2c_unlock(device->bus); return err < 0 ? err : -EIO; } err = snd_i2c_readbytes(device, ucontrol->value.bytes.data, 10); if (err != 10) { - snd_printk(KERN_ERR "unable to read Q-subcode bytes " - "from CS8427\n"); + dev_err(device->bus->card->dev, + "unable to read Q-subcode bytes from CS8427\n"); snd_i2c_unlock(device->bus); return err < 0 ? err : -EIO; } From 1ac6352e50780603bbae44eda7b3cc0036b1013f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:06 +0200 Subject: [PATCH 121/410] ALSA: i2c: pt2258: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-17-tiwai@suse.de --- sound/i2c/other/pt2258.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/i2c/other/pt2258.c b/sound/i2c/other/pt2258.c index c913f223892a..ba38c285241c 100644 --- a/sound/i2c/other/pt2258.c +++ b/sound/i2c/other/pt2258.c @@ -63,7 +63,7 @@ int snd_pt2258_reset(struct snd_pt2258 *pt) __error: snd_i2c_unlock(pt->i2c_bus); - snd_printk(KERN_ERR "PT2258 reset failed\n"); + dev_err(pt->card->dev, "PT2258 reset failed\n"); return -EIO; } @@ -124,7 +124,7 @@ static int pt2258_stereo_volume_put(struct snd_kcontrol *kcontrol, __error: snd_i2c_unlock(pt->i2c_bus); - snd_printk(KERN_ERR "PT2258 access failed\n"); + dev_err(pt->card->dev, "PT2258 access failed\n"); return -EIO; } @@ -161,7 +161,7 @@ static int pt2258_switch_put(struct snd_kcontrol *kcontrol, __error: snd_i2c_unlock(pt->i2c_bus); - snd_printk(KERN_ERR "PT2258 access failed 2\n"); + dev_err(pt->card->dev, "PT2258 access failed 2\n"); return -EIO; } From ae1873eeb8bac3f5c188637f24d8e8714e994929 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:07 +0200 Subject: [PATCH 122/410] ALSA: i2c: Drop commented old debug prints There are quite a few commented-out debug prints that have never been used in the production code. Let's rip them off for code cleanness. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-18-tiwai@suse.de --- sound/i2c/other/ak4113.c | 2 -- sound/i2c/other/ak4114.c | 12 ------------ sound/i2c/other/ak4117.c | 13 ------------- sound/i2c/other/ak4xxx-adda.c | 2 -- sound/i2c/tea6330t.c | 3 --- 5 files changed, 32 deletions(-) diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index e7213092eb4f..c1f7447a4d11 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -599,8 +599,6 @@ __rate: (runtime->rate != res)) { snd_pcm_stream_lock_irqsave(ak4113->substream, _flags); if (snd_pcm_running(ak4113->substream)) { - /*printk(KERN_DEBUG "rate changed (%i <- %i)\n", - * runtime->rate, res); */ snd_pcm_stop(ak4113->substream, SNDRV_PCM_STATE_DRAINING); wake_up(&runtime->sleep); diff --git a/sound/i2c/other/ak4114.c b/sound/i2c/other/ak4114.c index c0cffe28989b..7c493681f3cb 100644 --- a/sound/i2c/other/ak4114.c +++ b/sound/i2c/other/ak4114.c @@ -38,17 +38,6 @@ static inline unsigned char reg_read(struct ak4114 *ak4114, unsigned char reg) return ak4114->read(ak4114->private_data, reg); } -#if 0 -static void reg_dump(struct ak4114 *ak4114) -{ - int i; - - printk(KERN_DEBUG "AK4114 REG DUMP:\n"); - for (i = 0; i < 0x20; i++) - printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4114, i), i < ARRAY_SIZE(ak4114->regmap) ? ak4114->regmap[i] : 0); -} -#endif - static void snd_ak4114_free(struct ak4114 *chip) { atomic_inc(&chip->wq_processing); /* don't schedule new work */ @@ -589,7 +578,6 @@ int snd_ak4114_check_rate_and_errors(struct ak4114 *ak4114, unsigned int flags) if (!(flags & AK4114_CHECK_NO_RATE) && runtime && runtime->rate != res) { snd_pcm_stream_lock_irqsave(ak4114->capture_substream, _flags); if (snd_pcm_running(ak4114->capture_substream)) { - // printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res); snd_pcm_stop(ak4114->capture_substream, SNDRV_PCM_STATE_DRAINING); res = 1; } diff --git a/sound/i2c/other/ak4117.c b/sound/i2c/other/ak4117.c index 640501bb3ca6..165fdda8fda5 100644 --- a/sound/i2c/other/ak4117.c +++ b/sound/i2c/other/ak4117.c @@ -34,17 +34,6 @@ static inline unsigned char reg_read(struct ak4117 *ak4117, unsigned char reg) return ak4117->read(ak4117->private_data, reg); } -#if 0 -static void reg_dump(struct ak4117 *ak4117) -{ - int i; - - printk(KERN_DEBUG "AK4117 REG DUMP:\n"); - for (i = 0; i < 0x1b; i++) - printk(KERN_DEBUG "reg[%02x] = %02x (%02x)\n", i, reg_read(ak4117, i), i < sizeof(ak4117->regmap) ? ak4117->regmap[i] : 0); -} -#endif - static void snd_ak4117_free(struct ak4117 *chip) { timer_shutdown_sync(&chip->timer); @@ -452,7 +441,6 @@ int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags) goto __rate; rcs0 = reg_read(ak4117, AK4117_REG_RCS0); rcs2 = reg_read(ak4117, AK4117_REG_RCS2); - // printk(KERN_DEBUG "AK IRQ: rcs0 = 0x%x, rcs1 = 0x%x, rcs2 = 0x%x\n", rcs0, rcs1, rcs2); spin_lock_irqsave(&ak4117->lock, _flags); if (rcs0 & AK4117_PAR) ak4117->errors[AK4117_PARITY_ERRORS]++; @@ -505,7 +493,6 @@ int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags) if (!(flags & AK4117_CHECK_NO_RATE) && runtime && runtime->rate != res) { snd_pcm_stream_lock_irqsave(ak4117->substream, _flags); if (snd_pcm_running(ak4117->substream)) { - // printk(KERN_DEBUG "rate changed (%i <- %i)\n", runtime->rate, res); snd_pcm_stop(ak4117->substream, SNDRV_PCM_STATE_DRAINING); wake_up(&runtime->sleep); res = 1; diff --git a/sound/i2c/other/ak4xxx-adda.c b/sound/i2c/other/ak4xxx-adda.c index 7d15093844b9..b24c80410d45 100644 --- a/sound/i2c/other/ak4xxx-adda.c +++ b/sound/i2c/other/ak4xxx-adda.c @@ -391,8 +391,6 @@ static int put_ak_reg(struct snd_kcontrol *kcontrol, int addr, nval = mask - nval; if (AK_GET_NEEDSMSB(kcontrol->private_value)) nval |= 0x80; - /* printk(KERN_DEBUG "DEBUG - AK writing reg: chip %x addr %x, - nval %x\n", chip, addr, nval); */ snd_akm4xxx_write(ak, chip, addr, nval); return 1; } diff --git a/sound/i2c/tea6330t.c b/sound/i2c/tea6330t.c index 037d6293f728..676d58054944 100644 --- a/sound/i2c/tea6330t.c +++ b/sound/i2c/tea6330t.c @@ -56,9 +56,6 @@ int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer) static void snd_tea6330t_set(struct tea6330t *tea, unsigned char addr, unsigned char value) { -#if 0 - printk(KERN_DEBUG "set - 0x%x/0x%x\n", addr, value); -#endif snd_i2c_write(tea->bus, TEA6330T_ADDR, addr, value, 1); } #endif From 20869176d7a7509bad9ea6b895469aebea9c8f21 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:08 +0200 Subject: [PATCH 123/410] ALSA: ad1816a: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-19-tiwai@suse.de --- sound/isa/ad1816a/ad1816a.c | 16 ++++++++-------- sound/isa/ad1816a/ad1816a_lib.c | 16 ++++++++-------- 2 files changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 9ac873773129..99006dc4777e 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -17,8 +17,6 @@ #include #include -#define PFX "ad1816a: " - MODULE_AUTHOR("Massimo Piccioni "); MODULE_DESCRIPTION("AD1816A, AD1815"); MODULE_LICENSE("GPL"); @@ -87,7 +85,7 @@ static int snd_card_ad1816a_pnp(int dev, struct pnp_card_link *card, err = pnp_activate_dev(pdev); if (err < 0) { - printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); + dev_err(&pdev->dev, "AUDIO PnP configure failure\n"); return -EBUSY; } @@ -100,13 +98,13 @@ static int snd_card_ad1816a_pnp(int dev, struct pnp_card_link *card, pdev = pnp_request_card_device(card, id->devs[1].id, NULL); if (pdev == NULL) { mpu_port[dev] = -1; - snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); + dev_warn(&pdev->dev, "MPU401 device busy, skipping.\n"); return 0; } err = pnp_activate_dev(pdev); if (err < 0) { - printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); + dev_err(&pdev->dev, "MPU401 PnP configure failure\n"); mpu_port[dev] = -1; } else { mpu_port[dev] = pnp_port_start(pdev, 0); @@ -166,14 +164,16 @@ static int snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard, if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]); + dev_err(card->dev, "no MPU-401 device at 0x%lx.\n", + mpu_port[dev]); } if (fm_port[dev] > 0) { if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2); + dev_err(card->dev, "no OPL device at 0x%lx-0x%lx.\n", + fm_port[dev], fm_port[dev] + 2); } else { error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (error < 0) @@ -252,7 +252,7 @@ static int __init alsa_card_ad1816a_init(void) if (!ad1816a_devices) { pnp_unregister_card_driver(&ad1816a_pnpc_driver); #ifdef MODULE - printk(KERN_ERR "no AD1816A based soundcards found.\n"); + pr_err("no AD1816A based soundcards found.\n"); #endif /* MODULE */ return -ENODEV; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 132a095dca2c..2b87036cb94f 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -25,7 +25,7 @@ static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip) if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY) return 0; - snd_printk(KERN_WARNING "chip busy.\n"); + dev_warn(chip->card->dev, "chip busy.\n"); return -EBUSY; } @@ -186,7 +186,7 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, spin_unlock(&chip->lock); break; default: - snd_printk(KERN_WARNING "invalid trigger mode 0x%x.\n", what); + dev_warn(chip->card->dev, "invalid trigger mode 0x%x.\n", what); error = -EINVAL; } @@ -548,8 +548,8 @@ static const char *snd_ad1816a_chip_id(struct snd_ad1816a *chip) case AD1816A_HW_AD1815: return "AD1815"; case AD1816A_HW_AD18MAX10: return "AD18max10"; default: - snd_printk(KERN_WARNING "Unknown chip version %d:%d.\n", - chip->version, chip->hardware); + dev_warn(chip->card->dev, "Unknown chip version %d:%d.\n", + chip->version, chip->hardware); return "AD1816A - unknown"; } } @@ -566,23 +566,23 @@ int snd_ad1816a_create(struct snd_card *card, chip->res_port = devm_request_region(card->dev, port, 16, "AD1816A"); if (!chip->res_port) { - snd_printk(KERN_ERR "ad1816a: can't grab port 0x%lx\n", port); + dev_err(card->dev, "ad1816a: can't grab port 0x%lx\n", port); return -EBUSY; } if (devm_request_irq(card->dev, irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) { - snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq); + dev_err(card->dev, "ad1816a: can't grab IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; card->sync_irq = chip->irq; if (snd_devm_request_dma(card->dev, dma1, "AD1816A - 1")) { - snd_printk(KERN_ERR "ad1816a: can't grab DMA1 %d\n", dma1); + dev_err(card->dev, "ad1816a: can't grab DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (snd_devm_request_dma(card->dev, dma2, "AD1816A - 2")) { - snd_printk(KERN_ERR "ad1816a: can't grab DMA2 %d\n", dma2); + dev_err(card->dev, "ad1816a: can't grab DMA2 %d\n", dma2); return -EBUSY; } chip->dma2 = dma2; From 2508acd40301685f362eb6fdaf79da191ba8ec4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:09 +0200 Subject: [PATCH 124/410] ALSA: als100: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Use the standard print API instead of open-coded printk(). Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-20-tiwai@suse.de --- sound/isa/als100.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/isa/als100.c b/sound/isa/als100.c index d582eff64082..e70dbf0b099a 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -23,8 +23,6 @@ #include #include -#define PFX "als100: " - MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0"); MODULE_AUTHOR("Massimo Piccioni "); MODULE_LICENSE("GPL"); @@ -112,7 +110,7 @@ static int snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); + dev_err(&pdev->dev, "AUDIO pnp configure failure\n"); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -135,7 +133,7 @@ static int snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, __mpu_error: if (pdev) { pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "MPU401 pnp configure failure, skipping\n"); + dev_err(&pdev->dev, "MPU401 pnp configure failure, skipping\n"); } acard->devmpu = NULL; mpu_port[dev] = -1; @@ -151,7 +149,7 @@ static int snd_card_als100_pnp(int dev, struct snd_card_als100 *acard, __fm_error: if (pdev) { pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "OPL3 pnp configure failure, skipping\n"); + dev_err(&pdev->dev, "OPL3 pnp configure failure, skipping\n"); } acard->devopl = NULL; fm_port[dev] = -1; @@ -230,15 +228,15 @@ static int snd_card_als100_probe(int dev, mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); + dev_err(card->dev, "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n", - fm_port[dev], fm_port[dev] + 2); + dev_err(card->dev, "no OPL device at 0x%lx-0x%lx\n", + fm_port[dev], fm_port[dev] + 2); } else { error = snd_opl3_timer_new(opl3, 0, 1); if (error < 0) @@ -324,7 +322,7 @@ static int __init alsa_card_als100_init(void) if (!als100_devices) { pnp_unregister_card_driver(&als100_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no Avance Logic based soundcards found\n"); + pr_err("no Avance Logic based soundcards found\n"); #endif return -ENODEV; } From 80134f1bc7b34360ffb42629d4ad6e06b667bb00 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:10 +0200 Subject: [PATCH 125/410] ALSA: azt2320: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-21-tiwai@suse.de --- sound/isa/azt2320.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index 761cd198df2b..b937c9138d12 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -30,8 +30,6 @@ #include #include -#define PFX "azt2320: " - MODULE_AUTHOR("Massimo Piccioni "); MODULE_DESCRIPTION("Aztech Systems AZT2320"); MODULE_LICENSE("GPL"); @@ -99,7 +97,7 @@ static int snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); + dev_err(&pdev->dev, "AUDIO pnp configure failure\n"); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -120,7 +118,7 @@ static int snd_card_azt2320_pnp(int dev, struct snd_card_azt2320 *acard, __mpu_error: if (pdev) { pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "MPU401 pnp configure failure, skipping\n"); + dev_err(&pdev->dev, "MPU401 pnp configure failure, skipping\n"); } acard->devmpu = NULL; mpu_port[dev] = -1; @@ -210,15 +208,15 @@ static int snd_card_azt2320_probe(int dev, if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320, mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); + dev_err(card->dev, "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n", - fm_port[dev], fm_port[dev] + 2); + dev_err(card->dev, "no OPL device at 0x%lx-0x%lx\n", + fm_port[dev], fm_port[dev] + 2); } else { error = snd_opl3_timer_new(opl3, 1, 2); if (error < 0) @@ -303,7 +301,7 @@ static int __init alsa_card_azt2320_init(void) if (!azt2320_devices) { pnp_unregister_card_driver(&azt2320_pnpc_driver); #ifdef MODULE - snd_printk(KERN_ERR "no AZT2320 based soundcards found\n"); + pr_err("no AZT2320 based soundcards found\n"); #endif return -ENODEV; } From 09d1e9b4c18e46feb85f1e8738babf31b00632bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:11 +0200 Subject: [PATCH 126/410] ALSA: cmi8328: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-22-tiwai@suse.de --- sound/isa/cmi8328.c | 42 +++++++++++++++++++++--------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c index 8902cfb830f7..d30cce4cc7e3 100644 --- a/sound/isa/cmi8328.c +++ b/sound/isa/cmi8328.c @@ -159,7 +159,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip) strcpy(id2.name, "CD Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "error renaming control\n"); + dev_err(card->dev, "error renaming control\n"); return err; } /* rename AUX0 volume to CD */ @@ -167,7 +167,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip) strcpy(id2.name, "CD Playback Volume"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "error renaming control\n"); + dev_err(card->dev, "error renaming control\n"); return err; } /* rename AUX1 switch to Synth */ @@ -176,7 +176,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip) strcpy(id2.name, "Synth Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "error renaming control\n"); + dev_err(card->dev, "error renaming control\n"); return err; } /* rename AUX1 volume to Synth */ @@ -185,7 +185,7 @@ static int snd_cmi8328_mixer(struct snd_wss *chip) strcpy(id2.name, "Synth Playback Volume"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "error renaming control\n"); + dev_err(card->dev, "error renaming control\n"); return err; } @@ -251,35 +251,35 @@ static int snd_cmi8328_probe(struct device *pdev, unsigned int ndev) if (irq[ndev] == SNDRV_AUTO_IRQ) { irq[ndev] = snd_legacy_find_free_irq(irqs); if (irq[ndev] < 0) { - snd_printk(KERN_ERR "unable to find a free IRQ\n"); + dev_err(pdev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[ndev] == SNDRV_AUTO_DMA) { dma1[ndev] = snd_legacy_find_free_dma(dma1s); if (dma1[ndev] < 0) { - snd_printk(KERN_ERR "unable to find a free DMA1\n"); + dev_err(pdev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[ndev] == SNDRV_AUTO_DMA) { dma2[ndev] = snd_legacy_find_free_dma(dma2s[dma1[ndev] % 4]); if (dma2[ndev] < 0) { - snd_printk(KERN_WARNING "unable to find a free DMA2, full-duplex will not work\n"); + dev_warn(pdev, "unable to find a free DMA2, full-duplex will not work\n"); dma2[ndev] = -1; } } /* configure WSS IRQ... */ pos = array_find(irqs, irq[ndev]); if (pos < 0) { - snd_printk(KERN_ERR "invalid IRQ %d\n", irq[ndev]); + dev_err(pdev, "invalid IRQ %d\n", irq[ndev]); return -EINVAL; } val = irq_bits[pos] << 3; /* ...and DMA... */ pos = array_find(dma1s, dma1[ndev]); if (pos < 0) { - snd_printk(KERN_ERR "invalid DMA1 %d\n", dma1[ndev]); + dev_err(pdev, "invalid DMA1 %d\n", dma1[ndev]); return -EINVAL; } val |= dma_bits[pos]; @@ -287,7 +287,7 @@ static int snd_cmi8328_probe(struct device *pdev, unsigned int ndev) if (dma2[ndev] >= 0 && dma1[ndev] != dma2[ndev]) { pos = array_find(dma2s[dma1[ndev]], dma2[ndev]); if (pos < 0) { - snd_printk(KERN_ERR "invalid DMA2 %d\n", dma2[ndev]); + dev_err(pdev, "invalid DMA2 %d\n", dma2[ndev]); return -EINVAL; } val |= 0x04; /* enable separate capture DMA */ @@ -320,47 +320,47 @@ static int snd_cmi8328_probe(struct device *pdev, unsigned int ndev) return err; if (snd_wss_timer(cmi->wss, 0) < 0) - snd_printk(KERN_WARNING "error initializing WSS timer\n"); + dev_warn(pdev, "error initializing WSS timer\n"); if (mpuport[ndev] == SNDRV_AUTO_PORT) { mpuport[ndev] = snd_legacy_find_free_ioport(mpu_ports, 2); if (mpuport[ndev] < 0) - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + dev_err(pdev, "unable to find a free MPU401 port\n"); } if (mpuirq[ndev] == SNDRV_AUTO_IRQ) { mpuirq[ndev] = snd_legacy_find_free_irq(mpu_irqs); if (mpuirq[ndev] < 0) - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + dev_err(pdev, "unable to find a free MPU401 IRQ\n"); } /* enable and configure MPU401 */ if (mpuport[ndev] > 0 && mpuirq[ndev] > 0) { val = CFG2_MPU_ENABLE; pos = array_find_l(mpu_ports, mpuport[ndev]); if (pos < 0) - snd_printk(KERN_WARNING "invalid MPU401 port 0x%lx\n", - mpuport[ndev]); + dev_warn(pdev, "invalid MPU401 port 0x%lx\n", + mpuport[ndev]); else { val |= mpu_port_bits[pos] << 5; pos = array_find(mpu_irqs, mpuirq[ndev]); if (pos < 0) - snd_printk(KERN_WARNING "invalid MPU401 IRQ %d\n", - mpuirq[ndev]); + dev_warn(pdev, "invalid MPU401 IRQ %d\n", + mpuirq[ndev]); else { val |= mpu_irq_bits[pos] << 3; snd_cmi8328_cfg_write(port, CFG2, val); if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpuport[ndev], 0, mpuirq[ndev], NULL) < 0) - snd_printk(KERN_ERR "error initializing MPU401\n"); + dev_err(pdev, "error initializing MPU401\n"); } } } /* OPL3 is hardwired to 0x388 and cannot be disabled */ if (snd_opl3_create(card, 0x388, 0x38a, OPL3_HW_AUTO, 0, &opl3) < 0) - snd_printk(KERN_ERR "error initializing OPL3\n"); + dev_err(pdev, "error initializing OPL3\n"); else if (snd_opl3_hwdep_new(opl3, 0, 1, NULL) < 0) - snd_printk(KERN_WARNING "error initializing OPL3 hwdep\n"); + dev_warn(pdev, "error initializing OPL3 hwdep\n"); strcpy(card->driver, "CMI8328"); strcpy(card->shortname, "C-Media CMI8328"); @@ -378,7 +378,7 @@ static int snd_cmi8328_probe(struct device *pdev, unsigned int ndev) /* gameport is hardwired to 0x200 */ res = devm_request_region(pdev, 0x200, 8, "CMI8328 gameport"); if (!res) - snd_printk(KERN_WARNING "unable to allocate gameport I/O port\n"); + dev_warn(pdev, "unable to allocate gameport I/O port\n"); else { struct gameport *gp = cmi->gameport = gameport_allocate_port(); if (cmi->gameport) { From 6aa5cb8540d0763cd00c97eb7855c80c6eb4b7a7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:12 +0200 Subject: [PATCH 127/410] ALSA: cmi8330: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-23-tiwai@suse.de --- sound/isa/cmi8330.c | 36 +++++++++++++++++------------------- 1 file changed, 17 insertions(+), 19 deletions(-) diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index f209b16c5229..25b4dc181089 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -342,7 +342,7 @@ static int snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "AD1848 PnP configure failure\n"); + dev_err(&pdev->dev, "AD1848 PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -356,7 +356,7 @@ static int snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "SB16 PnP configure failure\n"); + dev_err(&pdev->dev, "SB16 PnP configure failure\n"); return -EBUSY; } sbport[dev] = pnp_port_start(pdev, 0); @@ -376,7 +376,7 @@ static int snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, err = pnp_activate_dev(pdev); if (err < 0) - snd_printk(KERN_ERR "MPU-401 PnP configure failure: will be disabled\n"); + dev_err(&pdev->dev, "MPU-401 PnP configure failure: will be disabled\n"); else { mpuport[dev] = pnp_port_start(pdev, 0); mpuirq[dev] = pnp_irq(pdev, 0); @@ -498,8 +498,6 @@ static int snd_cmi8330_resume(struct snd_card *card) #define is_isapnp_selected(dev) 0 #endif -#define PFX "cmi8330: " - static int snd_cmi8330_card_new(struct device *pdev, int dev, struct snd_card **cardp) { @@ -510,7 +508,7 @@ static int snd_cmi8330_card_new(struct device *pdev, int dev, err = snd_devm_card_new(pdev, index[dev], id[dev], THIS_MODULE, sizeof(struct snd_cmi8330), &card); if (err < 0) { - snd_printk(KERN_ERR PFX "could not get a new card\n"); + dev_err(pdev, "could not get a new card\n"); return err; } acard = card->private_data; @@ -531,11 +529,11 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "AD1848 device busy??\n"); + dev_err(card->dev, "AD1848 device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "AD1848 not found during probe\n"); + dev_err(card->dev, "AD1848 not found during probe\n"); return -ENODEV; } @@ -546,11 +544,11 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) sbdma16[dev], SB_HW_AUTO, &acard->sb); if (err < 0) { - snd_printk(KERN_ERR PFX "SB16 device busy??\n"); + dev_err(card->dev, "SB16 device busy??\n"); return err; } if (acard->sb->hardware != SB_HW_16) { - snd_printk(KERN_ERR PFX "SB16 not found during probe\n"); + dev_err(card->dev, "SB16 not found during probe\n"); return -ENODEV; } @@ -561,22 +559,22 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) err = snd_cmi8330_mixer(card, acard); if (err < 0) { - snd_printk(KERN_ERR PFX "failed to create mixers\n"); + dev_err(card->dev, "failed to create mixers\n"); return err; } err = snd_cmi8330_pcm(card, acard); if (err < 0) { - snd_printk(KERN_ERR PFX "failed to create pcms\n"); + dev_err(card->dev, "failed to create pcms\n"); return err; } if (fmport[dev] != SNDRV_AUTO_PORT) { if (snd_opl3_create(card, fmport[dev], fmport[dev] + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX - "no OPL device at 0x%lx-0x%lx ?\n", - fmport[dev], fmport[dev] + 2); + dev_err(card->dev, + "no OPL device at 0x%lx-0x%lx ?\n", + fmport[dev], fmport[dev] + 2); } else { err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) @@ -588,7 +586,7 @@ static int snd_cmi8330_probe(struct snd_card *card, int dev) if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpuport[dev], 0, mpuirq[dev], NULL) < 0) - printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", + dev_err(card->dev, "no MPU-401 device at 0x%lx.\n", mpuport[dev]); } @@ -609,11 +607,11 @@ static int snd_cmi8330_isa_match(struct device *pdev, if (!enable[dev] || is_isapnp_selected(dev)) return 0; if (wssport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify wssport\n"); + dev_err(pdev, "specify wssport\n"); return 0; } if (sbport[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify sbport\n"); + dev_err(pdev, "specify sbport\n"); return 0; } return 1; @@ -683,7 +681,7 @@ static int snd_cmi8330_pnp_detect(struct pnp_card_link *pcard, return res; res = snd_cmi8330_pnp(dev, card->private_data, pcard, pid); if (res < 0) { - snd_printk(KERN_ERR PFX "PnP detection failed\n"); + dev_err(card->dev, "PnP detection failed\n"); return res; } res = snd_cmi8330_probe(card, dev); From 257d0c813b65cd2afb5ee68c026053d011509063 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:13 +0200 Subject: [PATCH 128/410] ALSA: cs4236: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-24-tiwai@suse.de --- sound/isa/cs423x/cs4236.c | 31 ++++++++++--------- sound/isa/cs423x/cs4236_lib.c | 56 +++++++++++++++++------------------ 2 files changed, 45 insertions(+), 42 deletions(-) diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 7226cbf2d7de..ad20bb2649bd 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -204,7 +204,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); static int snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev) { if (pnp_activate_dev(pdev) < 0) { - printk(KERN_ERR IDENT " WSS PnP configure failed for WSS (out of resources?)\n"); + dev_err(&pdev->dev, IDENT " WSS PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } port[dev] = pnp_port_start(pdev, 0); @@ -214,10 +214,12 @@ static int snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev) irq[dev] = pnp_irq(pdev, 0); dma1[dev] = pnp_dma(pdev, 0); dma2[dev] = pnp_dma(pdev, 1) == 4 ? -1 : (int)pnp_dma(pdev, 1); - snd_printdd("isapnp WSS: wss port=0x%lx, fm port=0x%lx, sb port=0x%lx\n", - port[dev], fm_port[dev], sb_port[dev]); - snd_printdd("isapnp WSS: irq=%i, dma1=%i, dma2=%i\n", - irq[dev], dma1[dev], dma2[dev]); + dev_dbg(&pdev->dev, + "isapnp WSS: wss port=0x%lx, fm port=0x%lx, sb port=0x%lx\n", + port[dev], fm_port[dev], sb_port[dev]); + dev_dbg(&pdev->dev, + "isapnp WSS: irq=%i, dma1=%i, dma2=%i\n", + irq[dev], dma1[dev], dma2[dev]); return 0; } @@ -225,11 +227,11 @@ static int snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev) static int snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev) { if (pnp_activate_dev(pdev) < 0) { - printk(KERN_ERR IDENT " CTRL PnP configure failed for WSS (out of resources?)\n"); + dev_err(&pdev->dev, IDENT " CTRL PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } cport[dev] = pnp_port_start(pdev, 0); - snd_printdd("isapnp CTRL: control port=0x%lx\n", cport[dev]); + dev_dbg(&pdev->dev, "isapnp CTRL: control port=0x%lx\n", cport[dev]); return 0; } @@ -237,7 +239,7 @@ static int snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev) static int snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) { if (pnp_activate_dev(pdev) < 0) { - printk(KERN_ERR IDENT " MPU401 PnP configure failed for WSS (out of resources?)\n"); + dev_err(&pdev->dev, IDENT " MPU401 PnP configure failed for WSS (out of resources?)\n"); mpu_port[dev] = SNDRV_AUTO_PORT; mpu_irq[dev] = SNDRV_AUTO_IRQ; } else { @@ -250,7 +252,7 @@ static int snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) mpu_irq[dev] = -1; /* disable interrupt */ } } - snd_printdd("isapnp MPU: port=0x%lx, irq=%i\n", mpu_port[dev], mpu_irq[dev]); + dev_dbg(&pdev->dev, "isapnp MPU: port=0x%lx, irq=%i\n", mpu_port[dev], mpu_irq[dev]); return 0; } @@ -333,7 +335,8 @@ static int snd_cs423x_probe(struct snd_card *card, int dev) if (sb_port[dev] > 0 && sb_port[dev] != SNDRV_AUTO_PORT) { if (!devm_request_region(card->dev, sb_port[dev], 16, IDENT " SB")) { - printk(KERN_ERR IDENT ": unable to register SB port at 0x%lx\n", sb_port[dev]); + dev_err(card->dev, IDENT ": unable to register SB port at 0x%lx\n", + sb_port[dev]); return -EBUSY; } } @@ -384,7 +387,7 @@ static int snd_cs423x_probe(struct snd_card *card, int dev) if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_OPL3_CS, 0, &opl3) < 0) { - printk(KERN_WARNING IDENT ": OPL3 not detected\n"); + dev_warn(card->dev, IDENT ": OPL3 not detected\n"); } else { err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) @@ -398,7 +401,7 @@ static int snd_cs423x_probe(struct snd_card *card, int dev) if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - printk(KERN_WARNING IDENT ": MPU401 not detected\n"); + dev_warn(card->dev, IDENT ": MPU401 not detected\n"); } return snd_card_register(card); @@ -521,7 +524,7 @@ static int snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, return err; err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); if (err < 0) { - printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); + dev_err(card->dev, "PnP BIOS detection failed for " IDENT "\n"); return err; } err = snd_cs423x_probe(card, dev); @@ -573,7 +576,7 @@ static int snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, return res; res = snd_card_cs423x_pnpc(dev, card->private_data, pcard, pid); if (res < 0) { - printk(KERN_ERR "isapnp detection failed and probing for " IDENT + dev_err(card->dev, "isapnp detection failed and probing for " IDENT " is not supported\n"); return res; } diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 35f25911adcf..1a03cff6915b 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -279,8 +279,8 @@ int snd_cs4236_create(struct snd_card *card, return err; if ((chip->hardware & WSS_HW_CS4236B_MASK) == 0) { - snd_printd("chip is not CS4236+, hardware=0x%x\n", - chip->hardware); + dev_dbg(card->dev, "chip is not CS4236+, hardware=0x%x\n", + chip->hardware); *rchip = chip; return 0; } @@ -288,25 +288,25 @@ int snd_cs4236_create(struct snd_card *card, { int idx; for (idx = 0; idx < 8; idx++) - snd_printk(KERN_DEBUG "CD%i = 0x%x\n", - idx, inb(chip->cport + idx)); + dev_dbg(card->dev, "CD%i = 0x%x\n", + idx, inb(chip->cport + idx)); for (idx = 0; idx < 9; idx++) - snd_printk(KERN_DEBUG "C%i = 0x%x\n", - idx, snd_cs4236_ctrl_in(chip, idx)); + dev_dbg(card->dev, "C%i = 0x%x\n", + idx, snd_cs4236_ctrl_in(chip, idx)); } #endif if (cport < 0x100 || cport == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "please, specify control port " - "for CS4236+ chips\n"); + dev_err(card->dev, "please, specify control port for CS4236+ chips\n"); return -ENODEV; } ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); - snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", - cport, ver1, ver2); + dev_dbg(card->dev, "CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", + cport, ver1, ver2); if (ver1 != ver2) { - snd_printk(KERN_ERR "CS4236+ chip detected, but " - "control port 0x%lx is not valid\n", cport); + dev_err(card->dev, + "CS4236+ chip detected, but control port 0x%lx is not valid\n", + cport); return -ENODEV; } snd_cs4236_ctrl_out(chip, 0, 0x00); @@ -934,14 +934,14 @@ static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct sn spin_lock_irqsave(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0; #if 0 - printk(KERN_DEBUG "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " - "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", - snd_wss_in(chip, CS4231_ALT_FEATURE_1), - snd_cs4236_ctrl_in(chip, 3), - snd_cs4236_ctrl_in(chip, 4), - snd_cs4236_ctrl_in(chip, 5), - snd_cs4236_ctrl_in(chip, 6), - snd_cs4236_ctrl_in(chip, 8)); + dev_dbg(chip->card->dev, + "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + snd_wss_in(chip, CS4231_ALT_FEATURE_1), + snd_cs4236_ctrl_in(chip, 3), + snd_cs4236_ctrl_in(chip, 4), + snd_cs4236_ctrl_in(chip, 5), + snd_cs4236_ctrl_in(chip, 6), + snd_cs4236_ctrl_in(chip, 8)); #endif spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; @@ -972,14 +972,14 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn mutex_unlock(&chip->mce_mutex); #if 0 - printk(KERN_DEBUG "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " - "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", - snd_wss_in(chip, CS4231_ALT_FEATURE_1), - snd_cs4236_ctrl_in(chip, 3), - snd_cs4236_ctrl_in(chip, 4), - snd_cs4236_ctrl_in(chip, 5), - snd_cs4236_ctrl_in(chip, 6), - snd_cs4236_ctrl_in(chip, 8)); + dev_dbg(chip->card->dev, + "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + snd_wss_in(chip, CS4231_ALT_FEATURE_1), + snd_cs4236_ctrl_in(chip, 3), + snd_cs4236_ctrl_in(chip, 4), + snd_cs4236_ctrl_in(chip, 5), + snd_cs4236_ctrl_in(chip, 6), + snd_cs4236_ctrl_in(chip, 8)); #endif return change; } From 7f7eff209ee283eec9ae56e9c1446dd1ac3e1022 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:14 +0200 Subject: [PATCH 129/410] ALSA: es1688: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. For referring to the device, introduce snd_card pointer to struct snd_es1688. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-25-tiwai@suse.de --- include/sound/es1688.h | 1 + sound/isa/es1688/es1688.c | 2 +- sound/isa/es1688/es1688_lib.c | 55 ++++++++++++++++++----------------- 3 files changed, 31 insertions(+), 27 deletions(-) diff --git a/include/sound/es1688.h b/include/sound/es1688.h index 099569c31fbb..425a3717d77a 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -17,6 +17,7 @@ #define ES1688_HW_UNDEF 0x0003 struct snd_es1688 { + struct snd_card *card; unsigned long port; /* port of ESS chip */ struct resource *res_port; unsigned long mpu_port; /* MPU-401 port of ESS chip */ diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index 97728bf45474..6a95dfb7600a 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -213,7 +213,7 @@ static int snd_card_es968_pnp(struct snd_card *card, unsigned int n, error = pnp_activate_dev(pdev); if (error < 0) { - snd_printk(KERN_ERR "ES968 pnp configure failure\n"); + dev_err(card->dev, "ES968 pnp configure failure\n"); return error; } port[n] = pnp_port_start(pdev, 0); diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 8554cb2263c1..c0c230149a75 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -30,9 +30,7 @@ static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val) outb(val, ES1688P(chip, COMMAND)); return 1; } -#ifdef CONFIG_SND_DEBUG - printk(KERN_DEBUG "snd_es1688_dsp_command: timeout (0x%x)\n", val); -#endif + dev_dbg(chip->card->dev, "%s: timeout (0x%x)\n", __func__, val); return 0; } @@ -43,7 +41,8 @@ static int snd_es1688_dsp_get_byte(struct snd_es1688 *chip) for (i = 1000; i; i--) if (inb(ES1688P(chip, DATA_AVAIL)) & 0x80) return inb(ES1688P(chip, READ)); - snd_printd("es1688 get byte failed: 0x%lx = 0x%x!!!\n", ES1688P(chip, DATA_AVAIL), inb(ES1688P(chip, DATA_AVAIL))); + dev_dbg(chip->card->dev, "es1688 get byte failed: 0x%lx = 0x%x!!!\n", + ES1688P(chip, DATA_AVAIL), inb(ES1688P(chip, DATA_AVAIL))); return -ENODEV; } @@ -95,7 +94,8 @@ int snd_es1688_reset(struct snd_es1688 *chip) udelay(30); for (i = 0; i < 1000 && !(inb(ES1688P(chip, DATA_AVAIL)) & 0x80); i++); if (inb(ES1688P(chip, READ)) != 0xaa) { - snd_printd("ess_reset at 0x%lx: failed!!!\n", chip->port); + dev_dbg(chip->card->dev, "ess_reset at 0x%lx: failed!!!\n", + chip->port); return -ENODEV; } snd_es1688_dsp_command(chip, 0xc6); /* enable extended mode */ @@ -127,7 +127,8 @@ static int snd_es1688_probe(struct snd_es1688 *chip) inb(ES1688P(chip, ENABLE0)); /* ENABLE0 */ if (snd_es1688_reset(chip) < 0) { - snd_printdd("ESS: [0x%lx] reset failed... 0x%x\n", chip->port, inb(ES1688P(chip, READ))); + dev_dbg(chip->card->dev, "ESS: [0x%lx] reset failed... 0x%x\n", + chip->port, inb(ES1688P(chip, READ))); spin_unlock_irqrestore(&chip->reg_lock, flags); return -ENODEV; } @@ -145,7 +146,9 @@ static int snd_es1688_probe(struct snd_es1688 *chip) spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_printdd("ESS: [0x%lx] found.. major = 0x%x, minor = 0x%x\n", chip->port, major, minor); + dev_dbg(chip->card->dev, + "ESS: [0x%lx] found.. major = 0x%x, minor = 0x%x\n", + chip->port, major, minor); chip->version = (major << 8) | minor; if (!chip->version) @@ -153,15 +156,16 @@ static int snd_es1688_probe(struct snd_es1688 *chip) switch (chip->version & 0xfff0) { case 0x4880: - snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, " - "but driver is in another place\n", chip->port); + dev_err(chip->card->dev, + "[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", + chip->port); return -ENODEV; case 0x6880: break; default: - snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip " - "with version 0x%x (Jazz16 soundcard?)\n", - chip->port, chip->version); + dev_err(chip->card->dev, + "[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", + chip->port, chip->version); return -ENODEV; } @@ -210,9 +214,6 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) } } } -#if 0 - snd_printk(KERN_DEBUG "mpu cfg = 0x%x\n", cfg); -#endif spin_lock_irqsave(&chip->reg_lock, flags); snd_es1688_mixer_write(chip, 0x40, cfg); spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -225,9 +226,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* enable only DMA counter interrupt */ irq_bits = irqs[chip->irq & 0x0f]; if (irq_bits < 0) { - snd_printk(KERN_ERR "[0x%lx] ESS: bad IRQ %d " - "for ES1688 chip!!\n", - chip->port, chip->irq); + dev_err(chip->card->dev, + "[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", + chip->port, chip->irq); #if 0 irq_bits = 0; cfg = 0x10; @@ -240,8 +241,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* extended mode DMA enable */ dma = chip->dma8; if (dma > 3 || dma == 2) { - snd_printk(KERN_ERR "[0x%lx] ESS: bad DMA channel %d " - "for ES1688 chip!!\n", chip->port, dma); + dev_err(chip->card->dev, + "[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", + chip->port, dma); #if 0 dma_bits = 0; cfg = 0x00; /* disable all DMA */ @@ -326,9 +328,9 @@ static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char va return -EINVAL; /* something is wrong */ } #if 0 - printk(KERN_DEBUG "trigger: val = 0x%x, value = 0x%x\n", val, value); - printk(KERN_DEBUG "trigger: pointer = 0x%x\n", - snd_dma_pointer(chip->dma8, chip->dma_size)); + dev_dbg(chip->card->dev, "trigger: val = 0x%x, value = 0x%x\n", val, value); + dev_dbg(chip->card->dev, "trigger: pointer = 0x%x\n", + snd_dma_pointer(chip->dma8, chip->dma_size)); #endif snd_es1688_write(chip, 0xb8, (val & 0xf0) | value); spin_unlock(&chip->reg_lock); @@ -620,20 +622,21 @@ int snd_es1688_create(struct snd_card *card, if (chip == NULL) return -ENOMEM; + chip->card = card; chip->irq = -1; chip->dma8 = -1; chip->hardware = ES1688_HW_UNDEF; chip->res_port = request_region(port + 4, 12, "ES1688"); if (chip->res_port == NULL) { - snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); + dev_err(card->dev, "es1688: can't grab port 0x%lx\n", port + 4); err = -EBUSY; goto exit; } err = request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip); if (err < 0) { - snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); + dev_err(card->dev, "es1688: can't grab IRQ %d\n", irq); goto exit; } @@ -642,7 +645,7 @@ int snd_es1688_create(struct snd_card *card, err = request_dma(dma8, "ES1688"); if (err < 0) { - snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); + dev_err(card->dev, "es1688: can't grab DMA8 %d\n", dma8); goto exit; } chip->dma8 = dma8; From 12174dfee07e28c6e721d072174a128ec8433bb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:15 +0200 Subject: [PATCH 130/410] ALSA: es18xx: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. For referring to the device, introduce snd_card pointer to struct snd_es18xx. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-26-tiwai@suse.de --- sound/isa/es18xx.c | 87 ++++++++++++++++++++++++---------------------- 1 file changed, 46 insertions(+), 41 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 0a32845b1017..59c784a70ac1 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -82,9 +82,8 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include -#define PFX "es18xx: " - struct snd_es18xx { + struct snd_card *card; unsigned long port; /* port of ESS chip */ unsigned long ctrl_port; /* Control port of ESS chip */ int irq; /* IRQ number of ESS chip */ @@ -165,7 +164,7 @@ static int snd_es18xx_dsp_command(struct snd_es18xx *chip, unsigned char val) outb(val, chip->port + 0x0C); return 0; } - snd_printk(KERN_ERR "dsp_command: timeout (0x%x)\n", val); + dev_err(chip->card->dev, "dsp_command: timeout (0x%x)\n", val); return -EINVAL; } @@ -176,8 +175,8 @@ static int snd_es18xx_dsp_get_byte(struct snd_es18xx *chip) for(i = MILLISECOND/10; i; i--) if (inb(chip->port + 0x0C) & 0x40) return inb(chip->port + 0x0A); - snd_printk(KERN_ERR "dsp_get_byte failed: 0x%lx = 0x%x!!!\n", - chip->port + 0x0A, inb(chip->port + 0x0A)); + dev_err(chip->card->dev, "dsp_get_byte failed: 0x%lx = 0x%x!!!\n", + chip->port + 0x0A, inb(chip->port + 0x0A)); return -ENODEV; } @@ -197,7 +196,7 @@ static int snd_es18xx_write(struct snd_es18xx *chip, end: spin_unlock_irqrestore(&chip->reg_lock, flags); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Reg %02x set to %02x\n", reg, data); + dev_dbg(chip->card->dev, "Reg %02x set to %02x\n", reg, data); #endif return ret; } @@ -216,7 +215,7 @@ static int snd_es18xx_read(struct snd_es18xx *chip, unsigned char reg) data = snd_es18xx_dsp_get_byte(chip); ret = data; #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Reg %02x now is %02x (%d)\n", reg, data, ret); + dev_dbg(chip->card->dev, "Reg %02x now is %02x (%d)\n", reg, data, ret); #endif end: spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -252,8 +251,8 @@ static int snd_es18xx_bits(struct snd_es18xx *chip, unsigned char reg, if (ret < 0) goto end; #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Reg %02x was %02x, set to %02x (%d)\n", - reg, old, new, ret); + dev_dbg(chip->card->dev, "Reg %02x was %02x, set to %02x (%d)\n", + reg, old, new, ret); #endif } ret = oval; @@ -271,7 +270,7 @@ static inline void snd_es18xx_mixer_write(struct snd_es18xx *chip, outb(data, chip->port + 0x05); spin_unlock_irqrestore(&chip->mixer_lock, flags); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Mixer reg %02x set to %02x\n", reg, data); + dev_dbg(chip->card->dev, "Mixer reg %02x set to %02x\n", reg, data); #endif } @@ -284,7 +283,7 @@ static inline int snd_es18xx_mixer_read(struct snd_es18xx *chip, unsigned char r data = inb(chip->port + 0x05); spin_unlock_irqrestore(&chip->mixer_lock, flags); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Mixer reg %02x now is %02x\n", reg, data); + dev_dbg(chip->card->dev, "Mixer reg %02x now is %02x\n", reg, data); #endif return data; } @@ -303,8 +302,8 @@ static inline int snd_es18xx_mixer_bits(struct snd_es18xx *chip, unsigned char r new = (old & ~mask) | (val & mask); outb(new, chip->port + 0x05); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Mixer reg %02x was %02x, set to %02x\n", - reg, old, new); + dev_dbg(chip->card->dev, "Mixer reg %02x was %02x, set to %02x\n", + reg, old, new); #endif } spin_unlock_irqrestore(&chip->mixer_lock, flags); @@ -324,8 +323,8 @@ static inline int snd_es18xx_mixer_writable(struct snd_es18xx *chip, unsigned ch new = inb(chip->port + 0x05); spin_unlock_irqrestore(&chip->mixer_lock, flags); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Mixer reg %02x was %02x, set to %02x, now is %02x\n", - reg, old, expected, new); + dev_dbg(chip->card->dev, "Mixer reg %02x was %02x, set to %02x, now is %02x\n", + reg, old, expected, new); #endif return expected == new; } @@ -1356,7 +1355,7 @@ static void snd_es18xx_config_write(struct snd_es18xx *chip, outb(reg, chip->ctrl_port); outb(data, chip->ctrl_port + 1); #ifdef REG_DEBUG - snd_printk(KERN_DEBUG "Config reg %02x set to %02x\n", reg, data); + dev_dbg(chip->card->dev, "Config reg %02x set to %02x\n", reg, data); #endif } @@ -1425,7 +1424,7 @@ static int snd_es18xx_initialize(struct snd_es18xx *chip, irqmask = 3; break; default: - snd_printk(KERN_ERR "invalid irq %d\n", chip->irq); + dev_err(chip->card->dev, "invalid irq %d\n", chip->irq); return -ENODEV; } switch (chip->dma1) { @@ -1439,7 +1438,7 @@ static int snd_es18xx_initialize(struct snd_es18xx *chip, dma1mask = 3; break; default: - snd_printk(KERN_ERR "invalid dma1 %d\n", chip->dma1); + dev_err(chip->card->dev, "invalid dma1 %d\n", chip->dma1); return -ENODEV; } switch (chip->dma2) { @@ -1456,7 +1455,7 @@ static int snd_es18xx_initialize(struct snd_es18xx *chip, dma2mask = 3; break; default: - snd_printk(KERN_ERR "invalid dma2 %d\n", chip->dma2); + dev_err(chip->card->dev, "invalid dma2 %d\n", chip->dma2); return -ENODEV; } @@ -1531,7 +1530,7 @@ static int snd_es18xx_identify(struct snd_card *card, struct snd_es18xx *chip) /* reset */ if (snd_es18xx_reset(chip) < 0) { - snd_printk(KERN_ERR "reset at 0x%lx failed!!!\n", chip->port); + dev_err(card->dev, "reset at 0x%lx failed!!!\n", chip->port); return -ENODEV; } @@ -1569,7 +1568,7 @@ static int snd_es18xx_identify(struct snd_card *card, struct snd_es18xx *chip) if (!devm_request_region(card->dev, chip->ctrl_port, 8, "ES18xx - CTRL")) { - snd_printk(KERN_ERR PFX "unable go grab port 0x%lx\n", chip->ctrl_port); + dev_err(card->dev, "unable go grab port 0x%lx\n", chip->ctrl_port); return -EBUSY; } @@ -1601,7 +1600,7 @@ static int snd_es18xx_probe(struct snd_card *card, unsigned long fm_port) { if (snd_es18xx_identify(card, chip) < 0) { - snd_printk(KERN_ERR PFX "[0x%lx] ESS chip not found\n", chip->port); + dev_err(card->dev, "[0x%lx] ESS chip not found\n", chip->port); return -ENODEV; } @@ -1623,12 +1622,12 @@ static int snd_es18xx_probe(struct snd_card *card, chip->caps = ES18XX_PCM2 | ES18XX_RECMIX | ES18XX_AUXB | ES18XX_DUPLEX_SAME | ES18XX_GPO_2BIT; break; default: - snd_printk(KERN_ERR "[0x%lx] unsupported chip ES%x\n", - chip->port, chip->version); + dev_err(card->dev, "[0x%lx] unsupported chip ES%x\n", + chip->port, chip->version); return -ENODEV; } - snd_printd("[0x%lx] ESS%x chip found\n", chip->port, chip->version); + dev_dbg(card->dev, "[0x%lx] ESS%x chip found\n", chip->port, chip->version); if (chip->dma1 == chip->dma2) chip->caps &= ~(ES18XX_PCM2 | ES18XX_DUPLEX_SAME); @@ -1725,6 +1724,7 @@ static int snd_es18xx_new_device(struct snd_card *card, { struct snd_es18xx *chip = card->private_data; + chip->card = card; spin_lock_init(&chip->reg_lock); spin_lock_init(&chip->mixer_lock); chip->port = port; @@ -1735,27 +1735,27 @@ static int snd_es18xx_new_device(struct snd_card *card, chip->active = 0; if (!devm_request_region(card->dev, port, 16, "ES18xx")) { - snd_printk(KERN_ERR PFX "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1); + dev_err(card->dev, "unable to grap ports 0x%lx-0x%lx\n", port, port + 16 - 1); return -EBUSY; } if (devm_request_irq(card->dev, irq, snd_es18xx_interrupt, 0, "ES18xx", (void *) card)) { - snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); + dev_err(card->dev, "unable to grap IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; card->sync_irq = chip->irq; if (snd_devm_request_dma(card->dev, dma1, "ES18xx DMA 1")) { - snd_printk(KERN_ERR PFX "unable to grap DMA1 %d\n", dma1); + dev_err(card->dev, "unable to grap DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (dma2 != dma1 && snd_devm_request_dma(card->dev, dma2, "ES18xx DMA 2")) { - snd_printk(KERN_ERR PFX "unable to grap DMA2 %d\n", dma2); + dev_err(card->dev, "unable to grap DMA2 %d\n", dma2); return -EBUSY; } chip->dma2 = dma2; @@ -1954,7 +1954,7 @@ MODULE_DEVICE_TABLE(pnp, snd_audiodrive_pnpbiosids); static int snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) { if (pnp_activate_dev(pdev) < 0) { - snd_printk(KERN_ERR PFX "PnP configure failure (out of resources?)\n"); + dev_err(&pdev->dev, "PnP configure failure (out of resources?)\n"); return -EBUSY; } /* ok. hack using Vendor-Defined Card-Level registers */ @@ -1973,8 +1973,12 @@ static int snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) dma1[dev] = pnp_dma(pdev, 0); dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - snd_printdd("PnP ES18xx: port=0x%lx, fm port=0x%lx, mpu port=0x%lx\n", port[dev], fm_port[dev], mpu_port[dev]); - snd_printdd("PnP ES18xx: dma1=%i, dma2=%i, irq=%i\n", dma1[dev], dma2[dev], irq[dev]); + dev_dbg(&pdev->dev, + "PnP ES18xx: port=0x%lx, fm port=0x%lx, mpu port=0x%lx\n", + port[dev], fm_port[dev], mpu_port[dev]); + dev_dbg(&pdev->dev, + "PnP ES18xx: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); return 0; } @@ -2022,11 +2026,12 @@ static int snd_audiodrive_pnpc(int dev, struct snd_es18xx *chip, /* Control port initialization */ if (pnp_activate_dev(chip->devc) < 0) { - snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); + dev_err(chip->card->dev, + "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } - snd_printdd("pnp: port=0x%llx\n", - (unsigned long long)pnp_port_start(chip->devc, 0)); + dev_dbg(chip->card->dev, "pnp: port=0x%llx\n", + (unsigned long long)pnp_port_start(chip->devc, 0)); if (snd_audiodrive_pnp_init_main(dev, chip->dev) < 0) return -EBUSY; @@ -2084,9 +2089,9 @@ static int snd_audiodrive_probe(struct snd_card *card, int dev) if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) { if (snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_OPL3, 0, &opl3) < 0) { - snd_printk(KERN_WARNING PFX - "opl3 not detected at 0x%lx\n", - fm_port[dev]); + dev_warn(card->dev, + "opl3 not detected at 0x%lx\n", + fm_port[dev]); } else { err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) @@ -2134,21 +2139,21 @@ static int snd_es18xx_isa_probe(struct device *pdev, unsigned int dev) if (irq[dev] == SNDRV_AUTO_IRQ) { irq[dev] = snd_legacy_find_free_irq(possible_irqs); if (irq[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + dev_err(pdev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[dev] == SNDRV_AUTO_DMA) { dma1[dev] = snd_legacy_find_free_dma(possible_dmas); if (dma1[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA1\n"); + dev_err(pdev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[dev] == SNDRV_AUTO_DMA) { dma2[dev] = snd_legacy_find_free_dma(possible_dmas); if (dma2[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA2\n"); + dev_err(pdev, "unable to find a free DMA2\n"); return -EBUSY; } } From a6676811deb7e6b35d541437f49cd84c6b6d72bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:16 +0200 Subject: [PATCH 131/410] ALSA: gus: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some commented-out debug prints and dead code are dropped as well. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-27-tiwai@suse.de --- sound/isa/gus/gus_dma.c | 37 ++++--- sound/isa/gus/gus_io.c | 215 +++++++++++++++++++++++++------------ sound/isa/gus/gus_irq.c | 7 +- sound/isa/gus/gus_main.c | 29 ++--- sound/isa/gus/gus_mem.c | 2 +- sound/isa/gus/gus_pcm.c | 33 ++---- sound/isa/gus/gus_reset.c | 8 +- sound/isa/gus/gus_uart.c | 21 ++-- sound/isa/gus/gus_volume.c | 7 +- sound/isa/gus/gusclassic.c | 4 +- sound/isa/gus/gusextreme.c | 4 +- sound/isa/gus/gusmax.c | 16 ++- sound/isa/gus/interwave.c | 61 ++++++----- 13 files changed, 266 insertions(+), 178 deletions(-) diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index 6d664dd8dde0..bb5a4e2fbfb3 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -30,15 +30,18 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, unsigned char dma_cmd; unsigned int address_high; - snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", - addr, buf_addr, count); + dev_dbg(gus->card->dev, + "dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", + addr, buf_addr, count); if (gus->gf1.dma1 > 3) { if (gus->gf1.enh_mode) { address = addr >> 1; } else { if (addr & 0x1f) { - snd_printd("snd_gf1_dma_transfer: unaligned address (0x%x)?\n", addr); + dev_dbg(gus->card->dev, + "%s: unaligned address (0x%x)?\n", + __func__, addr); return; } address = (addr & 0x000c0000) | ((addr & 0x0003ffff) >> 1); @@ -63,8 +66,9 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, snd_gf1_dma_ack(gus); snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE); #if 0 - snd_printk(KERN_DEBUG "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", - address << 1, count, dma_cmd); + dev_dbg(gus->card->dev, + "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", + address << 1, count, dma_cmd); #endif spin_lock_irqsave(&gus->reg_lock, flags); if (gus->gf1.enh_mode) { @@ -131,9 +135,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 - snd_printd(KERN_DEBUG "program dma (IRQ) - " - "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", - block->addr, block->buf_addr, block->count, block->cmd); + dev_dbg(gus->card->dev, + "program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, block->buf_addr, block->count, block->cmd); #endif } @@ -194,14 +198,17 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, *block = *__block; block->next = NULL; - snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", - block->addr, (long) block->buffer, block->count, - block->cmd); + dev_dbg(gus->card->dev, + "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, (long) block->buffer, block->count, + block->cmd); - snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n", - (long)gus->gf1.dma_data_pcm_last); - snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n", - (long)gus->gf1.dma_data_pcm); + dev_dbg(gus->card->dev, + "gus->gf1.dma_data_pcm_last = 0x%lx\n", + (long)gus->gf1.dma_data_pcm_last); + dev_dbg(gus->card->dev, + "gus->gf1.dma_data_pcm = 0x%lx\n", + (long)gus->gf1.dma_data_pcm); spin_lock_irqsave(&gus->dma_lock, flags); if (synth) { diff --git a/sound/isa/gus/gus_io.c b/sound/isa/gus/gus_io.c index fb7b5e2636b8..1dc9e0edb3d9 100644 --- a/sound/isa/gus/gus_io.c +++ b/sound/isa/gus/gus_io.c @@ -325,10 +325,8 @@ void snd_gf1_pokew(struct snd_gus_card * gus, unsigned int addr, unsigned short { unsigned long flags; -#ifdef CONFIG_SND_DEBUG if (!gus->interwave) - snd_printk(KERN_DEBUG "snd_gf1_pokew - GF1!!!\n"); -#endif + dev_dbg(gus->card->dev, "%s - GF1!!!\n", __func__); spin_lock_irqsave(&gus->reg_lock, flags); outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel); mb(); @@ -349,10 +347,8 @@ unsigned short snd_gf1_peekw(struct snd_gus_card * gus, unsigned int addr) unsigned long flags; unsigned short res; -#ifdef CONFIG_SND_DEBUG if (!gus->interwave) - snd_printk(KERN_DEBUG "snd_gf1_peekw - GF1!!!\n"); -#endif + dev_dbg(gus->card->dev, "%s - GF1!!!\n", __func__); spin_lock_irqsave(&gus->reg_lock, flags); outb(SNDRV_GF1_GW_DRAM_IO_LOW, gus->gf1.reg_regsel); mb(); @@ -375,10 +371,8 @@ void snd_gf1_dram_setmem(struct snd_gus_card * gus, unsigned int addr, unsigned long port; unsigned long flags; -#ifdef CONFIG_SND_DEBUG if (!gus->interwave) - snd_printk(KERN_DEBUG "snd_gf1_dram_setmem - GF1!!!\n"); -#endif + dev_dbg(gus->card->dev, "%s - GF1!!!\n", __func__); addr &= ~1; count >>= 1; port = GUSP(gus, GF1DATALOW); @@ -433,30 +427,73 @@ void snd_gf1_print_voice_registers(struct snd_gus_card * gus) int voice, ctrl; voice = gus->gf1.active_voice; - printk(KERN_INFO " -%i- GF1 voice ctrl, ramp ctrl = 0x%x, 0x%x\n", voice, ctrl = snd_gf1_i_read8(gus, 0), snd_gf1_i_read8(gus, 0x0d)); - printk(KERN_INFO " -%i- GF1 frequency = 0x%x\n", voice, snd_gf1_i_read16(gus, 1)); - printk(KERN_INFO " -%i- GF1 loop start, end = 0x%x (0x%x), 0x%x (0x%x)\n", voice, snd_gf1_i_read_addr(gus, 2, ctrl & 4), snd_gf1_i_read_addr(gus, 2, (ctrl & 4) ^ 4), snd_gf1_i_read_addr(gus, 4, ctrl & 4), snd_gf1_i_read_addr(gus, 4, (ctrl & 4) ^ 4)); - printk(KERN_INFO " -%i- GF1 ramp start, end, rate = 0x%x, 0x%x, 0x%x\n", voice, snd_gf1_i_read8(gus, 7), snd_gf1_i_read8(gus, 8), snd_gf1_i_read8(gus, 6)); - printk(KERN_INFO" -%i- GF1 volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 9)); - printk(KERN_INFO " -%i- GF1 position = 0x%x (0x%x)\n", voice, snd_gf1_i_read_addr(gus, 0x0a, ctrl & 4), snd_gf1_i_read_addr(gus, 0x0a, (ctrl & 4) ^ 4)); + dev_info(gus->card->dev, + " -%i- GF1 voice ctrl, ramp ctrl = 0x%x, 0x%x\n", + voice, ctrl = snd_gf1_i_read8(gus, 0), snd_gf1_i_read8(gus, 0x0d)); + dev_info(gus->card->dev, + " -%i- GF1 frequency = 0x%x\n", + voice, snd_gf1_i_read16(gus, 1)); + dev_info(gus->card->dev, + " -%i- GF1 loop start, end = 0x%x (0x%x), 0x%x (0x%x)\n", + voice, snd_gf1_i_read_addr(gus, 2, ctrl & 4), + snd_gf1_i_read_addr(gus, 2, (ctrl & 4) ^ 4), + snd_gf1_i_read_addr(gus, 4, ctrl & 4), + snd_gf1_i_read_addr(gus, 4, (ctrl & 4) ^ 4)); + dev_info(gus->card->dev, + " -%i- GF1 ramp start, end, rate = 0x%x, 0x%x, 0x%x\n", + voice, snd_gf1_i_read8(gus, 7), snd_gf1_i_read8(gus, 8), + snd_gf1_i_read8(gus, 6)); + dev_info(gus->card->dev, + " -%i- GF1 volume = 0x%x\n", + voice, snd_gf1_i_read16(gus, 9)); + dev_info(gus->card->dev, + " -%i- GF1 position = 0x%x (0x%x)\n", + voice, snd_gf1_i_read_addr(gus, 0x0a, ctrl & 4), + snd_gf1_i_read_addr(gus, 0x0a, (ctrl & 4) ^ 4)); if (gus->interwave && snd_gf1_i_read8(gus, 0x19) & 0x01) { /* enhanced mode */ mode = snd_gf1_i_read8(gus, 0x15); - printk(KERN_INFO " -%i- GFA1 mode = 0x%x\n", voice, mode); + dev_info(gus->card->dev, + " -%i- GFA1 mode = 0x%x\n", + voice, mode); if (mode & 0x01) { /* Effect processor */ - printk(KERN_INFO " -%i- GFA1 effect address = 0x%x\n", voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); - printk(KERN_INFO " -%i- GFA1 effect volume = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x16)); - printk(KERN_INFO " -%i- GFA1 effect volume final = 0x%x\n", voice, snd_gf1_i_read16(gus, 0x1d)); - printk(KERN_INFO " -%i- GFA1 effect accumulator = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x14)); + dev_info(gus->card->dev, + " -%i- GFA1 effect address = 0x%x\n", + voice, snd_gf1_i_read_addr(gus, 0x11, ctrl & 4)); + dev_info(gus->card->dev, + " -%i- GFA1 effect volume = 0x%x\n", + voice, snd_gf1_i_read16(gus, 0x16)); + dev_info(gus->card->dev, + " -%i- GFA1 effect volume final = 0x%x\n", + voice, snd_gf1_i_read16(gus, 0x1d)); + dev_info(gus->card->dev, + " -%i- GFA1 effect accumulator = 0x%x\n", + voice, snd_gf1_i_read8(gus, 0x14)); } if (mode & 0x20) { - printk(KERN_INFO " -%i- GFA1 left offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x13), snd_gf1_i_read16(gus, 0x13) >> 4); - printk(KERN_INFO " -%i- GFA1 left offset final = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x1c), snd_gf1_i_read16(gus, 0x1c) >> 4); - printk(KERN_INFO " -%i- GFA1 right offset = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x0c), snd_gf1_i_read16(gus, 0x0c) >> 4); - printk(KERN_INFO " -%i- GFA1 right offset final = 0x%x (%i)\n", voice, snd_gf1_i_read16(gus, 0x1b), snd_gf1_i_read16(gus, 0x1b) >> 4); + dev_info(gus->card->dev, + " -%i- GFA1 left offset = 0x%x (%i)\n", + voice, snd_gf1_i_read16(gus, 0x13), + snd_gf1_i_read16(gus, 0x13) >> 4); + dev_info(gus->card->dev, + " -%i- GFA1 left offset final = 0x%x (%i)\n", + voice, snd_gf1_i_read16(gus, 0x1c), + snd_gf1_i_read16(gus, 0x1c) >> 4); + dev_info(gus->card->dev, + " -%i- GFA1 right offset = 0x%x (%i)\n", + voice, snd_gf1_i_read16(gus, 0x0c), + snd_gf1_i_read16(gus, 0x0c) >> 4); + dev_info(gus->card->dev, + " -%i- GFA1 right offset final = 0x%x (%i)\n", + voice, snd_gf1_i_read16(gus, 0x1b), + snd_gf1_i_read16(gus, 0x1b) >> 4); } else - printk(KERN_INFO " -%i- GF1 pan = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x0c)); + dev_info(gus->card->dev, + " -%i- GF1 pan = 0x%x\n", + voice, snd_gf1_i_read8(gus, 0x0c)); } else - printk(KERN_INFO " -%i- GF1 pan = 0x%x\n", voice, snd_gf1_i_read8(gus, 0x0c)); + dev_info(gus->card->dev, + " -%i- GF1 pan = 0x%x\n", + voice, snd_gf1_i_read8(gus, 0x0c)); } #if 0 @@ -465,61 +502,105 @@ void snd_gf1_print_global_registers(struct snd_gus_card * gus) { unsigned char global_mode = 0x00; - printk(KERN_INFO " -G- GF1 active voices = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_ACTIVE_VOICES)); + dev_info(gus->card->dev, + " -G- GF1 active voices = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_ACTIVE_VOICES)); if (gus->interwave) { global_mode = snd_gf1_i_read8(gus, SNDRV_GF1_GB_GLOBAL_MODE); - printk(KERN_INFO " -G- GF1 global mode = 0x%x\n", global_mode); + dev_info(gus->card->dev, + " -G- GF1 global mode = 0x%x\n", + global_mode); } if (global_mode & 0x02) /* LFO enabled? */ - printk(KERN_INFO " -G- GF1 LFO base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_LFO_BASE)); - printk(KERN_INFO " -G- GF1 voices IRQ read = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_VOICES_IRQ_READ)); - printk(KERN_INFO " -G- GF1 DRAM DMA control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL)); - printk(KERN_INFO " -G- GF1 DRAM DMA high/low = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_HIGH), snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_DMA_LOW)); - printk(KERN_INFO " -G- GF1 DRAM IO high/low = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_IO_HIGH), snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_IO_LOW)); + dev_info(gus->card->dev, + " -G- GF1 LFO base = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_LFO_BASE)); + dev_info(gus->card->dev, + " -G- GF1 voices IRQ read = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_VOICES_IRQ_READ)); + dev_info(gus->card->dev, + " -G- GF1 DRAM DMA control = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_CONTROL)); + dev_info(gus->card->dev, + " -G- GF1 DRAM DMA high/low = 0x%x/0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_DMA_HIGH), + snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_DMA_LOW)); + dev_info(gus->card->dev, + " -G- GF1 DRAM IO high/low = 0x%x/0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_DRAM_IO_HIGH), + snd_gf1_i_read16(gus, SNDRV_GF1_GW_DRAM_IO_LOW)); if (!gus->interwave) - printk(KERN_INFO " -G- GF1 record DMA control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL)); - printk(KERN_INFO " -G- GF1 DRAM IO 16 = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_DRAM_IO16)); + dev_info(gus->card->dev, + " -G- GF1 record DMA control = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_REC_DMA_CONTROL)); + dev_info(gus->card->dev, + " -G- GF1 DRAM IO 16 = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_DRAM_IO16)); if (gus->gf1.enh_mode) { - printk(KERN_INFO " -G- GFA1 memory config = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG)); - printk(KERN_INFO " -G- GFA1 memory control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_MEMORY_CONTROL)); - printk(KERN_INFO " -G- GFA1 FIFO record base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR)); - printk(KERN_INFO " -G- GFA1 FIFO playback base = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR)); - printk(KERN_INFO " -G- GFA1 interleave control = 0x%x\n", snd_gf1_i_look16(gus, SNDRV_GF1_GW_INTERLEAVE)); + dev_info(gus->card->dev, + " -G- GFA1 memory config = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG)); + dev_info(gus->card->dev, + " -G- GFA1 memory control = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_MEMORY_CONTROL)); + dev_info(gus->card->dev, + " -G- GFA1 FIFO record base = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_RECORD_BASE_ADDR)); + dev_info(gus->card->dev, + " -G- GFA1 FIFO playback base = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_FIFO_PLAY_BASE_ADDR)); + dev_info(gus->card->dev, + " -G- GFA1 interleave control = 0x%x\n", + snd_gf1_i_look16(gus, SNDRV_GF1_GW_INTERLEAVE)); } } void snd_gf1_print_setup_registers(struct snd_gus_card * gus) { - printk(KERN_INFO " -S- mix control = 0x%x\n", inb(GUSP(gus, MIXCNTRLREG))); - printk(KERN_INFO " -S- IRQ status = 0x%x\n", inb(GUSP(gus, IRQSTAT))); - printk(KERN_INFO " -S- timer control = 0x%x\n", inb(GUSP(gus, TIMERCNTRL))); - printk(KERN_INFO " -S- timer data = 0x%x\n", inb(GUSP(gus, TIMERDATA))); - printk(KERN_INFO " -S- status read = 0x%x\n", inb(GUSP(gus, REGCNTRLS))); - printk(KERN_INFO " -S- Sound Blaster control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL)); - printk(KERN_INFO " -S- AdLib timer 1/2 = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_1), snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_2)); - printk(KERN_INFO " -S- reset = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)); + dev_info(gus->card->dev, + " -S- mix control = 0x%x\n", + inb(GUSP(gus, MIXCNTRLREG))); + dev_info(gus->card->dev, + " -S- IRQ status = 0x%x\n", + inb(GUSP(gus, IRQSTAT))); + dev_info(gus->card->dev, + " -S- timer control = 0x%x\n", + inb(GUSP(gus, TIMERCNTRL))); + dev_info(gus->card->dev, + " -S- timer data = 0x%x\n", + inb(GUSP(gus, TIMERDATA))); + dev_info(gus->card->dev, + " -S- status read = 0x%x\n", + inb(GUSP(gus, REGCNTRLS))); + dev_info(gus->card->dev, + " -S- Sound Blaster control = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_SOUND_BLASTER_CONTROL)); + dev_info(gus->card->dev, + " -S- AdLib timer 1/2 = 0x%x/0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_1), + snd_gf1_i_look8(gus, SNDRV_GF1_GB_ADLIB_TIMER_2)); + dev_info(gus->card->dev, + " -S- reset = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET)); if (gus->interwave) { - printk(KERN_INFO " -S- compatibility = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_COMPATIBILITY)); - printk(KERN_INFO " -S- decode control = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_DECODE_CONTROL)); - printk(KERN_INFO " -S- version number = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER)); - printk(KERN_INFO " -S- MPU-401 emul. control A/B = 0x%x/0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A), snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B)); - printk(KERN_INFO " -S- emulation IRQ = 0x%x\n", snd_gf1_i_look8(gus, SNDRV_GF1_GB_EMULATION_IRQ)); + dev_info(gus->card->dev, + " -S- compatibility = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_COMPATIBILITY)); + dev_info(gus->card->dev, + " -S- decode control = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_DECODE_CONTROL)); + dev_info(gus->card->dev, + " -S- version number = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER)); + dev_info(gus->card->dev, + " -S- MPU-401 emul. control A/B = 0x%x/0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_A), + snd_gf1_i_look8(gus, SNDRV_GF1_GB_MPU401_CONTROL_B)); + dev_info(gus->card->dev, + " -S- emulation IRQ = 0x%x\n", + snd_gf1_i_look8(gus, SNDRV_GF1_GB_EMULATION_IRQ)); } } - -void snd_gf1_peek_print_block(struct snd_gus_card * gus, unsigned int addr, int count, int w_16bit) -{ - if (!w_16bit) { - while (count-- > 0) - printk(count > 0 ? "%02x:" : "%02x", snd_gf1_peek(gus, addr++)); - } else { - while (count-- > 0) { - printk(count > 0 ? "%04x:" : "%04x", snd_gf1_peek(gus, addr) | (snd_gf1_peek(gus, addr + 1) << 8)); - addr += 2; - } - } -} - #endif /* 0 */ #endif diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index 226b8438aa70..0e1054402c91 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -26,7 +26,6 @@ __again: if (status == 0) return IRQ_RETVAL(handled); handled = 1; - /* snd_printk(KERN_DEBUG "IRQ: status = 0x%x\n", status); */ if (status & 0x02) { STAT_ADD(gus->gf1.interrupt_stat_midi_in); if (gus->gf1.interrupt_handler_midi_in) @@ -50,9 +49,9 @@ __again: continue; /* multi request */ already |= _current_; /* mark request */ #if 0 - printk(KERN_DEBUG "voice = %i, voice_status = 0x%x, " - "voice_verify = %i\n", - voice, voice_status, inb(GUSP(gus, GF1PAGE))); + dev_dbg(gus->card->dev, + "voice = %i, voice_status = 0x%x, voice_verify = %i\n", + voice, voice_status, inb(GUSP(gus, GF1PAGE))); #endif pvoice = &gus->gf1.voices[voice]; if (pvoice->use) { diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 3b46490271fe..51ce405eba7a 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -158,32 +158,32 @@ int snd_gus_create(struct snd_card *card, /* allocate resources */ gus->gf1.res_port1 = request_region(port, 16, "GUS GF1 (Adlib/SB)"); if (!gus->gf1.res_port1) { - snd_printk(KERN_ERR "gus: can't grab SB port 0x%lx\n", port); + dev_err(card->dev, "gus: can't grab SB port 0x%lx\n", port); snd_gus_free(gus); return -EBUSY; } gus->gf1.res_port2 = request_region(port + 0x100, 12, "GUS GF1 (Synth)"); if (!gus->gf1.res_port2) { - snd_printk(KERN_ERR "gus: can't grab synth port 0x%lx\n", port + 0x100); + dev_err(card->dev, "gus: can't grab synth port 0x%lx\n", port + 0x100); snd_gus_free(gus); return -EBUSY; } if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { - snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); + dev_err(card->dev, "gus: can't grab irq %d\n", irq); snd_gus_free(gus); return -EBUSY; } gus->gf1.irq = irq; card->sync_irq = irq; if (request_dma(dma1, "GUS - 1")) { - snd_printk(KERN_ERR "gus: can't grab DMA1 %d\n", dma1); + dev_err(card->dev, "gus: can't grab DMA1 %d\n", dma1); snd_gus_free(gus); return -EBUSY; } gus->gf1.dma1 = dma1; if (dma2 >= 0 && dma1 != dma2) { if (request_dma(dma2, "GUS - 2")) { - snd_printk(KERN_ERR "gus: can't grab DMA2 %d\n", dma2); + dev_err(card->dev, "gus: can't grab DMA2 %d\n", dma2); snd_gus_free(gus); return -EBUSY; } @@ -229,7 +229,9 @@ static int snd_gus_detect_memory(struct snd_gus_card * gus) snd_gf1_poke(gus, 0L, 0xaa); snd_gf1_poke(gus, 1L, 0x55); if (snd_gf1_peek(gus, 0L) != 0xaa || snd_gf1_peek(gus, 1L) != 0x55) { - snd_printk(KERN_ERR "plain GF1 card at 0x%lx without onboard DRAM?\n", gus->gf1.port); + dev_err(gus->card->dev, + "plain GF1 card at 0x%lx without onboard DRAM?\n", + gus->gf1.port); return -ENOMEM; } for (idx = 1, d = 0xab; idx < 4; idx++, d++) { @@ -287,14 +289,14 @@ static int snd_gus_init_dma_irq(struct snd_gus_card * gus, int latches) dma1 |= gus->equal_dma ? 0x40 : (dma2 << 3); if ((dma1 & 7) == 0 || (dma2 & 7) == 0) { - snd_printk(KERN_ERR "Error! DMA isn't defined.\n"); + dev_err(gus->card->dev, "Error! DMA isn't defined.\n"); return -EINVAL; } irq = gus->gf1.irq; irq = abs(irq); irq = irqs[irq & 0x0f]; if (irq == 0) { - snd_printk(KERN_ERR "Error! IRQ isn't defined.\n"); + dev_err(gus->card->dev, "Error! IRQ isn't defined.\n"); return -EINVAL; } irq |= 0x40; @@ -357,7 +359,7 @@ static int snd_gus_check_version(struct snd_gus_card * gus) val = inb(GUSP(gus, REGCNTRLS)); rev = inb(GUSP(gus, BOARDVERSION)); spin_unlock_irqrestore(&gus->reg_lock, flags); - snd_printdd("GF1 [0x%lx] init - val = 0x%x, rev = 0x%x\n", gus->gf1.port, val, rev); + dev_dbg(card->dev, "GF1 [0x%lx] init - val = 0x%x, rev = 0x%x\n", gus->gf1.port, val, rev); strcpy(card->driver, "GUS"); strcpy(card->longname, "Gravis UltraSound Classic (2.4)"); if ((val != 255 && (val & 0x06)) || (rev >= 5 && rev != 255)) { @@ -382,8 +384,11 @@ static int snd_gus_check_version(struct snd_gus_card * gus) strcpy(card->longname, "Gravis UltraSound Extreme"); gus->ess_flag = 1; } else { - snd_printk(KERN_ERR "unknown GF1 revision number at 0x%lx - 0x%x (0x%x)\n", gus->gf1.port, rev, val); - snd_printk(KERN_ERR " please - report to \n"); + dev_err(card->dev, + "unknown GF1 revision number at 0x%lx - 0x%x (0x%x)\n", + gus->gf1.port, rev, val); + dev_err(card->dev, + " please - report to \n"); } } } @@ -400,7 +405,7 @@ int snd_gus_initialize(struct snd_gus_card *gus) if (!gus->interwave) { err = snd_gus_check_version(gus); if (err < 0) { - snd_printk(KERN_ERR "version check failed\n"); + dev_err(gus->card->dev, "version check failed\n"); return err; } err = snd_gus_detect_memory(gus); diff --git a/sound/isa/gus/gus_mem.c b/sound/isa/gus/gus_mem.c index 3e56c01c4544..054058779db6 100644 --- a/sound/isa/gus/gus_mem.c +++ b/sound/isa/gus/gus_mem.c @@ -189,7 +189,7 @@ struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owne if (nblock != NULL) { if (size != (int)nblock->size) { /* TODO: remove in the future */ - snd_printk(KERN_ERR "snd_gf1_mem_alloc - share: sizes differ\n"); + pr_err("%s - share: sizes differ\n", __func__); goto __std; } nblock->share++; diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 850544725da7..bcbcaa924c12 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -67,10 +67,6 @@ static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream, count += offset & 31; offset &= ~31; - /* - snd_printk(KERN_DEBUG "block change - offset = 0x%x, count = 0x%x\n", - offset, count); - */ memset(&block, 0, sizeof(block)); block.cmd = SNDRV_GF1_DMA_IRQ; if (snd_pcm_format_unsigned(runtime->format)) @@ -123,11 +119,6 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream) curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels; end = curr + (pcmp->block_size / runtime->channels); end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1; - /* - snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, " - "ctrl=0x%x, ramp=0x%x, rate=0x%x\n", - curr, begin, end, voice_ctrl, ramp_ctrl, rate); - */ pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8; vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; spin_lock_irqsave(&gus->reg_lock, flags); @@ -178,13 +169,13 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus, unsigned int end, step; if (!pvoice->private_data) { - snd_printd("snd_gf1_pcm: unknown wave irq?\n"); + dev_dbg(gus->card->dev, "%s: unknown wave irq?\n", __func__); snd_gf1_smart_stop_voice(gus, pvoice->number); return; } pcmp = pvoice->private_data; if (pcmp == NULL) { - snd_printd("snd_gf1_pcm: unknown wave irq?\n"); + dev_dbg(gus->card->dev, "%s: unknown wave irq?\n", __func__); snd_gf1_smart_stop_voice(gus, pvoice->number); return; } @@ -197,11 +188,11 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus, ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03; #if 0 snd_gf1_select_voice(gus, pvoice->number); - printk(KERN_DEBUG "position = 0x%x\n", - (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + dev_dbg(gus->card->dev, "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pcmp->pvoices[1]->number); - printk(KERN_DEBUG "position = 0x%x\n", - (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + dev_dbg(gus->card->dev, "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pvoice->number); #endif pcmp->bpos++; @@ -293,11 +284,6 @@ static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf, unsigned int len; unsigned long flags; - /* - printk(KERN_DEBUG - "poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", - (int)buf, pos, count, gus->gf1.port); - */ while (count > 0) { len = count; if (len > 512) /* limit, to allow IRQ */ @@ -673,8 +659,9 @@ static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_gf1_pcm_playback_free; #if 0 - printk(KERN_DEBUG "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", - (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); + dev_dbg(gus->card->dev, + "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", + (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); #endif err = snd_gf1_dma_init(gus); if (err < 0) @@ -695,7 +682,7 @@ static int snd_gf1_pcm_playback_close(struct snd_pcm_substream *substream) struct gus_pcm_private *pcmp = runtime->private_data; if (!wait_event_timeout(pcmp->sleep, (atomic_read(&pcmp->dma_count) <= 0), 2*HZ)) - snd_printk(KERN_ERR "gf1 pcm - serious DMA problem\n"); + dev_err(gus->card->dev, "gf1 pcm - serious DMA problem\n"); snd_gf1_dma_done(gus); return 0; diff --git a/sound/isa/gus/gus_reset.c b/sound/isa/gus/gus_reset.c index 9a1ab5872c4f..ac5da1281042 100644 --- a/sound/isa/gus/gus_reset.c +++ b/sound/isa/gus/gus_reset.c @@ -116,7 +116,9 @@ void snd_gf1_smart_stop_voice(struct snd_gus_card * gus, unsigned short voice) spin_lock_irqsave(&gus->reg_lock, flags); snd_gf1_select_voice(gus, voice); #if 0 - printk(KERN_DEBUG " -%i- smart stop voice - volume = 0x%x\n", voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME)); + dev_dbg(gus->card->dev, + " -%i- smart stop voice - volume = 0x%x\n", + voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME)); #endif snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL); snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); @@ -130,7 +132,9 @@ void snd_gf1_stop_voice(struct snd_gus_card * gus, unsigned short voice) spin_lock_irqsave(&gus->reg_lock, flags); snd_gf1_select_voice(gus, voice); #if 0 - printk(KERN_DEBUG " -%i- stop voice - volume = 0x%x\n", voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME)); + dev_dbg(gus->card->dev, + " -%i- stop voice - volume = 0x%x\n", + voice, snd_gf1_i_read16(gus, SNDRV_GF1_VW_VOLUME)); #endif snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_ADDRESS_CONTROL); snd_gf1_ctrl_stop(gus, SNDRV_GF1_VB_VOLUME_CONTROL); diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index 3975848160e7..08276509447f 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -89,7 +89,9 @@ static int snd_gf1_uart_output_open(struct snd_rawmidi_substream *substream) gus->midi_substream_output = substream; spin_unlock_irqrestore(&gus->uart_cmd_lock, flags); #if 0 - snd_printk(KERN_DEBUG "write init - cmd = 0x%x, stat = 0x%x\n", gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); + dev_dbg(gus->card->dev, + "write init - cmd = 0x%x, stat = 0x%x\n", + gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); #endif return 0; } @@ -111,18 +113,17 @@ static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream) for (i = 0; i < 1000 && (snd_gf1_uart_stat(gus) & 0x01); i++) snd_gf1_uart_get(gus); /* clean Rx */ if (i >= 1000) - snd_printk(KERN_ERR "gus midi uart init read - cleanup error\n"); + dev_err(gus->card->dev, "gus midi uart init read - cleanup error\n"); } spin_unlock_irqrestore(&gus->uart_cmd_lock, flags); #if 0 - snd_printk(KERN_DEBUG - "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", - gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); - snd_printk(KERN_DEBUG - "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x " - "(page = 0x%x)\n", - gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), - inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); + dev_dbg(gus->card->dev, + "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", + gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); + dev_dbg(gus->card->dev, + "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", + gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), + inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); #endif return 0; } diff --git a/sound/isa/gus/gus_volume.c b/sound/isa/gus/gus_volume.c index ed72196a361b..e729621756cf 100644 --- a/sound/isa/gus/gus_volume.c +++ b/sound/isa/gus/gus_volume.c @@ -104,7 +104,8 @@ unsigned short snd_gf1_translate_freq(struct snd_gus_card * gus, unsigned int fr freq16 = 50; if (freq16 & 0xf8000000) { freq16 = ~0xf8000000; - snd_printk(KERN_ERR "snd_gf1_translate_freq: overflow - freq = 0x%x\n", freq16); + dev_err(gus->card->dev, "%s: overflow - freq = 0x%x\n", + __func__, freq16); } return ((freq16 << 9) + (gus->gf1.playback_freq >> 1)) / gus->gf1.playback_freq; } @@ -189,14 +190,14 @@ unsigned short snd_gf1_compute_freq(unsigned int freq, fc = (freq << 10) / rate; if (fc > 97391L) { fc = 97391; - snd_printk(KERN_ERR "patch: (1) fc frequency overflow - %u\n", fc); + pr_err("patch: (1) fc frequency overflow - %u\n", fc); } fc = (fc * 44100UL) / mix_rate; while (scale--) fc <<= 1; if (fc > 65535L) { fc = 65535; - snd_printk(KERN_ERR "patch: (2) fc frequency overflow - %u\n", fc); + pr_err("patch: (2) fc frequency overflow - %u\n", fc); } return (unsigned short) fc; } diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c index 09cc53ceea2a..101202acefb3 100644 --- a/sound/isa/gus/gusclassic.c +++ b/sound/isa/gus/gusclassic.c @@ -115,7 +115,7 @@ static int snd_gusclassic_detect(struct snd_gus_card *gus) snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */ d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 0) { - snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } udelay(160); @@ -123,7 +123,7 @@ static int snd_gusclassic_detect(struct snd_gus_card *gus) udelay(160); d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 1) { - snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } return 0; diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 63d9f2d75df0..6eab95bd49c1 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -179,7 +179,7 @@ static int snd_gusextreme_detect(struct snd_gus_card *gus, snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */ d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 0) { - snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); return -EIO; } udelay(160); @@ -187,7 +187,7 @@ static int snd_gusextreme_detect(struct snd_gus_card *gus, udelay(160); d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 1) { - snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); return -EIO; } diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 6834c0560064..445fd2fb50f1 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -64,8 +64,6 @@ struct snd_gusmax { unsigned short pcm_status_reg; }; -#define PFX "gusmax: " - static int snd_gusmax_detect(struct snd_gus_card *gus) { unsigned char d; @@ -73,7 +71,7 @@ static int snd_gusmax_detect(struct snd_gus_card *gus) snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */ d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 0) { - snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } udelay(160); @@ -81,7 +79,7 @@ static int snd_gusmax_detect(struct snd_gus_card *gus) udelay(160); d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 1) { - snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } @@ -206,7 +204,7 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + dev_err(pdev, "unable to find a free IRQ\n"); return -EBUSY; } } @@ -214,7 +212,7 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) if (xdma1 == SNDRV_AUTO_DMA) { xdma1 = snd_legacy_find_free_dma(possible_dmas); if (xdma1 < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA1\n"); + dev_err(pdev, "unable to find a free DMA1\n"); return -EBUSY; } } @@ -222,7 +220,7 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) if (xdma2 == SNDRV_AUTO_DMA) { xdma2 = snd_legacy_find_free_dma(possible_dmas); if (xdma2 < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA2\n"); + dev_err(pdev, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -267,13 +265,13 @@ static int snd_gusmax_probe(struct device *pdev, unsigned int dev) return err; if (!gus->max_flag) { - snd_printk(KERN_ERR PFX "GUS MAX soundcard was not detected at 0x%lx\n", gus->gf1.port); + dev_err(pdev, "GUS MAX soundcard was not detected at 0x%lx\n", gus->gf1.port); return -ENODEV; } if (devm_request_irq(card->dev, xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) { - snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); + dev_err(pdev, "unable to grab IRQ %d\n", xirq); return -EBUSY; } maxcard->irq = xirq; diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index a04a9d3253f8..18a98123e286 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -52,11 +52,9 @@ static int pcm_channels[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; static int effect[SNDRV_CARDS]; #ifdef SNDRV_STB -#define PFX "interwave-stb: " #define INTERWAVE_DRIVER "snd_interwave_stb" #define INTERWAVE_PNP_DRIVER "interwave-stb" #else -#define PFX "interwave: " #define INTERWAVE_DRIVER "snd_interwave" #define INTERWAVE_PNP_DRIVER "interwave" #endif @@ -148,7 +146,7 @@ static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int da unsigned long port = bus->private_value; #if 0 - printk(KERN_DEBUG "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); + dev_dbg(bus->card->dev, "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); #endif outb((data << 1) | ctrl, port); udelay(10); @@ -161,7 +159,7 @@ static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus) res = inb(port) & 1; #if 0 - printk(KERN_DEBUG "i2c_getclockline - 0x%lx -> %i\n", port, res); + dev_dbg(bus->card->dev, "i2c_getclockline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -175,7 +173,7 @@ static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack) udelay(10); res = (inb(port) & 2) >> 1; #if 0 - printk(KERN_DEBUG "i2c_getdataline - 0x%lx -> %i\n", port, res); + dev_dbg(bus->card->dev, "i2c_getdataline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -215,7 +213,7 @@ static int snd_interwave_detect_stb(struct snd_interwave *iwcard, "InterWave (I2C bus)"); } if (iwcard->i2c_res == NULL) { - snd_printk(KERN_ERR "interwave: can't grab i2c bus port\n"); + dev_err(card->dev, "interwave: can't grab i2c bus port\n"); return -ENODEV; } @@ -248,7 +246,7 @@ static int snd_interwave_detect(struct snd_interwave *iwcard, snd_gf1_i_write8(gus, SNDRV_GF1_GB_RESET, 0); /* reset GF1 */ d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 0) { - snd_printdd("[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 1 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } udelay(160); @@ -256,7 +254,7 @@ static int snd_interwave_detect(struct snd_interwave *iwcard, udelay(160); d = snd_gf1_i_look8(gus, SNDRV_GF1_GB_RESET); if ((d & 0x07) != 1) { - snd_printdd("[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); + dev_dbg(gus->card->dev, "[0x%lx] check 2 failed - 0x%x\n", gus->gf1.port, d); return -ENODEV; } spin_lock_irqsave(&gus->reg_lock, flags); @@ -265,10 +263,13 @@ static int snd_interwave_detect(struct snd_interwave *iwcard, rev2 = snd_gf1_look8(gus, SNDRV_GF1_GB_VERSION_NUMBER); snd_gf1_write8(gus, SNDRV_GF1_GB_VERSION_NUMBER, rev1); spin_unlock_irqrestore(&gus->reg_lock, flags); - snd_printdd("[0x%lx] InterWave check - rev1=0x%x, rev2=0x%x\n", gus->gf1.port, rev1, rev2); + dev_dbg(gus->card->dev, + "[0x%lx] InterWave check - rev1=0x%x, rev2=0x%x\n", + gus->gf1.port, rev1, rev2); if ((rev1 & 0xf0) == (rev2 & 0xf0) && (rev1 & 0x0f) != (rev2 & 0x0f)) { - snd_printdd("[0x%lx] InterWave check - passed\n", gus->gf1.port); + dev_dbg(gus->card->dev, + "[0x%lx] InterWave check - passed\n", gus->gf1.port); gus->interwave = 1; strcpy(gus->card->shortname, "AMD InterWave"); gus->revision = rev1 >> 4; @@ -278,7 +279,7 @@ static int snd_interwave_detect(struct snd_interwave *iwcard, return snd_interwave_detect_stb(iwcard, gus, dev, rbus); #endif } - snd_printdd("[0x%lx] InterWave check - failed\n", gus->gf1.port); + dev_dbg(gus->card->dev, "[0x%lx] InterWave check - failed\n", gus->gf1.port); return -ENODEV; } @@ -327,7 +328,7 @@ static void snd_interwave_bank_sizes(struct snd_gus_card *gus, int *sizes) snd_gf1_poke(gus, local, d); snd_gf1_poke(gus, local + 1, d + 1); #if 0 - printk(KERN_DEBUG "d = 0x%x, local = 0x%x, " + dev_dbg(gus->card->dev, "d = 0x%x, local = 0x%x, " "local + 1 = 0x%x, idx << 22 = 0x%x\n", d, snd_gf1_peek(gus, local), @@ -342,7 +343,7 @@ static void snd_interwave_bank_sizes(struct snd_gus_card *gus, int *sizes) } } #if 0 - printk(KERN_DEBUG "sizes: %i %i %i %i\n", + dev_dbg(gus->card->dev, "sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]); #endif } @@ -397,12 +398,12 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) lmct = (psizes[3] << 24) | (psizes[2] << 16) | (psizes[1] << 8) | psizes[0]; #if 0 - printk(KERN_DEBUG "lmct = 0x%08x\n", lmct); + dev_dbg(gus->card->dev, "lmct = 0x%08x\n", lmct); #endif for (i = 0; i < ARRAY_SIZE(lmc); i++) if (lmct == lmc[i]) { #if 0 - printk(KERN_DEBUG "found !!! %i\n", i); + dev_dbg(gus->card->dev, "found !!! %i\n", i); #endif snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i); snd_interwave_bank_sizes(gus, psizes); @@ -566,12 +567,12 @@ static int snd_interwave_pnp(int dev, struct snd_interwave *iwcard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "InterWave PnP configure failure (out of resources?)\n"); + dev_err(&pdev->dev, "InterWave PnP configure failure (out of resources?)\n"); return err; } if (pnp_port_start(pdev, 0) + 0x100 != pnp_port_start(pdev, 1) || pnp_port_start(pdev, 0) + 0x10c != pnp_port_start(pdev, 2)) { - snd_printk(KERN_ERR "PnP configure failure (wrong ports)\n"); + dev_err(&pdev->dev, "PnP configure failure (wrong ports)\n"); return -ENOENT; } port[dev] = pnp_port_start(pdev, 0); @@ -579,22 +580,26 @@ static int snd_interwave_pnp(int dev, struct snd_interwave *iwcard, if (dma2[dev] >= 0) dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - snd_printdd("isapnp IW: sb port=0x%llx, gf1 port=0x%llx, codec port=0x%llx\n", - (unsigned long long)pnp_port_start(pdev, 0), - (unsigned long long)pnp_port_start(pdev, 1), - (unsigned long long)pnp_port_start(pdev, 2)); - snd_printdd("isapnp IW: dma1=%i, dma2=%i, irq=%i\n", dma1[dev], dma2[dev], irq[dev]); + dev_dbg(&pdev->dev, + "isapnp IW: sb port=0x%llx, gf1 port=0x%llx, codec port=0x%llx\n", + (unsigned long long)pnp_port_start(pdev, 0), + (unsigned long long)pnp_port_start(pdev, 1), + (unsigned long long)pnp_port_start(pdev, 2)); + dev_dbg(&pdev->dev, + "isapnp IW: dma1=%i, dma2=%i, irq=%i\n", + dma1[dev], dma2[dev], irq[dev]); #ifdef SNDRV_STB /* Tone Control initialization */ pdev = iwcard->devtc; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "InterWave ToneControl PnP configure failure (out of resources?)\n"); + dev_err(&pdev->dev, + "InterWave ToneControl PnP configure failure (out of resources?)\n"); return err; } port_tc[dev] = pnp_port_start(pdev, 0); - snd_printdd("isapnp IW: tone control port=0x%lx\n", port_tc[dev]); + dev_dbg(&pdev->dev, "isapnp IW: tone control port=0x%lx\n", port_tc[dev]); #endif return 0; } @@ -660,7 +665,7 @@ static int snd_interwave_probe(struct snd_card *card, int dev, if (devm_request_irq(card->dev, xirq, snd_interwave_interrupt, 0, "InterWave", iwcard)) { - snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); + dev_err(card->dev, "unable to grab IRQ %d\n", xirq); return -EBUSY; } iwcard->irq = xirq; @@ -780,21 +785,21 @@ static int snd_interwave_isa_probe(struct device *pdev, if (irq[dev] == SNDRV_AUTO_IRQ) { irq[dev] = snd_legacy_find_free_irq(possible_irqs); if (irq[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + dev_err(pdev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma1[dev] == SNDRV_AUTO_DMA) { dma1[dev] = snd_legacy_find_free_dma(possible_dmas); if (dma1[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA1\n"); + dev_err(pdev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2[dev] == SNDRV_AUTO_DMA) { dma2[dev] = snd_legacy_find_free_dma(possible_dmas); if (dma2[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA2\n"); + dev_err(pdev, "unable to find a free DMA2\n"); return -EBUSY; } } From b48601834da44698f56569f6660fa7dac17a6f12 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:17 +0200 Subject: [PATCH 132/410] ALSA: msnd: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-28-tiwai@suse.de --- sound/isa/msnd/msnd.c | 46 ++++----- sound/isa/msnd/msnd_midi.c | 4 - sound/isa/msnd/msnd_pinnacle.c | 184 +++++++++++++++++---------------- 3 files changed, 112 insertions(+), 122 deletions(-) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index c3fd1eb301bb..69c515421dd8 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -86,7 +86,7 @@ int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) } spin_unlock_irqrestore(&dev->lock, flags); - snd_printd(KERN_ERR LOGNAME ": Send DSP command timeout\n"); + dev_dbg(dev->card->dev, LOGNAME ": Send DSP command timeout\n"); return -EIO; } @@ -104,7 +104,7 @@ int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, return 0; } - snd_printd(KERN_ERR LOGNAME ": Send host word timeout\n"); + dev_dbg(dev->card->dev, LOGNAME ": Send host word timeout\n"); return -EIO; } @@ -115,7 +115,7 @@ int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) int i; if (len % 3 != 0) { - snd_printk(KERN_ERR LOGNAME + dev_err(dev->card->dev, LOGNAME ": Upload host data not multiple of 3!\n"); return -EINVAL; } @@ -138,7 +138,7 @@ int snd_msnd_enable_irq(struct snd_msnd *dev) if (dev->irq_ref++) return 0; - snd_printdd(LOGNAME ": Enabling IRQ\n"); + dev_dbg(dev->card->dev, LOGNAME ": Enabling IRQ\n"); spin_lock_irqsave(&dev->lock, flags); if (snd_msnd_wait_TXDE(dev) == 0) { @@ -156,7 +156,7 @@ int snd_msnd_enable_irq(struct snd_msnd *dev) } spin_unlock_irqrestore(&dev->lock, flags); - snd_printd(KERN_ERR LOGNAME ": Enable IRQ failed\n"); + dev_dbg(dev->card->dev, LOGNAME ": Enable IRQ failed\n"); return -EIO; } @@ -170,10 +170,10 @@ int snd_msnd_disable_irq(struct snd_msnd *dev) return 0; if (dev->irq_ref < 0) - snd_printd(KERN_WARNING LOGNAME ": IRQ ref count is %d\n", - dev->irq_ref); + dev_dbg(dev->card->dev, LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); - snd_printdd(LOGNAME ": Disabling IRQ\n"); + dev_dbg(dev->card->dev, LOGNAME ": Disabling IRQ\n"); spin_lock_irqsave(&dev->lock, flags); if (snd_msnd_wait_TXDE(dev) == 0) { @@ -186,7 +186,7 @@ int snd_msnd_disable_irq(struct snd_msnd *dev) } spin_unlock_irqrestore(&dev->lock, flags); - snd_printd(KERN_ERR LOGNAME ": Disable IRQ failed\n"); + dev_dbg(dev->card->dev, LOGNAME ": Disable IRQ failed\n"); return -EIO; } @@ -220,8 +220,8 @@ void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file) snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); snd_msnd_disable_irq(chip); if (file) { - snd_printd(KERN_INFO LOGNAME - ": Stopping read for %p\n", file); + dev_dbg(chip->card->dev, LOGNAME + ": Stopping read for %p\n", file); chip->mode &= ~FMODE_READ; } clear_bit(F_AUDIO_READ_INUSE, &chip->flags); @@ -233,8 +233,8 @@ void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file) } snd_msnd_disable_irq(chip); if (file) { - snd_printd(KERN_INFO - LOGNAME ": Stopping write for %p\n", file); + dev_dbg(chip->card->dev, + LOGNAME ": Stopping write for %p\n", file); chip->mode &= ~FMODE_WRITE; } clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); @@ -329,12 +329,6 @@ int snd_msnd_DAPQ(struct snd_msnd *chip, int start) ++nbanks; /* Then advance the tail */ - /* - if (protect) - snd_printd(KERN_INFO "B %X %lX\n", - bank_num, xtime.tv_usec); - */ - DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); writew(DAPQ_tail, chip->DAPQ + JQS_wTail); /* Tell the DSP to play the bank */ @@ -343,10 +337,6 @@ int snd_msnd_DAPQ(struct snd_msnd *chip, int start) if (2 == bank_num) break; } - /* - if (protect) - snd_printd(KERN_INFO "%lX\n", xtime.tv_usec); - */ /* spin_unlock_irqrestore(&chip->lock, flags); not necessary */ return nbanks; } @@ -406,7 +396,7 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, #endif chip->capturePeriodBytes = pcm_count; - snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + dev_dbg(chip->card->dev, "%s() %i\n", __func__, pcm_count); pDAQ = chip->mappedbase + DARQ_DATA_BUFF; @@ -533,21 +523,21 @@ static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, int result = 0; if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_printdd("snd_msnd_playback_trigger(START)\n"); + dev_dbg(chip->card->dev, "%s(START)\n", __func__); chip->banksPlayed = 0; set_bit(F_WRITING, &chip->flags); snd_msnd_DAPQ(chip, 1); } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { - snd_printdd("snd_msnd_playback_trigger(STop)\n"); + dev_dbg(chip->card->dev, "%s(STOP)\n", __func__); /* interrupt diagnostic, comment this out later */ clear_bit(F_WRITING, &chip->flags); snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); } else { - snd_printd(KERN_ERR "snd_msnd_playback_trigger(?????)\n"); + dev_dbg(chip->card->dev, "%s(?????)\n", __func__); result = -EINVAL; } - snd_printdd("snd_msnd_playback_trigger() ENDE\n"); + dev_dbg(chip->card->dev, "%s() ENDE\n", __func__); return result; } diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 7c61caaf99ad..3ffc8758bec2 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -42,8 +42,6 @@ static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) { struct snd_msndmidi *mpu; - snd_printdd("snd_msndmidi_input_open()\n"); - mpu = substream->rmidi->private_data; mpu->substream_input = substream; @@ -84,8 +82,6 @@ static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, unsigned long flags; struct snd_msndmidi *mpu; - snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up); - mpu = substream->rmidi->private_data; spin_lock_irqsave(&mpu->input_lock, flags); if (up) { diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 4433a92f08e7..635403301a15 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -81,11 +81,12 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) switch (HIBYTE(wMessage)) { case HIMT_PLAY_DONE: { if (chip->banksPlayed < 3) - snd_printdd("%08X: HIMT_PLAY_DONE: %i\n", + dev_dbg(chip->card->dev, "%08X: HIMT_PLAY_DONE: %i\n", (unsigned)jiffies, LOBYTE(wMessage)); if (chip->last_playbank == LOBYTE(wMessage)) { - snd_printdd("chip.last_playbank == LOBYTE(wMessage)\n"); + dev_dbg(chip->card->dev, + "chip.last_playbank == LOBYTE(wMessage)\n"); break; } chip->banksPlayed++; @@ -121,21 +122,22 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) case HIDSP_PLAY_UNDER: #endif case HIDSP_INT_PLAY_UNDER: - snd_printd(KERN_WARNING LOGNAME ": Play underflow %i\n", + dev_dbg(chip->card->dev, + LOGNAME ": Play underflow %i\n", chip->banksPlayed); if (chip->banksPlayed > 2) clear_bit(F_WRITING, &chip->flags); break; case HIDSP_INT_RECORD_OVER: - snd_printd(KERN_WARNING LOGNAME ": Record overflow\n"); + dev_dbg(chip->card->dev, LOGNAME ": Record overflow\n"); clear_bit(F_READING, &chip->flags); break; default: - snd_printd(KERN_WARNING LOGNAME - ": DSP message %d 0x%02x\n", - LOBYTE(wMessage), LOBYTE(wMessage)); + dev_dbg(chip->card->dev, LOGNAME + ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); break; } break; @@ -146,8 +148,8 @@ static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) break; default: - snd_printd(KERN_WARNING LOGNAME ": HIMT message %d 0x%02x\n", - HIBYTE(wMessage), HIBYTE(wMessage)); + dev_dbg(chip->card->dev, LOGNAME ": HIMT message %d 0x%02x\n", + HIBYTE(wMessage), HIBYTE(wMessage)); break; } } @@ -180,8 +182,9 @@ static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) } -static int snd_msnd_reset_dsp(long io, unsigned char *info) +static int snd_msnd_reset_dsp(struct snd_msnd *chip, unsigned char *info) { + long io = chip->io; int timeout = 100; outb(HPDSPRESET_ON, io + HP_DSPR); @@ -197,7 +200,7 @@ static int snd_msnd_reset_dsp(long io, unsigned char *info) return 0; msleep(1); } - snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + dev_err(chip->card->dev, LOGNAME ": Cannot reset DSP\n"); return -EIO; } @@ -213,11 +216,11 @@ static int snd_msnd_probe(struct snd_card *card) #endif if (!request_region(chip->io, DSP_NUMIO, "probing")) { - snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + dev_err(card->dev, LOGNAME ": I/O port conflict\n"); return -ENODEV; } - if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + if (snd_msnd_reset_dsp(chip, &info) < 0) { release_region(chip->io, DSP_NUMIO); return -ENODEV; } @@ -225,7 +228,7 @@ static int snd_msnd_probe(struct snd_card *card) #ifdef MSND_CLASSIC strcpy(card->shortname, "Classic/Tahiti/Monterey"); strcpy(card->longname, "Turtle Beach Multisound"); - printk(KERN_INFO LOGNAME ": %s, " + dev_info(card->dev, LOGNAME ": %s, " "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", card->shortname, chip->io, chip->io + DSP_NUMIO - 1, @@ -285,7 +288,7 @@ static int snd_msnd_probe(struct snd_card *card) break; } strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); - printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " + dev_info(card->dev, LOGNAME ": %s revision %s, Xilinx version %s, " "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", card->shortname, rev, xv, @@ -377,22 +380,22 @@ static int upload_dsp_code(struct snd_card *card) err = request_firmware(&init_fw, INITCODEFILE, card->dev); if (err < 0) { - printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + dev_err(card->dev, LOGNAME ": Error loading " INITCODEFILE); goto cleanup1; } err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); if (err < 0) { - printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + dev_err(card->dev, LOGNAME ": Error loading " PERMCODEFILE); goto cleanup; } memcpy_toio(chip->mappedbase, perm_fw->data, perm_fw->size); if (snd_msnd_upload_host(chip, init_fw->data, init_fw->size) < 0) { - printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + dev_warn(card->dev, LOGNAME ": Error uploading to DSP\n"); err = -ENODEV; goto cleanup; } - printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); + dev_info(card->dev, LOGNAME ": DSP firmware uploaded\n"); err = 0; cleanup: @@ -425,17 +428,17 @@ static int snd_msnd_initialize(struct snd_card *card) #endif err = snd_msnd_init_sma(chip); if (err < 0) { - printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + dev_warn(card->dev, LOGNAME ": Cannot initialize SMA\n"); return err; } - err = snd_msnd_reset_dsp(chip->io, NULL); + err = snd_msnd_reset_dsp(chip, NULL); if (err < 0) return err; err = upload_dsp_code(card); if (err < 0) { - printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + dev_warn(card->dev, LOGNAME ": Cannot upload DSP code\n"); return err; } @@ -444,7 +447,7 @@ static int snd_msnd_initialize(struct snd_card *card) while (readw(chip->mappedbase)) { msleep(1); if (!timeout--) { - snd_printd(KERN_ERR LOGNAME ": DSP reset timeout\n"); + dev_err(card->dev, LOGNAME ": DSP reset timeout\n"); return -EIO; } } @@ -466,7 +469,7 @@ static int snd_msnd_dsp_full_reset(struct snd_card *card) rv = snd_msnd_initialize(card); if (rv) - printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + dev_warn(card->dev, LOGNAME ": DSP reset failed\n"); snd_msndmix_force_recsrc(chip, 0); clear_bit(F_RESETTING, &chip->flags); return rv; @@ -483,7 +486,7 @@ static int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) static int snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) { - snd_printdd("snd_msnd_calibrate_adc(%i)\n", srate); + dev_dbg(chip->card->dev, "snd_msnd_calibrate_adc(%i)\n", srate); writew(srate, chip->SMA + SMA_wCalFreqAtoD); if (chip->calibrate_signal == 0) writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) @@ -496,7 +499,7 @@ static int snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) schedule_timeout_interruptible(msecs_to_jiffies(333)); return 0; } - printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + dev_warn(chip->card->dev, LOGNAME ": ADC calibration failed\n"); return -EIO; } @@ -527,7 +530,7 @@ static int snd_msnd_attach(struct snd_card *card) err = devm_request_irq(card->dev, chip->irq, snd_msnd_interrupt, 0, card->shortname, chip); if (err < 0) { - printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + dev_err(card->dev, LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); return err; } card->sync_irq = chip->irq; @@ -537,14 +540,14 @@ static int snd_msnd_attach(struct snd_card *card) if (!devm_request_mem_region(card->dev, chip->base, BUFFSIZE, card->shortname)) { - printk(KERN_ERR LOGNAME + dev_err(card->dev, LOGNAME ": unable to grab memory region 0x%lx-0x%lx\n", chip->base, chip->base + BUFFSIZE - 1); return -EBUSY; } chip->mappedbase = devm_ioremap(card->dev, chip->base, 0x8000); if (!chip->mappedbase) { - printk(KERN_ERR LOGNAME + dev_err(card->dev, LOGNAME ": unable to map memory region 0x%lx-0x%lx\n", chip->base, chip->base + BUFFSIZE - 1); return -EIO; @@ -556,13 +559,13 @@ static int snd_msnd_attach(struct snd_card *card) err = snd_msnd_pcm(card, 0); if (err < 0) { - printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + dev_err(card->dev, LOGNAME ": error creating new PCM device\n"); return err; } err = snd_msndmix_new(card); if (err < 0) { - printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + dev_err(card->dev, LOGNAME ": error creating new Mixer device\n"); return err; } @@ -577,7 +580,7 @@ static int snd_msnd_attach(struct snd_card *card) mpu_irq[0], &chip->rmidi); if (err < 0) { - printk(KERN_ERR LOGNAME + dev_err(card->dev, LOGNAME ": error creating new Midi device\n"); return err; } @@ -604,103 +607,104 @@ static int snd_msnd_attach(struct snd_card *card) /* Pinnacle/Fiji Logical Device Configuration */ -static int snd_msnd_write_cfg(int cfg, int reg, int value) +static int snd_msnd_write_cfg(struct snd_msnd *chip, int cfg, int reg, int value) { outb(reg, cfg); outb(value, cfg + 1); if (value != inb(cfg + 1)) { - printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + dev_err(chip->card->dev, LOGNAME ": %s: I/O error\n", __func__); return -EIO; } return 0; } -static int snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +static int snd_msnd_write_cfg_io0(struct snd_msnd *chip, int cfg, int num, u16 io) { - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + if (snd_msnd_write_cfg(chip, cfg, IREG_IO0_BASEHI, HIBYTE(io))) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + if (snd_msnd_write_cfg(chip, cfg, IREG_IO0_BASELO, LOBYTE(io))) return -EIO; return 0; } -static int snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +static int snd_msnd_write_cfg_io1(struct snd_msnd *chip, int cfg, int num, u16 io) { - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + if (snd_msnd_write_cfg(chip, cfg, IREG_IO1_BASEHI, HIBYTE(io))) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + if (snd_msnd_write_cfg(chip, cfg, IREG_IO1_BASELO, LOBYTE(io))) return -EIO; return 0; } -static int snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +static int snd_msnd_write_cfg_irq(struct snd_msnd *chip, int cfg, int num, u16 irq) { - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + if (snd_msnd_write_cfg(chip, cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + if (snd_msnd_write_cfg(chip, cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) return -EIO; return 0; } -static int snd_msnd_write_cfg_mem(int cfg, int num, int mem) +static int snd_msnd_write_cfg_mem(struct snd_msnd *chip, int cfg, int num, int mem) { u16 wmem; mem >>= 8; wmem = (u16)(mem & 0xfff); - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + if (snd_msnd_write_cfg(chip, cfg, IREG_MEMBASEHI, HIBYTE(wmem))) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + if (snd_msnd_write_cfg(chip, cfg, IREG_MEMBASELO, LOBYTE(wmem))) return -EIO; - if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, + if (wmem && snd_msnd_write_cfg(chip, cfg, IREG_MEMCONTROL, MEMTYPE_HIADDR | MEMTYPE_16BIT)) return -EIO; return 0; } -static int snd_msnd_activate_logical(int cfg, int num) +static int snd_msnd_activate_logical(struct snd_msnd *chip, int cfg, int num) { - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + if (snd_msnd_write_cfg(chip, cfg, IREG_ACTIVATE, LD_ACTIVATE)) return -EIO; return 0; } -static int snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, +static int snd_msnd_write_cfg_logical(struct snd_msnd *chip, + int cfg, int num, u16 io0, u16 io1, u16 irq, int mem) { - if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + if (snd_msnd_write_cfg(chip, cfg, IREG_LOGDEVICE, num)) return -EIO; - if (snd_msnd_write_cfg_io0(cfg, num, io0)) + if (snd_msnd_write_cfg_io0(chip, cfg, num, io0)) return -EIO; - if (snd_msnd_write_cfg_io1(cfg, num, io1)) + if (snd_msnd_write_cfg_io1(chip, cfg, num, io1)) return -EIO; - if (snd_msnd_write_cfg_irq(cfg, num, irq)) + if (snd_msnd_write_cfg_irq(chip, cfg, num, irq)) return -EIO; - if (snd_msnd_write_cfg_mem(cfg, num, mem)) + if (snd_msnd_write_cfg_mem(chip, cfg, num, mem)) return -EIO; - if (snd_msnd_activate_logical(cfg, num)) + if (snd_msnd_activate_logical(chip, cfg, num)) return -EIO; return 0; } -static int snd_msnd_pinnacle_cfg_reset(int cfg) +static int snd_msnd_pinnacle_cfg_reset(struct snd_msnd *chip, int cfg) { int i; /* Reset devices if told to */ - printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + dev_info(chip->card->dev, LOGNAME ": Resetting all devices\n"); for (i = 0; i < 4; ++i) - if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + if (snd_msnd_write_cfg_logical(chip, cfg, i, 0, 0, 0, 0)) return -EIO; return 0; @@ -779,7 +783,7 @@ static int snd_msnd_isa_match(struct device *pdev, unsigned int i) return 0; if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { - printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + dev_warn(pdev, LOGNAME ": io, irq and mem must be set\n"); return 0; } @@ -792,14 +796,14 @@ static int snd_msnd_isa_match(struct device *pdev, unsigned int i) io[i] == 0x220 || io[i] == 0x210 || io[i] == 0x3e0)) { - printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set " + dev_err(pdev, LOGNAME ": \"io\" - DSP I/O base must be set " " to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, " "or 0x3E0\n"); return 0; } #else if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { - printk(KERN_ERR LOGNAME + dev_err(pdev, LOGNAME ": \"io\" - DSP I/O base must within the range 0x100 " "to 0x3E0 and must be evenly divisible by 0x10\n"); return 0; @@ -812,7 +816,7 @@ static int snd_msnd_isa_match(struct device *pdev, unsigned int i) irq[i] == 10 || irq[i] == 11 || irq[i] == 12)) { - printk(KERN_ERR LOGNAME + dev_err(pdev, LOGNAME ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); return 0; } @@ -823,7 +827,7 @@ static int snd_msnd_isa_match(struct device *pdev, unsigned int i) mem[i] == 0xd8000 || mem[i] == 0xe0000 || mem[i] == 0xe8000)) { - printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + dev_err(pdev, LOGNAME ": \"mem\" - must be set to " "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or " "0xe8000\n"); return 0; @@ -831,9 +835,9 @@ static int snd_msnd_isa_match(struct device *pdev, unsigned int i) #ifndef MSND_CLASSIC if (cfg[i] == SNDRV_AUTO_PORT) { - printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + dev_info(pdev, LOGNAME ": Assuming PnP mode\n"); } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { - printk(KERN_INFO LOGNAME + dev_info(pdev, LOGNAME ": Config port must be 0x250, 0x260 or 0x270 " "(or unspecified for PnP mode)\n"); return 0; @@ -854,7 +858,7 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) || cfg[idx] == SNDRV_AUTO_PORT #endif ) { - printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + dev_info(pdev, LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } @@ -897,21 +901,21 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) chip->memid = HPMEM_E800; break; } #else - printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", - cfg[idx]); + dev_info(pdev, LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", + cfg[idx]); if (!devm_request_region(card->dev, cfg[idx], 2, "Pinnacle/Fiji Config")) { - printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", - cfg[idx]); + dev_err(pdev, LOGNAME ": Config port 0x%lx conflict\n", + cfg[idx]); return -EIO; } if (reset[idx]) - if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) + if (snd_msnd_pinnacle_cfg_reset(chip, cfg[idx])) return -EIO; /* DSP */ - err = snd_msnd_write_cfg_logical(cfg[idx], 0, + err = snd_msnd_write_cfg_logical(chip, cfg[idx], 0, io[idx], 0, irq[idx], mem[idx]); @@ -923,10 +927,10 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) /* MPU */ if (mpu_io[idx] != SNDRV_AUTO_PORT && mpu_irq[idx] != SNDRV_AUTO_IRQ) { - printk(KERN_INFO LOGNAME + dev_info(pdev, LOGNAME ": Configuring MPU to I/O 0x%lx IRQ %d\n", mpu_io[idx], mpu_irq[idx]); - err = snd_msnd_write_cfg_logical(cfg[idx], 1, + err = snd_msnd_write_cfg_logical(chip, cfg[idx], 1, mpu_io[idx], 0, mpu_irq[idx], 0); @@ -938,10 +942,10 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) if (ide_io0[idx] != SNDRV_AUTO_PORT && ide_io1[idx] != SNDRV_AUTO_PORT && ide_irq[idx] != SNDRV_AUTO_IRQ) { - printk(KERN_INFO LOGNAME + dev_info(pdev, LOGNAME ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", ide_io0[idx], ide_io1[idx], ide_irq[idx]); - err = snd_msnd_write_cfg_logical(cfg[idx], 2, + err = snd_msnd_write_cfg_logical(chip, cfg[idx], 2, ide_io0[idx], ide_io1[idx], ide_irq[idx], 0); @@ -951,10 +955,10 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) /* Joystick */ if (joystick_io[idx] != SNDRV_AUTO_PORT) { - printk(KERN_INFO LOGNAME + dev_info(pdev, LOGNAME ": Configuring joystick to I/O 0x%lx\n", joystick_io[idx]); - err = snd_msnd_write_cfg_logical(cfg[idx], 3, + err = snd_msnd_write_cfg_logical(chip, cfg[idx], 3, joystick_io[idx], 0, 0, 0); @@ -989,13 +993,13 @@ static int snd_msnd_isa_probe(struct device *pdev, unsigned int idx) spin_lock_init(&chip->lock); err = snd_msnd_probe(card); if (err < 0) { - printk(KERN_ERR LOGNAME ": Probe failed\n"); + dev_err(pdev, LOGNAME ": Probe failed\n"); return err; } err = snd_msnd_attach(card); if (err < 0) { - printk(KERN_ERR LOGNAME ": Attach failed\n"); + dev_err(pdev, LOGNAME ": Attach failed\n"); return err; } dev_set_drvdata(pdev, card); @@ -1042,12 +1046,12 @@ static int snd_msnd_pnp_detect(struct pnp_card_link *pcard, return -ENODEV; if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { - printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + dev_info(&pcard->card->dev, "msnd_pinnacle: device is inactive\n"); return -EBUSY; } if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { - printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + dev_info(&pcard->card->dev, "msnd_pinnacle: MPU device is inactive\n"); return -EBUSY; } @@ -1098,13 +1102,13 @@ static int snd_msnd_pnp_detect(struct pnp_card_link *pcard, spin_lock_init(&chip->lock); ret = snd_msnd_probe(card); if (ret < 0) { - printk(KERN_ERR LOGNAME ": Probe failed\n"); + dev_err(&pcard->card->dev, LOGNAME ": Probe failed\n"); return ret; } ret = snd_msnd_attach(card); if (ret < 0) { - printk(KERN_ERR LOGNAME ": Attach failed\n"); + dev_err(&pcard->card->dev, LOGNAME ": Attach failed\n"); return ret; } From 764a55bb8d41e7d4fc0686898c32e9a5b56493db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:18 +0200 Subject: [PATCH 133/410] ALSA: opl3sa2: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The card pointer is stored in struct snd_opl3sa2 to be referred for dev_*() calls. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-29-tiwai@suse.de --- sound/isa/opl3sa2.c | 46 ++++++++++++++++++++++----------------------- 1 file changed, 22 insertions(+), 24 deletions(-) diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index bad1490a66a0..a5ed5aa0606f 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -108,6 +108,7 @@ struct snd_opl3sa2 { int irq; int single_dma; spinlock_t reg_lock; + struct snd_card *card; struct snd_hwdep *synth; struct snd_rawmidi *rmidi; struct snd_wss *wss; @@ -157,12 +158,12 @@ static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char unsigned char result; #if 0 outb(0x1d, port); /* password */ - printk(KERN_DEBUG "read [0x%lx] = 0x%x\n", port, inb(port)); + dev_dbg(chip->card->dev, "read [0x%lx] = 0x%x\n", port, inb(port)); #endif outb(reg, chip->port); /* register */ result = inb(chip->port + 1); #if 0 - printk(KERN_DEBUG "read [0x%lx] = 0x%x [0x%x]\n", + dev_dbg(chip->card->dev, "read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port)); #endif return result; @@ -211,17 +212,13 @@ static int snd_opl3sa2_detect(struct snd_card *card) chip->res_port = devm_request_region(card->dev, port, 2, "OPL3-SA control"); if (!chip->res_port) { - snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); + dev_err(card->dev, "can't grab port 0x%lx\n", port); return -EBUSY; } - /* - snd_printk(KERN_DEBUG "REG 0A = 0x%x\n", - snd_opl3sa2_read(chip, 0x0a)); - */ chip->version = 0; tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC); if (tmp == 0xff) { - snd_printd("OPL3-SA [0x%lx] detect = 0x%x\n", port, tmp); + dev_dbg(card->dev, "OPL3-SA [0x%lx] detect = 0x%x\n", port, tmp); return -ENODEV; } switch (tmp & 0x07) { @@ -243,7 +240,7 @@ static int snd_opl3sa2_detect(struct snd_card *card) snd_opl3sa2_write(chip, OPL3SA2_MISC, tmp ^ 7); tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MISC); if (tmp1 != tmp) { - snd_printd("OPL3-SA [0x%lx] detect (1) = 0x%x (0x%x)\n", port, tmp, tmp1); + dev_dbg(card->dev, "OPL3-SA [0x%lx] detect (1) = 0x%x (0x%x)\n", port, tmp, tmp1); return -ENODEV; } /* try if the MIC register is accessible */ @@ -251,7 +248,7 @@ static int snd_opl3sa2_detect(struct snd_card *card) snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x8a); tmp1 = snd_opl3sa2_read(chip, OPL3SA2_MIC); if ((tmp1 & 0x9f) != 0x8a) { - snd_printd("OPL3-SA [0x%lx] detect (2) = 0x%x (0x%x)\n", port, tmp, tmp1); + dev_dbg(card->dev, "OPL3-SA [0x%lx] detect (2) = 0x%x (0x%x)\n", port, tmp, tmp1); return -ENODEV; } snd_opl3sa2_write(chip, OPL3SA2_MIC, 0x9f); @@ -495,14 +492,14 @@ static int snd_opl3sa2_mixer(struct snd_card *card) strcpy(id2.name, "CD Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n"); + dev_err(card->dev, "Cannot rename opl3sa2 control\n"); return err; } strcpy(id1.name, "Aux Playback Volume"); strcpy(id2.name, "CD Playback Volume"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n"); + dev_err(card->dev, "Cannot rename opl3sa2 control\n"); return err; } /* reassign AUX1 to FM */ @@ -510,14 +507,14 @@ static int snd_opl3sa2_mixer(struct snd_card *card) strcpy(id2.name, "FM Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n"); + dev_err(card->dev, "Cannot rename opl3sa2 control\n"); return err; } strcpy(id1.name, "Aux Playback Volume"); strcpy(id2.name, "FM Playback Volume"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opl3sa2 control\n"); + dev_err(card->dev, "Cannot rename opl3sa2 control\n"); return err; } /* add OPL3SA2 controls */ @@ -591,7 +588,7 @@ static int snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, struct pnp_dev *pdev) { if (pnp_activate_dev(pdev) < 0) { - snd_printk(KERN_ERR "PnP configure failure (out of resources?)\n"); + dev_err(chip->card->dev, "PnP configure failure (out of resources?)\n"); return -EBUSY; } sb_port[dev] = pnp_port_start(pdev, 0); @@ -602,9 +599,9 @@ static int snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, dma1[dev] = pnp_dma(pdev, 0); dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - snd_printdd("%sPnP OPL3-SA: sb port=0x%lx, wss port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", + dev_dbg(chip->card->dev, "%sPnP OPL3-SA: sb port=0x%lx, wss port=0x%lx, fm port=0x%lx, midi port=0x%lx\n", pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", sb_port[dev], wss_port[dev], fm_port[dev], midi_port[dev]); - snd_printdd("%sPnP OPL3-SA: control port=0x%lx, dma1=%i, dma2=%i, irq=%i\n", + dev_dbg(chip->card->dev, "%sPnP OPL3-SA: control port=0x%lx, dma1=%i, dma2=%i, irq=%i\n", pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", port[dev], dma1[dev], dma2[dev], irq[dev]); return 0; } @@ -640,6 +637,7 @@ static int snd_opl3sa2_probe(struct snd_card *card, int dev) /* initialise this card from supplied (or default) parameter*/ chip = card->private_data; + chip->card = card; chip->ymode = opl3sa3_ymode[dev] & 0x03 ; chip->port = port[dev]; xirq = irq[dev]; @@ -653,7 +651,7 @@ static int snd_opl3sa2_probe(struct snd_card *card, int dev) err = devm_request_irq(card->dev, xirq, snd_opl3sa2_interrupt, 0, "OPL3-SA2", card); if (err) { - snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); + dev_err(card->dev, "can't grab IRQ %d\n", xirq); return -ENODEV; } chip->irq = xirq; @@ -663,7 +661,7 @@ static int snd_opl3sa2_probe(struct snd_card *card, int dev) xirq, xdma1, xdma2, WSS_HW_OPL3SA2, WSS_HWSHARE_IRQ, &wss); if (err < 0) { - snd_printd("Oops, WSS not detected at 0x%lx\n", wss_port[dev] + 4); + dev_dbg(card->dev, "Oops, WSS not detected at 0x%lx\n", wss_port[dev] + 4); return err; } chip->wss = wss; @@ -770,7 +768,7 @@ static int snd_opl3sa2_pnp_cdetect(struct pnp_card_link *pcard, pdev = pnp_request_card_device(pcard, id->devs[0].id, NULL); if (pdev == NULL) { - snd_printk(KERN_ERR PFX "can't get pnp device from id '%s'\n", + dev_err(&pcard->card->dev, "can't get pnp device from id '%s'\n", id->devs[0].id); return -EBUSY; } @@ -828,19 +826,19 @@ static int snd_opl3sa2_isa_match(struct device *pdev, return 0; #endif if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify port\n"); + dev_err(pdev, "specify port\n"); return 0; } if (wss_port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify wss_port\n"); + dev_err(pdev, "specify wss_port\n"); return 0; } if (fm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify fm_port\n"); + dev_err(pdev, "specify fm_port\n"); return 0; } if (midi_port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR PFX "specify midi_port\n"); + dev_err(pdev, "specify midi_port\n"); return 0; } return 1; From 40b15de3c4f23994f53f930e94939c7b6a0cc52f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:19 +0200 Subject: [PATCH 134/410] ALSA: opti9xx: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The card pointer is stored in struct snd_opti9xx and snd_miro to be referred for dev_*() calls. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-30-tiwai@suse.de --- include/sound/aci.h | 1 + sound/isa/opti9xx/miro.c | 163 +++++++++++++++-------------- sound/isa/opti9xx/opti92x-ad1848.c | 62 +++++------ 3 files changed, 119 insertions(+), 107 deletions(-) diff --git a/include/sound/aci.h b/include/sound/aci.h index 6ebbd4223f12..36a761c9820d 100644 --- a/include/sound/aci.h +++ b/include/sound/aci.h @@ -72,6 +72,7 @@ #define ACI_SET_EQ7 0x46 /* ... to Treble */ struct snd_miro_aci { + struct snd_card *card; unsigned long aci_port; int aci_vendor; int aci_product; diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 59792f2fada1..31d736d1dd10 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -111,6 +111,7 @@ struct snd_miro { long mpu_port; int mpu_irq; + struct snd_card *card; struct snd_miro_aci *aci; }; @@ -151,8 +152,9 @@ static int aci_busy_wait(struct snd_miro_aci *aci) byte = inb(aci->aci_port + ACI_REG_BUSY); if ((byte & 1) == 0) { if (timeout >= ACI_MINTIME) - snd_printd("aci ready in round %ld.\n", - timeout-ACI_MINTIME); + dev_dbg(aci->card->dev, + "aci ready in round %ld.\n", + timeout-ACI_MINTIME); return byte; } if (timeout >= ACI_MINTIME) { @@ -174,7 +176,7 @@ static int aci_busy_wait(struct snd_miro_aci *aci) } } } - snd_printk(KERN_ERR "aci_busy_wait() time out\n"); + dev_err(aci->card->dev, "%s() time out\n", __func__); return -EBUSY; } @@ -184,7 +186,7 @@ static inline int aci_write(struct snd_miro_aci *aci, unsigned char byte) outb(byte, aci->aci_port + ACI_REG_COMMAND); return 0; } else { - snd_printk(KERN_ERR "aci busy, aci_write(0x%x) stopped.\n", byte); + dev_err(aci->card->dev, "aci busy, %s(0x%x) stopped.\n", __func__, byte); return -EBUSY; } } @@ -197,7 +199,7 @@ static inline int aci_read(struct snd_miro_aci *aci) byte = inb(aci->aci_port + ACI_REG_STATUS); return byte; } else { - snd_printk(KERN_ERR "aci busy, aci_read() stopped.\n"); + dev_err(aci->card->dev, "aci busy, %s() stopped.\n", __func__); return -EBUSY; } } @@ -260,8 +262,8 @@ static int snd_miro_get_capture(struct snd_kcontrol *kcontrol, value = aci_getvalue(miro->aci, ACI_S_GENERAL); if (value < 0) { - snd_printk(KERN_ERR "snd_miro_get_capture() failed: %d\n", - value); + dev_err(miro->card->dev, "%s() failed: %d\n", __func__, + value); return value; } @@ -280,8 +282,8 @@ static int snd_miro_put_capture(struct snd_kcontrol *kcontrol, error = aci_setvalue(miro->aci, ACI_SET_SOLOMODE, value); if (error < 0) { - snd_printk(KERN_ERR "snd_miro_put_capture() failed: %d\n", - error); + dev_err(miro->card->dev, "%s() failed: %d\n", __func__, + error); return error; } @@ -322,8 +324,8 @@ static int snd_miro_get_preamp(struct snd_kcontrol *kcontrol, value = aci_getvalue(miro->aci, ACI_GET_PREAMP); if (value < 0) { - snd_printk(KERN_ERR "snd_miro_get_preamp() failed: %d\n", - value); + dev_err(miro->card->dev, "%s() failed: %d\n", __func__, + value); return value; } @@ -342,8 +344,8 @@ static int snd_miro_put_preamp(struct snd_kcontrol *kcontrol, error = aci_setvalue(miro->aci, ACI_SET_PREAMP, value); if (error < 0) { - snd_printk(KERN_ERR "snd_miro_put_preamp() failed: %d\n", - error); + dev_err(miro->card->dev, "%s() failed: %d\n", __func__, + error); return error; } @@ -374,7 +376,8 @@ static int snd_miro_put_amp(struct snd_kcontrol *kcontrol, error = aci_setvalue(miro->aci, ACI_SET_POWERAMP, value); if (error < 0) { - snd_printk(KERN_ERR "snd_miro_put_amp() to %d failed: %d\n", value, error); + dev_err(miro->card->dev, "%s() to %d failed: %d\n", __func__, + value, error); return error; } @@ -430,13 +433,15 @@ static int snd_miro_get_double(struct snd_kcontrol *kcontrol, right_val = aci_getvalue(miro->aci, right_reg); if (right_val < 0) { - snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", right_reg, right_val); + dev_err(miro->card->dev, "aci_getvalue(%d) failed: %d\n", + right_reg, right_val); return right_val; } left_val = aci_getvalue(miro->aci, left_reg); if (left_val < 0) { - snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", left_reg, left_val); + dev_err(miro->card->dev, "aci_getvalue(%d) failed: %d\n", + left_reg, left_val); return left_val; } @@ -489,13 +494,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, left_old = aci_getvalue(aci, getreg_left); if (left_old < 0) { - snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_left, left_old); + dev_err(miro->card->dev, "aci_getvalue(%d) failed: %d\n", + getreg_left, left_old); return left_old; } right_old = aci_getvalue(aci, getreg_right); if (right_old < 0) { - snd_printk(KERN_ERR "aci_getvalue(%d) failed: %d\n", getreg_right, right_old); + dev_err(miro->card->dev, "aci_getvalue(%d) failed: %d\n", + getreg_right, right_old); return right_old; } @@ -515,15 +522,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, if (left >= 0) { error = aci_setvalue(aci, setreg_left, left); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - left, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + left, error); return error; } } else { error = aci_setvalue(aci, setreg_left, 0x80 - left); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - 0x80 - left, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + 0x80 - left, error); return error; } } @@ -531,15 +538,15 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, if (right >= 0) { error = aci_setvalue(aci, setreg_right, right); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - right, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + right, error); return error; } } else { error = aci_setvalue(aci, setreg_right, 0x80 - right); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - 0x80 - right, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + 0x80 - right, error); return error; } } @@ -557,14 +564,14 @@ static int snd_miro_put_double(struct snd_kcontrol *kcontrol, error = aci_setvalue(aci, setreg_left, 0x20 - left); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - 0x20 - left, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + 0x20 - left, error); return error; } error = aci_setvalue(aci, setreg_right, 0x20 - right); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - 0x20 - right, error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + 0x20 - right, error); return error; } } @@ -667,7 +674,7 @@ static int snd_set_aci_init_values(struct snd_miro *miro) if ((aci->aci_product == 'A') && wss) { error = aci_setvalue(aci, ACI_SET_WSS, wss); if (error < 0) { - snd_printk(KERN_ERR "enabling WSS mode failed\n"); + dev_err(miro->card->dev, "enabling WSS mode failed\n"); return error; } } @@ -677,7 +684,7 @@ static int snd_set_aci_init_values(struct snd_miro *miro) if (ide) { error = aci_setvalue(aci, ACI_SET_IDE, ide); if (error < 0) { - snd_printk(KERN_ERR "enabling IDE port failed\n"); + dev_err(miro->card->dev, "enabling IDE port failed\n"); return error; } } @@ -688,8 +695,8 @@ static int snd_set_aci_init_values(struct snd_miro *miro) error = aci_setvalue(aci, aci_init_values[idx][0], aci_init_values[idx][1]); if (error < 0) { - snd_printk(KERN_ERR "aci_setvalue(%d) failed: %d\n", - aci_init_values[idx][0], error); + dev_err(miro->card->dev, "aci_setvalue(%d) failed: %d\n", + aci_init_values[idx][0], error); return error; } } @@ -805,7 +812,7 @@ static int snd_miro_init(struct snd_miro *chip, break; default: - snd_printk(KERN_ERR "sorry, no support for %d\n", hardware); + dev_err(chip->card->dev, "sorry, no support for %d\n", hardware); return -ENODEV; } @@ -836,7 +843,7 @@ static unsigned char snd_miro_read(struct snd_miro *chip, break; default: - snd_printk(KERN_ERR "sorry, no support for %d\n", chip->hardware); + dev_err(chip->card->dev, "sorry, no support for %d\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -866,7 +873,7 @@ static void snd_miro_write(struct snd_miro *chip, unsigned char reg, break; default: - snd_printk(KERN_ERR "sorry, no support for %d\n", chip->hardware); + dev_err(chip->card->dev, "sorry, no support for %d\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -1022,7 +1029,7 @@ static int snd_miro_configure(struct snd_miro *chip) snd_miro_write_mask(chip, OPTi9XX_MC_REG(4), 0x00, 0x0c); break; default: - snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); + dev_err(chip->card->dev, "chip %d not supported\n", chip->hardware); return -EINVAL; } @@ -1045,7 +1052,7 @@ static int snd_miro_configure(struct snd_miro *chip) wss_base_bits = 0x02; break; default: - snd_printk(KERN_ERR "WSS port 0x%lx not valid\n", chip->wss_base); + dev_err(chip->card->dev, "WSS port 0x%lx not valid\n", chip->wss_base); goto __skip_base; } snd_miro_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -1068,7 +1075,7 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_ERR "WSS irq # %d not valid\n", chip->irq); + dev_err(chip->card->dev, "WSS irq # %d not valid\n", chip->irq); goto __skip_resources; } @@ -1083,12 +1090,12 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_ERR "WSS dma1 # %d not valid\n", chip->dma1); + dev_err(chip->card->dev, "WSS dma1 # %d not valid\n", chip->dma1); goto __skip_resources; } if (chip->dma1 == chip->dma2) { - snd_printk(KERN_ERR "don't want to share dmas\n"); + dev_err(chip->card->dev, "don't want to share dmas\n"); return -EBUSY; } @@ -1097,7 +1104,7 @@ __skip_base: case 1: break; default: - snd_printk(KERN_ERR "WSS dma2 # %d not valid\n", chip->dma2); + dev_err(chip->card->dev, "WSS dma2 # %d not valid\n", chip->dma2); goto __skip_resources; } dma_bits |= 0x04; @@ -1125,8 +1132,8 @@ __skip_resources: mpu_port_bits = 0x00; break; default: - snd_printk(KERN_ERR "MPU-401 port 0x%lx not valid\n", - chip->mpu_port); + dev_err(chip->card->dev, "MPU-401 port 0x%lx not valid\n", + chip->mpu_port); goto __skip_mpu; } @@ -1144,8 +1151,8 @@ __skip_resources: mpu_irq_bits = 0x01; break; default: - snd_printk(KERN_ERR "MPU-401 irq # %d not valid\n", - chip->mpu_irq); + dev_err(chip->card->dev, "MPU-401 irq # %d not valid\n", + chip->mpu_irq); goto __skip_mpu; } @@ -1208,6 +1215,7 @@ static int snd_card_miro_aci_detect(struct snd_card *card, miro->aci = aci; + aci->card = card; mutex_init(&aci->aci_mutex); /* get ACI port from OPTi9xx MC 4 */ @@ -1218,37 +1226,37 @@ static int snd_card_miro_aci_detect(struct snd_card *card, miro->res_aci_port = devm_request_region(card->dev, aci->aci_port, 3, "miro aci"); if (miro->res_aci_port == NULL) { - snd_printk(KERN_ERR "aci i/o area 0x%lx-0x%lx already used.\n", - aci->aci_port, aci->aci_port+2); + dev_err(card->dev, "aci i/o area 0x%lx-0x%lx already used.\n", + aci->aci_port, aci->aci_port+2); return -ENOMEM; } /* force ACI into a known state */ for (i = 0; i < 3; i++) if (snd_aci_cmd(aci, ACI_ERROR_OP, -1, -1) < 0) { - snd_printk(KERN_ERR "can't force aci into known state.\n"); + dev_err(card->dev, "can't force aci into known state.\n"); return -ENXIO; } aci->aci_vendor = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); aci->aci_product = snd_aci_cmd(aci, ACI_READ_IDCODE, -1, -1); if (aci->aci_vendor < 0 || aci->aci_product < 0) { - snd_printk(KERN_ERR "can't read aci id on 0x%lx.\n", - aci->aci_port); + dev_err(card->dev, "can't read aci id on 0x%lx.\n", + aci->aci_port); return -ENXIO; } aci->aci_version = snd_aci_cmd(aci, ACI_READ_VERSION, -1, -1); if (aci->aci_version < 0) { - snd_printk(KERN_ERR "can't read aci version on 0x%lx.\n", - aci->aci_port); + dev_err(card->dev, "can't read aci version on 0x%lx.\n", + aci->aci_port); return -ENXIO; } if (snd_aci_cmd(aci, ACI_INIT, -1, -1) < 0 || snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0 || snd_aci_cmd(aci, ACI_ERROR_OP, ACI_ERROR_OP, ACI_ERROR_OP) < 0) { - snd_printk(KERN_ERR "can't initialize aci.\n"); + dev_err(card->dev, "can't initialize aci.\n"); return -ENXIO; } @@ -1268,14 +1276,14 @@ static int snd_miro_probe(struct snd_card *card) miro->mc_base_size, "miro (OPTi9xx MC)"); if (miro->res_mc_base == NULL) { - snd_printk(KERN_ERR "request for OPTI9xx MC failed\n"); + dev_err(card->dev, "request for OPTI9xx MC failed\n"); return -ENOMEM; } } error = snd_card_miro_aci_detect(card, miro); if (error < 0) { - snd_printk(KERN_ERR "unable to detect aci chip\n"); + dev_err(card->dev, "unable to detect aci chip\n"); return -ENODEV; } @@ -1335,11 +1343,11 @@ static int snd_miro_probe(struct snd_card *card) default: sprintf(card->shortname, "unknown miro"); - snd_printk(KERN_INFO "unknown miro aci id\n"); + dev_info(card->dev, "unknown miro aci id\n"); break; } } else { - snd_printk(KERN_INFO "found unsupported aci card\n"); + dev_info(card->dev, "found unsupported aci card\n"); sprintf(card->shortname, "unknown Cardinal Technologies"); } @@ -1355,8 +1363,8 @@ static int snd_miro_probe(struct snd_card *card) error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port, 0, miro->mpu_irq, &rmidi); if (error < 0) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - mpu_port); + dev_warn(card->dev, "no MPU-401 device at 0x%lx?\n", + mpu_port); } if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { @@ -1365,8 +1373,8 @@ static int snd_miro_probe(struct snd_card *card) if (snd_opl4_create(card, fm_port, fm_port - 8, 2, &opl3, &opl4) < 0) - snd_printk(KERN_WARNING "no OPL4 device at 0x%lx\n", - fm_port); + dev_warn(card->dev, "no OPL4 device at 0x%lx\n", + fm_port); } error = snd_set_aci_init_values(miro); @@ -1410,14 +1418,14 @@ static int snd_miro_isa_probe(struct device *devptr, unsigned int n) error = snd_card_miro_detect(card, miro); if (error < 0) { - snd_printk(KERN_ERR "unable to detect OPTi9xx chip\n"); + dev_err(card->dev, "unable to detect OPTi9xx chip\n"); return -ENODEV; } if (port == SNDRV_AUTO_PORT) { port = snd_legacy_find_free_ioport(possible_ports, 4); if (port < 0) { - snd_printk(KERN_ERR "unable to find a free WSS port\n"); + dev_err(card->dev, "unable to find a free WSS port\n"); return -EBUSY; } } @@ -1425,8 +1433,8 @@ static int snd_miro_isa_probe(struct device *devptr, unsigned int n) if (mpu_port == SNDRV_AUTO_PORT) { mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); if (mpu_port < 0) { - snd_printk(KERN_ERR - "unable to find a free MPU401 port\n"); + dev_err(card->dev, + "unable to find a free MPU401 port\n"); return -EBUSY; } } @@ -1434,29 +1442,29 @@ static int snd_miro_isa_probe(struct device *devptr, unsigned int n) if (irq == SNDRV_AUTO_IRQ) { irq = snd_legacy_find_free_irq(possible_irqs); if (irq < 0) { - snd_printk(KERN_ERR "unable to find a free IRQ\n"); + dev_err(card->dev, "unable to find a free IRQ\n"); return -EBUSY; } } if (mpu_irq == SNDRV_AUTO_IRQ) { mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); if (mpu_irq < 0) { - snd_printk(KERN_ERR - "unable to find a free MPU401 IRQ\n"); + dev_err(card->dev, + "unable to find a free MPU401 IRQ\n"); return -EBUSY; } } if (dma1 == SNDRV_AUTO_DMA) { dma1 = snd_legacy_find_free_dma(possible_dma1s); if (dma1 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA1\n"); + dev_err(card->dev, "unable to find a free DMA1\n"); return -EBUSY; } } if (dma2 == SNDRV_AUTO_DMA) { dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); if (dma2 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA2\n"); + dev_err(card->dev, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -1505,14 +1513,14 @@ static int snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + dev_err(chip->card->dev, "AUDIO pnp configure failure: %d\n", err); return err; } err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "MC pnp configure failure: %d\n", - err); + dev_err(chip->card->dev, "MC pnp configure failure: %d\n", + err); return err; } @@ -1533,7 +1541,7 @@ static int snd_card_miro_pnp(struct snd_miro *chip, if (mpu_port > 0) { err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + dev_err(chip->card->dev, "MPU401 pnp configure failure\n"); mpu_port = -1; return err; } @@ -1560,6 +1568,7 @@ static int snd_miro_pnp_probe(struct pnp_card_link *pcard, return err; miro = card->private_data; + miro->card = card; err = snd_card_miro_pnp(miro, pcard, pid); if (err) @@ -1572,7 +1581,7 @@ static int snd_miro_pnp_probe(struct pnp_card_link *pcard, err = snd_miro_opti_check(card, miro); if (err) { - snd_printk(KERN_ERR "OPTI chip not found\n"); + dev_err(card->dev, "OPTI chip not found\n"); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index c33f67dd5133..220ea1952c1e 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -109,6 +109,7 @@ MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* OPTi93X */ struct snd_opti9xx { + struct snd_card *card; unsigned short hardware; unsigned char password; char name[7]; @@ -218,7 +219,7 @@ static int snd_opti9xx_init(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk(KERN_ERR "chip %d not supported\n", hardware); + dev_err(chip->card->dev, "chip %d not supported\n", hardware); return -ENODEV; } return 0; @@ -261,7 +262,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); + dev_err(chip->card->dev, "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -304,7 +305,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, #endif /* OPTi93X */ default: - snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); + dev_err(chip->card->dev, "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -400,7 +401,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); + dev_err(chip->card->dev, "chip %d not supported\n", chip->hardware); return -EINVAL; } @@ -423,7 +424,7 @@ static int snd_opti9xx_configure(struct snd_opti9xx *chip, wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); + dev_warn(chip->card->dev, "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -448,7 +449,7 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk(KERN_WARNING "WSS irq # %d not valid\n", irq); + dev_warn(chip->card->dev, "WSS irq # %d not valid\n", irq); goto __skip_resources; } @@ -463,13 +464,13 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", dma1); + dev_warn(chip->card->dev, "WSS dma1 # %d not valid\n", dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) if (dma1 == dma2) { - snd_printk(KERN_ERR "don't want to share dmas\n"); + dev_err(chip->card->dev, "don't want to share dmas\n"); return -EBUSY; } @@ -478,7 +479,7 @@ __skip_base: case 1: break; default: - snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", dma2); + dev_warn(chip->card->dev, "WSS dma2 # %d not valid\n", dma2); goto __skip_resources; } dma_bits |= 0x04; @@ -509,8 +510,8 @@ __skip_resources: mpu_port_bits = 0x00; break; default: - snd_printk(KERN_WARNING - "MPU-401 port 0x%lx not valid\n", mpu_port); + dev_warn(chip->card->dev, + "MPU-401 port 0x%lx not valid\n", mpu_port); goto __skip_mpu; } @@ -528,8 +529,8 @@ __skip_resources: mpu_irq_bits = 0x01; break; default: - snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", - mpu_irq); + dev_warn(chip->card->dev, "MPU-401 irq # %d not valid\n", + mpu_irq); goto __skip_mpu; } @@ -603,7 +604,7 @@ static int snd_opti93x_mixer(struct snd_wss *chip) strcpy(id2.name, "CD Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + dev_err(card->dev, "Cannot rename opti93x control\n"); return err; } /* reassign AUX1 switch to FM */ @@ -611,7 +612,7 @@ static int snd_opti93x_mixer(struct snd_wss *chip) strcpy(id2.name, "FM Playback Switch"); err = snd_ctl_rename_id(card, &id1, &id2); if (err < 0) { - snd_printk(KERN_ERR "Cannot rename opti93x control\n"); + dev_err(card->dev, "Cannot rename opti93x control\n"); return err; } /* remove AUX1 volume */ @@ -740,7 +741,7 @@ static int snd_card_opti9xx_pnp(struct snd_opti9xx *chip, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); + dev_err(chip->card->dev, "AUDIO pnp configure failure: %d\n", err); return err; } @@ -757,7 +758,7 @@ static int snd_card_opti9xx_pnp(struct snd_opti9xx *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + dev_err(chip->card->dev, "MC pnp configure failure: %d\n", err); return err; } @@ -781,7 +782,7 @@ static int snd_card_opti9xx_pnp(struct snd_opti9xx *chip, if (devmpu && mpu_port > 0) { err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); + dev_err(chip->card->dev, "MPU401 pnp configure failure\n"); mpu_port = -1; } else { mpu_port = pnp_port_start(devmpu, 0); @@ -811,7 +812,7 @@ static int snd_opti9xx_probe(struct snd_card *card) if (port == SNDRV_AUTO_PORT) { port = snd_legacy_find_free_ioport(possible_ports, 4); if (port < 0) { - snd_printk(KERN_ERR "unable to find a free WSS port\n"); + dev_err(card->dev, "unable to find a free WSS port\n"); return -EBUSY; } } @@ -850,7 +851,7 @@ static int snd_opti9xx_probe(struct snd_card *card) error = devm_request_irq(card->dev, irq, snd_opti93x_interrupt, 0, DEV_NAME" - WSS", chip); if (error < 0) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); + dev_err(card->dev, "opti9xx: can't grab IRQ %d\n", irq); return error; } #endif @@ -876,8 +877,8 @@ static int snd_opti9xx_probe(struct snd_card *card) error = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port, 0, mpu_irq, &rmidi); if (error) - snd_printk(KERN_WARNING "no MPU-401 device at 0x%lx?\n", - mpu_port); + dev_warn(card->dev, "no MPU-401 device at 0x%lx?\n", + mpu_port); } if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) { @@ -900,8 +901,8 @@ static int snd_opti9xx_probe(struct snd_card *card) #endif /* !OPTi93X */ if (!opl3 && snd_opl3_create(card, fm_port, fm_port + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - fm_port, fm_port + 4 - 1); + dev_warn(card->dev, "no OPL device at 0x%lx-0x%lx\n", + fm_port, fm_port + 4 - 1); } if (opl3) { error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); @@ -958,28 +959,28 @@ static int snd_opti9xx_isa_probe(struct device *devptr, if (mpu_port == SNDRV_AUTO_PORT) { mpu_port = snd_legacy_find_free_ioport(possible_mpu_ports, 2); if (mpu_port < 0) { - snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + dev_err(devptr, "unable to find a free MPU401 port\n"); return -EBUSY; } } if (irq == SNDRV_AUTO_IRQ) { irq = snd_legacy_find_free_irq(possible_irqs); if (irq < 0) { - snd_printk(KERN_ERR "unable to find a free IRQ\n"); + dev_err(devptr, "unable to find a free IRQ\n"); return -EBUSY; } } if (mpu_irq == SNDRV_AUTO_IRQ) { mpu_irq = snd_legacy_find_free_irq(possible_mpu_irqs); if (mpu_irq < 0) { - snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + dev_err(devptr, "unable to find a free MPU401 IRQ\n"); return -EBUSY; } } if (dma1 == SNDRV_AUTO_DMA) { dma1 = snd_legacy_find_free_dma(possible_dma1s); if (dma1 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA1\n"); + dev_err(devptr, "unable to find a free DMA1\n"); return -EBUSY; } } @@ -987,7 +988,7 @@ static int snd_opti9xx_isa_probe(struct device *devptr, if (dma2 == SNDRV_AUTO_DMA) { dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4]); if (dma2 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA2\n"); + dev_err(devptr, "unable to find a free DMA2\n"); return -EBUSY; } } @@ -1076,6 +1077,7 @@ static int snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, if (error < 0) return error; chip = card->private_data; + chip->card = card; hw = snd_card_opti9xx_pnp(chip, pcard, pid); switch (hw) { @@ -1097,7 +1099,7 @@ static int snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, return error; error = snd_opti9xx_read_check(card, chip); if (error) { - snd_printk(KERN_ERR "OPTI chip not found\n"); + dev_err(card->dev, "OPTI chip not found\n"); return error; } error = snd_opti9xx_probe(card); From b8986876e7196941c9723065e893af1a6528ecee Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:20 +0200 Subject: [PATCH 135/410] ALSA: sb: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some functions are changed to receive snd_card pointer for referring to the device pointer for dev_*() calls, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-31-tiwai@suse.de --- sound/isa/sb/emu8000.c | 11 ++++---- sound/isa/sb/emu8000_patch.c | 1 - sound/isa/sb/emu8000_synth.c | 2 +- sound/isa/sb/jazz16.c | 49 ++++++++++++++++++------------------ sound/isa/sb/sb16.c | 42 +++++++++++++++---------------- sound/isa/sb/sb16_csp.c | 38 ++++++++++++++++------------ sound/isa/sb/sb16_main.c | 13 +++++++--- sound/isa/sb/sb8.c | 12 ++++----- sound/isa/sb/sb_common.c | 27 ++++++++++---------- sound/isa/sb/sb_mixer.c | 4 +-- 10 files changed, 105 insertions(+), 94 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index af478c36ce5b..52884e6b9193 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -160,8 +160,8 @@ snd_emu8000_detect(struct snd_emu8000 *emu) if ((EMU8000_HWCF2_READ(emu) & 0x0003) != 0x0003) return -ENODEV; - snd_printdd("EMU8000 [0x%lx]: Synth chip found\n", - emu->port1); + dev_dbg(emu->card->dev, "EMU8000 [0x%lx]: Synth chip found\n", + emu->port1); return 0; } @@ -652,7 +652,7 @@ snd_emu8000_load_chorus_fx(struct snd_emu8000 *emu, int mode, const void __user { struct soundfont_chorus_fx rec; if (mode < SNDRV_EMU8000_CHORUS_PREDEFINED || mode >= SNDRV_EMU8000_CHORUS_NUMBERS) { - snd_printk(KERN_WARNING "invalid chorus mode %d for uploading\n", mode); + dev_warn(emu->card->dev, "invalid chorus mode %d for uploading\n", mode); return -EINVAL; } if (len < (long)sizeof(rec) || copy_from_user(&rec, buf, sizeof(rec))) @@ -780,7 +780,7 @@ snd_emu8000_load_reverb_fx(struct snd_emu8000 *emu, int mode, const void __user struct soundfont_reverb_fx rec; if (mode < SNDRV_EMU8000_REVERB_PREDEFINED || mode >= SNDRV_EMU8000_REVERB_NUMBERS) { - snd_printk(KERN_WARNING "invalid reverb mode %d for uploading\n", mode); + dev_warn(emu->card->dev, "invalid reverb mode %d for uploading\n", mode); return -EINVAL; } if (len < (long)sizeof(rec) || copy_from_user(&rec, buf, sizeof(rec))) @@ -1072,7 +1072,8 @@ snd_emu8000_new(struct snd_card *card, int index, long port, int seq_ports, if (!devm_request_region(card->dev, hw->port1, 4, "Emu8000-1") || !devm_request_region(card->dev, hw->port2, 4, "Emu8000-2") || !devm_request_region(card->dev, hw->port3, 4, "Emu8000-3")) { - snd_printk(KERN_ERR "sbawe: can't grab ports 0x%lx, 0x%lx, 0x%lx\n", hw->port1, hw->port2, hw->port3); + dev_err(card->dev, "sbawe: can't grab ports 0x%lx, 0x%lx, 0x%lx\n", + hw->port1, hw->port2, hw->port3); return -EBUSY; } hw->mem_size = 0; diff --git a/sound/isa/sb/emu8000_patch.c b/sound/isa/sb/emu8000_patch.c index ab4f988f080d..d60174ec8b39 100644 --- a/sound/isa/sb/emu8000_patch.c +++ b/sound/isa/sb/emu8000_patch.c @@ -157,7 +157,6 @@ snd_emu8000_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, sp->block = snd_util_mem_alloc(hdr, truesize * 2); if (sp->block == NULL) { - /*snd_printd("EMU8000: out of memory\n");*/ /* not ENOMEM (for compatibility) */ return -ENOSPC; } diff --git a/sound/isa/sb/emu8000_synth.c b/sound/isa/sb/emu8000_synth.c index 0edfb6875278..9bec85ec55b4 100644 --- a/sound/isa/sb/emu8000_synth.c +++ b/sound/isa/sb/emu8000_synth.c @@ -45,7 +45,7 @@ static int snd_emu8000_probe(struct device *_dev) emu->num_ports = hw->seq_ports; if (hw->memhdr) { - snd_printk(KERN_ERR "memhdr is already initialized!?\n"); + dev_err(hw->card->dev, "memhdr is already initialized!?\n"); snd_util_memhdr_free(hw->memhdr); } hw->memhdr = snd_util_memhdr_new(hw->mem_size); diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 64936c917170..b28490973892 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -75,13 +75,14 @@ static irqreturn_t jazz16_interrupt(int irq, void *chip) return snd_sb8dsp_interrupt(chip); } -static int jazz16_configure_ports(unsigned long port, +static int jazz16_configure_ports(struct snd_card *card, + unsigned long port, unsigned long mpu_port, int idx) { unsigned char val; if (!request_region(0x201, 1, "jazz16 config")) { - snd_printk(KERN_ERR "config port region is already in use.\n"); + dev_err(card->dev, "config port region is already in use.\n"); return -EBUSY; } outb(SB_JAZZ16_WAKEUP - idx, 0x201); @@ -96,15 +97,15 @@ static int jazz16_configure_ports(unsigned long port, return 0; } -static int jazz16_detect_board(unsigned long port, +static int jazz16_detect_board(struct snd_card *card, unsigned long port, unsigned long mpu_port) { int err; int val; - struct snd_sb chip; + struct snd_sb chip = {}; if (!request_region(port, 0x10, "jazz16")) { - snd_printk(KERN_ERR "I/O port region is already in use.\n"); + dev_err(card->dev, "I/O port region is already in use.\n"); return -EBUSY; } /* just to call snd_sbdsp_command/reset/get_byte() */ @@ -113,7 +114,7 @@ static int jazz16_detect_board(unsigned long port, err = snd_sbdsp_reset(&chip); if (err < 0) for (val = 0; val < 4; val++) { - err = jazz16_configure_ports(port, mpu_port, val); + err = jazz16_configure_ports(card, port, mpu_port, val); if (err < 0) break; @@ -143,8 +144,8 @@ static int jazz16_detect_board(unsigned long port, } snd_sbdsp_get_byte(&chip); err = snd_sbdsp_get_byte(&chip); - snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", - val, err); + dev_dbg(card->dev, "Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); err = 0; @@ -185,31 +186,31 @@ static int snd_jazz16_match(struct device *devptr, unsigned int dev) if (!enable[dev]) return 0; if (port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "please specify port\n"); + dev_err(devptr, "please specify port\n"); return 0; } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { - snd_printk(KERN_ERR "incorrect port specified\n"); + dev_err(devptr, "incorrect port specified\n"); return 0; } if (dma8[dev] != SNDRV_AUTO_DMA && dma8[dev] != 1 && dma8[dev] != 3) { - snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + dev_err(devptr, "dma8 must be 1 or 3\n"); return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + dev_err(devptr, "dma16 must be 5 or 7\n"); return 0; } if (mpu_port[dev] != SNDRV_AUTO_PORT && (mpu_port[dev] & ~0x030) != 0x300) { - snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + dev_err(devptr, "incorrect mpu_port specified\n"); return 0; } if (mpu_irq[dev] != SNDRV_AUTO_DMA && mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { - snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); + dev_err(devptr, "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; @@ -237,7 +238,7 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { - snd_printk(KERN_ERR "unable to find a free IRQ\n"); + dev_err(devptr, "unable to find a free IRQ\n"); return -EBUSY; } } @@ -245,7 +246,7 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) if (xdma8 == SNDRV_AUTO_DMA) { xdma8 = snd_legacy_find_free_dma(possible_dmas8); if (xdma8 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA8\n"); + dev_err(devptr, "unable to find a free DMA8\n"); return -EBUSY; } } @@ -253,7 +254,7 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) if (xdma16 == SNDRV_AUTO_DMA) { xdma16 = snd_legacy_find_free_dma(possible_dmas16); if (xdma16 < 0) { - snd_printk(KERN_ERR "unable to find a free DMA16\n"); + dev_err(devptr, "unable to find a free DMA16\n"); return -EBUSY; } } @@ -261,9 +262,9 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) xmpu_port = mpu_port[dev]; if (xmpu_port == SNDRV_AUTO_PORT) xmpu_port = 0; - err = jazz16_detect_board(port[dev], xmpu_port); + err = jazz16_detect_board(card, port[dev], xmpu_port); if (err < 0) { - printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + dev_err(devptr, "Media Vision Jazz16 board not detected\n"); return err; } err = snd_sbdsp_create(card, port[dev], irq[dev], @@ -279,7 +280,7 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) xmpu_irq = 0; err = jazz16_configure_board(chip, xmpu_irq); if (err < 0) { - printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + dev_err(devptr, "Media Vision Jazz16 configuration failed\n"); return err; } @@ -301,8 +302,8 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) err = snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_AUTO, 1, &opl3); if (err < 0) - snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", - chip->port, chip->port + 2); + dev_warn(devptr, "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); else { err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) @@ -317,8 +318,8 @@ static int snd_jazz16_probe(struct device *devptr, unsigned int dev) mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", - mpu_port[dev]); + dev_err(devptr, "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); } err = snd_card_register(card); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index e89b095aa282..2f7505ad855c 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -21,12 +21,6 @@ #define SNDRV_LEGACY_FIND_FREE_DMA #include -#ifdef SNDRV_SBAWE -#define PFX "sbawe: " -#else -#define PFX "sb16: " -#endif - MODULE_AUTHOR("Jaroslav Kysela "); MODULE_LICENSE("GPL"); #ifndef SNDRV_SBAWE @@ -246,7 +240,7 @@ static int snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); + dev_err(&pdev->dev, "AUDIO pnp configure failure\n"); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -255,10 +249,10 @@ static int snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, dma8[dev] = pnp_dma(pdev, 0); dma16[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - snd_printdd("pnp SB16: port=0x%lx, mpu port=0x%lx, fm port=0x%lx\n", - port[dev], mpu_port[dev], fm_port[dev]); - snd_printdd("pnp SB16: dma1=%i, dma2=%i, irq=%i\n", - dma8[dev], dma16[dev], irq[dev]); + dev_dbg(&pdev->dev, "pnp SB16: port=0x%lx, mpu port=0x%lx, fm port=0x%lx\n", + port[dev], mpu_port[dev], fm_port[dev]); + dev_dbg(&pdev->dev, "pnp SB16: dma1=%i, dma2=%i, irq=%i\n", + dma8[dev], dma16[dev], irq[dev]); #ifdef SNDRV_SBAWE_EMU8000 /* WaveTable initialization */ pdev = acard->devwt; @@ -268,13 +262,13 @@ static int snd_card_sb16_pnp(int dev, struct snd_card_sb16 *acard, goto __wt_error; } awe_port[dev] = pnp_port_start(pdev, 0); - snd_printdd("pnp SB16: wavetable port=0x%llx\n", - (unsigned long long)pnp_port_start(pdev, 0)); + dev_dbg(&pdev->dev, "pnp SB16: wavetable port=0x%llx\n", + (unsigned long long)pnp_port_start(pdev, 0)); } else { __wt_error: if (pdev) { pnp_release_card_device(pdev); - snd_printk(KERN_ERR PFX "WaveTable pnp configure failure\n"); + dev_err(&pdev->dev, "WaveTable pnp configure failure\n"); } acard->devwt = NULL; awe_port[dev] = -1; @@ -329,7 +323,7 @@ static int snd_sb16_probe(struct snd_card *card, int dev) acard->chip = chip; if (chip->hardware != SB_HW_16) { - snd_printk(KERN_ERR PFX "SB 16 chip was not detected at 0x%lx\n", port[dev]); + dev_err(card->dev, "SB 16 chip was not detected at 0x%lx\n", port[dev]); return -ENODEV; } chip->mpu_port = mpu_port[dev]; @@ -379,8 +373,8 @@ static int snd_sb16_probe(struct snd_card *card, int dev) OPL3_HW_OPL3, acard->fm_res != NULL || fm_port[dev] == port[dev], &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx\n", - fm_port[dev], fm_port[dev] + 2); + dev_err(card->dev, "no OPL device at 0x%lx-0x%lx\n", + fm_port[dev], fm_port[dev] + 2); } else { #ifdef SNDRV_SBAWE_EMU8000 int seqdev = awe_port[dev] > 0 ? 2 : 1; @@ -405,7 +399,9 @@ static int snd_sb16_probe(struct snd_card *card, int dev) chip->csp = xcsp->private_data; chip->hardware = SB_HW_16CSP; } else { - snd_printk(KERN_INFO PFX "warning - CSP chip not detected on soundcard #%i\n", dev + 1); + dev_info(card->dev, + "warning - CSP chip not detected on soundcard #%i\n", + dev + 1); } } #endif @@ -414,7 +410,9 @@ static int snd_sb16_probe(struct snd_card *card, int dev) err = snd_emu8000_new(card, 1, awe_port[dev], seq_ports[dev], NULL); if (err < 0) { - snd_printk(KERN_ERR PFX "fatal error - EMU-8000 synthesizer not detected at 0x%lx\n", awe_port[dev]); + dev_err(card->dev, + "fatal error - EMU-8000 synthesizer not detected at 0x%lx\n", + awe_port[dev]); return err; } @@ -502,21 +500,21 @@ static int snd_sb16_isa_probe(struct device *pdev, unsigned int dev) if (irq[dev] == SNDRV_AUTO_IRQ) { irq[dev] = snd_legacy_find_free_irq(possible_irqs); if (irq[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + dev_err(pdev, "unable to find a free IRQ\n"); return -EBUSY; } } if (dma8[dev] == SNDRV_AUTO_DMA) { dma8[dev] = snd_legacy_find_free_dma(possible_dmas8); if (dma8[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free 8-bit DMA\n"); + dev_err(pdev, "unable to find a free 8-bit DMA\n"); return -EBUSY; } } if (dma16[dev] == SNDRV_AUTO_DMA) { dma16[dev] = snd_legacy_find_free_dma(possible_dmas16); if (dma16[dev] < 0) { - snd_printk(KERN_ERR PFX "unable to find a free 16-bit DMA\n"); + dev_err(pdev, "unable to find a free 16-bit DMA\n"); return -EBUSY; } } diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index fdb992733bde..071ba85f76bb 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -296,6 +296,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, struct snd_sb_csp_microcode __user * mcode) { struct snd_sb_csp_mc_header info; + struct device *dev = p->chip->card->dev; unsigned char __user *data_ptr; unsigned char __user *data_end; @@ -316,7 +317,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if ((le32_to_cpu(file_h.name) != RIFF_HEADER) || (le32_to_cpu(file_h.len) >= SNDRV_SB_CSP_MAX_MICROCODE_FILE_SIZE - sizeof(file_h))) { - snd_printd("%s: Invalid RIFF header\n", __func__); + dev_dbg(dev, "%s: Invalid RIFF header\n", __func__); return -EINVAL; } data_ptr += sizeof(file_h); @@ -325,7 +326,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, if (copy_from_user(&item_type, data_ptr, sizeof(item_type))) return -EFAULT; if (le32_to_cpu(item_type) != CSP__HEADER) { - snd_printd("%s: Invalid RIFF file type\n", __func__); + dev_dbg(dev, "%s: Invalid RIFF file type\n", __func__); return -EINVAL; } data_ptr += sizeof (item_type); @@ -380,7 +381,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return -EFAULT; if (le32_to_cpu(code_h.name) != MAIN_HEADER) { - snd_printd("%s: Missing 'main' microcode\n", __func__); + dev_dbg(dev, "%s: Missing 'main' microcode\n", __func__); return -EINVAL; } data_ptr += sizeof(code_h); @@ -423,9 +424,9 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, default: /* other codecs are unsupported */ p->acc_format = p->acc_width = p->acc_rates = 0; p->mode = 0; - snd_printd("%s: Unsupported CSP codec type: 0x%04x\n", - __func__, - le16_to_cpu(funcdesc_h.VOC_type)); + dev_dbg(dev, "%s: Unsupported CSP codec type: 0x%04x\n", + __func__, + le16_to_cpu(funcdesc_h.VOC_type)); return -EINVAL; } p->acc_channels = le16_to_cpu(funcdesc_h.flags_stereo_mono); @@ -443,7 +444,7 @@ static int snd_sb_csp_riff_load(struct snd_sb_csp * p, return 0; } } - snd_printd("%s: Function #%d not found\n", __func__, info.func_req); + dev_dbg(dev, "%s: Function #%d not found\n", __func__, info.func_req); return -EINVAL; } @@ -597,7 +598,9 @@ static int get_version(struct snd_sb *chip) static int snd_sb_csp_check_version(struct snd_sb_csp * p) { if (p->version < 0x10 || p->version > 0x1f) { - snd_printd("%s: Invalid CSP version: 0x%x\n", __func__, p->version); + dev_dbg(p->chip->card->dev, + "%s: Invalid CSP version: 0x%x\n", + __func__, p->version); return 1; } return 0; @@ -616,7 +619,7 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int spin_lock_irqsave(&p->chip->reg_lock, flags); snd_sbdsp_command(p->chip, 0x01); /* CSP download command */ if (snd_sbdsp_get_byte(p->chip)) { - snd_printd("%s: Download command failed\n", __func__); + dev_dbg(p->chip->card->dev, "%s: Download command failed\n", __func__); goto __fail; } /* Send CSP low byte (size - 1) */ @@ -643,7 +646,9 @@ static int snd_sb_csp_load(struct snd_sb_csp * p, const unsigned char *buf, int udelay (10); } if (status != 0x55) { - snd_printd("%s: Microcode initialization failed\n", __func__); + dev_dbg(p->chip->card->dev, + "%s: Microcode initialization failed\n", + __func__); goto __fail; } } else { @@ -788,25 +793,26 @@ static int snd_sb_csp_autoload(struct snd_sb_csp * p, snd_pcm_format_t pcm_sfmt, */ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channels) { + struct device *dev = p->chip->card->dev; unsigned char s_type; /* sample type */ unsigned char mixL, mixR; int result = -EIO; unsigned long flags; if (!(p->running & (SNDRV_SB_CSP_ST_LOADED | SNDRV_SB_CSP_ST_AUTO))) { - snd_printd("%s: Microcode not loaded\n", __func__); + dev_dbg(dev, "%s: Microcode not loaded\n", __func__); return -ENXIO; } if (p->running & SNDRV_SB_CSP_ST_RUNNING) { - snd_printd("%s: CSP already running\n", __func__); + dev_dbg(dev, "%s: CSP already running\n", __func__); return -EBUSY; } if (!(sample_width & p->acc_width)) { - snd_printd("%s: Unsupported PCM sample width\n", __func__); + dev_dbg(dev, "%s: Unsupported PCM sample width\n", __func__); return -EINVAL; } if (!(channels & p->acc_channels)) { - snd_printd("%s: Invalid number of channels\n", __func__); + dev_dbg(dev, "%s: Invalid number of channels\n", __func__); return -EINVAL; } @@ -829,11 +835,11 @@ static int snd_sb_csp_start(struct snd_sb_csp * p, int sample_width, int channel s_type |= 0x22; /* 00dX 00dX (d = 1 if 8 bit samples) */ if (set_codec_parameter(p->chip, 0x81, s_type)) { - snd_printd("%s: Set sample type command failed\n", __func__); + dev_dbg(dev, "%s: Set sample type command failed\n", __func__); goto __fail; } if (set_codec_parameter(p->chip, 0x80, 0x00)) { - snd_printd("%s: Codec start command failed\n", __func__); + dev_dbg(dev, "%s: Codec start command failed\n", __func__); goto __fail; } p->run_width = sample_width; diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index a9b87e159b2d..74db11525003 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -733,7 +733,6 @@ int snd_sb16dsp_configure(struct snd_sb * chip) unsigned char realirq, realdma, realmpureg; /* note: mpu register should be present only on SB16 Vibra soundcards */ - // printk(KERN_DEBUG "codec->irq=%i, codec->dma8=%i, codec->dma16=%i\n", chip->irq, chip->dma8, chip->dma16); spin_lock_irqsave(&chip->mixer_lock, flags); mpureg = snd_sbmixer_read(chip, SB_DSP4_MPUSETUP) & ~0x06; spin_unlock_irqrestore(&chip->mixer_lock, flags); @@ -807,9 +806,15 @@ int snd_sb16dsp_configure(struct snd_sb * chip) spin_unlock_irqrestore(&chip->mixer_lock, flags); if ((~realirq) & irqreg || (~realdma) & dmareg) { - snd_printk(KERN_ERR "SB16 [0x%lx]: unable to set DMA & IRQ (PnP device?)\n", chip->port); - snd_printk(KERN_ERR "SB16 [0x%lx]: wanted: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", chip->port, realirq, realdma, realmpureg); - snd_printk(KERN_ERR "SB16 [0x%lx]: got: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", chip->port, irqreg, dmareg, mpureg); + dev_err(chip->card->dev, + "SB16 [0x%lx]: unable to set DMA & IRQ (PnP device?)\n", + chip->port); + dev_err(chip->card->dev, + "SB16 [0x%lx]: wanted: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", + chip->port, realirq, realdma, realmpureg); + dev_err(chip->card->dev, + "SB16 [0x%lx]: got: irqreg=0x%x, dmareg=0x%x, mpureg = 0x%x\n", + chip->port, irqreg, dmareg, mpureg); return -ENODEV; } return 0; diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index e5ef1777161f..8726778c815e 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -123,11 +123,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) if (chip->hardware >= SB_HW_16) { if (chip->hardware == SB_HW_ALS100) - snd_printk(KERN_WARNING "ALS100 chip detected at 0x%lx, try snd-als100 module\n", - port[dev]); + dev_warn(pdev, "ALS100 chip detected at 0x%lx, try snd-als100 module\n", + port[dev]); else - snd_printk(KERN_WARNING "SB 16 chip detected at 0x%lx, try snd-sb16 module\n", - port[dev]); + dev_warn(pdev, "SB 16 chip detected at 0x%lx, try snd-sb16 module\n", + port[dev]); return -ENODEV; } @@ -143,12 +143,12 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) err = snd_opl3_create(card, chip->port + 8, 0, OPL3_HW_AUTO, 1, &opl3); if (err < 0) - snd_printk(KERN_WARNING "sb8: no OPL device at 0x%lx\n", chip->port + 8); + dev_warn(pdev, "sb8: no OPL device at 0x%lx\n", chip->port + 8); } else { err = snd_opl3_create(card, chip->port, chip->port + 2, OPL3_HW_AUTO, 1, &opl3); if (err < 0) { - snd_printk(KERN_WARNING "sb8: no OPL device at 0x%lx-0x%lx\n", + dev_warn(pdev, "sb8: no OPL device at 0x%lx-0x%lx\n", chip->port, chip->port + 2); } } diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index c0e319d14210..a4d5bf3d145f 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -31,14 +31,14 @@ int snd_sbdsp_command(struct snd_sb *chip, unsigned char val) { int i; #ifdef IO_DEBUG - snd_printk(KERN_DEBUG "command 0x%x\n", val); + dev_dbg(chip->card->dev, "command 0x%x\n", val); #endif for (i = BUSY_LOOPS; i; i--) if ((inb(SBP(chip, STATUS)) & 0x80) == 0) { outb(val, SBP(chip, COMMAND)); return 1; } - snd_printd("%s [0x%lx]: timeout (0x%x)\n", __func__, chip->port, val); + dev_dbg(chip->card->dev, "%s [0x%lx]: timeout (0x%x)\n", __func__, chip->port, val); return 0; } @@ -50,12 +50,12 @@ int snd_sbdsp_get_byte(struct snd_sb *chip) if (inb(SBP(chip, DATA_AVAIL)) & 0x80) { val = inb(SBP(chip, READ)); #ifdef IO_DEBUG - snd_printk(KERN_DEBUG "get_byte 0x%x\n", val); + dev_dbg(chip->card->dev, "get_byte 0x%x\n", val); #endif return val; } } - snd_printd("%s [0x%lx]: timeout\n", __func__, chip->port); + dev_dbg(chip->card->dev, "%s [0x%lx]: timeout\n", __func__, chip->port); return -ENODEV; } @@ -74,7 +74,8 @@ int snd_sbdsp_reset(struct snd_sb *chip) else break; } - snd_printdd("%s [0x%lx] failed...\n", __func__, chip->port); + if (chip->card) + dev_dbg(chip->card->dev, "%s [0x%lx] failed...\n", __func__, chip->port); return -ENODEV; } @@ -112,8 +113,8 @@ static int snd_sbdsp_probe(struct snd_sb * chip) spin_unlock_irqrestore(&chip->reg_lock, flags); major = version >> 8; minor = version & 0xff; - snd_printdd("SB [0x%lx]: DSP chip found, version = %i.%i\n", - chip->port, major, minor); + dev_dbg(chip->card->dev, "SB [0x%lx]: DSP chip found, version = %i.%i\n", + chip->port, major, minor); switch (chip->hardware) { case SB_HW_AUTO: @@ -140,8 +141,8 @@ static int snd_sbdsp_probe(struct snd_sb * chip) str = "16"; break; default: - snd_printk(KERN_INFO "SB [0x%lx]: unknown DSP chip version %i.%i\n", - chip->port, major, minor); + dev_info(chip->card->dev, "SB [0x%lx]: unknown DSP chip version %i.%i\n", + chip->port, major, minor); return -ENODEV; } break; @@ -200,7 +201,7 @@ int snd_sbdsp_create(struct snd_card *card, hardware == SB_HW_CS5530) ? IRQF_SHARED : 0, "SoundBlaster", (void *) chip)) { - snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); + dev_err(card->dev, "sb: can't grab irq %d\n", irq); return -EBUSY; } chip->irq = irq; @@ -212,14 +213,14 @@ int snd_sbdsp_create(struct snd_card *card, chip->res_port = devm_request_region(card->dev, port, 16, "SoundBlaster"); if (!chip->res_port) { - snd_printk(KERN_ERR "sb: can't grab port 0x%lx\n", port); + dev_err(card->dev, "sb: can't grab port 0x%lx\n", port); return -EBUSY; } #ifdef CONFIG_ISA if (dma8 >= 0 && snd_devm_request_dma(card->dev, dma8, "SoundBlaster - 8bit")) { - snd_printk(KERN_ERR "sb: can't grab DMA8 %d\n", dma8); + dev_err(card->dev, "sb: can't grab DMA8 %d\n", dma8); return -EBUSY; } chip->dma8 = dma8; @@ -229,7 +230,7 @@ int snd_sbdsp_create(struct snd_card *card, dma16 = -1; } else if (snd_devm_request_dma(card->dev, dma16, "SoundBlaster - 16bit")) { - snd_printk(KERN_ERR "sb: can't grab DMA16 %d\n", dma16); + dev_err(card->dev, "sb: can't grab DMA16 %d\n", dma16); return -EBUSY; } } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index fffd681e5bf7..9d23b7a4570b 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -20,7 +20,7 @@ void snd_sbmixer_write(struct snd_sb *chip, unsigned char reg, unsigned char dat outb(data, SBP(chip, MIXER_DATA)); udelay(10); #ifdef IO_DEBUG - snd_printk(KERN_DEBUG "mixer_write 0x%x 0x%x\n", reg, data); + dev_dbg(chip->card->dev, "mixer_write 0x%x 0x%x\n", reg, data); #endif } @@ -33,7 +33,7 @@ unsigned char snd_sbmixer_read(struct snd_sb *chip, unsigned char reg) result = inb(SBP(chip, MIXER_DATA)); udelay(10); #ifdef IO_DEBUG - snd_printk(KERN_DEBUG "mixer_read 0x%x 0x%x\n", reg, result); + dev_dbg(chip->card->dev, "mixer_read 0x%x 0x%x\n", reg, result); #endif return result; } From 56887daf2fa9206fa0fec7cfe6428835e0b66264 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:21 +0200 Subject: [PATCH 136/410] ALSA: control_led: Use dev_err() Use the standard print API instead of open-coded printk(). Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-32-tiwai@suse.de --- sound/core/control_led.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 804805a95e2f..54dcfce983b8 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -677,7 +677,7 @@ static void snd_ctl_led_sysfs_add(struct snd_card *card) cerr: put_device(&led_card->dev); cerr2: - printk(KERN_ERR "snd_ctl_led: unable to add card%d", card->number); + dev_err(card->dev, "snd_ctl_led: unable to add card%d", card->number); } } From 55c531bd81a66e1c3a0dd35e8ec07237ba9dce49 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:22 +0200 Subject: [PATCH 137/410] ALSA: pcm: oss: Use pr_debug() Use the standard print API instead of open-coded printk(). Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-33-tiwai@suse.de --- sound/core/oss/pcm_plugin.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 50a6b50f5db4..55035dbf23f0 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -160,7 +160,7 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void **bufs, snd_pcm_uframes_t frames); #ifdef PLUGIN_DEBUG -#define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) +#define pdprintf(fmt, args...) pr_debug("plugin: " fmt, ##args) #else #define pdprintf(fmt, args...) #endif From e7c475b92043c02c3e6cd0c20e308fbb6f03ebde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:23 +0200 Subject: [PATCH 138/410] ALSA: sc6000: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some functions are changed to receive a device pointer to be passed to dev_*() calls. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-34-tiwai@suse.de --- sound/isa/sc6000.c | 177 +++++++++++++++++++++++---------------------- 1 file changed, 90 insertions(+), 87 deletions(-) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 60398fced046..3115c32b4061 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -204,7 +204,7 @@ static int sc6000_read(char __iomem *vport) } -static int sc6000_write(char __iomem *vport, int cmd) +static int sc6000_write(struct device *devptr, char __iomem *vport, int cmd) { unsigned char val; int loop = 500000; @@ -221,18 +221,19 @@ static int sc6000_write(char __iomem *vport, int cmd) cpu_relax(); } while (loop--); - snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd); + dev_err(devptr, "DSP Command (0x%x) timeout.\n", cmd); return -EIO; } -static int sc6000_dsp_get_answer(char __iomem *vport, int command, +static int sc6000_dsp_get_answer(struct device *devptr, + char __iomem *vport, int command, char *data, int data_len) { int len = 0; - if (sc6000_write(vport, command)) { - snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command); + if (sc6000_write(devptr, vport, command)) { + dev_err(devptr, "CMD 0x%x: failed!\n", command); return -EIO; } @@ -265,82 +266,86 @@ static int sc6000_dsp_reset(char __iomem *vport) } /* detection and initialization */ -static int sc6000_hw_cfg_write(char __iomem *vport, const int *cfg) +static int sc6000_hw_cfg_write(struct device *devptr, + char __iomem *vport, const int *cfg) { - if (sc6000_write(vport, COMMAND_6C) < 0) { - snd_printk(KERN_WARNING "CMD 0x%x: failed!\n", COMMAND_6C); + if (sc6000_write(devptr, vport, COMMAND_6C) < 0) { + dev_warn(devptr, "CMD 0x%x: failed!\n", COMMAND_6C); return -EIO; } - if (sc6000_write(vport, COMMAND_5C) < 0) { - snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_5C); + if (sc6000_write(devptr, vport, COMMAND_5C) < 0) { + dev_err(devptr, "CMD 0x%x: failed!\n", COMMAND_5C); return -EIO; } - if (sc6000_write(vport, cfg[0]) < 0) { - snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[0]); + if (sc6000_write(devptr, vport, cfg[0]) < 0) { + dev_err(devptr, "DATA 0x%x: failed!\n", cfg[0]); return -EIO; } - if (sc6000_write(vport, cfg[1]) < 0) { - snd_printk(KERN_ERR "DATA 0x%x: failed!\n", cfg[1]); + if (sc6000_write(devptr, vport, cfg[1]) < 0) { + dev_err(devptr, "DATA 0x%x: failed!\n", cfg[1]); return -EIO; } - if (sc6000_write(vport, COMMAND_C5) < 0) { - snd_printk(KERN_ERR "CMD 0x%x: failed!\n", COMMAND_C5); + if (sc6000_write(devptr, vport, COMMAND_C5) < 0) { + dev_err(devptr, "CMD 0x%x: failed!\n", COMMAND_C5); return -EIO; } return 0; } -static int sc6000_cfg_write(char __iomem *vport, unsigned char softcfg) +static int sc6000_cfg_write(struct device *devptr, + char __iomem *vport, unsigned char softcfg) { - if (sc6000_write(vport, WRITE_MDIRQ_CFG)) { - snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + if (sc6000_write(devptr, vport, WRITE_MDIRQ_CFG)) { + dev_err(devptr, "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); return -EIO; } - if (sc6000_write(vport, softcfg)) { - snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + if (sc6000_write(devptr, vport, softcfg)) { + dev_err(devptr, "%s: failed!\n", __func__); return -EIO; } return 0; } -static int sc6000_setup_board(char __iomem *vport, int config) +static int sc6000_setup_board(struct device *devptr, + char __iomem *vport, int config) { int loop = 10; do { - if (sc6000_write(vport, COMMAND_88)) { - snd_printk(KERN_ERR "CMD 0x%x: failed!\n", - COMMAND_88); + if (sc6000_write(devptr, vport, COMMAND_88)) { + dev_err(devptr, "CMD 0x%x: failed!\n", + COMMAND_88); return -EIO; } } while ((sc6000_wait_data(vport) < 0) && loop--); if (sc6000_read(vport) < 0) { - snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", - COMMAND_88); + dev_err(devptr, "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); return -EIO; } - if (sc6000_cfg_write(vport, config)) + if (sc6000_cfg_write(devptr, vport, config)) return -ENODEV; return 0; } -static int sc6000_init_mss(char __iomem *vport, int config, +static int sc6000_init_mss(struct device *devptr, + char __iomem *vport, int config, char __iomem *vmss_port, int mss_config) { - if (sc6000_write(vport, DSP_INIT_MSS)) { - snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n", - DSP_INIT_MSS); + if (sc6000_write(devptr, vport, DSP_INIT_MSS)) { + dev_err(devptr, "%s [0x%x]: failed!\n", __func__, + DSP_INIT_MSS); return -EIO; } msleep(10); - if (sc6000_cfg_write(vport, config)) + if (sc6000_cfg_write(devptr, vport, config)) return -EIO; iowrite8(mss_config, vmss_port); @@ -348,7 +353,8 @@ static int sc6000_init_mss(char __iomem *vport, int config, return 0; } -static void sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, +static void sc6000_hw_cfg_encode(struct device *devptr, + char __iomem *vport, int *cfg, long xport, long xmpu, long xmss_port, int joystick) { @@ -367,10 +373,11 @@ static void sc6000_hw_cfg_encode(char __iomem *vport, int *cfg, cfg[0] |= 0x02; cfg[1] |= 0x80; /* enable WSS system */ cfg[1] &= ~0x40; /* disable IDE */ - snd_printd("hw cfg %x, %x\n", cfg[0], cfg[1]); + dev_dbg(devptr, "hw cfg %x, %x\n", cfg[0], cfg[1]); } -static int sc6000_init_board(char __iomem *vport, +static int sc6000_init_board(struct device *devptr, + char __iomem *vport, char __iomem *vmss_port, int dev) { char answer[15]; @@ -384,14 +391,14 @@ static int sc6000_init_board(char __iomem *vport, err = sc6000_dsp_reset(vport); if (err < 0) { - snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n"); + dev_err(devptr, "sc6000_dsp_reset: failed!\n"); return err; } memset(answer, 0, sizeof(answer)); - err = sc6000_dsp_get_answer(vport, GET_DSP_COPYRIGHT, answer, 15); + err = sc6000_dsp_get_answer(devptr, vport, GET_DSP_COPYRIGHT, answer, 15); if (err <= 0) { - snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n"); + dev_err(devptr, "sc6000_dsp_copyright: failed!\n"); return -ENODEV; } /* @@ -399,52 +406,52 @@ static int sc6000_init_board(char __iomem *vport, * if we have something different, we have to be warned. */ if (strncmp("SC-6000", answer, 7)) - snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); + dev_warn(devptr, "Warning: non SC-6000 audio card!\n"); - if (sc6000_dsp_get_answer(vport, GET_DSP_VERSION, version, 2) < 2) { - snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n"); + if (sc6000_dsp_get_answer(devptr, vport, GET_DSP_VERSION, version, 2) < 2) { + dev_err(devptr, "sc6000_dsp_version: failed!\n"); return -ENODEV; } - printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n", + dev_info(devptr, "Detected model: %s, DSP version %d.%d\n", answer, version[0], version[1]); /* set configuration */ - sc6000_write(vport, COMMAND_5C); + sc6000_write(devptr, vport, COMMAND_5C); if (sc6000_read(vport) < 0) old = 1; if (!old) { int cfg[2]; - sc6000_hw_cfg_encode(vport, &cfg[0], port[dev], mpu_port[dev], + sc6000_hw_cfg_encode(devptr, + vport, &cfg[0], port[dev], mpu_port[dev], mss_port[dev], joystick[dev]); - if (sc6000_hw_cfg_write(vport, cfg) < 0) { - snd_printk(KERN_ERR "sc6000_hw_cfg_write: failed!\n"); + if (sc6000_hw_cfg_write(devptr, vport, cfg) < 0) { + dev_err(devptr, "sc6000_hw_cfg_write: failed!\n"); return -EIO; } } - err = sc6000_setup_board(vport, config); + err = sc6000_setup_board(devptr, vport, config); if (err < 0) { - snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + dev_err(devptr, "sc6000_setup_board: failed!\n"); return -ENODEV; } sc6000_dsp_reset(vport); if (!old) { - sc6000_write(vport, COMMAND_60); - sc6000_write(vport, 0x02); + sc6000_write(devptr, vport, COMMAND_60); + sc6000_write(devptr, vport, 0x02); sc6000_dsp_reset(vport); } - err = sc6000_setup_board(vport, config); + err = sc6000_setup_board(devptr, vport, config); if (err < 0) { - snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + dev_err(devptr, "sc6000_setup_board: failed!\n"); return -ENODEV; } - err = sc6000_init_mss(vport, config, vmss_port, mss_config); + err = sc6000_init_mss(devptr, vport, config, vmss_port, mss_config); if (err < 0) { - snd_printk(KERN_ERR "Cannot initialize " - "Microsoft Sound System mode.\n"); + dev_err(devptr, "Cannot initialize Microsoft Sound System mode.\n"); return -ENODEV; } @@ -491,39 +498,39 @@ static int snd_sc6000_match(struct device *devptr, unsigned int dev) if (!enable[dev]) return 0; if (port[dev] == SNDRV_AUTO_PORT) { - printk(KERN_ERR PFX "specify IO port\n"); + dev_err(devptr, "specify IO port\n"); return 0; } if (mss_port[dev] == SNDRV_AUTO_PORT) { - printk(KERN_ERR PFX "specify MSS port\n"); + dev_err(devptr, "specify MSS port\n"); return 0; } if (port[dev] != 0x220 && port[dev] != 0x240) { - printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n"); + dev_err(devptr, "Port must be 0x220 or 0x240\n"); return 0; } if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) { - printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n"); + dev_err(devptr, "MSS port must be 0x530 or 0xe80\n"); return 0; } if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) { - printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]); + dev_err(devptr, "invalid IRQ %d\n", irq[dev]); return 0; } if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) { - printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]); + dev_err(devptr, "invalid DMA %d\n", dma[dev]); return 0; } if (mpu_port[dev] != SNDRV_AUTO_PORT && (mpu_port[dev] & ~0x30L) != 0x300) { - printk(KERN_ERR PFX "invalid MPU-401 port %lx\n", + dev_err(devptr, "invalid MPU-401 port %lx\n", mpu_port[dev]); return 0; } if (mpu_port[dev] != SNDRV_AUTO_PORT && mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 && !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) { - printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); + dev_err(devptr, "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); return 0; } return 1; @@ -534,7 +541,7 @@ static void snd_sc6000_free(struct snd_card *card) char __iomem *vport = (char __force __iomem *)card->private_data; if (vport) - sc6000_setup_board(vport, 0); + sc6000_setup_board(card->dev, vport, 0); } static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) @@ -558,7 +565,7 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { - snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + dev_err(devptr, "unable to find a free IRQ\n"); return -EBUSY; } } @@ -566,42 +573,39 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) if (xdma == SNDRV_AUTO_DMA) { xdma = snd_legacy_find_free_dma(possible_dmas); if (xdma < 0) { - snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); + dev_err(devptr, "unable to find a free DMA\n"); return -EBUSY; } } if (!devm_request_region(devptr, port[dev], 0x10, DRV_NAME)) { - snd_printk(KERN_ERR PFX - "I/O port region is already in use.\n"); + dev_err(devptr, "I/O port region is already in use.\n"); return -EBUSY; } vport = devm_ioport_map(devptr, port[dev], 0x10); if (!vport) { - snd_printk(KERN_ERR PFX - "I/O port cannot be iomapped.\n"); + dev_err(devptr, "I/O port cannot be iomapped.\n"); return -EBUSY; } card->private_data = (void __force *)vport; /* to make it marked as used */ if (!devm_request_region(devptr, mss_port[dev], 4, DRV_NAME)) { - snd_printk(KERN_ERR PFX - "SC-6000 port I/O port region is already in use.\n"); + dev_err(devptr, + "SC-6000 port I/O port region is already in use.\n"); return -EBUSY; } vmss_port = devm_ioport_map(devptr, mss_port[dev], 4); if (!vmss_port) { - snd_printk(KERN_ERR PFX - "MSS port I/O cannot be iomapped.\n"); + dev_err(devptr, "MSS port I/O cannot be iomapped.\n"); return -EBUSY; } - snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", - port[dev], xirq, xdma, - mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); + dev_dbg(devptr, "Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", + port[dev], xirq, xdma, + mpu_irq[dev] == SNDRV_AUTO_IRQ ? 0 : mpu_irq[dev]); - err = sc6000_init_board(vport, vmss_port, dev); + err = sc6000_init_board(devptr, vport, vmss_port, dev); if (err < 0) return err; card->private_free = snd_sc6000_free; @@ -613,25 +617,24 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) err = snd_wss_pcm(chip, 0); if (err < 0) { - snd_printk(KERN_ERR PFX - "error creating new WSS PCM device\n"); + dev_err(devptr, "error creating new WSS PCM device\n"); return err; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR PFX "error creating new WSS mixer\n"); + dev_err(devptr, "error creating new WSS mixer\n"); return err; } err = snd_sc6000_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR PFX "the mixer rewrite failed\n"); + dev_err(devptr, "the mixer rewrite failed\n"); return err; } if (snd_opl3_create(card, 0x388, 0x388 + 2, OPL3_HW_AUTO, 0, &opl3) < 0) { - snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", - 0x388, 0x388 + 2); + dev_err(devptr, "no OPL device at 0x%x-0x%x ?\n", + 0x388, 0x388 + 2); } else { err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) @@ -645,8 +648,8 @@ static int __snd_sc6000_probe(struct device *devptr, unsigned int dev) MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], NULL) < 0) - snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", - mpu_port[dev]); + dev_err(devptr, "no MPU-401 device at 0x%lx ?\n", + mpu_port[dev]); } strcpy(card->driver, DRV_NAME); From 610f04ca710cef5ab35dd14accfb0d14eaa822a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:24 +0200 Subject: [PATCH 139/410] ALSA: sscape: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The device pointer is stored in struct soundscape for calling dev_*(). Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-35-tiwai@suse.de --- sound/isa/sscape.c | 96 +++++++++++++++++++++++----------------------- 1 file changed, 48 insertions(+), 48 deletions(-) diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index cc56fafd27b1..09120e38f4c2 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -138,6 +138,7 @@ struct soundscape { struct snd_wss *chip; unsigned char midi_vol; + struct device *dev; }; #define INVALID_IRQ ((unsigned)-1) @@ -161,9 +162,9 @@ static struct snd_dma_buffer *get_dmabuf(struct soundscape *s, if (snd_dma_alloc_pages_fallback(SNDRV_DMA_TYPE_DEV, s->chip->card->dev, size, buf) < 0) { - snd_printk(KERN_ERR "sscape: Failed to allocate " - "%lu bytes for DMA\n", - size); + dev_err(s->dev, + "sscape: Failed to allocate %lu bytes for DMA\n", + size); return NULL; } } @@ -463,8 +464,7 @@ static int upload_dma_data(struct soundscape *s, const unsigned char *data, */ spin_unlock_irqrestore(&s->lock, flags); - snd_printk(KERN_ERR - "sscape: DMA upload has timed out\n"); + dev_err(s->dev, "sscape: DMA upload has timed out\n"); ret = -EAGAIN; goto _release_dma; } @@ -487,12 +487,11 @@ static int upload_dma_data(struct soundscape *s, const unsigned char *data, */ ret = 0; if (!obp_startup_ack(s, 5000)) { - snd_printk(KERN_ERR "sscape: No response " - "from on-board processor after upload\n"); + dev_err(s->dev, + "sscape: No response from on-board processor after upload\n"); ret = -EAGAIN; } else if (!host_startup_ack(s, 5000)) { - snd_printk(KERN_ERR - "sscape: SoundScape failed to initialise\n"); + dev_err(s->dev, "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -521,7 +520,7 @@ static int sscape_upload_bootblock(struct snd_card *card) ret = request_firmware(&init_fw, "scope.cod", card->dev); if (ret < 0) { - snd_printk(KERN_ERR "sscape: Error loading scope.cod"); + dev_err(card->dev, "sscape: Error loading scope.cod"); return ret; } ret = upload_dma_data(sscape, init_fw->data, init_fw->size); @@ -539,8 +538,8 @@ static int sscape_upload_bootblock(struct snd_card *card) data &= 0xf; if (ret == 0 && data > 7) { - snd_printk(KERN_ERR - "sscape: timeout reading firmware version\n"); + dev_err(card->dev, + "sscape: timeout reading firmware version\n"); ret = -EAGAIN; } @@ -561,14 +560,14 @@ static int sscape_upload_microcode(struct snd_card *card, int version) err = request_firmware(&init_fw, name, card->dev); if (err < 0) { - snd_printk(KERN_ERR "sscape: Error loading sndscape.co%d", - version); + dev_err(card->dev, "sscape: Error loading sndscape.co%d", + version); return err; } err = upload_dma_data(sscape, init_fw->data, init_fw->size); if (err == 0) - snd_printk(KERN_INFO "sscape: MIDI firmware loaded %zu KBs\n", - init_fw->size >> 10); + dev_info(card->dev, "sscape: MIDI firmware loaded %zu KBs\n", + init_fw->size >> 10); release_firmware(init_fw); @@ -783,8 +782,8 @@ _done: static int mpu401_open(struct snd_mpu401 *mpu) { if (!verify_mpu401(mpu)) { - snd_printk(KERN_ERR "sscape: MIDI disabled, " - "please load firmware\n"); + dev_err(mpu->rmidi->card->dev, + "sscape: MIDI disabled, please load firmware\n"); return -ENODEV; } @@ -871,22 +870,22 @@ static int create_ad1845(struct snd_card *card, unsigned port, err = snd_wss_pcm(chip, 0); if (err < 0) { - snd_printk(KERN_ERR "sscape: No PCM device " - "for AD1845 chip\n"); + dev_err(card->dev, + "sscape: No PCM device for AD1845 chip\n"); goto _error; } err = snd_wss_mixer(chip); if (err < 0) { - snd_printk(KERN_ERR "sscape: No mixer device " - "for AD1845 chip\n"); + dev_err(card->dev, + "sscape: No mixer device for AD1845 chip\n"); goto _error; } if (chip->hardware != WSS_HW_AD1848) { err = snd_wss_timer(chip, 0); if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " - "for AD1845 chip\n"); + dev_err(card->dev, + "sscape: No timer device for AD1845 chip\n"); goto _error; } } @@ -895,8 +894,8 @@ static int create_ad1845(struct snd_card *card, unsigned port, err = snd_ctl_add(card, snd_ctl_new1(&midi_mixer_ctl, chip)); if (err < 0) { - snd_printk(KERN_ERR "sscape: Could not create " - "MIDI mixer control\n"); + dev_err(card->dev, + "sscape: Could not create MIDI mixer control\n"); goto _error; } } @@ -932,8 +931,8 @@ static int create_sscape(int dev, struct snd_card *card) */ io_res = devm_request_region(card->dev, port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR - "sscape: can't grab port 0x%lx\n", port[dev]); + dev_err(card->dev, + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -941,8 +940,8 @@ static int create_sscape(int dev, struct snd_card *card) wss_res = devm_request_region(card->dev, wss_port[dev], 4, "SoundScape"); if (!wss_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%lx\n", - wss_port[dev]); + dev_err(card->dev, "sscape: can't grab port 0x%lx\n", + wss_port[dev]); return -EBUSY; } } @@ -952,7 +951,7 @@ static int create_sscape(int dev, struct snd_card *card) */ err = snd_devm_request_dma(card->dev, dma[dev], "SoundScape"); if (err < 0) { - snd_printk(KERN_ERR "sscape: can't grab DMA %d\n", dma[dev]); + dev_err(card->dev, "sscape: can't grab DMA %d\n", dma[dev]); return err; } @@ -962,7 +961,7 @@ static int create_sscape(int dev, struct snd_card *card) sscape->io_base = port[dev]; if (!detect_sscape(sscape, wss_port[dev])) { - printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", + dev_err(card->dev, "sscape: hardware not detected at 0x%x\n", sscape->io_base); return -ENODEV; } @@ -985,21 +984,21 @@ static int create_sscape(int dev, struct snd_card *card) break; } - printk(KERN_INFO "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", - name, sscape->io_base, irq[dev], dma[dev]); + dev_info(card->dev, "sscape: %s card detected at 0x%x, using IRQ %d, DMA %d\n", + name, sscape->io_base, irq[dev], dma[dev]); /* * Check that the user didn't pass us garbage data ... */ irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); + dev_err(card->dev, "sscape: Invalid IRQ %d\n", irq[dev]); return -ENXIO; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { - snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); + dev_err(card->dev, "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } @@ -1043,9 +1042,9 @@ static int create_sscape(int dev, struct snd_card *card) err = create_ad1845(card, wss_port[dev], irq[dev], dma[dev], dma2[dev]); if (err < 0) { - snd_printk(KERN_ERR - "sscape: No AD1845 device at 0x%lx, IRQ %d\n", - wss_port[dev], irq[dev]); + dev_err(card->dev, + "sscape: No AD1845 device at 0x%lx, IRQ %d\n", + wss_port[dev], irq[dev]); return err; } strcpy(card->driver, "SoundScape"); @@ -1065,9 +1064,9 @@ static int create_sscape(int dev, struct snd_card *card) err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { - snd_printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%lx\n", - port[dev]); + dev_err(card->dev, + "sscape: Failed to create MPU-401 device at 0x%lx\n", + port[dev]); return err; } @@ -1110,9 +1109,8 @@ static int snd_sscape_match(struct device *pdev, unsigned int i) if (irq[i] == SNDRV_AUTO_IRQ || mpu_irq[i] == SNDRV_AUTO_IRQ || dma[i] == SNDRV_AUTO_DMA) { - printk(KERN_INFO - "sscape: insufficient parameters, " - "need IO, IRQ, MPU-IRQ and DMA\n"); + dev_info(pdev, + "sscape: insufficient parameters, need IO, IRQ, MPU-IRQ and DMA\n"); return 0; } @@ -1131,6 +1129,7 @@ static int snd_sscape_probe(struct device *pdev, unsigned int dev) return ret; sscape = get_card_soundscape(card); + sscape->dev = pdev; sscape->type = SSCAPE; dma[dev] &= 0x03; @@ -1141,7 +1140,7 @@ static int snd_sscape_probe(struct device *pdev, unsigned int dev) ret = snd_card_register(card); if (ret < 0) { - snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); + dev_err(pdev, "sscape: Failed to register sound card\n"); return ret; } dev_set_drvdata(pdev, card); @@ -1194,7 +1193,7 @@ static int sscape_pnp_detect(struct pnp_card_link *pcard, if (!pnp_is_active(dev)) { if (pnp_activate_dev(dev) < 0) { - snd_printk(KERN_INFO "sscape: device is inactive\n"); + dev_info(&dev->dev, "sscape: device is inactive\n"); return -EBUSY; } } @@ -1210,6 +1209,7 @@ static int sscape_pnp_detect(struct pnp_card_link *pcard, return ret; sscape = get_card_soundscape(card); + sscape->dev = card->dev; /* * Identify card model ... @@ -1240,7 +1240,7 @@ static int sscape_pnp_detect(struct pnp_card_link *pcard, ret = snd_card_register(card); if (ret < 0) { - snd_printk(KERN_ERR "sscape: Failed to register sound card\n"); + dev_err(card->dev, "sscape: Failed to register sound card\n"); return ret; } From 8b4ac5429938dd5f1fbf2eea0687f08cbcccb6be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:25 +0200 Subject: [PATCH 140/410] ALSA: wavefront: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-36-tiwai@suse.de --- include/sound/snd_wavefront.h | 4 - sound/isa/wavefront/wavefront.c | 61 ++++---- sound/isa/wavefront/wavefront_fx.c | 36 ++--- sound/isa/wavefront/wavefront_midi.c | 15 +- sound/isa/wavefront/wavefront_synth.c | 196 +++++++++++++------------- 5 files changed, 158 insertions(+), 154 deletions(-) diff --git a/include/sound/snd_wavefront.h b/include/sound/snd_wavefront.h index 55053557c898..27f7e8a477c2 100644 --- a/include/sound/snd_wavefront.h +++ b/include/sound/snd_wavefront.h @@ -137,8 +137,4 @@ extern int snd_wavefront_fx_ioctl (struct snd_hwdep *, extern int snd_wavefront_fx_open (struct snd_hwdep *, struct file *); extern int snd_wavefront_fx_release (struct snd_hwdep *, struct file *); -/* prefix in all snd_printk() delivered messages */ - -#define LOGNAME "WaveFront: " - #endif /* __SOUND_SND_WAVEFRONT_H__ */ diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index e6e46a0266b0..621ab420a60f 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -140,7 +140,7 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "PnP WSS pnp configure failure\n"); + dev_err(&pdev->dev, "PnP WSS pnp configure failure\n"); return err; } @@ -156,7 +156,7 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "PnP ICS2115 pnp configure failure\n"); + dev_err(&pdev->dev, "PnP ICS2115 pnp configure failure\n"); return err; } @@ -174,26 +174,27 @@ snd_wavefront_pnp (int dev, snd_wavefront_card_t *acard, struct pnp_card_link *c err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "PnP MPU401 pnp configure failure\n"); + dev_err(&pdev->dev, "PnP MPU401 pnp configure failure\n"); cs4232_mpu_port[dev] = SNDRV_AUTO_PORT; } else { cs4232_mpu_port[dev] = pnp_port_start(pdev, 0); cs4232_mpu_irq[dev] = pnp_irq(pdev, 0); } - snd_printk (KERN_INFO "CS4232 MPU: port=0x%lx, irq=%i\n", - cs4232_mpu_port[dev], - cs4232_mpu_irq[dev]); + dev_info(&pdev->dev, "CS4232 MPU: port=0x%lx, irq=%i\n", + cs4232_mpu_port[dev], + cs4232_mpu_irq[dev]); } - snd_printdd ("CS4232: pcm port=0x%lx, fm port=0x%lx, dma1=%i, dma2=%i, irq=%i\nICS2115: port=0x%lx, irq=%i\n", - cs4232_pcm_port[dev], - fm_port[dev], - dma1[dev], - dma2[dev], - cs4232_pcm_irq[dev], - ics2115_port[dev], - ics2115_irq[dev]); + dev_dbg(&pdev->dev, + "CS4232: pcm port=0x%lx, fm port=0x%lx, dma1=%i, dma2=%i, irq=%i\nICS2115: port=0x%lx, irq=%i\n", + cs4232_pcm_port[dev], + fm_port[dev], + dma1[dev], + dma2[dev], + cs4232_pcm_irq[dev], + ics2115_port[dev], + ics2115_irq[dev]); return 0; } @@ -251,7 +252,7 @@ static struct snd_hwdep *snd_wavefront_new_fx(struct snd_card *card, struct snd_hwdep *fx_processor; if (snd_wavefront_fx_start (&acard->wavefront)) { - snd_printk (KERN_ERR "cannot initialize YSS225 FX processor"); + dev_err(card->dev, "cannot initialize YSS225 FX processor"); return NULL; } @@ -282,7 +283,7 @@ static struct snd_rawmidi *snd_wavefront_new_midi(struct snd_card *card, first = 0; acard->wavefront.midi.base = port; if (snd_wavefront_midi_start (acard)) { - snd_printk (KERN_ERR "cannot initialize MIDI interface\n"); + dev_err(card->dev, "cannot initialize MIDI interface\n"); return NULL; } } @@ -349,7 +350,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) cs4232_pcm_irq[dev], dma1[dev], dma2[dev], WSS_HW_DETECT, 0, &chip); if (err < 0) { - snd_printk(KERN_ERR "can't allocate WSS device\n"); + dev_err(card->dev, "can't allocate WSS device\n"); return err; } @@ -369,7 +370,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) err = snd_opl3_create(card, fm_port[dev], fm_port[dev] + 2, OPL3_HW_OPL3_CS, 0, &opl3); if (err < 0) { - snd_printk (KERN_ERR "can't allocate or detect OPL3 synth\n"); + dev_err(card->dev, "can't allocate or detect OPL3 synth\n"); return err; } @@ -385,14 +386,14 @@ snd_wavefront_probe (struct snd_card *card, int dev) devm_request_region(card->dev, ics2115_port[dev], 16, "ICS2115"); if (acard->wavefront.res_base == NULL) { - snd_printk(KERN_ERR "unable to grab ICS2115 i/o region 0x%lx-0x%lx\n", - ics2115_port[dev], ics2115_port[dev] + 16 - 1); + dev_err(card->dev, "unable to grab ICS2115 i/o region 0x%lx-0x%lx\n", + ics2115_port[dev], ics2115_port[dev] + 16 - 1); return -EBUSY; } if (devm_request_irq(card->dev, ics2115_irq[dev], snd_wavefront_ics2115_interrupt, 0, "ICS2115", acard)) { - snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]); + dev_err(card->dev, "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]); return -EBUSY; } @@ -402,7 +403,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) wavefront_synth = snd_wavefront_new_synth(card, hw_dev, acard); if (wavefront_synth == NULL) { - snd_printk (KERN_ERR "can't create WaveFront synth device\n"); + dev_err(card->dev, "can't create WaveFront synth device\n"); return -ENOMEM; } @@ -414,7 +415,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) err = snd_wss_mixer(chip); if (err < 0) { - snd_printk (KERN_ERR "can't allocate mixer device\n"); + dev_err(card->dev, "can't allocate mixer device\n"); return err; } @@ -425,7 +426,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) cs4232_mpu_port[dev], 0, cs4232_mpu_irq[dev], NULL); if (err < 0) { - snd_printk (KERN_ERR "can't allocate CS4232 MPU-401 device\n"); + dev_err(card->dev, "can't allocate CS4232 MPU-401 device\n"); return err; } midi_dev++; @@ -441,7 +442,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) ics2115_port[dev], internal_mpu); if (ics2115_internal_rmidi == NULL) { - snd_printk (KERN_ERR "can't setup ICS2115 internal MIDI device\n"); + dev_err(card->dev, "can't setup ICS2115 internal MIDI device\n"); return -ENOMEM; } midi_dev++; @@ -457,7 +458,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) ics2115_port[dev], external_mpu); if (ics2115_external_rmidi == NULL) { - snd_printk (KERN_ERR "can't setup ICS2115 external MIDI device\n"); + dev_err(card->dev, "can't setup ICS2115 external MIDI device\n"); return -ENOMEM; } midi_dev++; @@ -471,7 +472,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) acard, ics2115_port[dev]); if (fx_processor == NULL) { - snd_printk (KERN_ERR "can't setup FX device\n"); + dev_err(card->dev, "can't setup FX device\n"); return -ENOMEM; } @@ -525,11 +526,11 @@ static int snd_wavefront_isa_match(struct device *pdev, return 0; #endif if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "specify CS4232 port\n"); + dev_err(pdev, "specify CS4232 port\n"); return 0; } if (ics2115_port[dev] == SNDRV_AUTO_PORT) { - snd_printk(KERN_ERR "specify ICS2115 port\n"); + dev_err(pdev, "specify ICS2115 port\n"); return 0; } return 1; @@ -585,7 +586,7 @@ static int snd_wavefront_pnp_detect(struct pnp_card_link *pcard, if (snd_wavefront_pnp (dev, card->private_data, pcard, pid) < 0) { if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk (KERN_ERR "isapnp detection failed\n"); + dev_err(card->dev, "isapnp detection failed\n"); return -ENODEV; } } diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index 0273b7dfaf12..beca35ce04f3 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -38,7 +38,7 @@ wavefront_fx_idle (snd_wavefront_t *dev) } if (x & 0x80) { - snd_printk ("FX device never idle.\n"); + dev_err(dev->card->dev, "FX device never idle.\n"); return 0; } @@ -64,14 +64,14 @@ wavefront_fx_memset (snd_wavefront_t *dev, unsigned short *data) { if (page < 0 || page > 7) { - snd_printk ("FX memset: " - "page must be >= 0 and <= 7\n"); + dev_err(dev->card->dev, + "FX memset: page must be >= 0 and <= 7\n"); return -EINVAL; } if (addr < 0 || addr > 0x7f) { - snd_printk ("FX memset: " - "addr must be >= 0 and <= 7f\n"); + dev_err(dev->card->dev, + "FX memset: addr must be >= 0 and <= 7f\n"); return -EINVAL; } @@ -83,7 +83,7 @@ wavefront_fx_memset (snd_wavefront_t *dev, outb ((data[0] >> 8), dev->fx_dsp_msb); outb ((data[0] & 0xff), dev->fx_dsp_lsb); - snd_printk ("FX: addr %d:%x set to 0x%x\n", + dev_err(dev->card->dev, "FX: addr %d:%x set to 0x%x\n", page, addr, data[0]); } else { @@ -102,9 +102,9 @@ wavefront_fx_memset (snd_wavefront_t *dev, } if (i != cnt) { - snd_printk ("FX memset " - "(0x%x, 0x%x, 0x%lx, %d) incomplete\n", - page, addr, (unsigned long) data, cnt); + dev_err(dev->card->dev, + "FX memset (0x%x, 0x%x, 0x%lx, %d) incomplete\n", + page, addr, (unsigned long) data, cnt); return -EIO; } } @@ -123,7 +123,7 @@ snd_wavefront_fx_detect (snd_wavefront_t *dev) */ if (inb (dev->fx_status) & 0x80) { - snd_printk ("Hmm, probably a Maui or Tropez.\n"); + dev_err(dev->card->dev, "Hmm, probably a Maui or Tropez.\n"); return -1; } @@ -180,15 +180,15 @@ snd_wavefront_fx_ioctl (struct snd_hwdep *sdev, struct file *file, case WFFX_MEMSET: if (r.data[2] <= 0) { - snd_printk ("cannot write " - "<= 0 bytes to FX\n"); + dev_err(dev->card->dev, + "cannot write <= 0 bytes to FX\n"); return -EIO; } else if (r.data[2] == 1) { pd = (unsigned short *) &r.data[3]; } else { if (r.data[2] > 256) { - snd_printk ("cannot write " - "> 512 bytes to FX\n"); + dev_err(dev->card->dev, + "cannot write > 512 bytes to FX\n"); return -EIO; } page_data = memdup_array_user((unsigned char __user *) @@ -208,8 +208,8 @@ snd_wavefront_fx_ioctl (struct snd_hwdep *sdev, struct file *file, break; default: - snd_printk ("FX: ioctl %d not yet supported\n", - r.request); + dev_err(dev->card->dev, "FX: ioctl %d not yet supported\n", + r.request); return -ENOTTY; } return err; @@ -254,8 +254,8 @@ snd_wavefront_fx_start (snd_wavefront_t *dev) goto out; } } else { - snd_printk(KERN_ERR "invalid address" - " in register data\n"); + dev_err(dev->card->dev, + "invalid address in register data\n"); err = -1; goto out; } diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c index 72e775ac7ad7..ead8cbe638de 100644 --- a/sound/isa/wavefront/wavefront_midi.c +++ b/sound/isa/wavefront/wavefront_midi.c @@ -501,7 +501,8 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card) for (i = 0; i < 30000 && !output_ready (midi); i++); if (!output_ready (midi)) { - snd_printk ("MIDI interface not ready for command\n"); + dev_err(card->wavefront.card->dev, + "MIDI interface not ready for command\n"); return -1; } @@ -523,7 +524,8 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card) } if (!ok) { - snd_printk ("cannot set UART mode for MIDI interface"); + dev_err(card->wavefront.card->dev, + "cannot set UART mode for MIDI interface"); dev->interrupts_are_midi = 0; return -1; } @@ -531,7 +533,8 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card) /* Route external MIDI to WaveFront synth (by default) */ if (snd_wavefront_cmd (dev, WFC_MISYNTH_ON, rbuf, wbuf)) { - snd_printk ("can't enable MIDI-IN-2-synth routing.\n"); + dev_warn(card->wavefront.card->dev, + "can't enable MIDI-IN-2-synth routing.\n"); /* XXX error ? */ } @@ -547,14 +550,16 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card) */ if (snd_wavefront_cmd (dev, WFC_VMIDI_OFF, rbuf, wbuf)) { - snd_printk ("virtual MIDI mode not disabled\n"); + dev_warn(card->wavefront.card->dev, + "virtual MIDI mode not disabled\n"); return 0; /* We're OK, but missing the external MIDI dev */ } snd_wavefront_midi_enable_virtual (card); if (snd_wavefront_cmd (dev, WFC_VMIDI_ON, rbuf, wbuf)) { - snd_printk ("cannot enable virtual MIDI mode.\n"); + dev_warn(card->wavefront.card->dev, + "cannot enable virtual MIDI mode.\n"); snd_wavefront_midi_disable_virtual (card); } return 0; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 13ce96148fa3..bd679e2da154 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -116,7 +116,7 @@ MODULE_PARM_DESC(osrun_time, "how many seconds to wait for the ICS2115 OS"); #define DPRINT(cond, ...) \ if ((dev->debug & (cond)) == (cond)) { \ - snd_printk (__VA_ARGS__); \ + pr_debug(__VA_ARGS__); \ } #else #define DPRINT(cond, args...) @@ -341,7 +341,7 @@ snd_wavefront_cmd (snd_wavefront_t *dev, wfcmd = wavefront_get_command(cmd); if (!wfcmd) { - snd_printk ("command 0x%x not supported.\n", + dev_err(dev->card->dev, "command 0x%x not supported.\n", cmd); return 1; } @@ -623,7 +623,7 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) /* check sample status */ if (snd_wavefront_cmd (dev, WFC_GET_NSAMPLES, rbuf, wbuf)) { - snd_printk ("cannot request sample count.\n"); + dev_err(dev->card->dev, "cannot request sample count.\n"); return -1; } @@ -635,8 +635,8 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) wbuf[1] = i >> 7; if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) { - snd_printk(KERN_WARNING "cannot identify sample " - "type of slot %d\n", i); + dev_warn(dev->card->dev, + "cannot identify sample type of slot %d\n", i); dev->sample_status[i] = WF_ST_EMPTY; continue; } @@ -661,9 +661,9 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) break; default: - snd_printk ("unknown sample type for " - "slot %d (0x%x)\n", - i, rbuf[0]); + dev_err(dev->card->dev, + "unknown sample type for slot %d (0x%x)\n", + i, rbuf[0]); } if (rbuf[0] != WF_ST_EMPTY) { @@ -671,9 +671,10 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) } } - snd_printk ("%d samples used (%d real, %d aliases, %d multi), " - "%d empty\n", dev->samples_used, sc_real, sc_alias, sc_multi, - WF_MAX_SAMPLE - dev->samples_used); + dev_info(dev->card->dev, + "%d samples used (%d real, %d aliases, %d multi), %d empty\n", + dev->samples_used, sc_real, sc_alias, sc_multi, + WF_MAX_SAMPLE - dev->samples_used); return (0); @@ -706,8 +707,8 @@ wavefront_get_patch_status (snd_wavefront_t *dev) } else if (x == 3) { /* Bad patch number */ dev->patch_status[i] = 0; } else { - snd_printk ("upload patch " - "error 0x%x\n", x); + dev_err(dev->card->dev, + "upload patch error 0x%x\n", x); dev->patch_status[i] = 0; return 1; } @@ -724,7 +725,8 @@ wavefront_get_patch_status (snd_wavefront_t *dev) } } - snd_printk ("%d patch slots filled, %d in use\n", cnt, cnt2); + dev_info(dev->card->dev, "%d patch slots filled, %d in use\n", + cnt, cnt2); return (0); } @@ -760,8 +762,8 @@ wavefront_get_program_status (snd_wavefront_t *dev) } else if (x == 1) { /* Bad program number */ dev->prog_status[i] = 0; } else { - snd_printk ("upload program " - "error 0x%x\n", x); + dev_err(dev->card->dev, + "upload program error 0x%x\n", x); dev->prog_status[i] = 0; } } @@ -772,7 +774,7 @@ wavefront_get_program_status (snd_wavefront_t *dev) } } - snd_printk ("%d programs slots in use\n", cnt); + dev_info(dev->card->dev, "%d programs slots in use\n", cnt); return (0); } @@ -796,7 +798,7 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) munge_buf ((unsigned char *)&header->hdr.p, bptr, WF_PATCH_BYTES); if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PATCH, NULL, buf)) { - snd_printk ("download patch failed\n"); + dev_err(dev->card->dev, "download patch failed\n"); return -EIO; } @@ -837,7 +839,7 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) munge_buf ((unsigned char *)&header->hdr.pr, &buf[1], WF_PROGRAM_BYTES); if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_PROGRAM, NULL, buf)) { - snd_printk ("download patch failed\n"); + dev_err(dev->card->dev, "download patch failed\n"); return -EIO; } @@ -851,7 +853,7 @@ wavefront_freemem (snd_wavefront_t *dev) char rbuf[8]; if (snd_wavefront_cmd (dev, WFC_REPORT_FREE_MEMORY, rbuf, NULL)) { - snd_printk ("can't get memory stats.\n"); + dev_err(dev->card->dev, "can't get memory stats.\n"); return -1; } else { return demunge_int32 (rbuf, 4); @@ -901,7 +903,7 @@ wavefront_send_sample (snd_wavefront_t *dev, x = wavefront_find_free_sample(dev); if (x < 0) return -ENOMEM; - snd_printk ("unspecified sample => %d\n", x); + dev_info(dev->card->dev, "unspecified sample => %d\n", x); header->number = x; } @@ -935,9 +937,9 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dev->rom_samples_rdonly) { if (dev->sample_status[header->number] & WF_SLOT_ROM) { - snd_printk ("sample slot %d " - "write protected\n", - header->number); + dev_err(dev->card->dev, + "sample slot %d write protected\n", + header->number); return -EACCES; } } @@ -949,9 +951,9 @@ wavefront_send_sample (snd_wavefront_t *dev, dev->freemem = wavefront_freemem (dev); if (dev->freemem < (int)header->size) { - snd_printk ("insufficient memory to " - "load %d byte sample.\n", - header->size); + dev_err(dev->card->dev, + "insufficient memory to load %d byte sample.\n", + header->size); return -ENOMEM; } @@ -960,8 +962,8 @@ wavefront_send_sample (snd_wavefront_t *dev, skip = WF_GET_CHANNEL(&header->hdr.s); if (skip > 0 && header->hdr.s.SampleResolution != LINEAR_16BIT) { - snd_printk ("channel selection only " - "possible on 16-bit samples"); + dev_err(dev->card->dev, + "channel selection only possible on 16-bit samples"); return -EINVAL; } @@ -1057,8 +1059,8 @@ wavefront_send_sample (snd_wavefront_t *dev, header->size ? WFC_DOWNLOAD_SAMPLE : WFC_DOWNLOAD_SAMPLE_HEADER, NULL, sample_hdr)) { - snd_printk ("sample %sdownload refused.\n", - header->size ? "" : "header "); + dev_err(dev->card->dev, "sample %sdownload refused.\n", + header->size ? "" : "header "); return -EIO; } @@ -1083,8 +1085,8 @@ wavefront_send_sample (snd_wavefront_t *dev, } if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_BLOCK, NULL, NULL)) { - snd_printk ("download block " - "request refused.\n"); + dev_err(dev->card->dev, + "download block request refused.\n"); return -EIO; } @@ -1145,13 +1147,13 @@ wavefront_send_sample (snd_wavefront_t *dev, dma_ack = wavefront_read(dev); if (dma_ack != WF_DMA_ACK) { if (dma_ack == -1) { - snd_printk ("upload sample " - "DMA ack timeout\n"); + dev_err(dev->card->dev, + "upload sample DMA ack timeout\n"); return -EIO; } else { - snd_printk ("upload sample " - "DMA ack error 0x%x\n", - dma_ack); + dev_err(dev->card->dev, + "upload sample DMA ack error 0x%x\n", + dma_ack); return -EIO; } } @@ -1195,7 +1197,7 @@ wavefront_send_alias (snd_wavefront_t *dev, wavefront_patch_info *header) munge_int32 (*(&header->hdr.a.FrequencyBias+1), &alias_hdr[23], 2); if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_SAMPLE_ALIAS, NULL, alias_hdr)) { - snd_printk ("download alias failed.\n"); + dev_err(dev->card->dev, "download alias failed.\n"); return -EIO; } @@ -1248,7 +1250,7 @@ wavefront_send_multisample (snd_wavefront_t *dev, wavefront_patch_info *header) if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_MULTISAMPLE, (unsigned char *) (long) ((num_samples*2)+3), msample_hdr)) { - snd_printk ("download of multisample failed.\n"); + dev_err(dev->card->dev, "download of multisample failed.\n"); kfree(msample_hdr); return -EIO; } @@ -1271,7 +1273,7 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, munge_int32 (header->number, number, 2); if (snd_wavefront_cmd (dev, WFC_UPLOAD_MULTISAMPLE, log_ns, number)) { - snd_printk ("upload multisample failed.\n"); + dev_err(dev->card->dev, "upload multisample failed.\n"); return -EIO; } @@ -1290,16 +1292,16 @@ wavefront_fetch_multisample (snd_wavefront_t *dev, val = wavefront_read(dev); if (val == -1) { - snd_printk ("upload multisample failed " - "during sample loop.\n"); + dev_err(dev->card->dev, + "upload multisample failed during sample loop.\n"); return -EIO; } d[0] = val; val = wavefront_read(dev); if (val == -1) { - snd_printk ("upload multisample failed " - "during sample loop.\n"); + dev_err(dev->card->dev, + "upload multisample failed during sample loop.\n"); return -EIO; } d[1] = val; @@ -1334,7 +1336,7 @@ wavefront_send_drum (snd_wavefront_t *dev, wavefront_patch_info *header) } if (snd_wavefront_cmd (dev, WFC_DOWNLOAD_EDRUM_PROGRAM, NULL, drumbuf)) { - snd_printk ("download drum failed.\n"); + dev_err(dev->card->dev, "download drum failed.\n"); return -EIO; } @@ -1352,7 +1354,7 @@ wavefront_find_free_sample (snd_wavefront_t *dev) return i; } } - snd_printk ("no free sample slots!\n"); + dev_err(dev->card->dev, "no free sample slots!\n"); return -1; } @@ -1368,7 +1370,7 @@ wavefront_find_free_patch (snd_wavefront_t *dev) return i; } } - snd_printk ("no free patch slots!\n"); + dev_err(dev->card->dev, "no free patch slots!\n"); return -1; } #endif @@ -1385,7 +1387,7 @@ wavefront_load_patch (snd_wavefront_t *dev, const char __user *addr) if (copy_from_user (header, addr, sizeof(wavefront_patch_info) - sizeof(wavefront_any))) { - snd_printk ("bad address for load patch.\n"); + dev_err(dev->card->dev, "bad address for load patch.\n"); err = -EFAULT; goto __error; } @@ -1463,8 +1465,8 @@ wavefront_load_patch (snd_wavefront_t *dev, const char __user *addr) break; default: - snd_printk ("unknown patch type %d.\n", - header->subkey); + dev_err(dev->card->dev, "unknown patch type %d.\n", + header->subkey); err = -EINVAL; break; } @@ -1527,13 +1529,13 @@ wavefront_synth_control (snd_wavefront_card_t *acard, switch (wc->cmd) { case WFC_DISABLE_INTERRUPTS: - snd_printk ("interrupts disabled.\n"); + dev_dbg(dev->card->dev, "interrupts disabled.\n"); outb (0x80|0x20, dev->control_port); dev->interrupts_are_midi = 1; return 0; case WFC_ENABLE_INTERRUPTS: - snd_printk ("interrupts enabled.\n"); + dev_dbg(dev->card->dev, "interrupts enabled.\n"); outb (0x80|0x40|0x20, dev->control_port); dev->interrupts_are_midi = 1; return 0; @@ -1550,7 +1552,7 @@ wavefront_synth_control (snd_wavefront_card_t *acard, case WFC_IDENTIFY_SLOT_TYPE: i = wc->wbuf[0] | (wc->wbuf[1] << 7); if (i <0 || i >= WF_MAX_SAMPLE) { - snd_printk ("invalid slot ID %d\n", + dev_err(dev->card->dev, "invalid slot ID %d\n", i); wc->status = EINVAL; return -EINVAL; @@ -1561,7 +1563,7 @@ wavefront_synth_control (snd_wavefront_card_t *acard, case WFC_DEBUG_DRIVER: dev->debug = wc->wbuf[0]; - snd_printk ("debug = 0x%x\n", dev->debug); + dev_dbg(dev->card->dev, "debug = 0x%x\n", dev->debug); return 0; case WFC_UPLOAD_PATCH: @@ -1578,8 +1580,8 @@ wavefront_synth_control (snd_wavefront_card_t *acard, return 0; case WFC_UPLOAD_SAMPLE_ALIAS: - snd_printk ("support for sample alias upload " - "being considered.\n"); + dev_err(dev->card->dev, + "support for sample alias upload being considered.\n"); wc->status = EINVAL; return -EINVAL; } @@ -1620,9 +1622,8 @@ wavefront_synth_control (snd_wavefront_card_t *acard, break; case WFC_UPLOAD_SAMPLE_ALIAS: - snd_printk ("support for " - "sample aliases still " - "being considered.\n"); + dev_err(dev->card->dev, + "support for sample aliases still being considered.\n"); break; case WFC_VMIDI_OFF: @@ -1760,7 +1761,7 @@ snd_wavefront_internal_interrupt (snd_wavefront_card_t *card) */ static int -snd_wavefront_interrupt_bits (int irq) +snd_wavefront_interrupt_bits(snd_wavefront_t *dev, int irq) { int bits; @@ -1780,7 +1781,7 @@ snd_wavefront_interrupt_bits (int irq) break; default: - snd_printk ("invalid IRQ %d\n", irq); + dev_err(dev->card->dev, "invalid IRQ %d\n", irq); bits = -1; } @@ -1815,7 +1816,7 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) /* IRQ already checked */ - bits = snd_wavefront_interrupt_bits (dev->irq); + bits = snd_wavefront_interrupt_bits(dev, dev->irq); /* try reset of port */ @@ -1885,7 +1886,7 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) */ if (!dev->irq_ok) { - snd_printk ("intr not received after h/w un-reset.\n"); + dev_err(dev->card->dev, "intr not received after h/w un-reset.\n"); goto gone_bad; } @@ -1909,18 +1910,18 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) dev->data_port, ramcheck_time*HZ); if (!dev->irq_ok) { - snd_printk ("post-RAM-check interrupt not received.\n"); + dev_err(dev->card->dev, "post-RAM-check interrupt not received.\n"); goto gone_bad; } if (!wavefront_wait (dev, STAT_CAN_READ)) { - snd_printk ("no response to HW version cmd.\n"); + dev_err(dev->card->dev, "no response to HW version cmd.\n"); goto gone_bad; } hwv[0] = wavefront_read(dev); if (hwv[0] == -1) { - snd_printk ("board not responding correctly.\n"); + dev_err(dev->card->dev, "board not responding correctly.\n"); goto gone_bad; } @@ -1932,11 +1933,11 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) hwv[0] = wavefront_read(dev); if (hwv[0] == -1) { - snd_printk ("on-board RAM test failed " - "(bad error code).\n"); + dev_err(dev->card->dev, + "on-board RAM test failed (bad error code).\n"); } else { - snd_printk ("on-board RAM test failed " - "(error code: 0x%x).\n", + dev_err(dev->card->dev, + "on-board RAM test failed (error code: 0x%x).\n", hwv[0]); } goto gone_bad; @@ -1946,12 +1947,12 @@ wavefront_reset_to_cleanliness (snd_wavefront_t *dev) hwv[1] = wavefront_read(dev); if (hwv[1] == -1) { - snd_printk ("incorrect h/w response.\n"); + dev_err(dev->card->dev, "incorrect h/w response.\n"); goto gone_bad; } - snd_printk ("hardware version %d.%d\n", - hwv[0], hwv[1]); + dev_info(dev->card->dev, "hardware version %d.%d\n", + hwv[0], hwv[1]); return 0; @@ -1971,7 +1972,7 @@ wavefront_download_firmware (snd_wavefront_t *dev, char *path) err = request_firmware(&firmware, path, dev->card->dev); if (err < 0) { - snd_printk(KERN_ERR "firmware (%s) download failed!!!\n", path); + dev_err(dev->card->dev, "firmware (%s) download failed!!!\n", path); return 1; } @@ -1982,16 +1983,16 @@ wavefront_download_firmware (snd_wavefront_t *dev, char *path) if (section_length == 0) break; if (section_length < 0 || section_length > WF_SECTION_MAX) { - snd_printk(KERN_ERR - "invalid firmware section length %d\n", - section_length); + dev_err(dev->card->dev, + "invalid firmware section length %d\n", + section_length); goto failure; } buf++; len++; if (firmware->size < len + section_length) { - snd_printk(KERN_ERR "firmware section read error.\n"); + dev_err(dev->card->dev, "firmware section read error.\n"); goto failure; } @@ -2008,15 +2009,14 @@ wavefront_download_firmware (snd_wavefront_t *dev, char *path) /* get ACK */ if (!wavefront_wait(dev, STAT_CAN_READ)) { - snd_printk(KERN_ERR "time out for firmware ACK.\n"); + dev_err(dev->card->dev, "time out for firmware ACK.\n"); goto failure; } err = inb(dev->data_port); if (err != WF_ACK) { - snd_printk(KERN_ERR - "download of section #%d not " - "acknowledged, ack = 0x%x\n", - section_cnt_downloaded + 1, err); + dev_err(dev->card->dev, + "download of section #%d not acknowledged, ack = 0x%x\n", + section_cnt_downloaded + 1, err); goto failure; } @@ -2028,7 +2028,7 @@ wavefront_download_firmware (snd_wavefront_t *dev, char *path) failure: release_firmware(firmware); - snd_printk(KERN_ERR "firmware download failed!!!\n"); + dev_err(dev->card->dev, "firmware download failed!!!\n"); return 1; } @@ -2040,7 +2040,7 @@ wavefront_do_reset (snd_wavefront_t *dev) char voices[1]; if (wavefront_reset_to_cleanliness (dev)) { - snd_printk ("hw reset failed.\n"); + dev_err(dev->card->dev, "hw reset failed.\n"); goto gone_bad; } @@ -2064,7 +2064,7 @@ wavefront_do_reset (snd_wavefront_t *dev) (osrun_time*HZ)); if (!dev->irq_ok) { - snd_printk ("no post-OS interrupt.\n"); + dev_err(dev->card->dev, "no post-OS interrupt.\n"); goto gone_bad; } @@ -2074,7 +2074,7 @@ wavefront_do_reset (snd_wavefront_t *dev) dev->data_port, (10*HZ)); if (!dev->irq_ok) { - snd_printk ("no post-OS interrupt(2).\n"); + dev_err(dev->card->dev, "no post-OS interrupt(2).\n"); goto gone_bad; } @@ -2094,20 +2094,20 @@ wavefront_do_reset (snd_wavefront_t *dev) if (dev->freemem < 0) goto gone_bad; - snd_printk ("available DRAM %dk\n", dev->freemem / 1024); + dev_info(dev->card->dev, "available DRAM %dk\n", dev->freemem / 1024); if (wavefront_write (dev, 0xf0) || wavefront_write (dev, 1) || (wavefront_read (dev) < 0)) { dev->debug = 0; - snd_printk ("MPU emulation mode not set.\n"); + dev_err(dev->card->dev, "MPU emulation mode not set.\n"); goto gone_bad; } voices[0] = 32; if (snd_wavefront_cmd (dev, WFC_SET_NVOICES, NULL, voices)) { - snd_printk ("cannot set number of voices to 32.\n"); + dev_err(dev->card->dev, "cannot set number of voices to 32.\n"); goto gone_bad; } @@ -2187,8 +2187,8 @@ snd_wavefront_detect (snd_wavefront_card_t *card) dev->fw_version[0] = rbuf[0]; dev->fw_version[1] = rbuf[1]; - snd_printk ("firmware %d.%d already loaded.\n", - rbuf[0], rbuf[1]); + dev_info(dev->card->dev, "firmware %d.%d already loaded.\n", + rbuf[0], rbuf[1]); /* check that a command actually works */ @@ -2197,22 +2197,24 @@ snd_wavefront_detect (snd_wavefront_card_t *card) dev->hw_version[0] = rbuf[0]; dev->hw_version[1] = rbuf[1]; } else { - snd_printk ("not raw, but no " - "hardware version!\n"); + dev_err(dev->card->dev, + "not raw, but no hardware version!\n"); return -1; } if (!wf_raw) { return 0; } else { - snd_printk ("reloading firmware as you requested.\n"); + dev_info(dev->card->dev, + "reloading firmware as you requested.\n"); dev->israw = 1; } } else { dev->israw = 1; - snd_printk ("no response to firmware probe, assume raw.\n"); + dev_info(dev->card->dev, + "no response to firmware probe, assume raw.\n"); } From 661c43fcdf622dffdf2d4f4e8d3276a4df59b920 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:26 +0200 Subject: [PATCH 141/410] ALSA: wss: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-37-tiwai@suse.de --- sound/isa/wss/wss_lib.c | 178 ++++++++++++++++++++-------------------- 1 file changed, 90 insertions(+), 88 deletions(-) diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 026061b55ee9..9c655789574d 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -187,15 +187,16 @@ void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk(KERN_DEBUG "out: auto calibration time out " - "- reg = 0x%x, value = 0x%x\n", reg, value); + dev_dbg(chip->card->dev, + "out: auto calibration time out - reg = 0x%x, value = 0x%x\n", + reg, value); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); wss_outb(chip, CS4231P(REG), value); chip->image[reg] = value; mb(); - snd_printdd("codec out - reg 0x%x = 0x%x\n", - chip->mce_bit | reg, value); + dev_dbg(chip->card->dev, "codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); } EXPORT_SYMBOL(snd_wss_out); @@ -204,8 +205,8 @@ unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk(KERN_DEBUG "in: auto calibration time out " - "- reg = 0x%x\n", reg); + dev_dbg(chip->card->dev, + "in: auto calibration time out - reg = 0x%x\n", reg); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); mb(); @@ -222,7 +223,7 @@ void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg, wss_outb(chip, CS4231P(REG), val); chip->eimage[CS4236_REG(reg)] = val; #if 0 - printk(KERN_DEBUG "ext out : reg = 0x%x, val = 0x%x\n", reg, val); + dev_dbg(chip->card->dev, "ext out : reg = 0x%x, val = 0x%x\n", reg, val); #endif } EXPORT_SYMBOL(snd_cs4236_ext_out); @@ -238,8 +239,8 @@ unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg) { unsigned char res; res = wss_inb(chip, CS4231P(REG)); - printk(KERN_DEBUG "ext in : reg = 0x%x, val = 0x%x\n", - reg, res); + dev_dbg(chip->card->dev, "ext in : reg = 0x%x, val = 0x%x\n", + reg, res); return res; } #endif @@ -250,87 +251,87 @@ EXPORT_SYMBOL(snd_cs4236_ext_in); static void snd_wss_debug(struct snd_wss *chip) { - printk(KERN_DEBUG + dev_dbg(chip->card->dev, "CS4231 REGS: INDEX = 0x%02x " " STATUS = 0x%02x\n", wss_inb(chip, CS4231P(REGSEL)), wss_inb(chip, CS4231P(STATUS))); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x00: left input = 0x%02x " " 0x10: alt 1 (CFIG 2) = 0x%02x\n", snd_wss_in(chip, 0x00), snd_wss_in(chip, 0x10)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x01: right input = 0x%02x " " 0x11: alt 2 (CFIG 3) = 0x%02x\n", snd_wss_in(chip, 0x01), snd_wss_in(chip, 0x11)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x02: GF1 left input = 0x%02x " " 0x12: left line in = 0x%02x\n", snd_wss_in(chip, 0x02), snd_wss_in(chip, 0x12)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x03: GF1 right input = 0x%02x " " 0x13: right line in = 0x%02x\n", snd_wss_in(chip, 0x03), snd_wss_in(chip, 0x13)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x04: CD left input = 0x%02x " " 0x14: timer low = 0x%02x\n", snd_wss_in(chip, 0x04), snd_wss_in(chip, 0x14)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x05: CD right input = 0x%02x " " 0x15: timer high = 0x%02x\n", snd_wss_in(chip, 0x05), snd_wss_in(chip, 0x15)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x06: left output = 0x%02x " " 0x16: left MIC (PnP) = 0x%02x\n", snd_wss_in(chip, 0x06), snd_wss_in(chip, 0x16)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x07: right output = 0x%02x " " 0x17: right MIC (PnP) = 0x%02x\n", snd_wss_in(chip, 0x07), snd_wss_in(chip, 0x17)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x08: playback format = 0x%02x " " 0x18: IRQ status = 0x%02x\n", snd_wss_in(chip, 0x08), snd_wss_in(chip, 0x18)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x09: iface (CFIG 1) = 0x%02x " " 0x19: left line out = 0x%02x\n", snd_wss_in(chip, 0x09), snd_wss_in(chip, 0x19)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0a: pin control = 0x%02x " " 0x1a: mono control = 0x%02x\n", snd_wss_in(chip, 0x0a), snd_wss_in(chip, 0x1a)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0b: init & status = 0x%02x " " 0x1b: right line out = 0x%02x\n", snd_wss_in(chip, 0x0b), snd_wss_in(chip, 0x1b)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0c: revision & mode = 0x%02x " " 0x1c: record format = 0x%02x\n", snd_wss_in(chip, 0x0c), snd_wss_in(chip, 0x1c)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0d: loopback = 0x%02x " " 0x1d: var freq (PnP) = 0x%02x\n", snd_wss_in(chip, 0x0d), snd_wss_in(chip, 0x1d)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0e: ply upr count = 0x%02x " " 0x1e: ply lwr count = 0x%02x\n", snd_wss_in(chip, 0x0e), snd_wss_in(chip, 0x1e)); - printk(KERN_DEBUG + dev_dbg(chip->card->dev, " 0x0f: rec upr count = 0x%02x " " 0x1f: rec lwr count = 0x%02x\n", snd_wss_in(chip, 0x0f), @@ -365,16 +366,16 @@ void snd_wss_mce_up(struct snd_wss *chip) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk(KERN_DEBUG - "mce_up - auto calibration time out (0)\n"); + dev_dbg(chip->card->dev, + "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= CS4231_MCE; timeout = wss_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) - snd_printk(KERN_DEBUG "mce_up [0x%lx]: " - "serious init problem - codec still busy\n", - chip->port); + dev_dbg(chip->card->dev, + "mce_up [0x%lx]: serious init problem - codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); @@ -393,9 +394,9 @@ void snd_wss_mce_down(struct snd_wss *chip) #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk(KERN_DEBUG "mce_down [0x%lx] - " - "auto calibration time out (0)\n", - (long)CS4231P(REGSEL)); + dev_dbg(chip->card->dev, + "mce_down [0x%lx] - auto calibration time out (0)\n", + (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; @@ -403,9 +404,9 @@ void snd_wss_mce_down(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk(KERN_DEBUG "mce_down [0x%lx]: " - "serious init problem - codec still busy\n", - chip->port); + dev_dbg(chip->card->dev, + "mce_down [0x%lx]: serious init problem - codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; @@ -416,7 +417,7 @@ void snd_wss_mce_down(struct snd_wss *chip) */ msleep(1); - snd_printdd("(1) jiffies = %lu\n", jiffies); + dev_dbg(chip->card->dev, "(1) jiffies = %lu\n", jiffies); /* check condition up to 250 ms */ end_time = jiffies + msecs_to_jiffies(250); @@ -424,27 +425,29 @@ void snd_wss_mce_down(struct snd_wss *chip) CS4231_CALIB_IN_PROGRESS) { if (time_after(jiffies, end_time)) { - snd_printk(KERN_ERR "mce_down - " - "auto calibration time out (2)\n"); + dev_err(chip->card->dev, + "mce_down - auto calibration time out (2)\n"); return; } msleep(1); } - snd_printdd("(2) jiffies = %lu\n", jiffies); + dev_dbg(chip->card->dev, "(2) jiffies = %lu\n", jiffies); /* check condition up to 100 ms */ end_time = jiffies + msecs_to_jiffies(100); while (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { if (time_after(jiffies, end_time)) { - snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); + dev_err(chip->card->dev, + "mce_down - auto calibration time out (3)\n"); return; } msleep(1); } - snd_printdd("(3) jiffies = %lu\n", jiffies); - snd_printd("mce_down - exit = 0x%x\n", wss_inb(chip, CS4231P(REGSEL))); + dev_dbg(chip->card->dev, "(3) jiffies = %lu\n", jiffies); + dev_dbg(chip->card->dev, "mce_down - exit = 0x%x\n", + wss_inb(chip, CS4231P(REGSEL))); } EXPORT_SYMBOL(snd_wss_mce_down); @@ -543,7 +546,7 @@ static unsigned char snd_wss_get_format(struct snd_wss *chip, if (channels > 1) rformat |= CS4231_STEREO; #if 0 - snd_printk(KERN_DEBUG "get_format: 0x%x (mode=0x%x)\n", format, mode); + dev_dbg(chip->card->dev, "get_format: 0x%x (mode=0x%x)\n", format, mode); #endif return rformat; } @@ -793,7 +796,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk(KERN_DEBUG "init: (1)\n"); + dev_dbg(chip->card->dev, "init: (1)\n"); #endif snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -808,7 +811,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk(KERN_DEBUG "init: (2)\n"); + dev_dbg(chip->card->dev, "init: (2)\n"); #endif snd_wss_mce_up(chip); @@ -821,8 +824,8 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk(KERN_DEBUG "init: (3) - afei = 0x%x\n", - chip->image[CS4231_ALT_FEATURE_1]); + dev_dbg(chip->card->dev, "init: (3) - afei = 0x%x\n", + chip->image[CS4231_ALT_FEATURE_1]); #endif spin_lock_irqsave(&chip->reg_lock, flags); @@ -838,7 +841,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk(KERN_DEBUG "init: (4)\n"); + dev_dbg(chip->card->dev, "init: (4)\n"); #endif snd_wss_mce_up(chip); @@ -851,7 +854,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_calibrate_mute(chip, 0); #ifdef SNDRV_DEBUG_MCE - snd_printk(KERN_DEBUG "init: (5)\n"); + dev_dbg(chip->card->dev, "init: (5)\n"); #endif } @@ -1256,12 +1259,13 @@ static int snd_wss_probe(struct snd_wss *chip) break; /* this is valid value */ } } - snd_printdd("wss: port = 0x%lx, id = 0x%x\n", chip->port, id); + dev_dbg(chip->card->dev, "wss: port = 0x%lx, id = 0x%x\n", + chip->port, id); if (id != 0x0a) return -ENODEV; /* no valid device found */ rev = snd_wss_in(chip, CS4231_VERSION) & 0xe7; - snd_printdd("CS4231: VERSION (I25) = 0x%x\n", rev); + dev_dbg(chip->card->dev, "CS4231: VERSION (I25) = 0x%x\n", rev); if (rev == 0x80) { unsigned char tmp = snd_wss_in(chip, 23); snd_wss_out(chip, 23, ~tmp); @@ -1280,8 +1284,8 @@ static int snd_wss_probe(struct snd_wss *chip) } else if (rev == 0x03) { chip->hardware = WSS_HW_CS4236B; } else { - snd_printk(KERN_ERR - "unknown CS chip with version 0x%x\n", rev); + dev_err(chip->card->dev, + "unknown CS chip with version 0x%x\n", rev); return -ENODEV; /* unknown CS4231 chip? */ } } @@ -1340,7 +1344,9 @@ static int snd_wss_probe(struct snd_wss *chip) snd_cs4236_ext_out(chip, CS4236_VERSION, 0xff); id = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_cs4236_ext_out(chip, CS4236_VERSION, rev); - snd_printdd("CS4231: ext version; rev = 0x%x, id = 0x%x\n", rev, id); + dev_dbg(chip->card->dev, + "CS4231: ext version; rev = 0x%x, id = 0x%x\n", + rev, id); if ((id & 0x1f) == 0x1d) { /* CS4235 */ chip->hardware = WSS_HW_CS4235; switch (id >> 5) { @@ -1349,10 +1355,9 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk(KERN_WARNING - "unknown CS4235 chip " - "(enhanced version = 0x%x)\n", - id); + dev_warn(chip->card->dev, + "unknown CS4235 chip (enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ switch (id >> 5) { @@ -1363,10 +1368,9 @@ static int snd_wss_probe(struct snd_wss *chip) chip->hardware = WSS_HW_CS4236B; break; default: - snd_printk(KERN_WARNING - "unknown CS4236 chip " - "(enhanced version = 0x%x)\n", - id); + dev_warn(chip->card->dev, + "unknown CS4236 chip (enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x08) { /* CS4237B */ chip->hardware = WSS_HW_CS4237B; @@ -1377,10 +1381,9 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk(KERN_WARNING - "unknown CS4237B chip " - "(enhanced version = 0x%x)\n", - id); + dev_warn(chip->card->dev, + "unknown CS4237B chip (enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x09) { /* CS4238B */ chip->hardware = WSS_HW_CS4238B; @@ -1390,10 +1393,9 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk(KERN_WARNING - "unknown CS4238B chip " - "(enhanced version = 0x%x)\n", - id); + dev_warn(chip->card->dev, + "unknown CS4238B chip (enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ chip->hardware = WSS_HW_CS4239; @@ -1403,15 +1405,14 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk(KERN_WARNING - "unknown CS4239 chip " - "(enhanced version = 0x%x)\n", - id); + dev_warn(chip->card->dev, + "unknown CS4239 chip (enhanced version = 0x%x)\n", + id); } } else { - snd_printk(KERN_WARNING - "unknown CS4236/CS423xB chip " - "(enhanced version = 0x%x)\n", id); + dev_warn(chip->card->dev, + "unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", + id); } } } @@ -1644,8 +1645,9 @@ static void snd_wss_resume(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk(KERN_ERR "down [0x%lx]: serious init problem " - "- codec still busy\n", chip->port); + dev_err(chip->card->dev + "down [0x%lx]: serious init problem - codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; @@ -1757,7 +1759,7 @@ int snd_wss_create(struct snd_card *card, chip->res_port = devm_request_region(card->dev, port, 4, "WSS"); if (!chip->res_port) { - snd_printk(KERN_ERR "wss: can't grab port 0x%lx\n", port); + dev_err(chip->card->dev, "wss: can't grab port 0x%lx\n", port); return -EBUSY; } chip->port = port; @@ -1765,7 +1767,7 @@ int snd_wss_create(struct snd_card *card, chip->res_cport = devm_request_region(card->dev, cport, 8, "CS4232 Control"); if (!chip->res_cport) { - snd_printk(KERN_ERR + dev_err(chip->card->dev, "wss: can't grab control port 0x%lx\n", cport); return -ENODEV; } @@ -1774,20 +1776,20 @@ int snd_wss_create(struct snd_card *card, if (!(hwshare & WSS_HWSHARE_IRQ)) if (devm_request_irq(card->dev, irq, snd_wss_interrupt, 0, "WSS", (void *) chip)) { - snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); + dev_err(chip->card->dev, "wss: can't grab IRQ %d\n", irq); return -EBUSY; } chip->irq = irq; card->sync_irq = chip->irq; if (!(hwshare & WSS_HWSHARE_DMA1) && snd_devm_request_dma(card->dev, dma1, "WSS - 1")) { - snd_printk(KERN_ERR "wss: can't grab DMA1 %d\n", dma1); + dev_err(chip->card->dev, "wss: can't grab DMA1 %d\n", dma1); return -EBUSY; } chip->dma1 = dma1; if (!(hwshare & WSS_HWSHARE_DMA2) && dma1 != dma2 && dma2 >= 0 && snd_devm_request_dma(card->dev, dma2, "WSS - 2")) { - snd_printk(KERN_ERR "wss: can't grab DMA2 %d\n", dma2); + dev_err(chip->card->dev, "wss: can't grab DMA2 %d\n", dma2); return -EBUSY; } if (dma1 == dma2 || dma2 < 0) { @@ -1810,8 +1812,8 @@ int snd_wss_create(struct snd_card *card, #if 0 if (chip->hardware & WSS_HW_CS4232_MASK) { if (chip->res_cport == NULL) - snd_printk(KERN_ERR "CS4232 control port features are " - "not accessible\n"); + dev_err(chip->card->dev, + "CS4232 control port features are not accessible\n"); } #endif From 8aee49444faa6dfe061ac1252f4901a179047899 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:27 +0200 Subject: [PATCH 142/410] ALSA: riptide: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The device pointer is stored in struct cmdif for calling dev_*() functions. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-38-tiwai@suse.de --- sound/pci/riptide/riptide.c | 193 ++++++++++++++++++------------------ 1 file changed, 98 insertions(+), 95 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 7e80686fb41a..329816f37b76 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -380,6 +380,7 @@ struct riptideport { }; struct cmdif { + struct device *dev; struct riptideport *hwport; spinlock_t lock; unsigned int cmdcnt; /* cmd statistics */ @@ -727,7 +728,7 @@ static int loadfirmware(struct cmdif *cif, const unsigned char *img, } } } - snd_printdd("load firmware return %d\n", err); + dev_dbg(cif->dev, "load firmware return %d\n", err); return err; } @@ -740,7 +741,7 @@ alloclbuspath(struct cmdif *cif, unsigned char source, sink = *path & (~SPLIT_PATH); if (sink != E2SINK_MAX) { - snd_printdd("alloc path 0x%x->0x%x\n", source, sink); + dev_dbg(cif->dev, "alloc path 0x%x->0x%x\n", source, sink); SEND_PSEL(cif, source, sink); source = lbusin2out[sink][0]; type = lbusin2out[sink][1]; @@ -778,7 +779,7 @@ freelbuspath(struct cmdif *cif, unsigned char source, const unsigned char *path) sink = *path & (~SPLIT_PATH); if (sink != E2SINK_MAX) { - snd_printdd("free path 0x%x->0x%x\n", source, sink); + dev_dbg(cif->dev, "free path 0x%x->0x%x\n", source, sink); SEND_PCLR(cif, source, sink); source = lbusin2out[sink][0]; } @@ -811,8 +812,8 @@ static int writearm(struct cmdif *cif, u32 addr, u32 data, u32 mask) } else rptr.retlongs[0] &= ~mask; } - snd_printdd("send arm 0x%x 0x%x 0x%x return %d\n", addr, data, mask, - flag); + dev_dbg(cif->dev, "send arm 0x%x 0x%x 0x%x return %d\n", addr, data, mask, + flag); return flag; } @@ -832,14 +833,14 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, hwport = cif->hwport; if (cif->errcnt > MAX_ERROR_COUNT) { if (cif->is_reset) { - snd_printk(KERN_ERR - "Riptide: Too many failed cmds, reinitializing\n"); + dev_err(cif->dev, + "Riptide: Too many failed cmds, reinitializing\n"); if (riptide_reset(cif, NULL) == 0) { cif->errcnt = 0; return -EIO; } } - snd_printk(KERN_ERR "Riptide: Initialization failed.\n"); + dev_err(cif->dev, "Riptide: Initialization failed.\n"); return -EINVAL; } if (ret) { @@ -899,21 +900,21 @@ static int sendcmd(struct cmdif *cif, u32 flags, u32 cmd, u32 parm, if (time < cif->cmdtimemin) cif->cmdtimemin = time; if ((cif->cmdcnt) % 1000 == 0) - snd_printdd - ("send cmd %d time: %d mintime: %d maxtime %d err: %d\n", - cif->cmdcnt, cif->cmdtime, cif->cmdtimemin, - cif->cmdtimemax, cif->errcnt); + dev_dbg(cif->dev, + "send cmd %d time: %d mintime: %d maxtime %d err: %d\n", + cif->cmdcnt, cif->cmdtime, cif->cmdtimemin, + cif->cmdtimemax, cif->errcnt); return 0; errout: cif->errcnt++; spin_unlock_irqrestore(&cif->lock, irqflags); - snd_printdd - ("send cmd %d hw: 0x%x flag: 0x%x cmd: 0x%x parm: 0x%x ret: 0x%x 0x%x CMDE: %d DATF: %d failed %d\n", - cif->cmdcnt, (int)((void *)&(cmdport->stat) - (void *)hwport), - flags, cmd, parm, ret ? ret->retlongs[0] : 0, - ret ? ret->retlongs[1] : 0, IS_CMDE(cmdport), IS_DATF(cmdport), - err); + dev_dbg(cif->dev, + "send cmd %d hw: 0x%x flag: 0x%x cmd: 0x%x parm: 0x%x ret: 0x%x 0x%x CMDE: %d DATF: %d failed %d\n", + cif->cmdcnt, (int)((void *)&(cmdport->stat) - (void *)hwport), + flags, cmd, parm, ret ? ret->retlongs[0] : 0, + ret ? ret->retlongs[1] : 0, IS_CMDE(cmdport), IS_DATF(cmdport), + err); return err; } @@ -923,14 +924,14 @@ setmixer(struct cmdif *cif, short num, unsigned short rval, unsigned short lval) union cmdret rptr = CMDRET_ZERO; int i = 0; - snd_printdd("sent mixer %d: 0x%x 0x%x\n", num, rval, lval); + dev_dbg(cif->dev, "sent mixer %d: 0x%x 0x%x\n", num, rval, lval); do { SEND_SDGV(cif, num, num, rval, lval); SEND_RDGV(cif, num, num, &rptr); if (rptr.retwords[0] == lval && rptr.retwords[1] == rval) return 0; } while (i++ < MAX_WRITE_RETRY); - snd_printdd("sent mixer failed\n"); + dev_dbg(cif->dev, "sent mixer failed\n"); return -EIO; } @@ -961,7 +962,7 @@ getsourcesink(struct cmdif *cif, unsigned char source, unsigned char sink, return -EIO; *a = rptr.retbytes[0]; *b = rptr.retbytes[1]; - snd_printdd("getsourcesink 0x%x 0x%x\n", *a, *b); + dev_dbg(cif->dev, "%s 0x%x 0x%x\n", __func__, *a, *b); return 0; } @@ -988,11 +989,11 @@ getsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int *rate) } if (p[0]) { if (p[1] != p[0]) - snd_printdd("rates differ %d %d\n", p[0], p[1]); + dev_dbg(cif->dev, "rates differ %d %d\n", p[0], p[1]); *rate = (unsigned int)p[0]; } else *rate = (unsigned int)p[1]; - snd_printdd("getsampleformat %d %d %d\n", intdec[0], intdec[1], *rate); + dev_dbg(cif->dev, "getsampleformat %d %d %d\n", intdec[0], intdec[1], *rate); return 0; } @@ -1003,9 +1004,9 @@ setsampleformat(struct cmdif *cif, { unsigned char w, ch, sig, order; - snd_printdd - ("setsampleformat mixer: %d id: %d channels: %d format: %d\n", - mixer, id, channels, format); + dev_dbg(cif->dev, + "%s mixer: %d id: %d channels: %d format: %d\n", + __func__, mixer, id, channels, format); ch = channels == 1; w = snd_pcm_format_width(format) == 8; sig = snd_pcm_format_unsigned(format) != 0; @@ -1013,7 +1014,7 @@ setsampleformat(struct cmdif *cif, if (SEND_SETF(cif, mixer, w, ch, order, sig, id) && SEND_SETF(cif, mixer, w, ch, order, sig, id)) { - snd_printdd("setsampleformat failed\n"); + dev_dbg(cif->dev, "%s failed\n", __func__); return -EIO; } return 0; @@ -1026,8 +1027,8 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) union cmdret rptr = CMDRET_ZERO; int i; - snd_printdd("setsamplerate intdec: %d,%d rate: %d\n", intdec[0], - intdec[1], rate); + dev_dbg(cif->dev, "%s intdec: %d,%d rate: %d\n", __func__, + intdec[0], intdec[1], rate); D = 48000; M = ((rate == 48000) ? 47999 : rate) * 65536; N = M % D; @@ -1042,8 +1043,8 @@ setsamplerate(struct cmdif *cif, unsigned char *intdec, unsigned int rate) rptr.retwords[3] != N && i++ < MAX_WRITE_RETRY); if (i > MAX_WRITE_RETRY) { - snd_printdd("sent samplerate %d: %d failed\n", - *intdec, rate); + dev_dbg(cif->dev, "sent samplerate %d: %d failed\n", + *intdec, rate); return -EIO; } } @@ -1062,7 +1063,7 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, return -EIO; *rval = rptr.retwords[0]; *lval = rptr.retwords[1]; - snd_printdd("got mixer %d: 0x%x 0x%x\n", num, *rval, *lval); + dev_dbg(cif->dev, "got mixer %d: 0x%x 0x%x\n", num, *rval, *lval); return 0; } @@ -1105,8 +1106,8 @@ static irqreturn_t riptide_handleirq(int irq, void *dev_id) if ((flag & EOS_STATUS) && (data->state == ST_PLAY)) { data->state = ST_STOP; - snd_printk(KERN_ERR - "Riptide: DMA stopped unexpectedly\n"); + dev_err(cif->dev, + "Riptide: DMA stopped unexpectedly\n"); } c->dwStat_Ctl = cpu_to_le32(flag & @@ -1119,11 +1120,11 @@ static irqreturn_t riptide_handleirq(int irq, void *dev_id) period_bytes = frames_to_bytes(runtime, runtime->period_size); - snd_printdd - ("interrupt 0x%x after 0x%lx of 0x%lx frames in period\n", - READ_AUDIO_STATUS(cif->hwport), - bytes_to_frames(runtime, pos), - runtime->period_size); + dev_dbg(cif->dev, + "interrupt 0x%x after 0x%lx of 0x%lx frames in period\n", + READ_AUDIO_STATUS(cif->hwport), + bytes_to_frames(runtime, pos), + runtime->period_size); j = 0; if (pos >= period_bytes) { j++; @@ -1184,23 +1185,23 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) break; } if (!timeout) { - snd_printk(KERN_ERR - "Riptide: device not ready, audio status: 0x%x " - "ready: %d gerr: %d\n", - READ_AUDIO_STATUS(cif->hwport), - IS_READY(cif->hwport), IS_GERR(cif->hwport)); + dev_err(cif->dev, + "Riptide: device not ready, audio status: 0x%x ready: %d gerr: %d\n", + READ_AUDIO_STATUS(cif->hwport), + IS_READY(cif->hwport), IS_GERR(cif->hwport)); return -EIO; } else { - snd_printdd - ("Riptide: audio status: 0x%x ready: %d gerr: %d\n", - READ_AUDIO_STATUS(cif->hwport), - IS_READY(cif->hwport), IS_GERR(cif->hwport)); + dev_dbg(cif->dev, + "Riptide: audio status: 0x%x ready: %d gerr: %d\n", + READ_AUDIO_STATUS(cif->hwport), + IS_READY(cif->hwport), IS_GERR(cif->hwport)); } SEND_GETV(cif, &firmware.ret); - snd_printdd("Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n", - firmware.firmware.ASIC, firmware.firmware.CODEC, - firmware.firmware.AUXDSP, firmware.firmware.PROG); + dev_dbg(cif->dev, + "Firmware version: ASIC: %d CODEC %d AUXDSP %d PROG %d\n", + firmware.firmware.ASIC, firmware.firmware.CODEC, + firmware.firmware.AUXDSP, firmware.firmware.PROG); if (!chip) return 1; @@ -1211,20 +1212,20 @@ static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip) } - snd_printdd("Writing Firmware\n"); + dev_dbg(cif->dev, "Writing Firmware\n"); if (!chip->fw_entry) { err = request_firmware(&chip->fw_entry, "riptide.hex", &chip->pci->dev); if (err) { - snd_printk(KERN_ERR - "Riptide: Firmware not available %d\n", err); + dev_err(cif->dev, + "Riptide: Firmware not available %d\n", err); return -EIO; } } err = loadfirmware(cif, chip->fw_entry->data, chip->fw_entry->size); if (err) { - snd_printk(KERN_ERR - "Riptide: Could not load firmware %d\n", err); + dev_err(cif->dev, + "Riptide: Could not load firmware %d\n", err); return err; } @@ -1257,7 +1258,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip) SEND_SACR(cif, 0, AC97_RESET); SEND_RACR(cif, AC97_RESET, &rptr); - snd_printdd("AC97: 0x%x 0x%x\n", rptr.retlongs[0], rptr.retlongs[1]); + dev_dbg(cif->dev, "AC97: 0x%x 0x%x\n", rptr.retlongs[0], rptr.retlongs[1]); SEND_PLST(cif, 0); SEND_SLST(cif, 0); @@ -1350,11 +1351,11 @@ static snd_pcm_uframes_t snd_riptide_pointer(struct snd_pcm_substream SEND_GPOS(cif, 0, data->id, &rptr); if (data->size && runtime->period_size) { - snd_printdd - ("pointer stream %d position 0x%x(0x%x in buffer) bytes 0x%lx(0x%lx in period) frames\n", - data->id, rptr.retlongs[1], rptr.retlongs[1] % data->size, - bytes_to_frames(runtime, rptr.retlongs[1]), - bytes_to_frames(runtime, + dev_dbg(cif->dev, + "pointer stream %d position 0x%x(0x%x in buffer) bytes 0x%lx(0x%lx in period) frames\n", + data->id, rptr.retlongs[1], rptr.retlongs[1] % data->size, + bytes_to_frames(runtime, rptr.retlongs[1]), + bytes_to_frames(runtime, rptr.retlongs[1]) % runtime->period_size); if (rptr.retlongs[1] > data->pointer) ret = @@ -1365,8 +1366,9 @@ static snd_pcm_uframes_t snd_riptide_pointer(struct snd_pcm_substream bytes_to_frames(runtime, data->pointer % data->size); } else { - snd_printdd("stream not started or strange parms (%d %ld)\n", - data->size, runtime->period_size); + dev_dbg(cif->dev, + "stream not started or strange parms (%d %ld)\n", + data->size, runtime->period_size); ret = bytes_to_frames(runtime, 0); } return ret; @@ -1410,7 +1412,7 @@ static int snd_riptide_trigger(struct snd_pcm_substream *substream, int cmd) udelay(1); } while (i != rptr.retlongs[1] && j++ < MAX_WRITE_RETRY); if (j > MAX_WRITE_RETRY) - snd_printk(KERN_ERR "Riptide: Could not stop stream!"); + dev_err(cif->dev, "Riptide: Could not stop stream!"); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (!(data->state & ST_PAUSE)) { @@ -1448,8 +1450,8 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream) if (snd_BUG_ON(!cif || !data)) return -EINVAL; - snd_printdd("prepare id %d ch: %d f:0x%x r:%d\n", data->id, - runtime->channels, runtime->format, runtime->rate); + dev_dbg(cif->dev, "prepare id %d ch: %d f:0x%x r:%d\n", data->id, + runtime->channels, runtime->format, runtime->rate); spin_lock_irq(&chip->lock); channels = runtime->channels; @@ -1469,8 +1471,7 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream) lbuspath = data->paths.stereo; break; } - snd_printdd("use sgdlist at 0x%p\n", - data->sgdlist.area); + dev_dbg(cif->dev, "use sgdlist at 0x%p\n", data->sgdlist.area); if (data->sgdlist.area) { unsigned int i, j, size, pages, f, pt, period; struct sgd *c, *p = NULL; @@ -1483,9 +1484,9 @@ static int snd_riptide_prepare(struct snd_pcm_substream *substream) pages = DIV_ROUND_UP(size, f); data->size = size; data->pages = pages; - snd_printdd - ("create sgd size: 0x%x pages %d of size 0x%x for period 0x%x\n", - size, pages, f, period); + dev_dbg(cif->dev, + "create sgd size: 0x%x pages %d of size 0x%x for period 0x%x\n", + size, pages, f, period); pt = 0; j = 0; for (i = 0; i < pages; i++) { @@ -1543,17 +1544,18 @@ snd_riptide_hw_params(struct snd_pcm_substream *substream, struct snd_dma_buffer *sgdlist = &data->sgdlist; int err; - snd_printdd("hw params id %d (sgdlist: 0x%p 0x%lx %d)\n", data->id, - sgdlist->area, (unsigned long)sgdlist->addr, - (int)sgdlist->bytes); + dev_dbg(chip->card->dev, "hw params id %d (sgdlist: 0x%p 0x%lx %d)\n", + data->id, sgdlist->area, (unsigned long)sgdlist->addr, + (int)sgdlist->bytes); if (sgdlist->area) snd_dma_free_pages(sgdlist); err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, &chip->pci->dev, sizeof(struct sgd) * (DESC_MAX_MASK + 1), sgdlist); if (err < 0) { - snd_printk(KERN_ERR "Riptide: failed to alloc %d dma bytes\n", - (int)sizeof(struct sgd) * (DESC_MAX_MASK + 1)); + dev_err(chip->card->dev, + "Riptide: failed to alloc %d dma bytes\n", + (int)sizeof(struct sgd) * (DESC_MAX_MASK + 1)); return err; } data->sgdbuf = (struct sgd *)sgdlist->area; @@ -1729,13 +1731,13 @@ snd_riptide_codec_write(struct snd_ac97 *ac97, unsigned short reg, if (snd_BUG_ON(!cif)) return; - snd_printdd("Write AC97 reg 0x%x 0x%x\n", reg, val); + dev_dbg(cif->dev, "Write AC97 reg 0x%x 0x%x\n", reg, val); do { SEND_SACR(cif, val, reg); SEND_RACR(cif, reg, &rptr); } while (rptr.retwords[1] != val && i++ < MAX_WRITE_RETRY); if (i > MAX_WRITE_RETRY) - snd_printdd("Write AC97 reg failed\n"); + dev_dbg(cif->dev, "Write AC97 reg failed\n"); } static unsigned short snd_riptide_codec_read(struct snd_ac97 *ac97, @@ -1750,7 +1752,7 @@ static unsigned short snd_riptide_codec_read(struct snd_ac97 *ac97, if (SEND_RACR(cif, reg, &rptr) != 0) SEND_RACR(cif, reg, &rptr); - snd_printdd("Read AC97 reg 0x%x got 0x%x\n", reg, rptr.retwords[1]); + dev_dbg(cif->dev, "Read AC97 reg 0x%x got 0x%x\n", reg, rptr.retwords[1]); return rptr.retwords[1]; } @@ -1768,6 +1770,7 @@ static int snd_riptide_initialize(struct snd_riptide *chip) cif = kzalloc(sizeof(struct cmdif), GFP_KERNEL); if (!cif) return -ENOMEM; + cif->dev = chip->card->dev; cif->hwport = (struct riptideport *)chip->port; spin_lock_init(&cif->lock); chip->cif = cif; @@ -1781,11 +1784,11 @@ static int snd_riptide_initialize(struct snd_riptide *chip) case 0x4310: case 0x4320: case 0x4330: - snd_printdd("Modem enable?\n"); + dev_dbg(cif->dev, "Modem enable?\n"); SEND_SETDPLL(cif); break; } - snd_printdd("Enabling MPU IRQs\n"); + dev_dbg(cif->dev, "Enabling MPU IRQs\n"); if (chip->rmidi) SET_EMPUIRQ(cif->hwport); return err; @@ -1838,8 +1841,8 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci) snd_riptide_interrupt, riptide_handleirq, IRQF_SHARED, KBUILD_MODNAME, chip)) { - snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n", - pci->irq); + dev_err(&pci->dev, "Riptide: unable to grab IRQ %d\n", + pci->irq); return -EBUSY; } chip->irq = pci->irq; @@ -1987,9 +1990,9 @@ snd_riptide_joystick_probe(struct pci_dev *pci, const struct pci_device_id *id) goto inc_dev; } if (!request_region(joystick_port[dev], 8, "Riptide gameport")) { - snd_printk(KERN_WARNING - "Riptide: cannot grab gameport 0x%x\n", - joystick_port[dev]); + dev_err(&pci->dev, + "Riptide: cannot grab gameport 0x%x\n", + joystick_port[dev]); gameport_free_port(gameport); ret = -EBUSY; goto inc_dev; @@ -2064,9 +2067,9 @@ __snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id val, MPU401_INFO_IRQ_HOOK, -1, &chip->rmidi); if (err < 0) - snd_printk(KERN_WARNING - "Riptide: Can't Allocate MPU at 0x%x\n", - val); + dev_warn(&pci->dev, + "Riptide: Can't Allocate MPU at 0x%x\n", + val); else chip->mpuaddr = val; } @@ -2076,15 +2079,15 @@ __snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id err = snd_opl3_create(card, val, val + 2, OPL3_HW_RIPTIDE, 0, &chip->opl3); if (err < 0) - snd_printk(KERN_WARNING - "Riptide: Can't Allocate OPL3 at 0x%x\n", - val); + dev_warn(&pci->dev, + "Riptide: Can't Allocate OPL3 at 0x%x\n", + val); else { chip->opladdr = val; err = snd_opl3_hwdep_new(chip->opl3, 0, 1, NULL); if (err < 0) - snd_printk(KERN_WARNING - "Riptide: Can't Allocate OPL3-HWDEP\n"); + dev_warn(&pci->dev, + "Riptide: Can't Allocate OPL3-HWDEP\n"); } } #ifdef SUPPORT_JOYSTICK From adf72c364816c16f448ecb3f0f193fc06bbb9939 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:28 +0200 Subject: [PATCH 143/410] ALSA: korg1212: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-39-tiwai@suse.de --- sound/pci/korg1212/korg1212.c | 30 ++++++++++++++++-------------- 1 file changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 5c2cac201a28..e62fb1ad6d77 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -28,14 +28,14 @@ // ---------------------------------------------------------------------------- #define K1212_DEBUG_LEVEL 0 #if K1212_DEBUG_LEVEL > 0 -#define K1212_DEBUG_PRINTK(fmt,args...) printk(KERN_DEBUG fmt,##args) +#define K1212_DEBUG_PRINTK(fmt, args...) pr_debug(fmt, ##args) #else -#define K1212_DEBUG_PRINTK(fmt,...) do { } while (0) +#define K1212_DEBUG_PRINTK(fmt, ...) do { } while (0) #endif #if K1212_DEBUG_LEVEL > 1 -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,args...) printk(KERN_DEBUG fmt,##args) +#define K1212_DEBUG_PRINTK_VERBOSE(fmt, args...) pr_debug(fmt, ##args) #else -#define K1212_DEBUG_PRINTK_VERBOSE(fmt,...) +#define K1212_DEBUG_PRINTK_VERBOSE(fmt, ...) #endif // ---------------------------------------------------------------------------- @@ -602,7 +602,7 @@ static void snd_korg1212_timer_func(struct timer_list *t) /* reprogram timer */ mod_timer(&korg1212->timer, jiffies + 1); } else { - snd_printd("korg1212_timer_func timeout\n"); + dev_dbg(korg1212->card->dev, "korg1212_timer_func timeout\n"); korg1212->sharedBufferPtr->cardCommand = 0; korg1212->dsp_stop_is_processed = 1; wake_up(&korg1212->wait); @@ -1131,7 +1131,7 @@ static irqreturn_t snd_korg1212_interrupt(int irq, void *dev_id) K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: IRQ DMAE count - %ld, %x, [%s].\n", korg1212->irqcount, doorbellValue, stateName[korg1212->cardState]); - snd_printk(KERN_ERR "korg1212: DMA Error\n"); + dev_err(korg1212->card->dev, "korg1212: DMA Error\n"); korg1212->errorcnt++; korg1212->totalerrorcnt++; korg1212->sharedBufferPtr->cardCommand = 0; @@ -1272,8 +1272,8 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun #if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_silence KERNEL EFAULT dst=%p iter=%d\n", - dst, i); + pr_debug("K1212_DEBUG: %s KERNEL EFAULT dst=%p iter=%d\n", + __func__, dst, i); return -EFAULT; } #endif @@ -1305,7 +1305,8 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst->kvec.iov_base, i); + pr_debug("K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, src, dst->kvec.iov_base, i); return -EFAULT; } #endif @@ -1330,8 +1331,8 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, size = korg1212->channels * 2; dst = korg1212->playDataBufsPtr[0].bufferData + pos; - K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: snd_korg1212_copy_from pos=%d size=%d count=%d\n", - pos, size, count); + K1212_DEBUG_PRINTK_VERBOSE("K1212_DEBUG: %s pos=%d size=%d count=%d\n", + __func__, pos, size, count); if (snd_BUG_ON(pos + count > K1212_MAX_SAMPLES)) return -EINVAL; @@ -1340,7 +1341,8 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src->kvec.iov_base, dst, i); + pr_debug("K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, src->kvec.iov_base, dst, i); return -EFAULT; } #endif @@ -2135,7 +2137,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci) KBUILD_MODNAME, korg1212); if (err) { - snd_printk(KERN_ERR "korg1212: unable to grab IRQ %d\n", pci->irq); + dev_err(&pci->dev, "korg1212: unable to grab IRQ %d\n", pci->irq); return -EBUSY; } @@ -2232,7 +2234,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci) err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { - snd_printk(KERN_ERR "firmware not available\n"); + dev_err(&pci->dev, "firmware not available\n"); return err; } From 8455587c892437aeac0f87d2d502f9c7fb60e8f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:29 +0200 Subject: [PATCH 144/410] ALSA: lx6464es: Cleanup the print API usages Use pr_debug() instead of open-coded printk(). Also drop a few snd_printdd() calls that can be better done via ftrace. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-40-tiwai@suse.de --- sound/pci/lx6464es/lx_core.c | 8 ++++---- sound/pci/lx6464es/lx_core.h | 3 --- 2 files changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index b5b0d43bb8dc..9d95ecb299ae 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -231,14 +231,14 @@ static void lx_message_dump(struct lx_rmh *rmh) u8 idx = rmh->cmd_idx; int i; - snd_printk(LXRMH "command %s\n", dsp_commands[idx].dcOpName); + pr_debug(LXRMH "command %s\n", dsp_commands[idx].dcOpName); for (i = 0; i != rmh->cmd_len; ++i) - snd_printk(LXRMH "\tcmd[%d] %08x\n", i, rmh->cmd[i]); + pr_debug(LXRMH "\tcmd[%d] %08x\n", i, rmh->cmd[i]); for (i = 0; i != rmh->stat_len; ++i) - snd_printk(LXRMH "\tstat[%d]: %08x\n", i, rmh->stat[i]); - snd_printk("\n"); + pr_debug(LXRMH "\tstat[%d]: %08x\n", i, rmh->stat[i]); + pr_debug("\n"); } #else static inline void lx_message_dump(struct lx_rmh *rmh) diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 296013c910b7..c1113439f7c9 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -129,21 +129,18 @@ int lx_stream_set_state(struct lx6464es *chip, u32 pipe, static inline int lx_stream_start(struct lx6464es *chip, u32 pipe, int is_capture) { - snd_printdd("->lx_stream_start\n"); return lx_stream_set_state(chip, pipe, is_capture, SSTATE_RUN); } static inline int lx_stream_pause(struct lx6464es *chip, u32 pipe, int is_capture) { - snd_printdd("->lx_stream_pause\n"); return lx_stream_set_state(chip, pipe, is_capture, SSTATE_PAUSE); } static inline int lx_stream_stop(struct lx6464es *chip, u32 pipe, int is_capture) { - snd_printdd("->lx_stream_stop\n"); return lx_stream_set_state(chip, pipe, is_capture, SSTATE_STOP); } From 594508d5f35876ab9b9385bc8bc1ec9e200af0df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:30 +0200 Subject: [PATCH 145/410] ALSA: azt3328: Use pr_warn() Replace with the standard print API from the home-baked one. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-41-tiwai@suse.de --- sound/pci/azt3328.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 84989c291cd7..8a895d838005 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -1220,7 +1220,7 @@ snd_azf3328_codec_setfmt(struct snd_azf3328_codec_data *codec, case AZF_FREQ_22050: freq = SOUNDFORMAT_FREQ_22050; break; case AZF_FREQ_32000: freq = SOUNDFORMAT_FREQ_32000; break; default: - snd_printk(KERN_WARNING "unknown bitrate %d, assuming 44.1kHz!\n", bitrate); + pr_warn("azf3328: unknown bitrate %d, assuming 44.1kHz!\n", bitrate); fallthrough; case AZF_FREQ_44100: freq = SOUNDFORMAT_FREQ_44100; break; case AZF_FREQ_48000: freq = SOUNDFORMAT_FREQ_48000; break; From 1b28f418e77979629afbe8bdb6bb8c61c96a07f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:31 +0200 Subject: [PATCH 146/410] ALSA: emu10k1: Use dev_warn() Replace an open-coded printk warning with dev_warn() for code simplicity and consistency. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-42-tiwai@suse.de --- sound/pci/emu10k1/emu10k1_patch.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/emu10k1/emu10k1_patch.c b/sound/pci/emu10k1/emu10k1_patch.c index dbfa89435ac2..806b4f95cad1 100644 --- a/sound/pci/emu10k1/emu10k1_patch.c +++ b/sound/pci/emu10k1/emu10k1_patch.c @@ -40,8 +40,9 @@ snd_emu10k1_sample_new(struct snd_emux *rec, struct snd_sf_sample *sp, if (sp->v.mode_flags & (SNDRV_SFNT_SAMPLE_BIDIR_LOOP | SNDRV_SFNT_SAMPLE_REVERSE_LOOP)) { /* should instead return -ENOTSUPP; but compatibility */ - printk(KERN_WARNING "Emu10k1 wavetable patch %d with unsupported loop feature\n", - sp->v.sample); + dev_warn(emu->card->dev, + "Emu10k1 wavetable patch %d with unsupported loop feature\n", + sp->v.sample); } if (sp->v.mode_flags & SNDRV_SFNT_SAMPLE_8BITS) { From def358f9baaa7ca7b04cd8fe1f72af2605f11209 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:32 +0200 Subject: [PATCH 147/410] ALSA: trident: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-43-tiwai@suse.de --- sound/pci/trident/trident_memory.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c index 05de2b9f4ed7..4a36f194c7f8 100644 --- a/sound/pci/trident/trident_memory.c +++ b/sound/pci/trident/trident_memory.c @@ -137,14 +137,14 @@ __found_pages: /* * check if the given pointer is valid for pages */ -static int is_valid_page(unsigned long ptr) +static int is_valid_page(struct snd_trident *trident, unsigned long ptr) { if (ptr & ~0x3fffffffUL) { - snd_printk(KERN_ERR "max memory size is 1GB!!\n"); + dev_err(trident->card->dev, "max memory size is 1GB!!\n"); return 0; } if (ptr & (SNDRV_TRIDENT_PAGE_SIZE-1)) { - snd_printk(KERN_ERR "page is not aligned\n"); + dev_err(trident->card->dev, "page is not aligned\n"); return 0; } return 1; @@ -184,7 +184,7 @@ snd_trident_alloc_sg_pages(struct snd_trident *trident, for (page = firstpg(blk); page <= lastpg(blk); page++, idx++) { unsigned long ofs = idx << PAGE_SHIFT; dma_addr_t addr = snd_pcm_sgbuf_get_addr(substream, ofs); - if (! is_valid_page(addr)) { + if (!is_valid_page(trident, addr)) { __snd_util_mem_free(hdr, blk); mutex_unlock(&hdr->block_mutex); return NULL; @@ -227,7 +227,7 @@ snd_trident_alloc_cont_pages(struct snd_trident *trident, addr = runtime->dma_addr; for (page = firstpg(blk); page <= lastpg(blk); page++, addr += SNDRV_TRIDENT_PAGE_SIZE) { - if (! is_valid_page(addr)) { + if (!is_valid_page(trident, addr)) { __snd_util_mem_free(hdr, blk); mutex_unlock(&hdr->block_mutex); return NULL; From fea1510719dd2a33e0ffa1eae6118c5437f7071d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:33 +0200 Subject: [PATCH 148/410] ALSA: emux: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some functions are changed to receive snd_card object for calling dev_*() functions, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-44-tiwai@suse.de --- include/sound/soundfont.h | 6 ++-- sound/synth/emux/emux_hwdep.c | 6 ++-- sound/synth/emux/emux_oss.c | 11 +++--- sound/synth/emux/emux_seq.c | 13 ++++--- sound/synth/emux/emux_synth.c | 12 +++---- sound/synth/emux/soundfont.c | 67 +++++++++++++++++++---------------- 6 files changed, 63 insertions(+), 52 deletions(-) diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h index 98ed98d89d6d..8a40cc15f66d 100644 --- a/include/sound/soundfont.h +++ b/include/sound/soundfont.h @@ -86,9 +86,11 @@ struct snd_sf_list { }; /* Prototypes for soundfont.c */ -int snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, +int snd_soundfont_load(struct snd_card *card, + struct snd_sf_list *sflist, const void __user *data, long count, int client); -int snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, +int snd_soundfont_load_guspatch(struct snd_card *card, + struct snd_sf_list *sflist, const char __user *data, long count); int snd_soundfont_close_check(struct snd_sf_list *sflist, int client); diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index fd8f978cde1c..9e98c4c73fd1 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -26,12 +26,14 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) return -EFAULT; if (patch.key == GUS_PATCH) - return snd_soundfont_load_guspatch(emu->sflist, arg, + return snd_soundfont_load_guspatch(emu->card, emu->sflist, arg, patch.len + sizeof(patch)); if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { - err = snd_soundfont_load(emu->sflist, arg, patch.len + sizeof(patch), TMP_CLIENT_ID); + err = snd_soundfont_load(emu->card, emu->sflist, arg, + patch.len + sizeof(patch), + TMP_CLIENT_ID); if (err < 0) return err; } else { diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index 04df46b269d3..5f98d5201ba2 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -115,7 +115,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) p = snd_emux_create_port(emu, tmpname, 32, 1, &callback); if (p == NULL) { - snd_printk(KERN_ERR "can't create port\n"); + dev_err(emu->card->dev, "can't create port\n"); snd_emux_dec_count(emu); return -ENOMEM; } @@ -205,7 +205,7 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -ENXIO; if (format == GUS_PATCH) - rc = snd_soundfont_load_guspatch(emu->sflist, buf, count); + rc = snd_soundfont_load_guspatch(emu->card, emu->sflist, buf, count); else if (format == SNDRV_OSS_SOUNDFONT_PATCH) { struct soundfont_patch_info patch; if (count < (int)sizeof(patch)) @@ -214,10 +214,13 @@ snd_emux_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -EFAULT; if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) - rc = snd_soundfont_load(emu->sflist, buf, count, SF_CLIENT_NO(p->chset.port)); + rc = snd_soundfont_load(emu->card, emu->sflist, buf, + count, + SF_CLIENT_NO(p->chset.port)); else { if (emu->ops.load_fx) - rc = emu->ops.load_fx(emu, patch.type, patch.optarg, buf, count); + rc = emu->ops.load_fx(emu, patch.type, + patch.optarg, buf, count); else rc = -EINVAL; } diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 1adaa75df2f6..9daced0e6c59 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -61,16 +61,17 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) emu->client = snd_seq_create_kernel_client(card, index, "%s WaveTable", emu->name); if (emu->client < 0) { - snd_printk(KERN_ERR "can't create client\n"); + dev_err(card->dev, "can't create client\n"); return -ENODEV; } if (emu->num_ports <= 0) { - snd_printk(KERN_WARNING "seqports must be greater than zero\n"); + dev_warn(card->dev, "seqports must be greater than zero\n"); emu->num_ports = 1; } else if (emu->num_ports > SNDRV_EMUX_MAX_PORTS) { - snd_printk(KERN_WARNING "too many ports. " - "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); + dev_warn(card->dev, + "too many ports. limited max. ports %d\n", + SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } @@ -87,7 +88,7 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS, 0, &pinfo); if (!p) { - snd_printk(KERN_ERR "can't create port\n"); + dev_err(card->dev, "can't create port\n"); return -ENOMEM; } @@ -376,12 +377,10 @@ int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card) goto __error; } emu->vmidi[i] = rmidi; - /* snd_printk(KERN_DEBUG "virmidi %d ok\n", i); */ } return 0; __error: - /* snd_printk(KERN_DEBUG "error init..\n"); */ snd_emux_delete_virmidi(emu); return -ENOMEM; } diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 075358a533a0..02e2c69d7f18 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -942,9 +942,9 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF) emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED; else - snd_printk(KERN_WARNING - "invalid voice for lock %d (state = %x)\n", - voice, emu->voices[voice].state); + dev_warn(emu->card->dev, + "invalid voice for lock %d (state = %x)\n", + voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } @@ -960,9 +960,9 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED) emu->voices[voice].state = SNDRV_EMUX_ST_OFF; else - snd_printk(KERN_WARNING - "invalid voice for unlock %d (state = %x)\n", - voice, emu->voices[voice].state); + dev_warn(emu->card->dev, + "invalid voice for unlock %d (state = %x)\n", + voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 2373ed580bf8..b38a4e231790 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -38,7 +38,8 @@ static struct snd_sf_sample *sf_sample_new(struct snd_sf_list *sflist, static void sf_sample_delete(struct snd_sf_list *sflist, struct snd_soundfont *sf, struct snd_sf_sample *sp); static int load_map(struct snd_sf_list *sflist, const void __user *data, int count); -static int load_info(struct snd_sf_list *sflist, const void __user *data, long count); +static int load_info(struct snd_card *card, struct snd_sf_list *sflist, + const void __user *data, long count); static int remove_info(struct snd_sf_list *sflist, struct snd_soundfont *sf, int bank, int instr); static void init_voice_info(struct soundfont_voice_info *avp); @@ -113,7 +114,8 @@ snd_soundfont_close_check(struct snd_sf_list *sflist, int client) * it wants to do with it. */ int -snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, +snd_soundfont_load(struct snd_card *card, + struct snd_sf_list *sflist, const void __user *data, long count, int client) { struct soundfont_patch_info patch; @@ -121,7 +123,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk(KERN_ERR "patch record too small %ld\n", count); + dev_err(card->dev, "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -131,16 +133,16 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, data += sizeof(patch); if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) { - snd_printk(KERN_ERR "The wrong kind of patch %x\n", patch.key); + dev_err(card->dev, "The wrong kind of patch %x\n", patch.key); return -EINVAL; } if (count < patch.len) { - snd_printk(KERN_ERR "Patch too short %ld, need %d\n", - count, patch.len); + dev_err(card->dev, "Patch too short %ld, need %d\n", + count, patch.len); return -EINVAL; } if (patch.len < 0) { - snd_printk(KERN_ERR "poor length %d\n", patch.len); + dev_err(card->dev, "poor length %d\n", patch.len); return -EINVAL; } @@ -164,7 +166,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, rc = -EINVAL; switch (patch.type) { case SNDRV_SFNT_LOAD_INFO: - rc = load_info(sflist, data, count); + rc = load_info(card, sflist, data, count); break; case SNDRV_SFNT_LOAD_DATA: rc = load_data(sflist, data, count); @@ -184,8 +186,8 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, case SNDRV_SFNT_REMOVE_INFO: /* patch must be opened */ if (!sflist->currsf) { - snd_printk(KERN_ERR "soundfont: remove_info: " - "patch not opened\n"); + dev_err(card->dev, + "soundfont: remove_info: patch not opened\n"); rc = -EINVAL; } else { int bank, instr; @@ -509,7 +511,8 @@ remove_info(struct snd_sf_list *sflist, struct snd_soundfont *sf, * open soundfont. */ static int -load_info(struct snd_sf_list *sflist, const void __user *data, long count) +load_info(struct snd_card *card, + struct snd_sf_list *sflist, const void __user *data, long count) { struct snd_soundfont *sf; struct snd_sf_zone *zone; @@ -525,7 +528,7 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; if (count < (long)sizeof(hdr)) { - printk(KERN_ERR "Soundfont error: invalid patch zone length\n"); + dev_err(card->dev, "Soundfont error: invalid patch zone length\n"); return -EINVAL; } if (copy_from_user((char*)&hdr, data, sizeof(hdr))) @@ -535,15 +538,15 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) count -= sizeof(hdr); if (hdr.nvoices <= 0 || hdr.nvoices >= 100) { - printk(KERN_ERR "Soundfont error: Illegal voice number %d\n", - hdr.nvoices); + dev_err(card->dev, "Soundfont error: Illegal voice number %d\n", + hdr.nvoices); return -EINVAL; } if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) { - printk(KERN_ERR "Soundfont Error: " - "patch length(%ld) is smaller than nvoices(%d)\n", - count, hdr.nvoices); + dev_err(card->dev, + "Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n", + count, hdr.nvoices); return -EINVAL; } @@ -974,7 +977,8 @@ int snd_sf_vol_table[128] = { /* load GUS patch */ static int -load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) +load_guspatch(struct snd_card *card, + struct snd_sf_list *sflist, const char __user *data, long count) { struct patch_info patch; struct snd_soundfont *sf; @@ -984,7 +988,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) int rc; if (count < (long)sizeof(patch)) { - snd_printk(KERN_ERR "patch record too small %ld\n", count); + dev_err(card->dev, "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -1076,10 +1080,10 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* panning position; -128 - 127 => 0-127 */ zone->v.pan = (patch.panning + 128) / 2; #if 0 - snd_printk(KERN_DEBUG - "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", - (int)patch.base_freq, zone->v.rate_offset, - zone->v.root, zone->v.tune, zone->v.low, zone->v.high); + pr_debug( + "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", + (int)patch.base_freq, zone->v.rate_offset, + zone->v.root, zone->v.tune, zone->v.low, zone->v.high); #endif /* detuning is ignored */ @@ -1111,12 +1115,12 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release); zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]); #if 0 - snd_printk(KERN_DEBUG - "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", - zone->v.parm.volatkhld, - zone->v.parm.voldcysus, - zone->v.parm.volrelease, - zone->v.attenuation); + dev_dbg(card->dev, + "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", + zone->v.parm.volatkhld, + zone->v.parm.voldcysus, + zone->v.parm.volrelease, + zone->v.attenuation); #endif } @@ -1160,12 +1164,13 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, long count) /* load GUS patch */ int -snd_soundfont_load_guspatch(struct snd_sf_list *sflist, const char __user *data, +snd_soundfont_load_guspatch(struct snd_card *card, + struct snd_sf_list *sflist, const char __user *data, long count) { int rc; lock_preset(sflist); - rc = load_guspatch(sflist, data, count); + rc = load_guspatch(card, sflist, data, count); unlock_preset(sflist); return rc; } From f8466d91f36d97e237ea1b3c6f19de980054a1b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:34 +0200 Subject: [PATCH 149/410] ALSA: usx2y: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Some superfluous debug prints are dropped, and some ambiguous messages are slightly rephrased. The sk.dev pointer is set properly for allowing to call dev_*() functions, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-45-tiwai@suse.de --- sound/usb/usx2y/us122l.c | 45 ++++++---------- sound/usb/usx2y/usX2Yhwdep.c | 25 +++------ sound/usb/usx2y/usb_stream.c | 92 +++++++++++++++------------------ sound/usb/usx2y/usb_stream.h | 1 + sound/usb/usx2y/usbusx2y.c | 7 ++- sound/usb/usx2y/usbusx2yaudio.c | 67 ++++++++++++++---------- sound/usb/usx2y/usx2yhwdeppcm.c | 54 ++++++++++--------- 7 files changed, 141 insertions(+), 150 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 709ccad972e2..1be0e980feb9 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -84,12 +84,9 @@ static int us144_create_usbmidi(struct snd_card *card) static void pt_info_set(struct usb_device *dev, u8 v) { - int ret; - - ret = usb_control_msg_send(dev, 0, 'I', - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - v, 0, NULL, 0, 1000, GFP_NOIO); - snd_printdd(KERN_DEBUG "%i\n", ret); + usb_control_msg_send(dev, 0, 'I', + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + v, 0, NULL, 0, 1000, GFP_NOIO); } static void usb_stream_hwdep_vm_open(struct vm_area_struct *area) @@ -97,7 +94,6 @@ static void usb_stream_hwdep_vm_open(struct vm_area_struct *area) struct us122l *us122l = area->vm_private_data; atomic_inc(&us122l->mmap_count); - snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count)); } static vm_fault_t usb_stream_hwdep_vm_fault(struct vm_fault *vmf) @@ -141,7 +137,6 @@ static void usb_stream_hwdep_vm_close(struct vm_area_struct *area) struct us122l *us122l = area->vm_private_data; atomic_dec(&us122l->mmap_count); - snd_printdd(KERN_DEBUG "%i\n", atomic_read(&us122l->mmap_count)); } static const struct vm_operations_struct usb_stream_hwdep_vm_ops = { @@ -155,7 +150,6 @@ static int usb_stream_hwdep_open(struct snd_hwdep *hw, struct file *file) struct us122l *us122l = hw->private_data; struct usb_interface *iface; - snd_printdd(KERN_DEBUG "%p %p\n", hw, file); if (hw->used >= 2) return -EBUSY; @@ -176,8 +170,6 @@ static int usb_stream_hwdep_release(struct snd_hwdep *hw, struct file *file) struct us122l *us122l = hw->private_data; struct usb_interface *iface; - snd_printdd(KERN_DEBUG "%p %p\n", hw, file); - if (us122l->is_us144) { iface = usb_ifnum_to_if(us122l->dev, 0); usb_autopm_put_interface(iface); @@ -213,12 +205,10 @@ static int usb_stream_hwdep_mmap(struct snd_hwdep *hw, err = -EPERM; goto out; } - snd_printdd(KERN_DEBUG "%lu %u\n", size, - read ? s->read_size : s->write_size); /* if userspace tries to mmap beyond end of our buffer, fail */ if (size > PAGE_ALIGN(read ? s->read_size : s->write_size)) { - snd_printk(KERN_WARNING "%lu > %u\n", size, - read ? s->read_size : s->write_size); + dev_warn(hw->card->dev, "%s: size %lu > %u\n", __func__, + size, read ? s->read_size : s->write_size); err = -EINVAL; goto out; } @@ -289,8 +279,8 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate) UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000, GFP_NOIO); if (err) - snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n", - dev->devnum, rate, ep); + dev_err(&dev->dev, "%d: cannot set freq %d to ep 0x%x\n", + dev->devnum, rate, ep); return err; } @@ -326,13 +316,13 @@ static bool us122l_start(struct us122l *us122l, err = us122l_set_sample_rate(us122l->dev, rate); if (err < 0) { us122l_stop(us122l); - snd_printk(KERN_ERR "us122l_set_sample_rate error\n"); + dev_err(&us122l->dev->dev, "us122l_set_sample_rate error\n"); goto out; } err = usb_stream_start(&us122l->sk); if (err < 0) { us122l_stop(us122l); - snd_printk(KERN_ERR "%s error %i\n", __func__, err); + dev_err(&us122l->dev->dev, "%s error %i\n", __func__, err); goto out; } list_for_each(p, &us122l->midi_list) @@ -445,13 +435,13 @@ static bool us122l_create_card(struct snd_card *card) if (us122l->is_us144) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(card->dev, "usb_set_interface error\n"); return false; } } err = usb_set_interface(us122l->dev, 1, 1); if (err) { - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(card->dev, "usb_set_interface error\n"); return false; } @@ -466,7 +456,7 @@ static bool us122l_create_card(struct snd_card *card) else err = us122l_create_usbmidi(card); if (err < 0) { - snd_printk(KERN_ERR "us122l_create_usbmidi error %i\n", err); + dev_err(card->dev, "us122l_create_usbmidi error %i\n", err); goto stop; } err = usb_stream_hwdep_new(card); @@ -517,6 +507,7 @@ static int usx2y_create_card(struct usb_device *device, card->private_free = snd_us122l_free; US122L(card)->dev = device; mutex_init(&US122L(card)->mutex); + US122L(card)->sk.dev = device; init_waitqueue_head(&US122L(card)->sk.sleep); US122L(card)->is_us144 = flags & US122L_FLAG_US144; INIT_LIST_HEAD(&US122L(card)->midi_list); @@ -572,12 +563,10 @@ static int snd_us122l_probe(struct usb_interface *intf, if (id->driver_info & US122L_FLAG_US144 && device->speed == USB_SPEED_HIGH) { - snd_printk(KERN_ERR "disable ehci-hcd to run US-144\n"); + dev_err(&device->dev, "disable ehci-hcd to run US-144\n"); return -ENODEV; } - snd_printdd(KERN_DEBUG"%p:%i\n", - intf, intf->cur_altsetting->desc.bInterfaceNumber); if (intf->cur_altsetting->desc.bInterfaceNumber != 1) return 0; @@ -668,13 +657,13 @@ static int snd_us122l_resume(struct usb_interface *intf) if (us122l->is_us144) { err = usb_set_interface(us122l->dev, 0, 1); if (err) { - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(&us122l->dev->dev, "usb_set_interface error\n"); goto unlock; } } err = usb_set_interface(us122l->dev, 1, 1); if (err) { - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(&us122l->dev->dev, "usb_set_interface error\n"); goto unlock; } @@ -684,7 +673,7 @@ static int snd_us122l_resume(struct usb_interface *intf) err = us122l_set_sample_rate(us122l->dev, us122l->sk.s->cfg.sample_rate); if (err < 0) { - snd_printk(KERN_ERR "us122l_set_sample_rate error\n"); + dev_err(&us122l->dev->dev, "us122l_set_sample_rate error\n"); goto unlock; } err = usb_stream_start(&us122l->sk); diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 4937ede0b5d7..9fd6a86cc08e 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -24,19 +24,12 @@ static vm_fault_t snd_us428ctls_vm_fault(struct vm_fault *vmf) struct page *page; void *vaddr; - snd_printdd("ENTER, start %lXh, pgoff %ld\n", - vmf->vma->vm_start, - vmf->pgoff); - offset = vmf->pgoff << PAGE_SHIFT; vaddr = (char *)((struct usx2ydev *)vmf->vma->vm_private_data)->us428ctls_sharedmem + offset; page = virt_to_page(vaddr); get_page(page); vmf->page = page; - snd_printdd("vaddr=%p made us428ctls_vm_fault() page %p\n", - vaddr, page); - return 0; } @@ -56,7 +49,8 @@ static int snd_us428ctls_mmap(struct snd_hwdep *hw, struct file *filp, struct vm /* if userspace tries to mmap beyond end of our buffer, fail */ if (size > US428_SHAREDMEM_PAGES) { - snd_printd("%lu > %lu\n", size, (unsigned long)US428_SHAREDMEM_PAGES); + dev_dbg(hw->card->dev, "%s: mmap size %lu > %lu\n", __func__, + size, (unsigned long)US428_SHAREDMEM_PAGES); return -EINVAL; } @@ -150,7 +144,6 @@ static int usx2y_create_usbmidi(struct snd_card *card) le16_to_cpu(dev->descriptor.idProduct) == USB_ID_US428 ? &quirk_2 : &quirk_1; - snd_printdd("%s\n", __func__); return snd_usbmidi_create(card, iface, &usx2y(card)->midi_list, quirk); } @@ -160,7 +153,8 @@ static int usx2y_create_alsa_devices(struct snd_card *card) err = usx2y_create_usbmidi(card); if (err < 0) { - snd_printk(KERN_ERR "%s: usx2y_create_usbmidi error %i\n", __func__, err); + dev_err(card->dev, "%s: usx2y_create_usbmidi error %i\n", + __func__, err); return err; } err = usx2y_audio_create(card); @@ -183,15 +177,13 @@ static int snd_usx2y_hwdep_dsp_load(struct snd_hwdep *hw, int lret, err; char *buf; - snd_printdd("dsp_load %s\n", dsp->name); - buf = memdup_user(dsp->image, dsp->length); if (IS_ERR(buf)) return PTR_ERR(buf); err = usb_set_interface(dev, 0, 1); if (err) - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(&dev->dev, "usb_set_interface error\n"); else err = usb_bulk_msg(dev, usb_sndbulkpipe(dev, 2), buf, dsp->length, &lret, 6000); kfree(buf); @@ -201,21 +193,20 @@ static int snd_usx2y_hwdep_dsp_load(struct snd_hwdep *hw, msleep(250); // give the device some time err = usx2y_async_seq04_init(priv); if (err) { - snd_printk(KERN_ERR "usx2y_async_seq04_init error\n"); + dev_err(&dev->dev, "usx2y_async_seq04_init error\n"); return err; } err = usx2y_in04_init(priv); if (err) { - snd_printk(KERN_ERR "usx2y_in04_init error\n"); + dev_err(&dev->dev, "usx2y_in04_init error\n"); return err; } err = usx2y_create_alsa_devices(hw->card); if (err) { - snd_printk(KERN_ERR "usx2y_create_alsa_devices error %i\n", err); + dev_err(&dev->dev, "usx2y_create_alsa_devices error %i\n", err); return err; } priv->chip_status |= USX2Y_STAT_CHIP_INIT; - snd_printdd("%s: alsa all started\n", hw->name); } return err; } diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index a4d32e8a1d36..66ea610aa433 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -34,14 +34,11 @@ static void playback_prep_freqn(struct usb_stream_kernel *sk, struct urb *urb) urb->iso_frame_desc[pack].length = l; lb += l; } - snd_printdd(KERN_DEBUG "%i\n", lb); check: urb->number_of_packets = pack; urb->transfer_buffer_length = lb; s->idle_outsize += lb - s->period_size; - snd_printdd(KERN_DEBUG "idle=%i ul=%i ps=%i\n", s->idle_outsize, - lb, s->period_size); } static int init_pipe_urbs(struct usb_stream_kernel *sk, @@ -191,14 +188,14 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk, write_size = max_packsize * packets * USB_STREAM_URBDEPTH; if (read_size >= 256*PAGE_SIZE || write_size >= 256*PAGE_SIZE) { - snd_printk(KERN_WARNING "a size exceeds 128*PAGE_SIZE\n"); + dev_warn(&dev->dev, "%s: a size exceeds 128*PAGE_SIZE\n", __func__); goto out; } sk->s = alloc_pages_exact(read_size, GFP_KERNEL | __GFP_ZERO | __GFP_NOWARN); if (!sk->s) { - pr_warn("us122l: couldn't allocate read buffer\n"); + dev_warn(&dev->dev, "us122l: couldn't allocate read buffer\n"); goto out; } sk->s->cfg.version = USB_STREAM_INTERFACE_VERSION; @@ -217,7 +214,7 @@ struct usb_stream *usb_stream_new(struct usb_stream_kernel *sk, sk->write_page = alloc_pages_exact(write_size, GFP_KERNEL | __GFP_ZERO | __GFP_NOWARN); if (!sk->write_page) { - pr_warn("us122l: couldn't allocate write buffer\n"); + dev_warn(&dev->dev, "us122l: couldn't allocate write buffer\n"); usb_stream_free(sk); return NULL; } @@ -246,7 +243,8 @@ static bool balance_check(struct usb_stream_kernel *sk, struct urb *urb) if (unlikely(urb->status)) { if (urb->status != -ESHUTDOWN && urb->status != -ENOENT) - snd_printk(KERN_WARNING "status=%i\n", urb->status); + dev_warn(&sk->dev->dev, "%s: status=%i\n", __func__, + urb->status); sk->iso_frame_balance = 0x7FFFFFFF; return false; } @@ -318,17 +316,18 @@ static int usb_stream_prepare_playback(struct usb_stream_kernel *sk, check_ok: s->sync_packet -= inurb->number_of_packets; if (unlikely(s->sync_packet < -2 || s->sync_packet > 0)) { - snd_printk(KERN_WARNING "invalid sync_packet = %i;" - " p=%i nop=%i %i %x %x %x > %x\n", - s->sync_packet, p, inurb->number_of_packets, - s->idle_outsize + lb + l, - s->idle_outsize, lb, l, - s->period_size); + dev_warn(&sk->dev->dev, + "%s: invalid sync_packet = %i; p=%i nop=%i %i %x %x %x > %x\n", + __func__, + s->sync_packet, p, inurb->number_of_packets, + s->idle_outsize + lb + l, + s->idle_outsize, lb, l, + s->period_size); return -1; } if (unlikely(lb % s->cfg.frame_size)) { - snd_printk(KERN_WARNING"invalid outsize = %i\n", - lb); + dev_warn(&sk->dev->dev, "%s: invalid outsize = %i\n", + __func__, lb); return -1; } s->idle_outsize += lb - s->period_size; @@ -337,7 +336,7 @@ check_ok: if (s->idle_outsize <= 0) return 0; - snd_printk(KERN_WARNING "idle=%i\n", s->idle_outsize); + dev_warn(&sk->dev->dev, "%s: idle=%i\n", __func__, s->idle_outsize); return -1; } @@ -377,7 +376,7 @@ static int submit_urbs(struct usb_stream_kernel *sk, return 0; report_failure: - snd_printk(KERN_ERR "%i\n", err); + dev_err(&sk->dev->dev, "%s error: %i\n", __func__, err); return err; } @@ -424,8 +423,8 @@ loop: } if (iu == sk->completed_inurb) { if (l != s->period_size) - printk(KERN_DEBUG"%s:%i %i\n", __func__, __LINE__, - l/(int)s->cfg.frame_size); + dev_dbg(&sk->dev->dev, "%s:%i %i\n", __func__, __LINE__, + l/(int)s->cfg.frame_size); return; } @@ -460,8 +459,8 @@ static void stream_idle(struct usb_stream_kernel *sk, l = id[p].actual_length; if (unlikely(l == 0 || id[p].status)) { - snd_printk(KERN_WARNING "underrun, status=%u\n", - id[p].status); + dev_warn(&sk->dev->dev, "%s: underrun, status=%u\n", + __func__, id[p].status); goto err_out; } s->inpacket_head++; @@ -482,8 +481,8 @@ static void stream_idle(struct usb_stream_kernel *sk, } s->idle_insize += urb_size - s->period_size; if (s->idle_insize < 0) { - snd_printk(KERN_WARNING "%i\n", - (s->idle_insize)/(int)s->cfg.frame_size); + dev_warn(&sk->dev->dev, "%s error: %i\n", __func__, + (s->idle_insize)/(int)s->cfg.frame_size); goto err_out; } s->insize_done += urb_size; @@ -558,19 +557,15 @@ static void stream_start(struct usb_stream_kernel *sk, min_frames += frames_per_packet; diff = urb_size - (min_frames >> 8) * s->cfg.frame_size; - if (diff < max_diff) { - snd_printdd(KERN_DEBUG "%i %i %i %i\n", - s->insize_done, - urb_size / (int)s->cfg.frame_size, - inurb->number_of_packets, diff); + if (diff < max_diff) max_diff = diff; - } } s->idle_insize -= max_diff - max_diff_0; s->idle_insize += urb_size - s->period_size; if (s->idle_insize < 0) { - snd_printk(KERN_WARNING "%i %i %i\n", - s->idle_insize, urb_size, s->period_size); + dev_warn(&sk->dev->dev, "%s idle_insize: %i %i %i\n", + __func__, + s->idle_insize, urb_size, s->period_size); return; } else if (s->idle_insize == 0) { s->next_inpacket_split = @@ -620,7 +615,7 @@ static void i_capture_start(struct urb *urb) int empty = 0; if (urb->status) { - snd_printk(KERN_WARNING "status=%i\n", urb->status); + dev_warn(&sk->dev->dev, "%s: status=%i\n", __func__, urb->status); return; } @@ -630,7 +625,7 @@ static void i_capture_start(struct urb *urb) if (l < s->cfg.frame_size) { ++empty; if (s->state >= usb_stream_sync0) { - snd_printk(KERN_WARNING "%i\n", l); + dev_warn(&sk->dev->dev, "%s: length %i\n", __func__, l); return; } } @@ -642,8 +637,8 @@ static void i_capture_start(struct urb *urb) } #ifdef SHOW_EMPTY if (empty) { - printk(KERN_DEBUG"%s:%i: %i", __func__, __LINE__, - urb->iso_frame_desc[0].actual_length); + dev_dbg(&sk->dev->dev, "%s:%i: %i", __func__, __LINE__, + urb->iso_frame_desc[0].actual_length); for (pack = 1; pack < urb->number_of_packets; ++pack) { int l = urb->iso_frame_desc[pack].actual_length; @@ -710,38 +705,38 @@ dotry: } err = usb_submit_urb(inurb, GFP_ATOMIC); if (err < 0) { - snd_printk(KERN_ERR - "usb_submit_urb(sk->inurb[%i]) returned %i\n", - u, err); + dev_err(&sk->dev->dev, + "%s: usb_submit_urb(sk->inurb[%i]) returned %i\n", + __func__, u, err); return err; } err = usb_submit_urb(outurb, GFP_ATOMIC); if (err < 0) { - snd_printk(KERN_ERR - "usb_submit_urb(sk->outurb[%i]) returned %i\n", - u, err); + dev_err(&sk->dev->dev, + "%s: usb_submit_urb(sk->outurb[%i]) returned %i\n", + __func__, u, err); return err; } if (inurb->start_frame != outurb->start_frame) { - snd_printd(KERN_DEBUG - "u[%i] start_frames differ in:%u out:%u\n", - u, inurb->start_frame, outurb->start_frame); + dev_dbg(&sk->dev->dev, + "%s: u[%i] start_frames differ in:%u out:%u\n", + __func__, u, inurb->start_frame, outurb->start_frame); goto check_retry; } } - snd_printdd(KERN_DEBUG "%i %i\n", frame, iters); try = 0; check_retry: if (try) { usb_stream_stop(sk); if (try < 5) { msleep(1500); - snd_printd(KERN_DEBUG "goto dotry;\n"); + dev_dbg(&sk->dev->dev, "goto dotry;\n"); goto dotry; } - snd_printk(KERN_WARNING - "couldn't start all urbs on the same start_frame.\n"); + dev_warn(&sk->dev->dev, + "%s: couldn't start all urbs on the same start_frame.\n", + __func__); return -EFAULT; } @@ -755,7 +750,6 @@ check_retry: int wait_ms = 3000; while (s->state != usb_stream_ready && wait_ms > 0) { - snd_printdd(KERN_DEBUG "%i\n", s->state); msleep(200); wait_ms -= 200; } diff --git a/sound/usb/usx2y/usb_stream.h b/sound/usb/usx2y/usb_stream.h index 73e57b341adc..2ed10add49f1 100644 --- a/sound/usb/usx2y/usb_stream.h +++ b/sound/usb/usx2y/usb_stream.h @@ -9,6 +9,7 @@ struct usb_stream_kernel { struct usb_stream *s; + struct usb_device *dev; void *write_page; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 52f4e6652407..2f9cede242b3 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -163,7 +163,7 @@ static void i_usx2y_out04_int(struct urb *urb) for (i = 0; i < 10 && usx2y->as04.urb[i] != urb; i++) ; - snd_printdd("%s urb %i status=%i\n", __func__, i, urb->status); + dev_dbg(&urb->dev->dev, "%s urb %i status=%i\n", __func__, i, urb->status); } #endif } @@ -179,11 +179,10 @@ static void i_usx2y_in04_int(struct urb *urb) usx2y->in04_int_calls++; if (urb->status) { - snd_printdd("Interrupt Pipe 4 came back with status=%i\n", urb->status); + dev_dbg(&urb->dev->dev, "Interrupt Pipe 4 came back with status=%i\n", urb->status); return; } - // printk("%i:0x%02X ", 8, (int)((unsigned char*)usx2y->in04_buf)[8]); Master volume shows 0 here if fader is at max during boot ?!? if (us428ctls) { diff = -1; if (us428ctls->ctl_snapshot_last == -2) { @@ -239,7 +238,7 @@ static void i_usx2y_in04_int(struct urb *urb) } if (err) - snd_printk(KERN_ERR "in04_int() usb_submit_urb err=%i\n", err); + dev_err(&urb->dev->dev, "in04_int() usb_submit_urb err=%i\n", err); urb->dev = usx2y->dev; usb_submit_urb(urb, GFP_ATOMIC); diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index ca7888495a9f..f540f46a0b14 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -67,14 +67,15 @@ static int usx2y_urb_capt_retire(struct snd_usx2y_substream *subs) for (i = 0; i < nr_of_packs(); i++) { cp = (unsigned char *)urb->transfer_buffer + urb->iso_frame_desc[i].offset; if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ - snd_printk(KERN_ERR - "active frame status %i. Most probably some hardware problem.\n", - urb->iso_frame_desc[i].status); + dev_err(&usx2y->dev->dev, + "%s: active frame status %i. Most probably some hardware problem.\n", + __func__, + urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } len = urb->iso_frame_desc[i].actual_length / usx2y->stride; if (!len) { - snd_printd("0 == len ERROR!\n"); + dev_dbg(&usx2y->dev->dev, "%s: 0 == len ERROR!\n", __func__); continue; } @@ -128,7 +129,8 @@ static int usx2y_urb_play_prepare(struct snd_usx2y_substream *subs, counts = cap_urb->iso_frame_desc[pack].actual_length / usx2y->stride; count += counts; if (counts < 43 || counts > 50) { - snd_printk(KERN_ERR "should not be here with counts=%i\n", counts); + dev_err(&usx2y->dev->dev, "%s: should not be here with counts=%i\n", + __func__, counts); return -EPIPE; } /* set up descriptor */ @@ -196,7 +198,8 @@ static int usx2y_urb_submit(struct snd_usx2y_substream *subs, struct urb *urb, i urb->dev = subs->usx2y->dev; /* we need to set this at each time */ err = usb_submit_urb(urb, GFP_ATOMIC); if (err < 0) { - snd_printk(KERN_ERR "usb_submit_urb() returned %i\n", err); + dev_err(&urb->dev->dev, "%s: usb_submit_urb() returned %i\n", + __func__, err); return err; } return 0; @@ -264,7 +267,8 @@ static void usx2y_clients_stop(struct usx2ydev *usx2y) for (s = 0; s < 4; s++) { subs = usx2y->subs[s]; if (subs) { - snd_printdd("%i %p state=%i\n", s, subs, atomic_read(&subs->state)); + dev_dbg(&usx2y->dev->dev, "%s: %i %p state=%i\n", + __func__, s, subs, atomic_read(&subs->state)); atomic_set(&subs->state, STATE_STOPPED); } } @@ -276,8 +280,9 @@ static void usx2y_clients_stop(struct usx2ydev *usx2y) for (u = 0; u < NRURBS; u++) { urb = subs->urb[u]; if (urb) - snd_printdd("%i status=%i start_frame=%i\n", - u, urb->status, urb->start_frame); + dev_dbg(&usx2y->dev->dev, + "%s: %i status=%i start_frame=%i\n", + __func__, u, urb->status, urb->start_frame); } } } @@ -288,7 +293,8 @@ static void usx2y_clients_stop(struct usx2ydev *usx2y) static void usx2y_error_urb_status(struct usx2ydev *usx2y, struct snd_usx2y_substream *subs, struct urb *urb) { - snd_printk(KERN_ERR "ep=%i stalled with status=%i\n", subs->endpoint, urb->status); + dev_err(&usx2y->dev->dev, "%s: ep=%i stalled with status=%i\n", + __func__, subs->endpoint, urb->status); urb->status = 0; usx2y_clients_stop(usx2y); } @@ -300,10 +306,12 @@ static void i_usx2y_urb_complete(struct urb *urb) struct snd_usx2y_substream *capsubs, *playbacksubs; if (unlikely(atomic_read(&subs->state) < STATE_PREPARED)) { - snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usx2y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - urb->status, urb->start_frame); + dev_dbg(&usx2y->dev->dev, + "%s: hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", + __func__, + usb_get_current_frame_number(usx2y->dev), + subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", + urb->status, urb->start_frame); return; } if (unlikely(urb->status)) { @@ -323,7 +331,6 @@ static void i_usx2y_urb_complete(struct urb *urb) if (!usx2y_usbframe_complete(capsubs, playbacksubs, urb->start_frame)) { usx2y->wait_iso_frame += nr_of_packs(); } else { - snd_printdd("\n"); usx2y_clients_stop(usx2y); } } @@ -373,8 +380,9 @@ static void i_usx2y_subs_startup(struct urb *urb) static void usx2y_subs_prepare(struct snd_usx2y_substream *subs) { - snd_printdd("usx2y_substream_prepare(%p) ep=%i urb0=%p urb1=%p\n", - subs, subs->endpoint, subs->urb[0], subs->urb[1]); + dev_dbg(&subs->usx2y->dev->dev, + "%s(%p) ep=%i urb0=%p urb1=%p\n", + __func__, subs, subs->endpoint, subs->urb[0], subs->urb[1]); /* reset the pointer */ subs->hwptr = 0; subs->hwptr_done = 0; @@ -399,7 +407,7 @@ static void usx2y_urbs_release(struct snd_usx2y_substream *subs) { int i; - snd_printdd("%s %i\n", __func__, subs->endpoint); + dev_dbg(&subs->usx2y->dev->dev, "%s %i\n", __func__, subs->endpoint); for (i = 0; i < NRURBS; i++) usx2y_urb_release(subs->urb + i, subs != subs->usx2y->subs[SNDRV_PCM_STREAM_PLAYBACK]); @@ -505,7 +513,8 @@ static int usx2y_urbs_start(struct snd_usx2y_substream *subs) urb->transfer_buffer_length = subs->maxpacksize * nr_of_packs(); err = usb_submit_urb(urb, GFP_ATOMIC); if (err < 0) { - snd_printk(KERN_ERR "cannot submit datapipe for urb %d, err = %d\n", i, err); + dev_err(&urb->dev->dev, "%s: cannot submit datapipe for urb %d, err = %d\n", + __func__, i, err); err = -EPIPE; goto cleanup; } else { @@ -550,17 +559,16 @@ static int snd_usx2y_pcm_trigger(struct snd_pcm_substream *substream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_printdd("%s(START)\n", __func__); + dev_dbg(&subs->usx2y->dev->dev, "%s(START)\n", __func__); if (atomic_read(&subs->state) == STATE_PREPARED && atomic_read(&subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]->state) >= STATE_PREPARED) { atomic_set(&subs->state, STATE_PRERUNNING); } else { - snd_printdd("\n"); return -EPIPE; } break; case SNDRV_PCM_TRIGGER_STOP: - snd_printdd("%s(STOP)\n", __func__); + dev_dbg(&subs->usx2y->dev->dev, "%s(STOP)\n", __func__); if (atomic_read(&subs->state) >= STATE_PRERUNNING) atomic_set(&subs->state, STATE_PREPARED); break; @@ -661,7 +669,8 @@ static void i_usx2y_04int(struct urb *urb) struct usx2ydev *usx2y = urb->context; if (urb->status) - snd_printk(KERN_ERR "snd_usx2y_04int() urb->status=%i\n", urb->status); + dev_err(&urb->dev->dev, "%s() urb->status=%i\n", + __func__, urb->status); if (!--usx2y->us04->len) wake_up(&usx2y->in04_wait_queue); } @@ -751,7 +760,8 @@ static int usx2y_format_set(struct usx2ydev *usx2y, snd_pcm_format_t format) usb_kill_urb(usx2y->in04_urb); err = usb_set_interface(usx2y->dev, 0, alternate); if (err) { - snd_printk(KERN_ERR "usb_set_interface error\n"); + dev_err(&usx2y->dev->dev, "%s: usb_set_interface error\n", + __func__); return err; } usx2y->in04_urb->dev = usx2y->dev; @@ -778,7 +788,7 @@ static int snd_usx2y_pcm_hw_params(struct snd_pcm_substream *substream, int i; mutex_lock(&usx2y(card)->pcm_mutex); - snd_printdd("snd_usx2y_hw_params(%p, %p)\n", substream, hw_params); + dev_dbg(&dev->dev->dev, "%s(%p, %p)\n", __func__, substream, hw_params); /* all pcm substreams off one usx2y have to operate at the same * rate & format */ @@ -814,7 +824,7 @@ static int snd_usx2y_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_usx2y_substream *cap_subs, *playback_subs; mutex_lock(&subs->usx2y->pcm_mutex); - snd_printdd("snd_usx2y_hw_free(%p)\n", substream); + dev_dbg(&subs->usx2y->dev->dev, "%s(%p)\n", __func__, substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { cap_subs = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]; @@ -850,7 +860,7 @@ static int snd_usx2y_pcm_prepare(struct snd_pcm_substream *substream) struct snd_usx2y_substream *capsubs = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]; int err = 0; - snd_printdd("%s(%p)\n", __func__, substream); + dev_dbg(&usx2y->dev->dev, "%s(%p)\n", __func__, substream); mutex_lock(&usx2y->pcm_mutex); usx2y_subs_prepare(subs); @@ -867,7 +877,8 @@ static int snd_usx2y_pcm_prepare(struct snd_pcm_substream *substream) if (err < 0) goto up_prepare_mutex; } - snd_printdd("starting capture pipe for %s\n", subs == capsubs ? "self" : "playpipe"); + dev_dbg(&usx2y->dev->dev, "%s: starting capture pipe for %s\n", + __func__, subs == capsubs ? "self" : "playpipe"); err = usx2y_urbs_start(capsubs); if (err < 0) goto up_prepare_mutex; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 36f2e31168fb..1b1496adb47e 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -59,13 +59,13 @@ static int usx2y_usbpcm_urb_capt_retire(struct snd_usx2y_substream *subs) if (head >= ARRAY_SIZE(usx2y->hwdep_pcm_shm->captured_iso)) head = 0; usx2y->hwdep_pcm_shm->capture_iso_start = head; - snd_printdd("cap start %i\n", head); + dev_dbg(&usx2y->dev->dev, "cap start %i\n", head); } for (i = 0; i < nr_of_packs(); i++) { if (urb->iso_frame_desc[i].status) { /* active? hmm, skip this */ - snd_printk(KERN_ERR - "active frame status %i. Most probably some hardware problem.\n", - urb->iso_frame_desc[i].status); + dev_err(&usx2y->dev->dev, + "active frame status %i. Most probably some hardware problem.\n", + urb->iso_frame_desc[i].status); return urb->iso_frame_desc[i].status; } lens += urb->iso_frame_desc[i].actual_length / usx2y->stride; @@ -120,7 +120,7 @@ static int usx2y_hwdep_urb_play_prepare(struct snd_usx2y_substream *subs, /* calculate the size of a packet */ counts = shm->captured_iso[shm->playback_iso_head].length / usx2y->stride; if (counts < 43 || counts > 50) { - snd_printk(KERN_ERR "should not be here with counts=%i\n", counts); + dev_err(&usx2y->dev->dev, "should not be here with counts=%i\n", counts); return -EPIPE; } /* set up descriptor */ @@ -234,10 +234,11 @@ static void i_usx2y_usbpcm_urb_complete(struct urb *urb) struct snd_usx2y_substream *capsubs, *capsubs2, *playbacksubs; if (unlikely(atomic_read(&subs->state) < STATE_PREPARED)) { - snd_printdd("hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", - usb_get_current_frame_number(usx2y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - urb->status, urb->start_frame); + dev_dbg(&usx2y->dev->dev, + "hcd_frame=%i ep=%i%s status=%i start_frame=%i\n", + usb_get_current_frame_number(usx2y->dev), + subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", + urb->status, urb->start_frame); return; } if (unlikely(urb->status)) { @@ -255,7 +256,6 @@ static void i_usx2y_usbpcm_urb_complete(struct urb *urb) if (!usx2y_usbpcm_usbframe_complete(capsubs, capsubs2, playbacksubs, urb->start_frame)) { usx2y->wait_iso_frame += nr_of_packs(); } else { - snd_printdd("\n"); usx2y_clients_stop(usx2y); } } @@ -275,7 +275,8 @@ static void usx2y_usbpcm_urbs_release(struct snd_usx2y_substream *subs) { int i; - snd_printdd("snd_usx2y_urbs_release() %i\n", subs->endpoint); + dev_dbg(&subs->usx2y->dev->dev, + "snd_usx2y_urbs_release() %i\n", subs->endpoint); for (i = 0; i < NRURBS; i++) usx2y_hwdep_urb_release(subs->urb + i); } @@ -365,7 +366,7 @@ static int snd_usx2y_usbpcm_hw_free(struct snd_pcm_substream *substream) struct snd_usx2y_substream *cap_subs2; mutex_lock(&subs->usx2y->pcm_mutex); - snd_printdd("%s(%p)\n", __func__, substream); + dev_dbg(&subs->usx2y->dev->dev, "%s(%p)\n", __func__, substream); cap_subs2 = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -456,11 +457,12 @@ static int usx2y_usbpcm_urbs_start(struct snd_usx2y_substream *subs) urb->transfer_buffer_length = subs->maxpacksize * nr_of_packs(); err = usb_submit_urb(urb, GFP_KERNEL); if (err < 0) { - snd_printk(KERN_ERR "cannot usb_submit_urb() for urb %d, err = %d\n", u, err); + dev_err(&urb->dev->dev, + "cannot usb_submit_urb() for urb %d, err = %d\n", + u, err); err = -EPIPE; goto cleanup; } else { - snd_printdd("%i\n", urb->start_frame); if (!u) usx2y->wait_iso_frame = urb->start_frame; } @@ -500,7 +502,7 @@ static int snd_usx2y_usbpcm_prepare(struct snd_pcm_substream *substream) struct snd_usx2y_substream *capsubs = subs->usx2y->subs[SNDRV_PCM_STREAM_CAPTURE]; int err = 0; - snd_printdd("snd_usx2y_pcm_prepare(%p)\n", substream); + dev_dbg(&usx2y->dev->dev, "snd_usx2y_pcm_prepare(%p)\n", substream); mutex_lock(&usx2y->pcm_mutex); @@ -528,8 +530,9 @@ static int snd_usx2y_usbpcm_prepare(struct snd_pcm_substream *substream) if (err < 0) goto up_prepare_mutex; } - snd_printdd("starting capture pipe for %s\n", subs == capsubs ? - "self" : "playpipe"); + dev_dbg(&usx2y->dev->dev, + "starting capture pipe for %s\n", subs == capsubs ? + "self" : "playpipe"); err = usx2y_usbpcm_urbs_start(capsubs); if (err < 0) goto up_prepare_mutex; @@ -540,9 +543,10 @@ static int snd_usx2y_usbpcm_prepare(struct snd_pcm_substream *substream) if (atomic_read(&subs->state) < STATE_PREPARED) { while (usx2y_iso_frames_per_buffer(runtime, usx2y) > usx2y->hwdep_pcm_shm->captured_iso_frames) { - snd_printdd("Wait: iso_frames_per_buffer=%i,captured_iso_frames=%i\n", - usx2y_iso_frames_per_buffer(runtime, usx2y), - usx2y->hwdep_pcm_shm->captured_iso_frames); + dev_dbg(&usx2y->dev->dev, + "Wait: iso_frames_per_buffer=%i,captured_iso_frames=%i\n", + usx2y_iso_frames_per_buffer(runtime, usx2y), + usx2y->hwdep_pcm_shm->captured_iso_frames); if (msleep_interruptible(10)) { err = -ERESTARTSYS; goto up_prepare_mutex; @@ -552,9 +556,10 @@ static int snd_usx2y_usbpcm_prepare(struct snd_pcm_substream *substream) if (err < 0) goto up_prepare_mutex; } - snd_printdd("Ready: iso_frames_per_buffer=%i,captured_iso_frames=%i\n", - usx2y_iso_frames_per_buffer(runtime, usx2y), - usx2y->hwdep_pcm_shm->captured_iso_frames); + dev_dbg(&usx2y->dev->dev, + "Ready: iso_frames_per_buffer=%i,captured_iso_frames=%i\n", + usx2y_iso_frames_per_buffer(runtime, usx2y), + usx2y->hwdep_pcm_shm->captured_iso_frames); } else { usx2y->hwdep_pcm_shm->capture_iso_start = -1; } @@ -698,7 +703,8 @@ static int snd_usx2y_hwdep_pcm_mmap(struct snd_hwdep *hw, struct file *filp, str /* if userspace tries to mmap beyond end of our buffer, fail */ if (size > USX2Y_HWDEP_PCM_PAGES) { - snd_printd("%lu > %lu\n", size, (unsigned long)USX2Y_HWDEP_PCM_PAGES); + dev_dbg(hw->card->dev, "%s: %lu > %lu\n", __func__, + size, (unsigned long)USX2Y_HWDEP_PCM_PAGES); return -EINVAL; } From df04b43fee68367130a17ad2346ff74f36645068 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:35 +0200 Subject: [PATCH 150/410] ALSA: usb-audio: Use standard print API Use pr_*() instead of open-coded printk() for code simplicity. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-46-tiwai@suse.de --- sound/usb/midi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index c1f2e5a03de9..737dd00e97b1 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -224,10 +224,10 @@ static void snd_usbmidi_input_data(struct snd_usb_midi_in_endpoint *ep, #ifdef DUMP_PACKETS static void dump_urb(const char *type, const u8 *data, int length) { - snd_printk(KERN_DEBUG "%s packet: [", type); + pr_debug("%s packet: [", type); for (; length > 0; ++data, --length) - printk(KERN_CONT " %02x", *data); - printk(KERN_CONT " ]\n"); + pr_cont(" %02x", *data); + pr_cont(" ]\n"); } #else #define dump_urb(type, data, length) /* nothing */ From c7e58049a20b2d63c5964f229b11333e5a82168b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:36 +0200 Subject: [PATCH 151/410] ALSA: intel8x0: Drop unused snd_printd() calls There are commented out debug prints, which are utterly superfluous. Let's drop them. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-47-tiwai@suse.de --- sound/pci/intel8x0.c | 3 --- sound/pci/intel8x0m.c | 3 --- 2 files changed, 6 deletions(-) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index dae3e15ba534..e4bb99f71c2c 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -703,7 +703,6 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich if (!(status & ICH_BCIS)) { step = 0; } else if (civ == ichdev->civ) { - // snd_printd("civ same %d\n", civ); step = 1; ichdev->civ++; ichdev->civ &= ICH_REG_LVI_MASK; @@ -711,8 +710,6 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich step = civ - ichdev->civ; if (step < 0) step += ICH_REG_LVI_MASK + 1; - // if (step != 1) - // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); ichdev->civ = civ; } diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 3d6f5b3cc73e..38f8de51d641 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -423,7 +423,6 @@ static inline void snd_intel8x0m_update(struct intel8x0m *chip, struct ichdev *i civ = igetbyte(chip, port + ICH_REG_OFF_CIV); if (civ == ichdev->civ) { - // snd_printd("civ same %d\n", civ); step = 1; ichdev->civ++; ichdev->civ &= ICH_REG_LVI_MASK; @@ -431,8 +430,6 @@ static inline void snd_intel8x0m_update(struct intel8x0m *chip, struct ichdev *i step = civ - ichdev->civ; if (step < 0) step += ICH_REG_LVI_MASK + 1; - // if (step != 1) - // snd_printd("step = %d, %d -> %d\n", step, ichdev->civ, civ); ichdev->civ = civ; } From 2acbb5e57230830016e0fb2d61886e7a8c7f59ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:37 +0200 Subject: [PATCH 152/410] ALSA: vxpocket: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-48-tiwai@suse.de --- sound/pcmcia/vx/vxp_ops.c | 10 ++++++---- sound/pcmcia/vx/vxpocket.c | 25 ++++++++----------------- 2 files changed, 14 insertions(+), 21 deletions(-) diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 4176abd73799..0bc5c5d9d157 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -84,7 +84,7 @@ static int vx_check_magic(struct vx_core *chip) return 0; msleep(10); } while (time_after_eq(end_time, jiffies)); - snd_printk(KERN_ERR "cannot find xilinx magic word (%x)\n", c); + dev_err(chip->card->dev, "cannot find xilinx magic word (%x)\n", c); return -EIO; } @@ -153,7 +153,6 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * vx_outb(chip, ICR, 0); /* Wait for answer HF2 equal to 1 */ - snd_printdd(KERN_DEBUG "check ISR_HF2\n"); if (vx_check_isr(_chip, ISR_HF2, ISR_HF2, 20) < 0) goto _error; @@ -170,7 +169,9 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * goto _error; c = vx_inb(chip, RXL); if (c != (int)data) - snd_printk(KERN_ERR "vxpocket: load xilinx mismatch at %d: 0x%x != 0x%x\n", i, c, (int)data); + dev_err(_chip->card->dev, + "vxpocket: load xilinx mismatch at %d: 0x%x != 0x%x\n", + i, c, (int)data); } /* reset HF1 */ @@ -188,7 +189,8 @@ static int vxp_load_xilinx_binary(struct vx_core *_chip, const struct firmware * c |= (int)vx_inb(chip, RXM) << 8; c |= vx_inb(chip, RXL); - snd_printdd(KERN_DEBUG "xilinx: dsp size received 0x%x, orig 0x%zx\n", c, fw->size); + dev_dbg(_chip->card->dev, + "xilinx: dsp size received 0x%x, orig 0x%zx\n", c, fw->size); vx_outb(chip, ICR, ICR_HF0); diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7a0f0e73ceb2..2ea62efa0064 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -151,7 +151,8 @@ static int snd_vxpocket_assign_resources(struct vx_core *chip, int port, int irq struct snd_card *card = chip->card; struct snd_vxpocket *vxp = to_vxpocket(chip); - snd_printdd(KERN_DEBUG "vxpocket assign resources: port = 0x%x, irq = %d\n", port, irq); + dev_dbg(chip->card->dev, + "vxpocket assign resources: port = 0x%x, irq = %d\n", port, irq); vxp->port = port; sprintf(card->shortname, "Digigram %s", card->driver); @@ -178,13 +179,11 @@ static int vxpocket_config(struct pcmcia_device *link) struct vx_core *chip = link->priv; int ret; - snd_printdd(KERN_DEBUG "vxpocket_config called\n"); - /* redefine hardware record according to the VERSION1 string */ if (!strcmp(link->prod_id[1], "VX-POCKET")) { - snd_printdd("VX-pocket is detected\n"); + dev_dbg(chip->card->dev, "VX-pocket is detected\n"); } else { - snd_printdd("VX-pocket 440 is detected\n"); + dev_dbg(chip->card->dev, "VX-pocket 440 is detected\n"); /* overwrite the hardware information */ chip->hw = &vxp440_hw; chip->type = vxp440_hw.type; @@ -226,11 +225,8 @@ static int vxp_suspend(struct pcmcia_device *link) { struct vx_core *chip = link->priv; - snd_printdd(KERN_DEBUG "SUSPEND\n"); - if (chip) { - snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n"); + if (chip) snd_vx_suspend(chip); - } return 0; } @@ -239,15 +235,10 @@ static int vxp_resume(struct pcmcia_device *link) { struct vx_core *chip = link->priv; - snd_printdd(KERN_DEBUG "RESUME\n"); if (pcmcia_dev_present(link)) { - //struct snd_vxpocket *vxp = (struct snd_vxpocket *)chip; - if (chip) { - snd_printdd(KERN_DEBUG "calling snd_vx_resume\n"); + if (chip) snd_vx_resume(chip); - } } - snd_printdd(KERN_DEBUG "resume done!\n"); return 0; } @@ -269,7 +260,7 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) break; } if (i >= SNDRV_CARDS) { - snd_printk(KERN_ERR "vxpocket: too many cards found\n"); + dev_err(&p_dev->dev, "vxpocket: too many cards found\n"); return -EINVAL; } if (! enable[i]) @@ -279,7 +270,7 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n"); + dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); return err; } From 7ba0212231bc89571cbb916545ac1237084e1ad6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:38 +0200 Subject: [PATCH 153/410] ALSA: pdaudiocf: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-49-tiwai@suse.de --- sound/pcmcia/pdaudiocf/pdaudiocf.c | 21 ++++----------- sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 36 ++++++++++++------------- sound/pcmcia/pdaudiocf/pdaudiocf_irq.c | 3 +-- 3 files changed, 24 insertions(+), 36 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c index 8363ec08df5d..494460746614 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c @@ -85,14 +85,13 @@ static int snd_pdacf_probe(struct pcmcia_device *link) .dev_free = snd_pdacf_dev_free, }; - snd_printdd(KERN_DEBUG "pdacf_attach called\n"); /* find an empty slot from the card list */ for (i = 0; i < SNDRV_CARDS; i++) { if (! card_list[i]) break; } if (i >= SNDRV_CARDS) { - snd_printk(KERN_ERR "pdacf: too many cards found\n"); + dev_err(&link->dev, "pdacf: too many cards found\n"); return -EINVAL; } if (! enable[i]) @@ -102,7 +101,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link) err = snd_card_new(&link->dev, index[i], id[i], THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR "pdacf: cannot create a card instance\n"); + dev_err(&link->dev, "pdacf: cannot create a card instance\n"); return err; } @@ -152,7 +151,7 @@ static int snd_pdacf_assign_resources(struct snd_pdacf *pdacf, int port, int irq int err; struct snd_card *card = pdacf->card; - snd_printdd(KERN_DEBUG "pdacf assign resources: port = 0x%x, irq = %d\n", port, irq); + dev_dbg(card->dev, "pdacf assign resources: port = 0x%x, irq = %d\n", port, irq); pdacf->port = port; pdacf->irq = irq; pdacf->chip_status |= PDAUDIOCF_STAT_IS_CONFIGURED; @@ -185,8 +184,6 @@ static void snd_pdacf_detach(struct pcmcia_device *link) { struct snd_pdacf *chip = link->priv; - snd_printdd(KERN_DEBUG "pdacf_detach called\n"); - if (chip->chip_status & PDAUDIOCF_STAT_IS_CONFIGURED) snd_pdacf_powerdown(chip); chip->chip_status |= PDAUDIOCF_STAT_IS_STALE; /* to be sure */ @@ -203,7 +200,6 @@ static int pdacf_config(struct pcmcia_device *link) struct snd_pdacf *pdacf = link->priv; int ret; - snd_printdd(KERN_DEBUG "pdacf_config called\n"); link->config_index = 0x5; link->config_flags |= CONF_ENABLE_IRQ | CONF_ENABLE_PULSE_IRQ; @@ -241,11 +237,8 @@ static int pdacf_suspend(struct pcmcia_device *link) { struct snd_pdacf *chip = link->priv; - snd_printdd(KERN_DEBUG "SUSPEND\n"); - if (chip) { - snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n"); + if (chip) snd_pdacf_suspend(chip); - } return 0; } @@ -254,14 +247,10 @@ static int pdacf_resume(struct pcmcia_device *link) { struct snd_pdacf *chip = link->priv; - snd_printdd(KERN_DEBUG "RESUME\n"); if (pcmcia_dev_present(link)) { - if (chip) { - snd_printdd(KERN_DEBUG "calling snd_pdacf_resume\n"); + if (chip) snd_pdacf_resume(chip); - } } - snd_printdd(KERN_DEBUG "resume done!\n"); return 0; } diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 5537c0882aa5..11aacc7e3f0b 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -28,7 +28,7 @@ static unsigned char pdacf_ak4117_read(void *private_data, unsigned char reg) udelay(5); if (--timeout == 0) { spin_unlock_irqrestore(&chip->ak4117_lock, flags); - snd_printk(KERN_ERR "AK4117 ready timeout (read)\n"); + dev_err(chip->card->dev, "AK4117 ready timeout (read)\n"); return 0; } } @@ -38,7 +38,7 @@ static unsigned char pdacf_ak4117_read(void *private_data, unsigned char reg) udelay(5); if (--timeout == 0) { spin_unlock_irqrestore(&chip->ak4117_lock, flags); - snd_printk(KERN_ERR "AK4117 read timeout (read2)\n"); + dev_err(chip->card->dev, "AK4117 read timeout (read2)\n"); return 0; } } @@ -59,7 +59,7 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c udelay(5); if (--timeout == 0) { spin_unlock_irqrestore(&chip->ak4117_lock, flags); - snd_printk(KERN_ERR "AK4117 ready timeout (write)\n"); + dev_err(chip->card->dev, "AK4117 ready timeout (write)\n"); return; } } @@ -70,21 +70,21 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c #if 0 void pdacf_dump(struct snd_pdacf *chip) { - printk(KERN_DEBUG "PDAUDIOCF DUMP (0x%lx):\n", chip->port); - printk(KERN_DEBUG "WPD : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_WDP)); - printk(KERN_DEBUG "RDP : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_RDP)); - printk(KERN_DEBUG "TCR : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_TCR)); - printk(KERN_DEBUG "SCR : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_SCR)); - printk(KERN_DEBUG "ISR : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_ISR)); - printk(KERN_DEBUG "IER : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_IER)); - printk(KERN_DEBUG "AK_IFR : 0x%x\n", - inw(chip->port + PDAUDIOCF_REG_AK_IFR)); + dev_dbg(chip->card->dev, "PDAUDIOCF DUMP (0x%lx):\n", chip->port); + dev_dbg(chip->card->dev, "WPD : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_WDP)); + dev_dbg(chip->card->dev, "RDP : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_RDP)); + dev_dbg(chip->card->dev, "TCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_TCR)); + dev_dbg(chip->card->dev, "SCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_SCR)); + dev_dbg(chip->card->dev, "ISR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_ISR)); + dev_dbg(chip->card->dev, "IER : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_IER)); + dev_dbg(chip->card->dev, "AK_IFR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_AK_IFR)); } #endif diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index f134b4a4622f..af40a2c8789a 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -27,7 +27,7 @@ irqreturn_t pdacf_interrupt(int irq, void *dev) stat = inw(chip->port + PDAUDIOCF_REG_ISR); if (stat & (PDAUDIOCF_IRQLVL|PDAUDIOCF_IRQOVR)) { if (stat & PDAUDIOCF_IRQOVR) /* should never happen */ - snd_printk(KERN_ERR "PDAUDIOCF SRAM buffer overrun detected!\n"); + dev_err(chip->card->dev, "PDAUDIOCF SRAM buffer overrun detected!\n"); if (chip->pcm_substream) wake_thread = true; if (!(stat & PDAUDIOCF_IRQAKM)) @@ -257,7 +257,6 @@ irqreturn_t pdacf_threaded_irq(int irq, void *dev) rdp = inw(chip->port + PDAUDIOCF_REG_RDP); wdp = inw(chip->port + PDAUDIOCF_REG_WDP); - /* printk(KERN_DEBUG "TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); */ size = wdp - rdp; if (size < 0) size += 0x10000; From 76a6ef90d5400dbd282803a2d4de70b1bef593f7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:39 +0200 Subject: [PATCH 154/410] ALSA: ppc: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-50-tiwai@suse.de --- sound/ppc/awacs.c | 4 ++-- sound/ppc/daca.c | 2 +- sound/ppc/keywest.c | 5 +++-- sound/ppc/pmac.c | 52 ++++++++++++++++++++------------------------ sound/ppc/powermac.c | 2 +- sound/ppc/tumbler.c | 21 +++++++++--------- 6 files changed, 41 insertions(+), 45 deletions(-) diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 659866cfe3b4..49399a4a290d 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -39,7 +39,7 @@ static void snd_pmac_screamer_wait(struct snd_pmac *chip) while (!(in_le32(&chip->awacs->codec_stat) & MASK_VALID)) { mdelay(1); if (! --timeout) { - snd_printd("snd_pmac_screamer_wait timeout\n"); + dev_dbg(chip->card->dev, "%s timeout\n", __func__); break; } } @@ -58,7 +58,7 @@ snd_pmac_awacs_write(struct snd_pmac *chip, int val) out_le32(&chip->awacs->codec_ctrl, val | (chip->subframe << 22)); while (in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) { if (! --timeout) { - snd_printd("snd_pmac_awacs_write timeout\n"); + dev_dbg(chip->card->dev, "%s timeout\n", __func__); break; } } diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index 4da9278dd58a..d5766952f3db 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -69,7 +69,7 @@ static int daca_set_volume(struct pmac_daca *mix) data[1] |= mix->deemphasis ? 0x40 : 0; if (i2c_smbus_write_block_data(mix->i2c.client, DACA_REG_AVOL, 2, data) < 0) { - snd_printk(KERN_ERR "failed to set volume \n"); + dev_err(&mix->i2c.client->dev, "failed to set volume\n"); return -EINVAL; } return 0; diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index 2894d041b2f5..3d3513d9def5 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -113,7 +113,8 @@ int snd_pmac_tumbler_post_init(void) err = keywest_ctx->init_client(keywest_ctx); if (err < 0) { - snd_printk(KERN_ERR "tumbler: %i :cannot initialize the MCS\n", err); + dev_err(&keywest_ctx->client->dev, + "tumbler: %i :cannot initialize the MCS\n", err); return err; } return 0; @@ -136,7 +137,7 @@ int snd_pmac_keywest_init(struct pmac_keywest *i2c) err = i2c_add_driver(&keywest_driver); if (err) { - snd_printk(KERN_ERR "cannot register keywest i2c driver\n"); + dev_err(&i2c->client->dev, "cannot register keywest i2c driver\n"); i2c_put_adapter(adap); return err; } diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index 84058bbf9d12..76674c43fa7e 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -269,7 +269,6 @@ static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec, case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&chip->reg_lock); rec->running = 0; - /*printk(KERN_DEBUG "stopped!!\n");*/ snd_pmac_dma_stop(rec); for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++) out_le16(&cp->command, DBDMA_STOP); @@ -304,7 +303,6 @@ static snd_pcm_uframes_t snd_pmac_pcm_pointer(struct snd_pmac *chip, } #endif count += rec->cur_period * rec->period_size; - /*printk(KERN_DEBUG "pointer=%d\n", count);*/ return bytes_to_frames(subs->runtime, count); } @@ -384,8 +382,6 @@ static inline void snd_pmac_pcm_dead_xfer(struct pmac_stream *rec, unsigned short req, res ; unsigned int phy ; - /* printk(KERN_WARNING "snd-powermac: DMA died - patching it up!\n"); */ - /* to clear DEAD status we must first clear RUN set it to quiescent to be on the safe side */ (void)in_le32(&rec->dma->status); @@ -456,7 +452,6 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) if (! (stat & ACTIVE)) break; - /*printk(KERN_DEBUG "update frag %d\n", rec->cur_period);*/ cp->xfer_status = cpu_to_le16(0); cp->req_count = cpu_to_le16(rec->period_size); /*cp->res_count = cpu_to_le16(0);*/ @@ -770,7 +765,6 @@ snd_pmac_ctrl_intr(int irq, void *devid) struct snd_pmac *chip = devid; int ctrl = in_le32(&chip->awacs->control); - /*printk(KERN_DEBUG "pmac: control interrupt.. 0x%x\n", ctrl);*/ if (ctrl & MASK_PORTCHG) { /* do something when headphone is plugged/unplugged? */ if (chip->update_automute) @@ -779,7 +773,7 @@ snd_pmac_ctrl_intr(int irq, void *devid) if (ctrl & MASK_CNTLERR) { int err = (in_le32(&chip->awacs->codec_stat) & MASK_ERRCODE) >> 16; if (err && chip->model <= PMAC_SCREAMER) - snd_printk(KERN_DEBUG "error %x\n", err); + dev_dbg(chip->card->dev, "%s: error %x\n", __func__, err); } /* Writing 1s to the CNTLERR and PORTCHG bits clears them... */ out_le32(&chip->awacs->control, ctrl); @@ -964,9 +958,8 @@ static int snd_pmac_detect(struct snd_pmac *chip) if (prop) { /* partly deprecate snd-powermac, for those machines * that have a layout-id property for now */ - printk(KERN_INFO "snd-powermac no longer handles any " - "machines with a layout-id property " - "in the device-tree, use snd-aoa.\n"); + dev_info(chip->card->dev, + "snd-powermac no longer handles any machines with a layout-id property in the device-tree, use snd-aoa.\n"); of_node_put(sound); of_node_put(chip->node); chip->node = NULL; @@ -1021,7 +1014,7 @@ static int snd_pmac_detect(struct snd_pmac *chip) */ macio = macio_find(chip->node, macio_unknown); if (macio == NULL) - printk(KERN_WARNING "snd-powermac: can't locate macio !\n"); + dev_warn(chip->card->dev, "snd-powermac: can't locate macio !\n"); else { struct pci_dev *pdev = NULL; @@ -1034,8 +1027,8 @@ static int snd_pmac_detect(struct snd_pmac *chip) } } if (chip->pdev == NULL) - printk(KERN_WARNING "snd-powermac: can't locate macio PCI" - " device !\n"); + dev_warn(chip->card->dev, + "snd-powermac: can't locate macio PCI device !\n"); detect_byte_swap(chip); @@ -1125,7 +1118,8 @@ int snd_pmac_add_automute(struct snd_pmac *chip) chip->auto_mute = 1; err = snd_ctl_add(chip->card, snd_ctl_new1(&auto_mute_controls[0], chip)); if (err < 0) { - printk(KERN_ERR "snd-powermac: Failed to add automute control\n"); + dev_err(chip->card->dev, + "snd-powermac: Failed to add automute control\n"); return err; } chip->hp_detect_ctl = snd_ctl_new1(&auto_mute_controls[1], chip); @@ -1180,17 +1174,18 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) for (i = 0; i < 2; i ++) { if (of_address_to_resource(np->parent, i, &chip->rsrc[i])) { - printk(KERN_ERR "snd: can't translate rsrc " - " %d (%s)\n", i, rnames[i]); + dev_err(chip->card->dev, + "snd: can't translate rsrc %d (%s)\n", + i, rnames[i]); err = -ENODEV; goto __error; } if (request_mem_region(chip->rsrc[i].start, resource_size(&chip->rsrc[i]), rnames[i]) == NULL) { - printk(KERN_ERR "snd: can't request rsrc " - " %d (%s: %pR)\n", - i, rnames[i], &chip->rsrc[i]); + dev_err(chip->card->dev, + "snd: can't request rsrc %d (%s: %pR)\n", + i, rnames[i], &chip->rsrc[i]); err = -ENODEV; goto __error; } @@ -1205,17 +1200,18 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) for (i = 0; i < 3; i ++) { if (of_address_to_resource(np, i, &chip->rsrc[i])) { - printk(KERN_ERR "snd: can't translate rsrc " - " %d (%s)\n", i, rnames[i]); + dev_err(chip->card->dev, + "snd: can't translate rsrc %d (%s)\n", + i, rnames[i]); err = -ENODEV; goto __error; } if (request_mem_region(chip->rsrc[i].start, resource_size(&chip->rsrc[i]), rnames[i]) == NULL) { - printk(KERN_ERR "snd: can't request rsrc " - " %d (%s: %pR)\n", - i, rnames[i], &chip->rsrc[i]); + dev_err(chip->card->dev, + "snd: can't request rsrc %d (%s: %pR)\n", + i, rnames[i], &chip->rsrc[i]); err = -ENODEV; goto __error; } @@ -1233,8 +1229,8 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) irq = irq_of_parse_and_map(np, 0); if (request_irq(irq, snd_pmac_ctrl_intr, 0, "PMac", (void*)chip)) { - snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n", - irq); + dev_err(chip->card->dev, + "pmac: unable to grab IRQ %d\n", irq); err = -EBUSY; goto __error; } @@ -1242,14 +1238,14 @@ int snd_pmac_new(struct snd_card *card, struct snd_pmac **chip_return) } irq = irq_of_parse_and_map(np, 1); if (request_irq(irq, snd_pmac_tx_intr, 0, "PMac Output", (void*)chip)){ - snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n", irq); + dev_err(chip->card->dev, "pmac: unable to grab IRQ %d\n", irq); err = -EBUSY; goto __error; } chip->tx_irq = irq; irq = irq_of_parse_and_map(np, 2); if (request_irq(irq, snd_pmac_rx_intr, 0, "PMac Input", (void*)chip)) { - snd_printk(KERN_ERR "pmac: unable to grab IRQ %d\n", irq); + dev_err(chip->card->dev, "pmac: unable to grab IRQ %d\n", irq); err = -EBUSY; goto __error; } diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index e17af46abddd..8e29c92830ad 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -104,7 +104,7 @@ static int snd_pmac_probe(struct platform_device *devptr) goto __error; break; default: - snd_printk(KERN_ERR "unsupported hardware %d\n", chip->model); + dev_err(&devptr->dev, "unsupported hardware %d\n", chip->model); err = -EINVAL; goto __error; } diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 12f1e10db1c4..3c09660e1522 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -29,7 +29,7 @@ #undef DEBUG #ifdef DEBUG -#define DBG(fmt...) printk(KERN_DEBUG fmt) +#define DBG(fmt...) pr_debug(fmt) #else #define DBG(fmt...) #endif @@ -230,7 +230,7 @@ static int tumbler_set_master_volume(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_VOL, 6, block) < 0) { - snd_printk(KERN_ERR "failed to set volume \n"); + dev_err(&mix->i2c.client->dev, "failed to set volume\n"); return -EINVAL; } DBG("(I) succeeded to set volume (%u, %u)\n", left_vol, right_vol); @@ -341,7 +341,7 @@ static int tumbler_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 2, val) < 0) { - snd_printk(KERN_ERR "failed to set DRC\n"); + dev_err(&mix->i2c.client->dev, "failed to set DRC\n"); return -EINVAL; } DBG("(I) succeeded to set DRC (%u, %u)\n", val[0], val[1]); @@ -378,7 +378,7 @@ static int snapper_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 6, val) < 0) { - snd_printk(KERN_ERR "failed to set DRC\n"); + dev_err(&mix->i2c.client->dev, "failed to set DRC\n"); return -EINVAL; } DBG("(I) succeeded to set DRC (%u, %u)\n", val[0], val[1]); @@ -503,8 +503,8 @@ static int tumbler_set_mono_volume(struct pmac_tumbler *mix, block[i] = (vol >> ((info->bytes - i - 1) * 8)) & 0xff; if (i2c_smbus_write_i2c_block_data(mix->i2c.client, info->reg, info->bytes, block) < 0) { - snd_printk(KERN_ERR "failed to set mono volume %d\n", - info->index); + dev_err(&mix->i2c.client->dev, "failed to set mono volume %d\n", + info->index); return -EINVAL; } return 0; @@ -643,7 +643,8 @@ static int snapper_set_mix_vol1(struct pmac_tumbler *mix, int idx, int ch, int r } if (i2c_smbus_write_i2c_block_data(mix->i2c.client, reg, 9, block) < 0) { - snd_printk(KERN_ERR "failed to set mono volume %d\n", reg); + dev_err(&mix->i2c.client->dev, + "failed to set mono volume %d\n", reg); return -EINVAL; } return 0; @@ -1102,7 +1103,6 @@ static long tumbler_find_device(const char *device, const char *platform, node = find_audio_device(device); if (! node) { DBG("(W) cannot find audio device %s !\n", device); - snd_printdd("cannot find device %s\n", device); return -ENODEV; } @@ -1111,7 +1111,6 @@ static long tumbler_find_device(const char *device, const char *platform, base = of_get_property(node, "reg", NULL); if (!base) { DBG("(E) cannot find address for device %s !\n", device); - snd_printd("cannot find address for device %s\n", device); of_node_put(node); return -ENODEV; } @@ -1232,9 +1231,9 @@ static void tumbler_resume(struct snd_pmac *chip) tumbler_reset_audio(chip); if (mix->i2c.client && mix->i2c.init_client) { if (mix->i2c.init_client(&mix->i2c) < 0) - printk(KERN_ERR "tumbler_init_client error\n"); + dev_err(chip->card->dev, "tumbler_init_client error\n"); } else - printk(KERN_ERR "tumbler: i2c is not initialized\n"); + dev_err(chip->card->dev, "tumbler: i2c is not initialized\n"); if (chip->model == PMAC_TUMBLER) { tumbler_set_mono_volume(mix, &tumbler_pcm_vol_info); tumbler_set_mono_volume(mix, &tumbler_bass_vol_info); From 7e88541f006b1c685b8ea5ab2d157724215fd7ac Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:40 +0200 Subject: [PATCH 155/410] ALSA: sh: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-51-tiwai@suse.de --- sound/sh/aica.c | 7 +++---- sound/sh/sh_dac_audio.c | 8 ++++---- 2 files changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 3182c634464d..936cd6e91529 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -75,8 +75,7 @@ static void spu_write_wait(void) /* To ensure hardware failure doesn't wedge kernel */ time_count++; if (time_count > 0x10000) { - snd_printk - ("WARNING: G2 FIFO appears to be blocked.\n"); + pr_warn("WARNING: G2 FIFO appears to be blocked.\n"); break; } } @@ -591,8 +590,8 @@ static int snd_aica_probe(struct platform_device *devptr) if (unlikely(err < 0)) goto freedreamcast; platform_set_drvdata(devptr, dreamcastcard); - snd_printk - ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + dev_info(&devptr->dev, + "ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); return 0; freedreamcast: snd_card_free(dreamcastcard->card); diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 95ba3abd4e47..e7b6ce7bd086 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -348,8 +348,8 @@ static int snd_sh_dac_probe(struct platform_device *devptr) err = snd_card_new(&devptr->dev, index, id, THIS_MODULE, 0, &card); if (err < 0) { - snd_printk(KERN_ERR "cannot allocate the card\n"); - return err; + dev_err(&devptr->dev, "cannot allocate the card\n"); + return err; } err = snd_sh_dac_create(card, devptr, &chip); @@ -362,13 +362,13 @@ static int snd_sh_dac_probe(struct platform_device *devptr) strcpy(card->driver, "snd_sh_dac"); strcpy(card->shortname, "SuperH DAC audio driver"); - printk(KERN_INFO "%s %s", card->longname, card->shortname); + dev_info(&devptr->dev, "%s %s\n", card->longname, card->shortname); err = snd_card_register(card); if (err < 0) goto probe_error; - snd_printk(KERN_INFO "ALSA driver for SuperH DAC audio"); + dev_info(&devptr->dev, "ALSA driver for SuperH DAC audio\n"); platform_set_drvdata(devptr, card); return 0; From d41abde894830bfb77252653f606473728e930eb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:41 +0200 Subject: [PATCH 156/410] ALSA: sparc: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-52-tiwai@suse.de --- sound/sparc/amd7930.c | 8 ++--- sound/sparc/cs4231.c | 78 +++++++++++++++++++++++++------------------ 2 files changed, 49 insertions(+), 37 deletions(-) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 0fea04acc3ea..9bdf3db51d62 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -935,8 +935,8 @@ static int snd_amd7930_create(struct snd_card *card, amd->regs = of_ioremap(&op->resource[0], 0, resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { - snd_printk(KERN_ERR - "amd7930-%d: Unable to map chip registers.\n", dev); + dev_err(card->dev, + "amd7930-%d: Unable to map chip registers.\n", dev); kfree(amd); return -EIO; } @@ -945,8 +945,8 @@ static int snd_amd7930_create(struct snd_card *card, if (request_irq(irq, snd_amd7930_interrupt, IRQF_SHARED, "amd7930", amd)) { - snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", - dev, irq); + dev_err(card->dev, "amd7930-%d: Unable to grab IRQ %d\n", + dev, irq); snd_amd7930_free(amd); return -EBUSY; } diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index c2ad3fa2f25a..5f39b7847d31 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -292,9 +292,9 @@ static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) - snd_printdd("out: auto calibration time out - reg = 0x%x, " - "value = 0x%x\n", - reg, value); + dev_dbg(chip->card->dev, + "out: auto calibration time out - reg = 0x%x, value = 0x%x\n", + reg, value); #endif __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL)); wmb(); @@ -325,8 +325,9 @@ static unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) - snd_printdd("in: auto calibration time out - reg = 0x%x\n", - reg); + dev_dbg(chip->card->dev, + "in: auto calibration time out - reg = 0x%x\n", + reg); #endif __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL)); mb(); @@ -363,14 +364,15 @@ static void snd_cs4231_mce_up(struct snd_cs4231 *chip) snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) - snd_printdd("mce_up - auto calibration time out (0)\n"); + dev_dbg(chip->card->dev, + "mce_up - auto calibration time out (0)\n"); #endif chip->mce_bit |= CS4231_MCE; timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); if (timeout == 0x80) - snd_printdd("mce_up [%p]: serious init problem - " - "codec still busy\n", - chip->port); + dev_dbg(chip->card->dev, + "mce_up [%p]: serious init problem - codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231U(chip, REGSEL)); @@ -386,16 +388,18 @@ static void snd_cs4231_mce_down(struct snd_cs4231 *chip) spin_lock_irqsave(&chip->lock, flags); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) - snd_printdd("mce_down [%p] - auto calibration time out (0)\n", - CS4231U(chip, REGSEL)); + dev_dbg(chip->card->dev, + "mce_down [%p] - auto calibration time out (0)\n", + CS4231U(chip, REGSEL)); #endif chip->mce_bit &= ~CS4231_MCE; reg = __cs4231_readb(chip, CS4231U(chip, REGSEL)); __cs4231_writeb(chip, chip->mce_bit | (reg & 0x1f), CS4231U(chip, REGSEL)); if (reg == 0x80) - snd_printdd("mce_down [%p]: serious init problem " - "- codec still busy\n", chip->port); + dev_dbg(chip->card->dev, + "mce_down [%p]: serious init problem - codec still busy\n", + chip->port); if ((reg & CS4231_MCE) == 0) { spin_unlock_irqrestore(&chip->lock, flags); return; @@ -415,8 +419,8 @@ static void snd_cs4231_mce_down(struct snd_cs4231 *chip) spin_unlock_irqrestore(&chip->lock, flags); if (reg) - snd_printk(KERN_ERR - "mce_down - auto calibration time out (2)\n"); + dev_err(chip->card->dev, + "mce_down - auto calibration time out (2)\n"); } static void snd_cs4231_advance_dma(struct cs4231_dma_control *dma_cont, @@ -710,7 +714,7 @@ static void snd_cs4231_init(struct snd_cs4231 *chip) snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (1)\n"); + pr_debug("init: (1)\n"); #endif snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); @@ -725,7 +729,7 @@ static void snd_cs4231_init(struct snd_cs4231 *chip) snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (2)\n"); + pr_debug("init: (2)\n"); #endif snd_cs4231_mce_up(chip); @@ -736,8 +740,8 @@ static void snd_cs4231_init(struct snd_cs4231 *chip) snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (3) - afei = 0x%x\n", - chip->image[CS4231_ALT_FEATURE_1]); + pr_debug("init: (3) - afei = 0x%x\n", + chip->image[CS4231_ALT_FEATURE_1]); #endif spin_lock_irqsave(&chip->lock, flags); @@ -753,7 +757,7 @@ static void snd_cs4231_init(struct snd_cs4231 *chip) snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (4)\n"); + pr_debug("init: (4)\n"); #endif snd_cs4231_mce_up(chip); @@ -763,7 +767,7 @@ static void snd_cs4231_init(struct snd_cs4231 *chip) snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (5)\n"); + pr_debug("init: (5)\n"); #endif } @@ -1038,7 +1042,8 @@ static int snd_cs4231_probe(struct snd_cs4231 *chip) break; /* this is valid value */ } } - snd_printdd("cs4231: port = %p, id = 0x%x\n", chip->port, id); + dev_dbg(chip->card->dev, + "cs4231: port = %p, id = 0x%x\n", chip->port, id); if (id != 0x0a) return -ENODEV; /* no valid device found */ @@ -1794,7 +1799,8 @@ static int snd_cs4231_sbus_create(struct snd_card *card, chip->port = of_ioremap(&op->resource[0], 0, chip->regs_size, "cs4231"); if (!chip->port) { - snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; } @@ -1815,8 +1821,9 @@ static int snd_cs4231_sbus_create(struct snd_card *card, if (request_irq(op->archdata.irqs[0], snd_cs4231_sbus_interrupt, IRQF_SHARED, "cs4231", chip)) { - snd_printdd("cs4231-%d: Unable to grab SBUS IRQ %d\n", - dev, op->archdata.irqs[0]); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to grab SBUS IRQ %d\n", + dev, op->archdata.irqs[0]); snd_cs4231_sbus_free(chip); return -EBUSY; } @@ -1986,32 +1993,37 @@ static int snd_cs4231_ebus_create(struct snd_card *card, if (!chip->port || !chip->p_dma.ebus_info.regs || !chip->c_dma.ebus_info.regs) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; } if (ebus_dma_register(&chip->c_dma.ebus_info)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to register EBUS capture DMA\n", - dev); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to register EBUS capture DMA\n", + dev); return -EBUSY; } if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to enable EBUS capture IRQ\n", - dev); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to enable EBUS capture IRQ\n", + dev); return -EBUSY; } if (ebus_dma_register(&chip->p_dma.ebus_info)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to register EBUS play DMA\n", - dev); + dev_dbg(chpi->card->dev, + "cs4231-%d: Unable to register EBUS play DMA\n", + dev); return -EBUSY; } if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to enable EBUS play IRQ\n", dev); + dev_dbg(chip->card->dev, + "cs4231-%d: Unable to enable EBUS play IRQ\n", dev); return -EBUSY; } From 7f5485c4d3190dfc9294f347b2bfcce8c946b966 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:42 +0200 Subject: [PATCH 157/410] ALSA: asihpi: Use standard print API Use the standard print API with dev_*() instead of the old house-baked one. It gives better information and allows dynamically control of debug prints. The debug prints with a special macro snd_printddd() are replaced with the conditional pr_debug(). Unfortunately dev_dbg() can't be applied well due to the lack of the reference to the device pointer. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-53-tiwai@suse.de --- sound/pci/asihpi/asihpi.c | 101 ++++++++++++++++--------------------- sound/pci/asihpi/hpioctl.c | 2 +- 2 files changed, 44 insertions(+), 59 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 001786e2aba1..fdd4fe16225f 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -36,19 +36,10 @@ MODULE_AUTHOR("AudioScience inc. "); MODULE_DESCRIPTION("AudioScience ALSA ASI5xxx ASI6xxx ASI87xx ASI89xx " HPI_VER_STRING); -#if defined CONFIG_SND_DEBUG_VERBOSE -/** - * snd_printddd - very verbose debug printk - * @format: format string - * - * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. - * Must set snd module debug parameter to 3 to enable at runtime. - */ -#define snd_printddd(format, args...) \ - __snd_printk(3, __FILE__, __LINE__, format, ##args) +#ifdef ASIHPI_VERBOSE_DEBUG +#define asihpi_dbg(format, args...) pr_debug(format, ##args) #else -#define snd_printddd(format, args...) do { } while (0) +#define asihpi_dbg(format, args...) do { } while (0) #endif static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* index 0-MAX */ @@ -260,8 +251,7 @@ static inline u16 hpi_stream_group_reset(u32 h_stream) static u16 handle_error(u16 err, int line, char *filename) { if (err) - printk(KERN_WARNING - "in file %s, line %d: HPI error %d\n", + pr_warn("in file %s, line %d: HPI error %d\n", filename, line, err); return err; } @@ -273,16 +263,18 @@ static u16 handle_error(u16 err, int line, char *filename) static void print_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *p) { + struct device *dev = substream->pcm->card->dev; char name[16]; + snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printdd("%s HWPARAMS\n", name); - snd_printdd(" samplerate=%dHz channels=%d format=%d subformat=%d\n", + dev_dbg(dev, "%s HWPARAMS\n", name); + dev_dbg(dev, " samplerate=%dHz channels=%d format=%d subformat=%d\n", params_rate(p), params_channels(p), params_format(p), params_subformat(p)); - snd_printdd(" buffer=%dB period=%dB period_size=%dB periods=%d\n", + dev_dbg(dev, " buffer=%dB period=%dB period_size=%dB periods=%d\n", params_buffer_bytes(p), params_period_bytes(p), params_period_size(p), params_periods(p)); - snd_printdd(" buffer_size=%d access=%d data_rate=%dB/s\n", + dev_dbg(dev, " buffer_size=%d access=%d data_rate=%dB/s\n", params_buffer_size(p), params_access(p), params_rate(p) * params_channels(p) * snd_pcm_format_width(params_format(p)) / 8); @@ -317,7 +309,8 @@ static const snd_pcm_format_t hpi_to_alsa_formats[] = { }; -static int snd_card_asihpi_format_alsa2hpi(snd_pcm_format_t alsa_format, +static int snd_card_asihpi_format_alsa2hpi(struct snd_card_asihpi *asihpi, + snd_pcm_format_t alsa_format, u16 *hpi_format) { u16 format; @@ -330,8 +323,8 @@ static int snd_card_asihpi_format_alsa2hpi(snd_pcm_format_t alsa_format, } } - snd_printd(KERN_WARNING "failed match for alsa format %d\n", - alsa_format); + dev_dbg(asihpi->card->dev, "failed match for alsa format %d\n", + alsa_format); *hpi_format = 0; return -EINVAL; } @@ -439,7 +432,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, unsigned int bytes_per_sec; print_hwparams(substream, params); - err = snd_card_asihpi_format_alsa2hpi(params_format(params), &format); + err = snd_card_asihpi_format_alsa2hpi(card, params_format(params), &format); if (err) return err; @@ -461,13 +454,13 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, err = hpi_stream_host_buffer_attach(dpcm->h_stream, params_buffer_bytes(params), runtime->dma_addr); if (err == 0) { - snd_printdd( + dev_dbg(card->card->dev, "stream_host_buffer_attach success %u %lu\n", params_buffer_bytes(params), (unsigned long)runtime->dma_addr); } else { - snd_printd("stream_host_buffer_attach error %d\n", - err); + dev_dbg(card->card->dev, + "stream_host_buffer_attach error %d\n", err); return -ENOMEM; } @@ -569,7 +562,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: - snd_printdd("%s trigger start\n", name); + dev_dbg(card->card->dev, "%s trigger start\n", name); snd_pcm_group_for_each_entry(s, substream) { struct snd_pcm_runtime *runtime = s->runtime; struct snd_card_asihpi_pcm *ds = runtime->private_data; @@ -590,7 +583,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, * data?? */ unsigned int preload = ds->period_bytes * 1; - snd_printddd("%d preload %d\n", s->number, preload); + asihpi_dbg("%d preload %d\n", s->number, preload); hpi_handle_error(hpi_outstream_write_buf( ds->h_stream, &runtime->dma_area[0], @@ -600,7 +593,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, } if (card->support_grouping) { - snd_printdd("%d group\n", s->number); + dev_dbg(card->card->dev, "%d group\n", s->number); e = hpi_stream_group_add( dpcm->h_stream, ds->h_stream); @@ -621,7 +614,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_STOP: - snd_printdd("%s trigger stop\n", name); + dev_dbg(card->card->dev, "%s trigger stop\n", name); card->pcm_stop(substream); snd_pcm_group_for_each_entry(s, substream) { if (snd_pcm_substream_chip(s) != card) @@ -635,7 +628,7 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, __snd_pcm_set_state(s->runtime, SNDRV_PCM_STATE_SETUP); if (card->support_grouping) { - snd_printdd("%d group\n", s->number); + dev_dbg(card->card->dev, "%d group\n", s->number); snd_pcm_trigger_done(s, substream); } else break; @@ -652,17 +645,17 @@ static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream, break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_printdd("%s trigger pause release\n", name); + dev_dbg(card->card->dev, "%s trigger pause release\n", name); card->pcm_start(substream); hpi_handle_error(hpi_stream_start(dpcm->h_stream)); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - snd_printdd("%s trigger pause push\n", name); + dev_dbg(card->card->dev, "%s trigger pause push\n", name); card->pcm_stop(substream); hpi_handle_error(hpi_stream_stop(dpcm->h_stream)); break; default: - snd_printd(KERN_ERR "\tINVALID\n"); + dev_dbg(card->card->dev, "\tINVALID\n"); return -EINVAL; } @@ -760,12 +753,13 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) if (state == HPI_STATE_STOPPED) { if (bytes_avail == 0) { hpi_handle_error(hpi_stream_start(ds->h_stream)); - snd_printdd("P%d start\n", s->number); + dev_dbg(card->card->dev, + "P%d start\n", s->number); ds->drained_count = 0; } } else if (state == HPI_STATE_DRAINED) { - snd_printd(KERN_WARNING "P%d drained\n", - s->number); + dev_dbg(card->card->dev, + "P%d drained\n", s->number); ds->drained_count++; if (ds->drained_count > 20) { snd_pcm_stop_xrun(s); @@ -790,7 +784,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) newdata); } - snd_printddd( + asihpi_dbg( "timer1, %s, %d, S=%d, elap=%d, rw=%d, dsp=%d, left=%d, aux=%d, space=%d, hw_ptr=%ld, appl_ptr=%ld\n", name, s->number, state, ds->pcm_buf_elapsed_dma_ofs, @@ -821,7 +815,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) next_jiffies = max(next_jiffies, 1U); dpcm->timer.expires = jiffies + next_jiffies; - snd_printddd("timer2, jif=%d, buf_pos=%d, newdata=%d, xfer=%d\n", + asihpi_dbg("timer2, jif=%d, buf_pos=%d, newdata=%d, xfer=%d\n", next_jiffies, pcm_buf_dma_ofs, newdata, xfercount); snd_pcm_group_for_each_entry(s, substream) { @@ -854,7 +848,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) } if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { - snd_printddd("write1, P=%d, xfer=%d, buf_ofs=%d\n", + asihpi_dbg("write1, P=%d, xfer=%d, buf_ofs=%d\n", s->number, xfer1, buf_ofs); hpi_handle_error( hpi_outstream_write_buf( @@ -864,7 +858,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) if (xfer2) { pd = s->runtime->dma_area; - snd_printddd("write2, P=%d, xfer=%d, buf_ofs=%d\n", + asihpi_dbg("write2, P=%d, xfer=%d, buf_ofs=%d\n", s->number, xfercount - xfer1, buf_ofs); hpi_handle_error( @@ -874,7 +868,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) &ds->format)); } } else { - snd_printddd("read1, C=%d, xfer=%d\n", + asihpi_dbg("read1, C=%d, xfer=%d\n", s->number, xfer1); hpi_handle_error( hpi_instream_read_buf( @@ -882,7 +876,7 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) pd, xfer1)); if (xfer2) { pd = s->runtime->dma_area; - snd_printddd("read2, C=%d, xfer=%d\n", + asihpi_dbg("read2, C=%d, xfer=%d\n", s->number, xfer2); hpi_handle_error( hpi_instream_read_buf( @@ -919,8 +913,6 @@ static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream * struct snd_pcm_runtime *runtime = substream->runtime; struct snd_card_asihpi_pcm *dpcm = runtime->private_data; - snd_printdd("P%d prepare\n", substream->number); - hpi_handle_error(hpi_outstream_reset(dpcm->h_stream)); dpcm->pcm_buf_host_rw_ofs = 0; dpcm->pcm_buf_dma_ofs = 0; @@ -938,7 +930,7 @@ snd_card_asihpi_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_debug_name(substream, name, sizeof(name)); ptr = bytes_to_frames(runtime, dpcm->pcm_buf_dma_ofs % dpcm->buffer_bytes); - snd_printddd("%s, pointer=%ld\n", name, (unsigned long)ptr); + asihpi_dbg("%s, pointer=%ld\n", name, (unsigned long)ptr); return ptr; } @@ -1060,8 +1052,6 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, card->update_interval_frames, UINT_MAX); - snd_printdd("playback open\n"); - return 0; } @@ -1071,8 +1061,6 @@ static int snd_card_asihpi_playback_close(struct snd_pcm_substream *substream) struct snd_card_asihpi_pcm *dpcm = runtime->private_data; hpi_handle_error(hpi_outstream_close(dpcm->h_stream)); - snd_printdd("playback close\n"); - return 0; } @@ -1095,7 +1083,7 @@ snd_card_asihpi_capture_pointer(struct snd_pcm_substream *substream) char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); - snd_printddd("%s, pointer=%d\n", name, dpcm->pcm_buf_dma_ofs); + asihpi_dbg("%s, pointer=%d\n", name, dpcm->pcm_buf_dma_ofs); /* NOTE Unlike playback can't use actual samples_played for the capture position, because those samples aren't yet in the local buffer available for reading. @@ -1113,7 +1101,6 @@ static int snd_card_asihpi_capture_prepare(struct snd_pcm_substream *substream) dpcm->pcm_buf_dma_ofs = 0; dpcm->pcm_buf_elapsed_dma_ofs = 0; - snd_printdd("Capture Prepare %d\n", substream->number); return 0; } @@ -1162,8 +1149,9 @@ static int snd_card_asihpi_capture_open(struct snd_pcm_substream *substream) if (dpcm == NULL) return -ENOMEM; - snd_printdd("capture open adapter %d stream %d\n", - card->hpi->adapter->index, substream->number); + + dev_dbg(card->card->dev, "capture open adapter %d stream %d\n", + card->hpi->adapter->index, substream->number); err = hpi_handle_error( hpi_instream_open(card->hpi->adapter->index, @@ -1413,8 +1401,6 @@ static void asihpi_ctl_init(struct snd_kcontrol_new *snd_control, hpi_ctl->src_node_index, dir, name); } - /* printk(KERN_INFO "Adding %s %d to %d ", hpi_ctl->name, - hpi_ctl->wSrcNodeType, hpi_ctl->wDstNodeType); */ } /*------------------------------------------------------------ @@ -2175,9 +2161,8 @@ static int snd_asihpi_mux_get(struct snd_kcontrol *kcontrol, return 0; } } - snd_printd(KERN_WARNING - "Control %x failed to match mux source %hu %hu\n", - h_control, source_type, source_index); + pr_warn("%s: Control %x failed to match mux source %hu %hu\n", + __func__, h_control, source_type, source_index); ucontrol->value.enumerated.item[0] = 0; return 0; } diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 477a5b4b50bc..9fb0c8e503df 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -356,7 +356,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, memset(&adapter, 0, sizeof(adapter)); - dev_printk(KERN_DEBUG, &pci_dev->dev, + dev_dbg(&pci_dev->dev, "probe %04x:%04x,%04x:%04x,%04x\n", pci_dev->vendor, pci_dev->device, pci_dev->subsystem_vendor, pci_dev->subsystem_device, pci_dev->devfn); From 9acb51e9617c28a92f9ce2af767db6bd660a6d4f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:43 +0200 Subject: [PATCH 158/410] ALSA: docs: Drop snd_print*() stuff The legacy snd_print*() helpers will be dropped now. Clean up the corresponding documentation, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-54-tiwai@suse.de --- Documentation/sound/hd-audio/notes.rst | 6 ----- .../kernel-api/writing-an-alsa-driver.rst | 25 ------------------- 2 files changed, 31 deletions(-) diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst index ef6a4513cce7..e199131bf5ab 100644 --- a/Documentation/sound/hd-audio/notes.rst +++ b/Documentation/sound/hd-audio/notes.rst @@ -321,12 +321,6 @@ Kernel Configuration -------------------- In general, I recommend you to enable the sound debug option, ``CONFIG_SND_DEBUG=y``, no matter whether you are debugging or not. -This enables snd_printd() macro and others, and you'll get additional -kernel messages at probing. - -In addition, you can enable ``CONFIG_SND_DEBUG_VERBOSE=y``. But this -will give you far more messages. Thus turn this on only when you are -sure to want it. Don't forget to turn on the appropriate ``CONFIG_SND_HDA_CODEC_*`` options. Note that each of them corresponds to the codec chip, not diff --git a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst index 801b0bb57e97..895752cbcedd 100644 --- a/Documentation/sound/kernel-api/writing-an-alsa-driver.rst +++ b/Documentation/sound/kernel-api/writing-an-alsa-driver.rst @@ -4030,31 +4030,6 @@ located in the new subdirectory, sound/pci/xyz. Useful Functions ================ -:c:func:`snd_printk()` and friends ----------------------------------- - -.. note:: This subsection describes a few helper functions for - decorating a bit more on the standard :c:func:`printk()` & co. - However, in general, the use of such helpers is no longer recommended. - If possible, try to stick with the standard functions like - :c:func:`dev_err()` or :c:func:`pr_err()`. - -ALSA provides a verbose version of the :c:func:`printk()` function. -If a kernel config ``CONFIG_SND_VERBOSE_PRINTK`` is set, this function -prints the given message together with the file name and the line of the -caller. The ``KERN_XXX`` prefix is processed as well as the original -:c:func:`printk()` does, so it's recommended to add this prefix, -e.g. snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\\n"); - -There are also :c:func:`printk()`'s for debugging. -:c:func:`snd_printd()` can be used for general debugging purposes. -If ``CONFIG_SND_DEBUG`` is set, this function is compiled, and works -just like :c:func:`snd_printk()`. If the ALSA is compiled without -the debugging flag, it's ignored. - -:c:func:`snd_printdd()` is compiled in only when -``CONFIG_SND_DEBUG_VERBOSE`` is set. - :c:func:`snd_BUG()` ------------------- From 504dc9f5e62e3beef2554c453576e84993053647 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 15:34:44 +0200 Subject: [PATCH 159/410] ALSA: core: Drop snd_print stuff and co Now that all users of snd_print*() are gone, let's drop the functions completely. This also makes CONFIG_SND_VERBOSE_PRINTK redundant, and it's dropped, too. Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807133452.9424-55-tiwai@suse.de --- include/sound/core.h | 67 ----------------------------------------- sound/core/Kconfig | 9 ------ sound/core/misc.c | 71 -------------------------------------------- 3 files changed, 147 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index dfef0c9d4b9f..23401aaf3dc2 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -345,45 +345,7 @@ void release_and_free_resource(struct resource *res); /* --- */ -/* sound printk debug levels */ -enum { - SND_PR_ALWAYS, - SND_PR_DEBUG, - SND_PR_VERBOSE, -}; - -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) -__printf(4, 5) -void __snd_printk(unsigned int level, const char *file, int line, - const char *format, ...); -#else -#define __snd_printk(level, file, line, format, ...) \ - printk(format, ##__VA_ARGS__) -#endif - -/** - * snd_printk - printk wrapper - * @fmt: format string - * - * Works like printk() but prints the file and the line of the caller - * when configured with CONFIG_SND_VERBOSE_PRINTK. - */ -#define snd_printk(fmt, ...) \ - __snd_printk(0, __FILE__, __LINE__, fmt, ##__VA_ARGS__) - #ifdef CONFIG_SND_DEBUG -/** - * snd_printd - debug printk - * @fmt: format string - * - * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG is not set. - */ -#define snd_printd(fmt, ...) \ - __snd_printk(1, __FILE__, __LINE__, fmt, ##__VA_ARGS__) -#define _snd_printd(level, fmt, ...) \ - __snd_printk(level, __FILE__, __LINE__, fmt, ##__VA_ARGS__) - /** * snd_BUG - give a BUG warning message and stack trace * @@ -392,12 +354,6 @@ void __snd_printk(unsigned int level, const char *file, int line, */ #define snd_BUG() WARN(1, "BUG?\n") -/** - * snd_printd_ratelimit - Suppress high rates of output when - * CONFIG_SND_DEBUG is enabled. - */ -#define snd_printd_ratelimit() printk_ratelimit() - /** * snd_BUG_ON - debugging check macro * @cond: condition to evaluate @@ -409,11 +365,6 @@ void __snd_printk(unsigned int level, const char *file, int line, #else /* !CONFIG_SND_DEBUG */ -__printf(1, 2) -static inline void snd_printd(const char *format, ...) {} -__printf(2, 3) -static inline void _snd_printd(int level, const char *format, ...) {} - #define snd_BUG() do { } while (0) #define snd_BUG_ON(condition) ({ \ @@ -421,26 +372,8 @@ static inline void _snd_printd(int level, const char *format, ...) {} unlikely(__ret_warn_on); \ }) -static inline bool snd_printd_ratelimit(void) { return false; } - #endif /* CONFIG_SND_DEBUG */ -#ifdef CONFIG_SND_DEBUG_VERBOSE -/** - * snd_printdd - debug printk - * @format: format string - * - * Works like snd_printk() for debugging purposes. - * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set. - */ -#define snd_printdd(format, ...) \ - __snd_printk(2, __FILE__, __LINE__, format, ##__VA_ARGS__) -#else -__printf(1, 2) -static inline void snd_printdd(const char *format, ...) {} -#endif - - #define SNDRV_OSS_VERSION ((3<<16)|(8<<8)|(1<<4)|(0)) /* 3.8.1a */ /* for easier backward-porting */ diff --git a/sound/core/Kconfig b/sound/core/Kconfig index b970a1734647..f0ba87d4e504 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -175,15 +175,6 @@ config SND_VERBOSE_PROCFS useful information to developers when a problem occurs). On the other side, it makes the ALSA subsystem larger. -config SND_VERBOSE_PRINTK - bool "Verbose printk" - help - Say Y here to enable verbose log messages. These messages - will help to identify source file and position containing - printed messages. - - You don't need this unless you're debugging ALSA. - config SND_CTL_FAST_LOOKUP bool "Fast lookup of control elements" if EXPERT default y diff --git a/sound/core/misc.c b/sound/core/misc.c index d32a19976a2b..c2fda3bd90a0 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -13,20 +13,6 @@ #include #include -#ifdef CONFIG_SND_DEBUG - -#ifdef CONFIG_SND_DEBUG_VERBOSE -#define DEFAULT_DEBUG_LEVEL 2 -#else -#define DEFAULT_DEBUG_LEVEL 1 -#endif - -static int debug = DEFAULT_DEBUG_LEVEL; -module_param(debug, int, 0644); -MODULE_PARM_DESC(debug, "Debug level (0 = disable)"); - -#endif /* CONFIG_SND_DEBUG */ - void release_and_free_resource(struct resource *res) { if (res) { @@ -36,63 +22,6 @@ void release_and_free_resource(struct resource *res) } EXPORT_SYMBOL(release_and_free_resource); -#ifdef CONFIG_SND_VERBOSE_PRINTK -/* strip the leading path if the given path is absolute */ -static const char *sanity_file_name(const char *path) -{ - if (*path == '/') - return strrchr(path, '/') + 1; - else - return path; -} -#endif - -#if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) -void __snd_printk(unsigned int level, const char *path, int line, - const char *format, ...) -{ - va_list args; -#ifdef CONFIG_SND_VERBOSE_PRINTK - int kern_level; - struct va_format vaf; - char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; - bool level_found = false; -#endif - -#ifdef CONFIG_SND_DEBUG - if (debug < level) - return; -#endif - - va_start(args, format); -#ifdef CONFIG_SND_VERBOSE_PRINTK - vaf.fmt = format; - vaf.va = &args; - - while ((kern_level = printk_get_level(vaf.fmt)) != 0) { - const char *end_of_header = printk_skip_level(vaf.fmt); - - /* Ignore KERN_CONT. We print filename:line for each piece. */ - if (kern_level >= '0' && kern_level <= '7') { - memcpy(verbose_fmt, vaf.fmt, end_of_header - vaf.fmt); - level_found = true; - } - - vaf.fmt = end_of_header; - } - - if (!level_found && level) - memcpy(verbose_fmt, KERN_DEBUG, sizeof(KERN_DEBUG) - 1); - - printk(verbose_fmt, sanity_file_name(path), line, &vaf); -#else - vprintk(format, args); -#endif - va_end(args); -} -EXPORT_SYMBOL_GPL(__snd_printk); -#endif - #ifdef CONFIG_PCI #include /** From 7063a710830a09b01734be7f4ffd23f0ef72a57e Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 7 Aug 2024 14:27:15 +0000 Subject: [PATCH 160/410] ASoC: cs35l56: Use regmap_read_bypassed() to wake the device Now that regmap_read_bypassed() has been added to the kernel it is preferable to wake the device with a read rather than a write as the utility function can be called at a time before the device has been identified. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240807142715.47077-1-simont@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l56-shared.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e7e8d617da94..91b3c1c8575c 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -450,32 +450,23 @@ static const struct reg_sequence cs35l56_hibernate_seq[] = { REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_ALLOW_AUTO_HIBERNATE), }; -static const struct reg_sequence cs35l56_hibernate_wake_seq[] = { - REG_SEQ0(CS35L56_DSP_VIRTUAL1_MBOX_1, CS35L56_MBOX_CMD_WAKEUP), -}; - static void cs35l56_issue_wake_event(struct cs35l56_base *cs35l56_base) { + unsigned int val; + /* * Dummy transactions to trigger I2C/SPI auto-wake. Issue two * transactions to meet the minimum required time from the rising edge * to the last falling edge of wake. * - * It uses bypassed write because we must wake the chip before + * It uses bypassed read because we must wake the chip before * disabling regmap cache-only. - * - * This can NAK on I2C which will terminate the write sequence so the - * single-write sequence is issued twice. */ - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); usleep_range(CS35L56_WAKE_HOLD_TIME_US, 2 * CS35L56_WAKE_HOLD_TIME_US); - regmap_multi_reg_write_bypassed(cs35l56_base->regmap, - cs35l56_hibernate_wake_seq, - ARRAY_SIZE(cs35l56_hibernate_wake_seq)); + regmap_read_bypassed(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); cs35l56_wait_control_port_ready(); } From be942e3d20cf666f6174f47826914c3b5ea61d85 Mon Sep 17 00:00:00 2001 From: Zhang Yi Date: Wed, 7 Aug 2024 10:53:56 +0800 Subject: [PATCH 161/410] ASoC: codecs: ES8326: input issue after init We found an input issue after initiation.So we added the default input source at initiation. Signed-off-by: Zhang Yi Link: https://patch.msgid.link/20240807025356.24904-3-zhangyi@everest-semi.com Signed-off-by: Mark Brown --- sound/soc/codecs/es8326.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 60877116c0ef..132aed28b1aa 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -1068,6 +1068,8 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); + regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); msleep(200); From 20288905e1ee33af82570b79adee3f15018030d4 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 7 Aug 2024 10:38:39 +0530 Subject: [PATCH 162/410] ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver As module device table added for AMD SoundWire machine driver MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD SoundWire machine driver. Signed-off-by: Vijendar Mukunda Suggested-by: Krzysztof Kozlowski Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240807050846.1616725-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 3419675e45a9..f7e4af850309 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -738,5 +738,4 @@ module_platform_driver(sof_sdw_driver); MODULE_DESCRIPTION("ASoC AMD SoundWire Generic Machine driver"); MODULE_AUTHOR("Vijendar Mukunda Date: Wed, 7 Aug 2024 14:21:48 +0530 Subject: [PATCH 163/410] ASoC: amd: acp: add ZSC control register programming sequence Add ZSC Control register programming sequence for ACP D0 and D3 state transitions for ACP7.0 onwards. This will allow ACP to enter low power state when ACP enters D3 state. When ACP enters D0 State, ZSC control should be disabled. Tested-by: Leo Li Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240807085154.1987681-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-legacy-common.c | 5 +++++ sound/soc/amd/acp/amd.h | 2 ++ 2 files changed, 7 insertions(+) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 4422cec81e3c..04bd605fdce3 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -321,6 +321,8 @@ int acp_init(struct acp_chip_info *chip) pr_err("ACP reset failed\n"); return ret; } + if (chip->acp_rev >= ACP70_DEV) + writel(0, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_init, SND_SOC_ACP_COMMON); @@ -336,6 +338,9 @@ int acp_deinit(struct acp_chip_info *chip) if (chip->acp_rev != ACP70_DEV) writel(0, chip->base + ACP_CONTROL); + + if (chip->acp_rev >= ACP70_DEV) + writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } EXPORT_SYMBOL_NS_GPL(acp_deinit, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 87a4813783f9..c095a34a7229 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -103,6 +103,8 @@ #define ACP70_PGFSM_CONTROL ACP6X_PGFSM_CONTROL #define ACP70_PGFSM_STATUS ACP6X_PGFSM_STATUS +#define ACP_ZSC_DSP_CTRL 0x0001014 +#define ACP_ZSC_STS 0x0001018 #define ACP_SOFT_RST_DONE_MASK 0x00010001 #define ACP_PGFSM_CNTL_POWER_ON_MASK 0xffffffff From 5dde0cd2433dc6ef7b82e04276335137cc4c3105 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" Date: Mon, 5 Aug 2024 09:28:55 -0600 Subject: [PATCH 164/410] ASoC: SOF: sof-audio: Avoid -Wflex-array-member-not-at-end warnings -Wflex-array-member-not-at-end was introduced in GCC-14, and we are getting ready to enable it, globally. Move the conflicting declaration to the end of the structure. Notice that `struct snd_sof_pcm` ends in a flexible-array member through `struct snd_soc_tplg_pcm` -> `struct snd_soc_tplg_private`. Whith this, fix the following warnings: sound/soc/sof/sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] ./include/trace/events/../../../sound/soc/sof/sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] sound/soc/amd/../sof/amd/../sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] sound/soc/sof/amd/../sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] sound/soc/sof/intel/../sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] sound/soc/sof/mediatek/mt8186/../../sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] sound/soc/sof/mediatek/mt8195/../../sof-audio.h:350:33: warning: structure containing a flexible array member is not at the end of another structure [-Wflex-array-member-not-at-end] Signed-off-by: Gustavo A. R. Silva Link: https://patch.msgid.link/ZrDvt6eyeFyajq6l@cute Signed-off-by: Mark Brown --- sound/soc/sof/sof-audio.h | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index b9b8b64768b9..01b819dd8498 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -347,12 +347,14 @@ struct snd_sof_pcm_stream { /* ALSA SOF PCM device */ struct snd_sof_pcm { struct snd_soc_component *scomp; - struct snd_soc_tplg_pcm pcm; struct snd_sof_pcm_stream stream[2]; struct list_head list; /* list in sdev pcm list */ struct snd_pcm_hw_params params[2]; bool prepared[2]; /* PCM_PARAMS set successfully */ bool pending_stop[2]; /* only used if (!pcm_ops->platform_stop_during_hw_free) */ + + /* Must be last - ends in a flex-array member. */ + struct snd_soc_tplg_pcm pcm; }; struct snd_sof_led_control { From 001f8443d480773117a013a07f774d252f369bea Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 7 Aug 2024 10:43:17 +0530 Subject: [PATCH 165/410] ASoC: SOF: amd: update conditional check for cache register update Instead of desc->rev, use acp pci revision id(pci_rev) for cache register conditional check. Signed-off-by: Vijendar Mukunda Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240807051341.1616925-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-loader.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp-loader.c b/sound/soc/sof/amd/acp-loader.c index 2d5e58846499..19f10dd77e4b 100644 --- a/sound/soc/sof/amd/acp-loader.c +++ b/sound/soc/sof/amd/acp-loader.c @@ -219,7 +219,7 @@ int acp_dsp_pre_fw_run(struct snd_sof_dev *sdev) dev_err(sdev->dev, "acp dma transfer status: %d\n", ret); } - if (desc->rev > 3) { + if (adata->pci_rev > ACP_RN_PCI_ID) { /* Cache Window enable */ snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_OFFSET0, desc->sram_pte_offset); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DSP0_CACHE_SIZE0, SRAM1_SIZE | BIT(31)); From 7b986c7430a6bb68d523dac7bfc74cbd5b44ef96 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 11:14:42 +0200 Subject: [PATCH 166/410] ALSA: asihpi: Fix potential OOB array access ASIHPI driver stores some values in the static array upon a response from the driver, and its index depends on the firmware. We shouldn't trust it blindly. This patch adds a sanity check of the array index to fit in the array size. Link: https://patch.msgid.link/20240808091454.30846-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/asihpi/hpimsgx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index d0caef299481..b68e6bfbbfba 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -708,7 +708,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm, phr->error = HPI_ERROR_PROCESSING_MESSAGE; return phr->error; } - if (hr.error == 0) { + if (hr.error == 0 && hr.u.s.adapter_index < HPI_MAX_ADAPTERS) { /* the adapter was created successfully save the mapping for future use */ hpi_entry_points[hr.u.s.adapter_index] = entry_point_func; From c01f3815453e2d5f699ccd8c8c1f93a5b8669e59 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 11:15:12 +0200 Subject: [PATCH 167/410] ALSA: hdsp: Break infinite MIDI input flush loop The current MIDI input flush on HDSP and HDSPM drivers relies on the hardware reporting the right value. If the hardware doesn't give the proper value but returns -1, it may be stuck at an infinite loop. Add a counter and break if the loop is unexpectedly too long. Link: https://patch.msgid.link/20240808091513.31380-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdsp.c | 6 ++++-- sound/pci/rme9652/hdspm.c | 6 ++++-- 2 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index e7d1b43471a2..713ca262a0e9 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -1298,8 +1298,10 @@ static int snd_hdsp_midi_output_possible (struct hdsp *hdsp, int id) static void snd_hdsp_flush_midi_input (struct hdsp *hdsp, int id) { - while (snd_hdsp_midi_input_available (hdsp, id)) - snd_hdsp_midi_read_byte (hdsp, id); + int count = 256; + + while (snd_hdsp_midi_input_available(hdsp, id) && --count) + snd_hdsp_midi_read_byte(hdsp, id); } static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 56d335f0e196..c3f930a8f78d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1838,8 +1838,10 @@ static inline int snd_hdspm_midi_output_possible (struct hdspm *hdspm, int id) static void snd_hdspm_flush_midi_input(struct hdspm *hdspm, int id) { - while (snd_hdspm_midi_input_available (hdspm, id)) - snd_hdspm_midi_read_byte (hdspm, id); + int count = 256; + + while (snd_hdspm_midi_input_available(hdspm, id) && --count) + snd_hdspm_midi_read_byte(hdspm, id); } static int snd_hdspm_midi_output_write (struct hdspm_midi *hmidi) From 9b88d0890ed9adab413ea991fac90842688e1017 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 11:15:21 +0200 Subject: [PATCH 168/410] ALSA: usb-audio: Check shutdown at endpoint_set_interface() The call of usb_set_interface() and a delay are superfluous when the device has been already disconnected. Add a disconnection check before doing it. Link: https://patch.msgid.link/20240808091522.31415-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 8f65349a06d3..568099467dbb 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -921,6 +921,9 @@ static int endpoint_set_interface(struct snd_usb_audio *chip, if (ep->iface_ref->altset == altset) return 0; + /* already disconnected? */ + if (unlikely(atomic_read(&chip->shutdown))) + return -ENODEV; usb_audio_dbg(chip, "Setting usb interface %d:%d for EP 0x%x\n", ep->iface, altset, ep->ep_num); From c37f7cd7e5b6c2aabe0d5b2220545789517b064a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 18:29:01 +0200 Subject: [PATCH 169/410] ALSA: vxpocket: Drop no longer existent chip->dev assignment The recent cleanup change for vx_core overlooked the code in vxpocket pcmcia driver. Kill the superfluous line as well. Fixes: b426b3ba9f6f ("ALSA: vx_core: Drop unused dev field") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202408090016.kW0TA6fc-lkp@intel.com/ Link: https://patch.msgid.link/20240808162902.20082-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxpocket.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 2ea62efa0064..5f72c73eac85 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -204,8 +204,6 @@ static int vxpocket_config(struct pcmcia_device *link) if (ret) goto failed; - chip->dev = &link->dev; - if (snd_vxpocket_assign_resources(chip, link->resource[0]->start, link->irq) < 0) goto failed; From a1066453b5e49a28523f3ecbbfe4e06c6a29561c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 18:31:27 +0200 Subject: [PATCH 170/410] ALSA: control: Fix power_ref lock order for compat code, too In the previous change for swapping the power_ref and controls_rwsem lock order, the code path for the compat layer was forgotten. This patch covers the remaining code. Fixes: fcc62b19104a ("ALSA: control: Take power_ref lock primarily") Link: https://patch.msgid.link/20240808163128.20383-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/control_compat.c | 45 ++++++++++++++++++++++++++++++------- 1 file changed, 37 insertions(+), 8 deletions(-) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 934bb945e702..ff0031cc7dfb 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -79,6 +79,7 @@ struct snd_ctl_elem_info32 { static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, struct snd_ctl_elem_info32 __user *data32) { + struct snd_card *card = ctl->card; struct snd_ctl_elem_info *data __free(kfree) = NULL; int err; @@ -95,7 +96,11 @@ static int snd_ctl_elem_info_compat(struct snd_ctl_file *ctl, if (get_user(data->value.enumerated.item, &data32->value.enumerated.item)) return -EFAULT; + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; err = snd_ctl_elem_info(ctl, data); + snd_power_unref(card); if (err < 0) return err; /* restore info to 32bit */ @@ -175,10 +180,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, if (info == NULL) return -ENOMEM; info->id = *id; - err = snd_power_ref_and_wait(card); - if (!err) - err = kctl->info(kctl, info); - snd_power_unref(card); + err = kctl->info(kctl, info); if (err >= 0) { err = info->type; *countp = info->count; @@ -275,8 +277,8 @@ static int copy_ctl_value_to_user(void __user *userdata, return 0; } -static int ctl_elem_read_user(struct snd_card *card, - void __user *userdata, void __user *valuep) +static int __ctl_elem_read_user(struct snd_card *card, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data __free(kfree) = NULL; int err, type, count; @@ -296,8 +298,21 @@ static int ctl_elem_read_user(struct snd_card *card, return copy_ctl_value_to_user(userdata, valuep, data, type, count); } -static int ctl_elem_write_user(struct snd_ctl_file *file, - void __user *userdata, void __user *valuep) +static int ctl_elem_read_user(struct snd_card *card, + void __user *userdata, void __user *valuep) +{ + int err; + + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; + err = __ctl_elem_read_user(card, userdata, valuep); + snd_power_unref(card); + return err; +} + +static int __ctl_elem_write_user(struct snd_ctl_file *file, + void __user *userdata, void __user *valuep) { struct snd_ctl_elem_value *data __free(kfree) = NULL; struct snd_card *card = file->card; @@ -318,6 +333,20 @@ static int ctl_elem_write_user(struct snd_ctl_file *file, return copy_ctl_value_to_user(userdata, valuep, data, type, count); } +static int ctl_elem_write_user(struct snd_ctl_file *file, + void __user *userdata, void __user *valuep) +{ + struct snd_card *card = file->card; + int err; + + err = snd_power_ref_and_wait(card); + if (err < 0) + return err; + err = __ctl_elem_write_user(file, userdata, valuep); + snd_power_unref(card); + return err; +} + static int snd_ctl_elem_read_user_compat(struct snd_card *card, struct snd_ctl_elem_value32 __user *data32) { From e95b9f7f2ee05bbef3bf6a4cd7da6e887f17f652 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 8 Aug 2024 15:48:54 +0200 Subject: [PATCH 171/410] ALSA: snd-usb-caiaq: use snd_pcm_rate_to_rate_bit Use snd_pcm_rate_to_rate_bit() helper provided by Alsa instead re-implementing it. This reduce code duplication and helps when changing some Alsa definition is necessary. Signed-off-by: Jerome Brunet Link: https://patch.msgid.link/20240808134857.86749-1-jbrunet@baylibre.com Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 4981753652a7..e62a4ea1d19c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -174,14 +174,6 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) return 0; } -/* this should probably go upstream */ -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 -#error "Change this table" -#endif - -static const unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; - static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { int bytes_per_sample, bpp, ret, i; @@ -233,10 +225,7 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) /* the first client that opens a stream defines the sample rate * setting for all subsequent calls, until the last client closed. */ - for (i=0; i < ARRAY_SIZE(rates); i++) - if (runtime->rate == rates[i]) - cdev->pcm_info.rates = 1 << i; - + cdev->pcm_info.rates = snd_pcm_rate_to_rate_bit(runtime->rate); snd_pcm_limit_hw_rates(runtime); bytes_per_sample = BYTES_PER_SAMPLE; From c46fc83e3f3c89f18962e43890de90b1c304747a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2024 20:23:07 +0200 Subject: [PATCH 172/410] ALSA: vxpocket: Fix a typo at conversion to dev_*() There was a typo in the previous conversion to dev_*() that caused a build error. Fix it. Fixes: 2acbb5e57230 ("ALSA: vxpocket: Use standard print API") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202408090110.t0mWbTyh-lkp@intel.com/ Link: https://patch.msgid.link/20240808182308.28418-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pcmcia/vx/vxpocket.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 5f72c73eac85..d2d5f64d63b4 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -268,7 +268,7 @@ static int vxpocket_probe(struct pcmcia_device *p_dev) err = snd_card_new(&p_dev->dev, index[i], id[i], THIS_MODULE, 0, &card); if (err < 0) { - dev_err(&pdev->dev, "vxpocket: cannot create a card instance\n"); + dev_err(&p_dev->dev, "vxpocket: cannot create a card instance\n"); return err; } From 528baf4f62dd57dee1a1076390b0921ad32b88a8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 09:39:21 +0200 Subject: [PATCH 173/410] ALSA: sparc: Fix a typo at dev_*() conversion There was a stupid typo left that broke the build. Fix it. Fixes: d41abde89483 ("ALSA: sparc: Use standard print API") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202408090648.J2EAijjH-lkp@intel.com/ Link: https://patch.msgid.link/20240809073923.3735-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/sparc/cs4231.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 5f39b7847d31..a1339f9ef12a 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -2015,7 +2015,7 @@ static int snd_cs4231_ebus_create(struct snd_card *card, if (ebus_dma_register(&chip->p_dma.ebus_info)) { snd_cs4231_ebus_free(chip); - dev_dbg(chpi->card->dev, + dev_dbg(chip->card->dev, "cs4231-%d: Unable to register EBUS play DMA\n", dev); return -EBUSY; From af1d53b6e0ebef5ea57887e23619b2bb33a6f4de Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 09:42:51 +0200 Subject: [PATCH 174/410] ALSA: caiaq: Fix unused variable warning The recent cleanup of caiaq driver forgot to remove the unused loop variable: sound/usb/caiaq/audio.c: In function 'snd_usb_caiaq_pcm_prepare': sound/usb/caiaq/audio.c:179:41: error: unused variable 'i' [-Werror=unused-variable] Fixes: e95b9f7f2ee0 ("ALSA: snd-usb-caiaq: use snd_pcm_rate_to_rate_bit") Reported-by: Stephen Rothwell Closes: https://lore.kernel.org/20240809112252.5af8025f@canb.auug.org.au Link: https://patch.msgid.link/20240809074254.5290-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/caiaq/audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index e62a4ea1d19c..772c0ecb7077 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -176,7 +176,7 @@ static int snd_usb_caiaq_pcm_hw_free(struct snd_pcm_substream *sub) static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) { - int bytes_per_sample, bpp, ret, i; + int bytes_per_sample, bpp, ret; int index = substream->number; struct snd_usb_caiaqdev *cdev = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; From f6c9a097b55e1955e3dd35f1de4828d3ed67534c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 09:56:59 +0200 Subject: [PATCH 175/410] ALSA: usx2y: Drop no longer used variable The recent conversion to the standard print API included some cleanups and that changed the code no longer referring to a variable iters at usb_stream_start(). This caused a compiler warning in the end. Let's drop the unused variable. Fixes: f8466d91f36d ("ALSA: usx2y: Use standard print API") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202408090249.8LE9qrae-lkp@intel.com/ Link: https://patch.msgid.link/20240809075700.7320-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usb_stream.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 66ea610aa433..3122cf653273 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -665,7 +665,7 @@ static void i_playback_start(struct urb *urb) int usb_stream_start(struct usb_stream_kernel *sk) { struct usb_stream *s = sk->s; - int frame = 0, iters = 0; + int frame = 0; int u, err; int try = 0; @@ -700,7 +700,6 @@ dotry: frame = usb_get_current_frame_number(dev); do { now = usb_get_current_frame_number(dev); - ++iters; } while (now > -1 && now == frame); } err = usb_submit_urb(inurb, GFP_ATOMIC); From f428cc9eac6e29d57579be4978ba210c344322ea Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 12:42:29 +0200 Subject: [PATCH 176/410] ALSA: control: Rename ctl_files_rwlock to controls_rwlock We'll re-use the existing rwlock for the protection of control list lookup, too, and now rename it to a more generic name. This is a preliminary change, only the rename of the struct field here, no functional changes. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240809104234.8488-2-tiwai@suse.de --- include/sound/core.h | 2 +- sound/core/control.c | 10 +++++----- sound/core/init.c | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index dfef0c9d4b9f..55607e91d5fd 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -99,7 +99,7 @@ struct snd_card { struct device *ctl_dev; /* control device */ unsigned int last_numid; /* last used numeric ID */ struct rw_semaphore controls_rwsem; /* controls lock (list and values) */ - rwlock_t ctl_files_rwlock; /* ctl_files list lock */ + rwlock_t controls_rwlock; /* lock for ctl_files list */ int controls_count; /* count of all controls */ size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ diff --git a/sound/core/control.c b/sound/core/control.c index f64a555f404f..7c4410d1eeeb 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -79,7 +79,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) ctl->preferred_subdevice[i] = -1; ctl->pid = get_pid(task_pid(current)); file->private_data = ctl; - scoped_guard(write_lock_irqsave, &card->ctl_files_rwlock) + scoped_guard(write_lock_irqsave, &card->controls_rwlock) list_add_tail(&ctl->list, &card->ctl_files); snd_card_unref(card); return 0; @@ -117,7 +117,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) file->private_data = NULL; card = ctl->card; - scoped_guard(write_lock_irqsave, &card->ctl_files_rwlock) + scoped_guard(write_lock_irqsave, &card->controls_rwlock) list_del(&ctl->list); scoped_guard(rwsem_write, &card->controls_rwsem) { @@ -157,7 +157,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, if (card->shutdown) return; - guard(read_lock_irqsave)(&card->ctl_files_rwlock); + guard(read_lock_irqsave)(&card->controls_rwlock); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif @@ -2178,7 +2178,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) struct snd_ctl_file *kctl; int subdevice = -1; - guard(read_lock_irqsave)(&card->ctl_files_rwlock); + guard(read_lock_irqsave)(&card->controls_rwlock); list_for_each_entry(kctl, &card->ctl_files, list) { if (kctl->pid == task_pid(current)) { subdevice = kctl->preferred_subdevice[type]; @@ -2328,7 +2328,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; - scoped_guard(read_lock_irqsave, &card->ctl_files_rwlock) { + scoped_guard(read_lock_irqsave, &card->controls_rwlock) { list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); diff --git a/sound/core/init.c b/sound/core/init.c index b9b708cf980d..b92aa7103589 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -315,7 +315,7 @@ static int snd_card_init(struct snd_card *card, struct device *parent, card->module = module; INIT_LIST_HEAD(&card->devices); init_rwsem(&card->controls_rwsem); - rwlock_init(&card->ctl_files_rwlock); + rwlock_init(&card->controls_rwlock); INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); #ifdef CONFIG_SND_CTL_FAST_LOOKUP From 38ea4c3dc306edf6f4e483e8ad9cb8d33943afde Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 12:42:30 +0200 Subject: [PATCH 177/410] ALSA: control: Optimize locking for look-up For a fast look-up of a control element via either numid or name matching (enabled via CONFIG_SND_CTL_FAST_LOOKUP), a locking isn't needed at all thanks to Xarray. OTOH, the locking is still needed for a slow linked-list traversal, and that's rather a rare case. In this patch, we reduce the use of locking at snd_ctl_find_*() API functions, and switch from controls_rwsem to controls_rwlock for avoiding unnecessary lock inversions. This also resulted in a nice cleanup, as *_unlocked() version of snd_ctl_find_*() APIs can be dropped. snd_ctl_find_id_mixer_unlocked() is still left just as an alias of snd_ctl_find_id_mixer(), since soc-card.c has a wrapper and there are several users. Once after converting there, we can remove it later. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240809104234.8488-3-tiwai@suse.de --- include/sound/control.h | 22 +----- include/sound/core.h | 2 +- sound/core/control.c | 141 ++++++++++++++---------------------- sound/core/control_compat.c | 2 +- sound/core/control_led.c | 2 +- sound/core/oss/mixer_oss.c | 10 +-- 6 files changed, 64 insertions(+), 115 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index c1659036c4a7..4851a86c72ef 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -140,9 +140,7 @@ int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id); void snd_ctl_rename(struct snd_card *card, struct snd_kcontrol *kctl, const char *name); int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int active); -struct snd_kcontrol *snd_ctl_find_numid_locked(struct snd_card *card, unsigned int numid); struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid); -struct snd_kcontrol *snd_ctl_find_id_locked(struct snd_card *card, const struct snd_ctl_elem_id *id); struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, const struct snd_ctl_elem_id *id); /** @@ -167,27 +165,11 @@ snd_ctl_find_id_mixer(struct snd_card *card, const char *name) return snd_ctl_find_id(card, &id); } -/** - * snd_ctl_find_id_mixer_locked - find the control instance with the given name string - * @card: the card instance - * @name: the name string - * - * Finds the control instance with the given name and - * @SNDRV_CTL_ELEM_IFACE_MIXER. Other fields are set to zero. - * - * This is merely a wrapper to snd_ctl_find_id_locked(). - * The caller must down card->controls_rwsem before calling this function. - * - * Return: The pointer of the instance if found, or %NULL if not. - */ +/* to be removed */ static inline struct snd_kcontrol * snd_ctl_find_id_mixer_locked(struct snd_card *card, const char *name) { - struct snd_ctl_elem_id id = {}; - - id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; - strscpy(id.name, name, sizeof(id.name)); - return snd_ctl_find_id_locked(card, &id); + return snd_ctl_find_id_mixer(card, name); } int snd_ctl_create(struct snd_card *card); diff --git a/include/sound/core.h b/include/sound/core.h index 55607e91d5fd..5c418ab24aa4 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -99,7 +99,7 @@ struct snd_card { struct device *ctl_dev; /* control device */ unsigned int last_numid; /* last used numeric ID */ struct rw_semaphore controls_rwsem; /* controls lock (list and values) */ - rwlock_t controls_rwlock; /* lock for ctl_files list */ + rwlock_t controls_rwlock; /* lock for lookup and ctl_files list */ int controls_count; /* count of all controls */ size_t user_ctl_alloc_size; // current memory allocation by user controls. struct list_head controls; /* all controls for this card */ diff --git a/sound/core/control.c b/sound/core/control.c index 7c4410d1eeeb..38728b00f59c 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -470,7 +470,7 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (id.index > UINT_MAX - kcontrol->count) return -EINVAL; - old = snd_ctl_find_id_locked(card, &id); + old = snd_ctl_find_id(card, &id); if (!old) { if (mode == CTL_REPLACE) return -EINVAL; @@ -491,10 +491,12 @@ static int __snd_ctl_add_replace(struct snd_card *card, if (snd_ctl_find_hole(card, kcontrol->count) < 0) return -ENOMEM; - list_add_tail(&kcontrol->list, &card->controls); - card->controls_count += kcontrol->count; - kcontrol->id.numid = card->last_numid + 1; - card->last_numid += kcontrol->count; + scoped_guard(write_lock_irq, &card->controls_rwlock) { + list_add_tail(&kcontrol->list, &card->controls); + card->controls_count += kcontrol->count; + kcontrol->id.numid = card->last_numid + 1; + card->last_numid += kcontrol->count; + } add_hash_entries(card, kcontrol); @@ -579,12 +581,15 @@ static int __snd_ctl_remove(struct snd_card *card, if (snd_BUG_ON(!card || !kcontrol)) return -EINVAL; - list_del(&kcontrol->list); if (remove_hash) remove_hash_entries(card, kcontrol); - card->controls_count -= kcontrol->count; + scoped_guard(write_lock_irq, &card->controls_rwlock) { + list_del(&kcontrol->list); + card->controls_count -= kcontrol->count; + } + for (idx = 0; idx < kcontrol->count; idx++) snd_ctl_notify_one(card, SNDRV_CTL_EVENT_MASK_REMOVE, kcontrol, idx); snd_ctl_free_one(kcontrol); @@ -634,7 +639,7 @@ int snd_ctl_remove_id(struct snd_card *card, struct snd_ctl_elem_id *id) struct snd_kcontrol *kctl; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); + kctl = snd_ctl_find_id(card, id); if (kctl == NULL) return -ENOENT; return snd_ctl_remove_locked(card, kctl); @@ -659,7 +664,7 @@ static int snd_ctl_remove_user_ctl(struct snd_ctl_file * file, int idx; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); + kctl = snd_ctl_find_id(card, id); if (kctl == NULL) return -ENOENT; if (!(kctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_USER)) @@ -691,7 +696,7 @@ int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id, int ret; down_write(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); + kctl = snd_ctl_find_id(card, id); if (kctl == NULL) { ret = -ENOENT; goto unlock; @@ -745,7 +750,7 @@ int snd_ctl_rename_id(struct snd_card *card, struct snd_ctl_elem_id *src_id, int saved_numid; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, src_id); + kctl = snd_ctl_find_id(card, src_id); if (kctl == NULL) return -ENOENT; saved_numid = kctl->id.numid; @@ -787,6 +792,7 @@ snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) { struct snd_kcontrol *kctl; + guard(read_lock_irqsave)(&card->controls_rwlock); list_for_each_entry(kctl, &card->controls, list) { if (kctl->id.numid <= numid && kctl->id.numid + kctl->count > numid) return kctl; @@ -795,32 +801,6 @@ snd_ctl_find_numid_slow(struct snd_card *card, unsigned int numid) } #endif /* !CONFIG_SND_CTL_FAST_LOOKUP */ -/** - * snd_ctl_find_numid_locked - find the control instance with the given number-id - * @card: the card instance - * @numid: the number-id to search - * - * Finds the control instance with the given number-id from the card. - * - * The caller must down card->controls_rwsem before calling this function - * (if the race condition can happen). - * - * Return: The pointer of the instance if found, or %NULL if not. - */ -struct snd_kcontrol * -snd_ctl_find_numid_locked(struct snd_card *card, unsigned int numid) -{ - if (snd_BUG_ON(!card || !numid)) - return NULL; - lockdep_assert_held(&card->controls_rwsem); -#ifdef CONFIG_SND_CTL_FAST_LOOKUP - return xa_load(&card->ctl_numids, numid); -#else - return snd_ctl_find_numid_slow(card, numid); -#endif -} -EXPORT_SYMBOL(snd_ctl_find_numid_locked); - /** * snd_ctl_find_numid - find the control instance with the given number-id * @card: the card instance @@ -830,54 +810,22 @@ EXPORT_SYMBOL(snd_ctl_find_numid_locked); * * Return: The pointer of the instance if found, or %NULL if not. * - * Note that this function takes card->controls_rwsem lock internally. + * Note that this function takes card->controls_rwlock lock internally. */ struct snd_kcontrol *snd_ctl_find_numid(struct snd_card *card, unsigned int numid) { - guard(rwsem_read)(&card->controls_rwsem); - return snd_ctl_find_numid_locked(card, numid); + if (snd_BUG_ON(!card || !numid)) + return NULL; + +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + return xa_load(&card->ctl_numids, numid); +#else + return snd_ctl_find_numid_slow(card, numid); +#endif } EXPORT_SYMBOL(snd_ctl_find_numid); -/** - * snd_ctl_find_id_locked - find the control instance with the given id - * @card: the card instance - * @id: the id to search - * - * Finds the control instance with the given id from the card. - * - * The caller must down card->controls_rwsem before calling this function - * (if the race condition can happen). - * - * Return: The pointer of the instance if found, or %NULL if not. - */ -struct snd_kcontrol *snd_ctl_find_id_locked(struct snd_card *card, - const struct snd_ctl_elem_id *id) -{ - struct snd_kcontrol *kctl; - - if (snd_BUG_ON(!card || !id)) - return NULL; - lockdep_assert_held(&card->controls_rwsem); - if (id->numid != 0) - return snd_ctl_find_numid_locked(card, id->numid); -#ifdef CONFIG_SND_CTL_FAST_LOOKUP - kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); - if (kctl && elem_id_matches(kctl, id)) - return kctl; - if (!card->ctl_hash_collision) - return NULL; /* we can rely on only hash table */ -#endif - /* no matching in hash table - try all as the last resort */ - list_for_each_entry(kctl, &card->controls, list) - if (elem_id_matches(kctl, id)) - return kctl; - - return NULL; -} -EXPORT_SYMBOL(snd_ctl_find_id_locked); - /** * snd_ctl_find_id - find the control instance with the given id * @card: the card instance @@ -887,13 +835,32 @@ EXPORT_SYMBOL(snd_ctl_find_id_locked); * * Return: The pointer of the instance if found, or %NULL if not. * - * Note that this function takes card->controls_rwsem lock internally. + * Note that this function takes card->controls_rwlock lock internally. */ struct snd_kcontrol *snd_ctl_find_id(struct snd_card *card, const struct snd_ctl_elem_id *id) { - guard(rwsem_read)(&card->controls_rwsem); - return snd_ctl_find_id_locked(card, id); + struct snd_kcontrol *kctl; + + if (snd_BUG_ON(!card || !id)) + return NULL; + + if (id->numid != 0) + return snd_ctl_find_numid(card, id->numid); +#ifdef CONFIG_SND_CTL_FAST_LOOKUP + kctl = xa_load(&card->ctl_hash, get_ctl_id_hash(id)); + if (kctl && elem_id_matches(kctl, id)) + return kctl; + if (!card->ctl_hash_collision) + return NULL; /* we can rely on only hash table */ +#endif + /* no matching in hash table - try all as the last resort */ + guard(read_lock_irqsave)(&card->controls_rwlock); + list_for_each_entry(kctl, &card->controls, list) + if (elem_id_matches(kctl, id)) + return kctl; + + return NULL; } EXPORT_SYMBOL(snd_ctl_find_id); @@ -1199,7 +1166,7 @@ static int snd_ctl_elem_info(struct snd_ctl_file *ctl, struct snd_kcontrol *kctl; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, &info->id); + kctl = snd_ctl_find_id(card, &info->id); if (!kctl) return -ENOENT; return __snd_ctl_elem_info(card, kctl, info, ctl); @@ -1235,7 +1202,7 @@ static int snd_ctl_elem_read(struct snd_card *card, int ret; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, &control->id); + kctl = snd_ctl_find_id(card, &control->id); if (!kctl) return -ENOENT; @@ -1303,7 +1270,7 @@ static int snd_ctl_elem_write(struct snd_card *card, struct snd_ctl_file *file, int result; down_write(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, &control->id); + kctl = snd_ctl_find_id(card, &control->id); if (kctl == NULL) { up_write(&card->controls_rwsem); return -ENOENT; @@ -1381,7 +1348,7 @@ static int snd_ctl_elem_lock(struct snd_ctl_file *file, if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, &id); + kctl = snd_ctl_find_id(card, &id); if (!kctl) return -ENOENT; vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; @@ -1402,7 +1369,7 @@ static int snd_ctl_elem_unlock(struct snd_ctl_file *file, if (copy_from_user(&id, _id, sizeof(id))) return -EFAULT; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, &id); + kctl = snd_ctl_find_id(card, &id); if (!kctl) return -ENOENT; vd = &kctl->vd[snd_ctl_get_ioff(kctl, &id)]; @@ -1903,7 +1870,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, container_size = header.length; container = buf->tlv; - kctl = snd_ctl_find_numid_locked(file->card, header.numid); + kctl = snd_ctl_find_numid(file->card, header.numid); if (kctl == NULL) return -ENOENT; diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 934bb945e702..401c12e62816 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -168,7 +168,7 @@ static int get_ctl_type(struct snd_card *card, struct snd_ctl_elem_id *id, int err; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); + kctl = snd_ctl_find_id(card, id); if (!kctl) return -ENOENT; info = kzalloc(sizeof(*info), GFP_KERNEL); diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 804805a95e2f..14eb3e6cc94c 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -254,7 +254,7 @@ static int snd_ctl_led_set_id(int card_number, struct snd_ctl_elem_id *id, if (!card) return -ENXIO; guard(rwsem_write)(&card->controls_rwsem); - kctl = snd_ctl_find_id_locked(card, id); + kctl = snd_ctl_find_id(card, id); if (!kctl) return -ENOENT; ioff = snd_ctl_get_ioff(kctl, id); diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 6a0508093ea6..33bf9a220ada 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -510,7 +510,7 @@ static struct snd_kcontrol *snd_mixer_oss_test_id(struct snd_mixer_oss *mixer, c id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; strscpy(id.name, name, sizeof(id.name)); id.index = index; - return snd_ctl_find_id_locked(card, &id); + return snd_ctl_find_id(card, &id); } static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer, @@ -526,7 +526,7 @@ static void snd_mixer_oss_get_volume1_vol(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_numid_locked(card, numid); + kctl = snd_ctl_find_numid(card, numid); if (!kctl) return; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); @@ -559,7 +559,7 @@ static void snd_mixer_oss_get_volume1_sw(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_numid_locked(card, numid); + kctl = snd_ctl_find_numid(card, numid); if (!kctl) return; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); @@ -619,7 +619,7 @@ static void snd_mixer_oss_put_volume1_vol(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_numid_locked(card, numid); + kctl = snd_ctl_find_numid(card, numid); if (!kctl) return; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); @@ -656,7 +656,7 @@ static void snd_mixer_oss_put_volume1_sw(struct snd_mixer_oss_file *fmixer, if (numid == ID_UNKNOWN) return; guard(rwsem_read)(&card->controls_rwsem); - kctl = snd_ctl_find_numid_locked(card, numid); + kctl = snd_ctl_find_numid(card, numid); if (!kctl) return; uinfo = kzalloc(sizeof(*uinfo), GFP_KERNEL); From 9cacb32a0ba691c2859a20bd5e30b71cc592fad2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2024 12:42:31 +0200 Subject: [PATCH 178/410] ASoC: Drop snd_soc_*_get_kcontrol_locked() The recent cleanup in ALSA control core made no difference between snd_ctl_find_id_mixer() and snd_ctl_find_id_mixer_locked(), and the latter is to be dropped. The only user of the left API was ASoC, and that's snd_soc_card_get_kcontrol_locked() and snd_soc_component_get_kcontrol_locked(). This patch drops those functions and rewrites those users to call the variant without locked instead. The test of the API became superfluous, hence dropped as well. As all callers of snd_ctl_find_id_mixer_locked() are gone, snd_ctl_find_id_mixer_locked() is finally dropped, too. Reviewed-by: Mark Brown Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240809104234.8488-4-tiwai@suse.de --- include/sound/control.h | 7 ----- include/sound/soc-card.h | 2 -- include/sound/soc-component.h | 3 -- sound/soc/codecs/cs35l45.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/soc-card-test.c | 57 ----------------------------------- sound/soc/soc-card.c | 10 ------ sound/soc/soc-component.c | 12 -------- 8 files changed, 2 insertions(+), 93 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 4851a86c72ef..e31bd8e36bfa 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -165,13 +165,6 @@ snd_ctl_find_id_mixer(struct snd_card *card, const char *name) return snd_ctl_find_id(card, &id); } -/* to be removed */ -static inline struct snd_kcontrol * -snd_ctl_find_id_mixer_locked(struct snd_card *card, const char *name) -{ - return snd_ctl_find_id_mixer(card, name); -} - int snd_ctl_create(struct snd_card *card); int snd_ctl_register_ioctl(snd_kctl_ioctl_func_t fcn); diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h index 1f4c39922d82..ecc02e955279 100644 --- a/include/sound/soc-card.h +++ b/include/sound/soc-card.h @@ -30,8 +30,6 @@ static inline void snd_soc_card_mutex_unlock(struct snd_soc_card *card) struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name); -struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, - const char *name); int snd_soc_card_jack_new(struct snd_soc_card *card, const char *id, int type, struct snd_soc_jack *jack); int snd_soc_card_jack_new_pins(struct snd_soc_card *card, const char *id, diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index bf2e381cd124..61534ac0edd1 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -464,9 +464,6 @@ int snd_soc_component_force_enable_pin_unlocked( /* component controls */ struct snd_kcontrol *snd_soc_component_get_kcontrol(struct snd_soc_component *component, const char * const ctl); -struct snd_kcontrol * -snd_soc_component_get_kcontrol_locked(struct snd_soc_component *component, - const char * const ctl); int snd_soc_component_notify_control(struct snd_soc_component *component, const char * const ctl); diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index 1e9d73bee3b4..fa1d9d9151f9 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -177,7 +177,7 @@ static int cs35l45_activate_ctl(struct snd_soc_component *component, struct snd_kcontrol_volatile *vd; unsigned int index_offset; - kcontrol = snd_soc_component_get_kcontrol_locked(component, ctl_name); + kcontrol = snd_soc_component_get_kcontrol(component, ctl_name); if (!kcontrol) { dev_err(component->dev, "Can't find kcontrol %s\n", ctl_name); return -EINVAL; diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index bf9a4e90978e..c98e3db74fe7 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -186,7 +186,7 @@ static int fsl_xcvr_activate_ctl(struct snd_soc_dai *dai, const char *name, lockdep_assert_held(&card->snd_card->controls_rwsem); - kctl = snd_soc_card_get_kcontrol_locked(card, name); + kctl = snd_soc_card_get_kcontrol(card, name); if (kctl == NULL) return -ENOENT; diff --git a/sound/soc/soc-card-test.c b/sound/soc/soc-card-test.c index e4a4b101d743..47dfd8b1babc 100644 --- a/sound/soc/soc-card-test.c +++ b/sound/soc/soc-card-test.c @@ -67,62 +67,6 @@ static void test_snd_soc_card_get_kcontrol(struct kunit *test) KUNIT_EXPECT_NULL(test, kc); } -static void test_snd_soc_card_get_kcontrol_locked(struct kunit *test) -{ - struct soc_card_test_priv *priv = test->priv; - struct snd_soc_card *card = priv->card; - struct snd_kcontrol *kc, *kcw; - struct soc_mixer_control *mc; - int i, ret; - - ret = snd_soc_add_card_controls(card, test_card_controls, ARRAY_SIZE(test_card_controls)); - KUNIT_ASSERT_EQ(test, ret, 0); - - /* Look up every control */ - for (i = 0; i < ARRAY_SIZE(test_card_controls); ++i) { - down_read(&card->snd_card->controls_rwsem); - kc = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name); - up_read(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kc, "Failed to find '%s'\n", - test_card_controls[i].name); - if (!kc) - continue; - - /* Test that it is the correct control */ - mc = (struct soc_mixer_control *)kc->private_value; - KUNIT_EXPECT_EQ_MSG(test, mc->shift, i, "For '%s'\n", test_card_controls[i].name); - - down_write(&card->snd_card->controls_rwsem); - kcw = snd_soc_card_get_kcontrol_locked(card, test_card_controls[i].name); - up_write(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NOT_ERR_OR_NULL_MSG(test, kcw, "Failed to find '%s'\n", - test_card_controls[i].name); - - KUNIT_EXPECT_PTR_EQ(test, kc, kcw); - } - - /* Test some names that should not be found */ - down_read(&card->snd_card->controls_rwsem); - kc = snd_soc_card_get_kcontrol_locked(card, "None"); - up_read(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NULL(test, kc); - - down_read(&card->snd_card->controls_rwsem); - kc = snd_soc_card_get_kcontrol_locked(card, "Left None"); - up_read(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NULL(test, kc); - - down_read(&card->snd_card->controls_rwsem); - kc = snd_soc_card_get_kcontrol_locked(card, "Left"); - up_read(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NULL(test, kc); - - down_read(&card->snd_card->controls_rwsem); - kc = snd_soc_card_get_kcontrol_locked(card, NULL); - up_read(&card->snd_card->controls_rwsem); - KUNIT_EXPECT_NULL(test, kc); -} - static int soc_card_test_case_init(struct kunit *test) { struct soc_card_test_priv *priv; @@ -169,7 +113,6 @@ static void soc_card_test_case_exit(struct kunit *test) static struct kunit_case soc_card_test_cases[] = { KUNIT_CASE(test_snd_soc_card_get_kcontrol), - KUNIT_CASE(test_snd_soc_card_get_kcontrol_locked), {} }; diff --git a/sound/soc/soc-card.c b/sound/soc/soc-card.c index 0a3104d4ad23..8e9546fe7428 100644 --- a/sound/soc/soc-card.c +++ b/sound/soc/soc-card.c @@ -29,16 +29,6 @@ static inline int _soc_card_ret(struct snd_soc_card *card, return ret; } -struct snd_kcontrol *snd_soc_card_get_kcontrol_locked(struct snd_soc_card *soc_card, - const char *name) -{ - if (unlikely(!name)) - return NULL; - - return snd_ctl_find_id_mixer_locked(soc_card->snd_card, name); -} -EXPORT_SYMBOL_GPL(snd_soc_card_get_kcontrol_locked); - struct snd_kcontrol *snd_soc_card_get_kcontrol(struct snd_soc_card *soc_card, const char *name) { diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 42f481321919..b3d7bb91e294 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -257,18 +257,6 @@ struct snd_kcontrol *snd_soc_component_get_kcontrol(struct snd_soc_component *co } EXPORT_SYMBOL_GPL(snd_soc_component_get_kcontrol); -struct snd_kcontrol * -snd_soc_component_get_kcontrol_locked(struct snd_soc_component *component, - const char * const ctl) -{ - char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; - - soc_get_kcontrol_name(component, name, ARRAY_SIZE(name), ctl); - - return snd_soc_card_get_kcontrol_locked(component->card, name); -} -EXPORT_SYMBOL_GPL(snd_soc_component_get_kcontrol_locked); - int snd_soc_component_notify_control(struct snd_soc_component *component, const char * const ctl) { From e9606148a6712f7a73dc81d69e19325b61bd4d09 Mon Sep 17 00:00:00 2001 From: Stefan Stistrup Date: Fri, 9 Aug 2024 22:49:22 +0200 Subject: [PATCH 179/410] ALSA: usb-audio: Add input gain and master output mixer elements for RME Babyface Pro Add missing input gain and master output mixer controls for RME Babyface Pro. This patch implements: 1. Input gain controls for 2 mic and 2 line inputs 2. Master output volume controls for all 12 output channels These additions allow for more complete control of the Babyface Pro under Linux. Signed-off-by: Stefan Stistrup Link: https://patch.msgid.link/20240809204922.20112-1-sstistrup@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 163 ++++++++++++++++++++++++++++++++++++++- 1 file changed, 162 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2bc344cf54a8..d56e76fd5cbe 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2541,14 +2541,23 @@ enum { #define SND_BBFPRO_CTL_REG2_PAD_AN1 4 #define SND_BBFPRO_CTL_REG2_PAD_AN2 5 -#define SND_BBFPRO_MIXER_IDX_MASK 0x1ff +#define SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET 992 +#define SND_BBFPRO_MIXER_IDX_MASK 0x3ff #define SND_BBFPRO_MIXER_VAL_MASK 0x3ffff #define SND_BBFPRO_MIXER_VAL_SHIFT 9 #define SND_BBFPRO_MIXER_VAL_MIN 0 // -inf #define SND_BBFPRO_MIXER_VAL_MAX 65536 // +6dB +#define SND_BBFPRO_GAIN_CHANNEL_MASK 0x03 +#define SND_BBFPRO_GAIN_CHANNEL_SHIFT 7 +#define SND_BBFPRO_GAIN_VAL_MASK 0x7f +#define SND_BBFPRO_GAIN_VAL_MIN 0 +#define SND_BBFPRO_GAIN_VAL_MIC_MAX 65 +#define SND_BBFPRO_GAIN_VAL_LINE_MAX 18 // 9db in 0.5db incraments + #define SND_BBFPRO_USBREQ_CTL_REG1 0x10 #define SND_BBFPRO_USBREQ_CTL_REG2 0x17 +#define SND_BBFPRO_USBREQ_GAIN 0x1a #define SND_BBFPRO_USBREQ_MIXER 0x12 static int snd_bbfpro_ctl_update(struct usb_mixer_interface *mixer, u8 reg, @@ -2695,6 +2704,114 @@ static int snd_bbfpro_ctl_resume(struct usb_mixer_elem_list *list) return snd_bbfpro_ctl_update(list->mixer, reg, idx, value); } +static int snd_bbfpro_gain_update(struct usb_mixer_interface *mixer, + u8 channel, u8 gain) +{ + int err; + struct snd_usb_audio *chip = mixer->chip; + + if (channel < 2) { + // XLR preamp: 3-bit fine, 5-bit coarse; special case >60 + if (gain < 60) + gain = ((gain % 3) << 5) | (gain / 3); + else + gain = ((gain % 6) << 5) | (60 / 3); + } + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), + SND_BBFPRO_USBREQ_GAIN, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + gain, channel, NULL, 0); + + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_bbfpro_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = kcontrol->private_value & SND_BBFPRO_GAIN_VAL_MASK; + + ucontrol->value.integer.value[0] = value; + return 0; +} + +static int snd_bbfpro_gain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int pv, channel; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = SND_BBFPRO_GAIN_VAL_MIN; + + if (channel < 2) + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_MIC_MAX; + else + uinfo->value.integer.max = SND_BBFPRO_GAIN_VAL_LINE_MAX; + + return 0; +} + +static int snd_bbfpro_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int pv, channel, old_value, value, err; + + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + pv = kcontrol->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + old_value = pv & SND_BBFPRO_GAIN_VAL_MASK; + value = ucontrol->value.integer.value[0]; + + if (value < SND_BBFPRO_GAIN_VAL_MIN) + return -EINVAL; + + if (channel < 2) { + if (value > SND_BBFPRO_GAIN_VAL_MIC_MAX) + return -EINVAL; + } else { + if (value > SND_BBFPRO_GAIN_VAL_LINE_MAX) + return -EINVAL; + } + + if (value == old_value) + return 0; + + err = snd_bbfpro_gain_update(mixer, channel, value); + if (err < 0) + return err; + + kcontrol->private_value = + (channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT) | value; + return 1; +} + +static int snd_bbfpro_gain_resume(struct usb_mixer_elem_list *list) +{ + int pv, channel, value; + struct snd_kcontrol *kctl = list->kctl; + + pv = kctl->private_value; + channel = (pv >> SND_BBFPRO_GAIN_CHANNEL_SHIFT) & + SND_BBFPRO_GAIN_CHANNEL_MASK; + value = pv & SND_BBFPRO_GAIN_VAL_MASK; + + return snd_bbfpro_gain_update(list->mixer, channel, value); +} + static int snd_bbfpro_vol_update(struct usb_mixer_interface *mixer, u16 index, u32 value) { @@ -2790,6 +2907,15 @@ static const struct snd_kcontrol_new snd_bbfpro_ctl_control = { .put = snd_bbfpro_ctl_put }; +static const struct snd_kcontrol_new snd_bbfpro_gain_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_bbfpro_gain_info, + .get = snd_bbfpro_gain_get, + .put = snd_bbfpro_gain_put +}; + static const struct snd_kcontrol_new snd_bbfpro_vol_control = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -2813,6 +2939,18 @@ static int snd_bbfpro_ctl_add(struct usb_mixer_interface *mixer, u8 reg, &knew, NULL); } +static int snd_bbfpro_gain_add(struct usb_mixer_interface *mixer, u8 channel, + char *name) +{ + struct snd_kcontrol_new knew = snd_bbfpro_gain_control; + + knew.name = name; + knew.private_value = channel << SND_BBFPRO_GAIN_CHANNEL_SHIFT; + + return add_single_ctl_with_resume(mixer, 0, snd_bbfpro_gain_resume, + &knew, NULL); +} + static int snd_bbfpro_vol_add(struct usb_mixer_interface *mixer, u16 index, char *name) { @@ -2860,6 +2998,29 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) } } + // Main out volume + for (i = 0 ; i < 12 ; ++i) { + snprintf(name, sizeof(name), "Main-Out %s", output[i]); + // Main outs are offset to 992 + err = snd_bbfpro_vol_add(mixer, + i + SND_BBFPRO_MIXER_MAIN_OUT_CH_OFFSET, + name); + if (err < 0) + return err; + } + + // Input gain + for (i = 0 ; i < 4 ; ++i) { + if (i < 2) + snprintf(name, sizeof(name), "Mic-%s Gain", input[i]); + else + snprintf(name, sizeof(name), "Line-%s Gain", input[i]); + + err = snd_bbfpro_gain_add(mixer, i, name); + if (err < 0) + return err; + } + // Control Reg 1 err = snd_bbfpro_ctl_add(mixer, SND_BBFPRO_CTL_REG1, SND_BBFPRO_CTL_REG1_CLK_OPTICAL, From 72c0f57dbe8bf625108dc67e34f3472f28501776 Mon Sep 17 00:00:00 2001 From: Norman Bintang Date: Fri, 9 Aug 2024 22:06:45 +0800 Subject: [PATCH 180/410] ALSA: pcm: Add xrun counter for snd_pcm_substream This patch adds an xrun counter to snd_pcm_substream as an alternative to using logs from XRUN_DEBUG_BASIC. The counter provides a way to track the number of xrun occurences, accessible through the /proc interface. The counter is enabled when CONFIG_SND_PCM_XRUN_DEBUG is set. Example output: $ cat /proc/asound/card0/pcm9p/sub0/status owner_pid : 1425 trigger_time: 235.248957291 tstamp : 0.000000000 delay : 1912 avail : 480 avail_max : 1920 ----- hw_ptr : 672000 appl_ptr : 673440 xrun_counter: 3 # (new row) Signed-off-by: Norman Bintang Reviewed-by: Chih-Yang Hsia Tested-by: Chih-Yang Hsia Reviewed-by: David Riley Link: https://patch.msgid.link/20240809140648.3414349-1-normanbt@chromium.org Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 3 +++ sound/core/pcm.c | 6 ++++++ sound/core/pcm_lib.c | 3 +++ 3 files changed, 12 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index ac8f3aef9205..384032b6c59c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -498,6 +498,9 @@ struct snd_pcm_substream { /* misc flags */ unsigned int hw_opened: 1; unsigned int managed_buffer_alloc:1; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + unsigned int xrun_counter; /* number of times xrun happens */ +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ }; #define SUBSTREAM_BUSY(substream) ((substream)->ref_count > 0) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index dc37f3508dc7..290690fc2abc 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -462,6 +462,9 @@ static void snd_pcm_substream_proc_status_read(struct snd_info_entry *entry, snd_iprintf(buffer, "-----\n"); snd_iprintf(buffer, "hw_ptr : %ld\n", runtime->status->hw_ptr); snd_iprintf(buffer, "appl_ptr : %ld\n", runtime->control->appl_ptr); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + snd_iprintf(buffer, "xrun_counter: %d\n", substream->xrun_counter); +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG @@ -970,6 +973,9 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, substream->pid = get_pid(task_pid(current)); pstr->substream_opened++; *rsubstream = substream; +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter = 0; +#endif /* CONFIG_SND_PCM_XRUN_DEBUG */ return 0; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e7905749c4a..6eaa950504cf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,6 +184,9 @@ void __snd_pcm_xrun(struct snd_pcm_substream *substream) pcm_warn(substream->pcm, "XRUN: %s\n", name); dump_stack_on_xrun(substream); } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + substream->xrun_counter++; +#endif } #ifdef CONFIG_SND_PCM_XRUN_DEBUG From ebfb5a57caa4e2e2203883ac6669bf8a294e7c89 Mon Sep 17 00:00:00 2001 From: Jonathan LoBue Date: Sun, 11 Aug 2024 21:53:25 -0700 Subject: [PATCH 181/410] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich Signed-off-by: Kyle Gospodnetich Co-developed by: Jan Drogehoff Signed-off-by: Jan Drogehoff Signed-off-by: Antheas Kapenekakis Tested-by: Richard Alvarez Tested-by: Miles Montierth Signed-off-by: Jonathan LoBue Link: https://patch.msgid.link/20240812045325.47736-1-jlobue10@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d022a25635f9..2f080e02f1db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7492,6 +7492,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8960,6 +8961,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10411,6 +10418,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), From 6aa8700150f7dc62f60b4cf5b1624e2e3d9ed78e Mon Sep 17 00:00:00 2001 From: Karol Kosik Date: Sun, 11 Aug 2024 17:29:56 -0700 Subject: [PATCH 182/410] ALSA: usb-audio: Support multiple control interfaces Registering Numark Party Mix II fails with error 'bogus bTerminalLink 1'. The problem stems from the driver not being able to find input/output terminals required to configure audio streaming. The information about those terminals is stored in AudioControl Interface. Numark device contains 2 AudioControl Interfaces and the driver checks only one of them. According to the USB standard, a device can have multiple audio functions, each represented by Audio Interface Collection. Every audio function is considered to be closed box and will contain unique AudioControl Interface and zero or more AudioStreaming and MIDIStreaming Interfaces. The Numark device adheres to the standard and defines two audio functions: - MIDIStreaming function - AudioStreaming function It starts with MIDI function, followed by the audio function. The driver saves the first AudioControl Interface in `snd_usb_audio` structure associated with the entire device. It then attempts to use this interface to query for terminals and clocks. However, this fails because the correct information is stored in the second AudioControl Interface, defined in the second Audio Interface Collection. This patch introduces a structure holding association between each MIDI/Audio Interface and its corresponding AudioControl Interface, instead of relying on AudioControl Interface defined for the entire device. This structure is populated during usb probing phase and leveraged later when querying for terminals and when sending USB requests. Alternative solutions considered include: - defining a quirk for Numark where the order of interface is manually changed, or terminals are hardcoded in the driver. This solution would have fixed only this model, though it seems that device is USB compliant, and it also seems that other devices from this company may be affected. What's more, it looks like products from other manufacturers have similar problems, i.e. Rane One DJ console - keeping a list of all AudioControl Interfaces and querying all of them to find required information. That would have solved my problem and have low probability of breaking other devices, as we would always start with the same logic of querying first AudioControl Interface. This solution would not have followed the standard though. This patch preserves the `snd_usb_audio.ctrl_intf` variable, which holds the first AudioControl Interface, and uses it as a fallback when some interfaces are not parsed correctly and lack an associated AudioControl Interface, i.e., when configured via quirks. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217865 Signed-off-by: Karol Kosik Link: https://patch.msgid.link/AS8P190MB1285893F4735C8B32AD3886BEC852@AS8P190MB1285.EURP190.PROD.OUTLOOK.COM Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 ++ sound/usb/clock.c | 62 ++++++++++++++++++++++++-------------- sound/usb/format.c | 6 ++-- sound/usb/helper.c | 34 +++++++++++++++++++++ sound/usb/helper.h | 10 ++++-- sound/usb/mixer.c | 2 +- sound/usb/mixer_quirks.c | 17 ++++++----- sound/usb/mixer_scarlett.c | 4 +-- sound/usb/power.c | 3 +- sound/usb/power.h | 1 + sound/usb/stream.c | 21 ++++++++----- sound/usb/usbaudio.h | 12 ++++++++ 12 files changed, 127 insertions(+), 47 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index bdb04fa37a71..778de9244f1e 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -206,6 +206,8 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } + snd_usb_add_ctrl_interface_link(chip, interface, ctrlif); + if (! snd_usb_parse_audio_interface(chip, interface)) { usb_set_interface(dev, interface, 0); /* reset the current interface */ return usb_driver_claim_interface(&usb_audio_driver, iface, diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 60fcb872a80b..8f85200292f3 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -76,11 +76,14 @@ static bool validate_clock_multiplier(void *p, int id, int proto) } #define DEFINE_FIND_HELPER(name, obj, validator, type2, type3) \ -static obj *name(struct snd_usb_audio *chip, int id, int proto) \ +static obj *name(struct snd_usb_audio *chip, int id, \ + const struct audioformat *fmt) \ { \ - return find_uac_clock_desc(chip->ctrl_intf, id, validator, \ - proto == UAC_VERSION_3 ? (type3) : (type2), \ - proto); \ + struct usb_host_interface *ctrl_intf = \ + snd_usb_find_ctrl_interface(chip, fmt->iface); \ + return find_uac_clock_desc(ctrl_intf, id, validator, \ + fmt->protocol == UAC_VERSION_3 ? (type3) : (type2), \ + fmt->protocol); \ } DEFINE_FIND_HELPER(snd_usb_find_clock_source, @@ -93,16 +96,19 @@ DEFINE_FIND_HELPER(snd_usb_find_clock_multiplier, union uac23_clock_multiplier_desc, validate_clock_multiplier, UAC2_CLOCK_MULTIPLIER, UAC3_CLOCK_MULTIPLIER); -static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id) +static int uac_clock_selector_get_val(struct snd_usb_audio *chip, + int selector_id, int iface_no) { + struct usb_host_interface *ctrl_intf; unsigned char buf; int ret; + ctrl_intf = snd_usb_find_ctrl_interface(chip, iface_no); ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, UAC2_CX_CLOCK_SELECTOR << 8, - snd_usb_ctrl_intf(chip) | (selector_id << 8), + snd_usb_ctrl_intf(ctrl_intf) | (selector_id << 8), &buf, sizeof(buf)); if (ret < 0) @@ -111,16 +117,18 @@ static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_i return buf; } -static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_id, - unsigned char pin) +static int uac_clock_selector_set_val(struct snd_usb_audio *chip, + int selector_id, unsigned char pin, int iface_no) { + struct usb_host_interface *ctrl_intf; int ret; + ctrl_intf = snd_usb_find_ctrl_interface(chip, iface_no); ret = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, UAC2_CX_CLOCK_SELECTOR << 8, - snd_usb_ctrl_intf(chip) | (selector_id << 8), + snd_usb_ctrl_intf(ctrl_intf) | (selector_id << 8), &pin, sizeof(pin)); if (ret < 0) return ret; @@ -132,7 +140,7 @@ static int uac_clock_selector_set_val(struct snd_usb_audio *chip, int selector_i return -EINVAL; } - ret = uac_clock_selector_get_val(chip, selector_id); + ret = uac_clock_selector_get_val(chip, selector_id, iface_no); if (ret < 0) return ret; @@ -155,8 +163,10 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, unsigned char data; struct usb_device *dev = chip->dev; union uac23_clock_source_desc *cs_desc; + struct usb_host_interface *ctrl_intf; - cs_desc = snd_usb_find_clock_source(chip, source_id, fmt->protocol); + ctrl_intf = snd_usb_find_ctrl_interface(chip, fmt->iface); + cs_desc = snd_usb_find_clock_source(chip, source_id, fmt); if (!cs_desc) return false; @@ -191,7 +201,7 @@ static bool uac_clock_source_is_valid_quirk(struct snd_usb_audio *chip, err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_CLOCK_VALID << 8, - snd_usb_ctrl_intf(chip) | (source_id << 8), + snd_usb_ctrl_intf(ctrl_intf) | (source_id << 8), &data, sizeof(data)); if (err < 0) { dev_warn(&dev->dev, @@ -217,8 +227,10 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; u32 bmControls; union uac23_clock_source_desc *cs_desc; + struct usb_host_interface *ctrl_intf; - cs_desc = snd_usb_find_clock_source(chip, source_id, fmt->protocol); + ctrl_intf = snd_usb_find_ctrl_interface(chip, fmt->iface); + cs_desc = snd_usb_find_clock_source(chip, source_id, fmt); if (!cs_desc) return false; @@ -235,7 +247,7 @@ static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_CLOCK_VALID << 8, - snd_usb_ctrl_intf(chip) | (source_id << 8), + snd_usb_ctrl_intf(ctrl_intf) | (source_id << 8), &data, sizeof(data)); if (err < 0) { @@ -274,7 +286,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, } /* first, see if the ID we're looking at is a clock source already */ - source = snd_usb_find_clock_source(chip, entity_id, proto); + source = snd_usb_find_clock_source(chip, entity_id, fmt); if (source) { entity_id = GET_VAL(source, proto, bClockID); if (validate && !uac_clock_source_is_valid(chip, fmt, @@ -287,7 +299,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return entity_id; } - selector = snd_usb_find_clock_selector(chip, entity_id, proto); + selector = snd_usb_find_clock_selector(chip, entity_id, fmt); if (selector) { pins = GET_VAL(selector, proto, bNrInPins); clock_id = GET_VAL(selector, proto, bClockID); @@ -317,7 +329,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, /* the entity ID we are looking at is a selector. * find out what it currently selects */ - ret = uac_clock_selector_get_val(chip, clock_id); + ret = uac_clock_selector_get_val(chip, clock_id, fmt->iface); if (ret < 0) { if (!chip->autoclock) return ret; @@ -346,7 +358,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (chip->quirk_flags & QUIRK_FLAG_SKIP_CLOCK_SELECTOR || !writeable) return ret; - err = uac_clock_selector_set_val(chip, entity_id, cur); + err = uac_clock_selector_set_val(chip, entity_id, cur, fmt->iface); if (err < 0) { if (pins == 1) { usb_audio_dbg(chip, @@ -377,7 +389,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, if (ret < 0) continue; - err = uac_clock_selector_set_val(chip, entity_id, i); + err = uac_clock_selector_set_val(chip, entity_id, i, fmt->iface); if (err < 0) continue; @@ -391,7 +403,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, } /* FIXME: multipliers only act as pass-thru element for now */ - multiplier = snd_usb_find_clock_multiplier(chip, entity_id, proto); + multiplier = snd_usb_find_clock_multiplier(chip, entity_id, fmt); if (multiplier) return __uac_clock_find_source(chip, fmt, GET_VAL(multiplier, proto, bCSourceID), @@ -491,11 +503,13 @@ static int get_sample_rate_v2v3(struct snd_usb_audio *chip, int iface, struct usb_device *dev = chip->dev; __le32 data; int err; + struct usb_host_interface *ctrl_intf; + ctrl_intf = snd_usb_find_ctrl_interface(chip, iface); err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), + snd_usb_ctrl_intf(ctrl_intf) | (clock << 8), &data, sizeof(data)); if (err < 0) { dev_warn(&dev->dev, "%d:%d: cannot get freq (v2/v3): err %d\n", @@ -524,8 +538,10 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, __le32 data; int err; union uac23_clock_source_desc *cs_desc; + struct usb_host_interface *ctrl_intf; - cs_desc = snd_usb_find_clock_source(chip, clock, fmt->protocol); + ctrl_intf = snd_usb_find_ctrl_interface(chip, fmt->iface); + cs_desc = snd_usb_find_clock_source(chip, clock, fmt); if (!cs_desc) return 0; @@ -544,7 +560,7 @@ int snd_usb_set_sample_rate_v2v3(struct snd_usb_audio *chip, err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT, UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), + snd_usb_ctrl_intf(ctrl_intf) | (clock << 8), &data, sizeof(data)); if (err < 0) return err; diff --git a/sound/usb/format.c b/sound/usb/format.c index 1bb6a455a1b4..0cbf1d4fbe6e 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -545,7 +545,9 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, unsigned char tmp[2], *data; int nr_triplets, data_size, ret = 0, ret_l6; int clock = snd_usb_clock_find_source(chip, fp, false); + struct usb_host_interface *ctrl_intf; + ctrl_intf = snd_usb_find_ctrl_interface(chip, fp->iface); if (clock < 0) { dev_err(&dev->dev, "%s(): unable to find clock source (clock %d)\n", @@ -557,7 +559,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), + snd_usb_ctrl_intf(ctrl_intf) | (clock << 8), tmp, sizeof(tmp)); if (ret < 0) { @@ -592,7 +594,7 @@ static int parse_audio_format_rates_v2v3(struct snd_usb_audio *chip, ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_CS_CONTROL_SAM_FREQ << 8, - snd_usb_ctrl_intf(chip) | (clock << 8), + snd_usb_ctrl_intf(ctrl_intf) | (clock << 8), data, data_size); if (ret < 0) { diff --git a/sound/usb/helper.c b/sound/usb/helper.c index bf80e55d013a..72b671fb2c84 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -130,3 +130,37 @@ snd_usb_get_host_interface(struct snd_usb_audio *chip, int ifnum, int altsetting return NULL; return usb_altnum_to_altsetting(iface, altsetting); } + +int snd_usb_add_ctrl_interface_link(struct snd_usb_audio *chip, int ifnum, + int ctrlif) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *host_iface; + + if (chip->num_intf_to_ctrl >= MAX_CARD_INTERFACES) { + dev_info(&dev->dev, "Too many interfaces assigned to the single USB-audio card\n"); + return -EINVAL; + } + + /* find audiocontrol interface */ + host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; + + chip->intf_to_ctrl[chip->num_intf_to_ctrl].interface = ifnum; + chip->intf_to_ctrl[chip->num_intf_to_ctrl].ctrl_intf = host_iface; + chip->num_intf_to_ctrl++; + + return 0; +} + +struct usb_host_interface *snd_usb_find_ctrl_interface(struct snd_usb_audio *chip, + int ifnum) +{ + int i; + + for (i = 0; i < chip->num_intf_to_ctrl; ++i) + if (chip->intf_to_ctrl[i].interface == ifnum) + return chip->intf_to_ctrl[i].ctrl_intf; + + /* Fallback to first audiocontrol interface */ + return chip->ctrl_intf; +} diff --git a/sound/usb/helper.h b/sound/usb/helper.h index e2b51ec96ec6..0372e050b3dc 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -17,6 +17,12 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip, struct usb_host_interface * snd_usb_get_host_interface(struct snd_usb_audio *chip, int ifnum, int altsetting); +int snd_usb_add_ctrl_interface_link(struct snd_usb_audio *chip, int ifnum, + int ctrlif); + +struct usb_host_interface *snd_usb_find_ctrl_interface(struct snd_usb_audio *chip, + int ifnum); + /* * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) @@ -28,9 +34,9 @@ snd_usb_get_host_interface(struct snd_usb_audio *chip, int ifnum, int altsetting #define snd_usb_get_speed(dev) ((dev)->speed) -static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip) +static inline int snd_usb_ctrl_intf(struct usb_host_interface *ctrl_intf) { - return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber; + return get_iface_desc(ctrl_intf)->bInterfaceNumber; } /* in validate.c */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 2d27d729c3be..9945ae55b0d0 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -728,7 +728,7 @@ static int get_cluster_channels_v3(struct mixer_build *state, unsigned int clust UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, cluster_id, - snd_usb_ctrl_intf(state->chip), + snd_usb_ctrl_intf(state->mixer->hostif), &c_header, sizeof(c_header)); if (err < 0) goto error; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d56e76fd5cbe..5d6792f51f4c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1043,7 +1043,7 @@ static int snd_ftu_eff_switch_init(struct usb_mixer_interface *mixer, err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, pval & 0xff00, - snd_usb_ctrl_intf(mixer->chip) | ((pval & 0xff) << 8), + snd_usb_ctrl_intf(mixer->hostif) | ((pval & 0xff) << 8), value, 2); if (err < 0) return err; @@ -1077,7 +1077,7 @@ static int snd_ftu_eff_switch_update(struct usb_mixer_elem_list *list) UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, pval & 0xff00, - snd_usb_ctrl_intf(chip) | ((pval & 0xff) << 8), + snd_usb_ctrl_intf(list->mixer->hostif) | ((pval & 0xff) << 8), value, 2); snd_usb_unlock_shutdown(chip); return err; @@ -2115,24 +2115,25 @@ static int dell_dock_mixer_create(struct usb_mixer_interface *mixer) return 0; } -static void dell_dock_init_vol(struct snd_usb_audio *chip, int ch, int id) +static void dell_dock_init_vol(struct usb_mixer_interface *mixer, int ch, int id) { + struct snd_usb_audio *chip = mixer->chip; u16 buf = 0; snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), UAC_SET_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, (UAC_FU_VOLUME << 8) | ch, - snd_usb_ctrl_intf(chip) | (id << 8), + snd_usb_ctrl_intf(mixer->hostif) | (id << 8), &buf, 2); } static int dell_dock_mixer_init(struct usb_mixer_interface *mixer) { /* fix to 0dB playback volumes */ - dell_dock_init_vol(mixer->chip, 1, 16); - dell_dock_init_vol(mixer->chip, 2, 16); - dell_dock_init_vol(mixer->chip, 1, 19); - dell_dock_init_vol(mixer->chip, 2, 19); + dell_dock_init_vol(mixer, 1, 16); + dell_dock_init_vol(mixer, 2, 16); + dell_dock_init_vol(mixer, 1, 19); + dell_dock_init_vol(mixer, 2, 19); return 0; } diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 0d6e4f15bf77..ff548041679b 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -460,7 +460,7 @@ static int scarlett_ctl_meter_get(struct snd_kcontrol *kctl, struct snd_usb_audio *chip = elem->head.mixer->chip; unsigned char buf[2 * MAX_CHANNELS] = {0, }; int wValue = (elem->control << 8) | elem->idx_off; - int idx = snd_usb_ctrl_intf(chip) | (elem->head.id << 8); + int idx = snd_usb_ctrl_intf(elem->head.mixer->hostif) | (elem->head.id << 8); int err; err = snd_usb_ctl_msg(chip->dev, @@ -1002,7 +1002,7 @@ int snd_scarlett_controls_create(struct usb_mixer_interface *mixer) err = snd_usb_ctl_msg(mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), UAC2_CS_CUR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | - USB_DIR_OUT, 0x0100, snd_usb_ctrl_intf(mixer->chip) | + USB_DIR_OUT, 0x0100, snd_usb_ctrl_intf(mixer->hostif) | (0x29 << 8), sample_rate_buffer, 4); if (err < 0) return err; diff --git a/sound/usb/power.c b/sound/usb/power.c index 606a2cb23eab..66bd4daa68fd 100644 --- a/sound/usb/power.c +++ b/sound/usb/power.c @@ -40,6 +40,7 @@ snd_usb_find_power_domain(struct usb_host_interface *ctrl_iface, le16_to_cpu(pd_desc->waRecoveryTime1); pd->pd_d2d0_rec = le16_to_cpu(pd_desc->waRecoveryTime2); + pd->ctrl_iface = ctrl_iface; return pd; } } @@ -57,7 +58,7 @@ int snd_usb_power_domain_set(struct snd_usb_audio *chip, unsigned char current_state; int err, idx; - idx = snd_usb_ctrl_intf(chip) | (pd->pd_id << 8); + idx = snd_usb_ctrl_intf(pd->ctrl_iface) | (pd->pd_id << 8); err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), UAC2_CS_CUR, diff --git a/sound/usb/power.h b/sound/usb/power.h index 396e3e51440a..1fa92ad0ca92 100644 --- a/sound/usb/power.h +++ b/sound/usb/power.h @@ -6,6 +6,7 @@ struct snd_usb_power_domain { int pd_id; /* UAC3 Power Domain ID */ int pd_d1d0_rec; /* D1 to D0 recovery time */ int pd_d2d0_rec; /* D2 to D0 recovery time */ + struct usb_host_interface *ctrl_iface; /* Control interface */ }; enum { diff --git a/sound/usb/stream.c b/sound/usb/stream.c index e14c725acebf..d70c140813d6 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -713,10 +713,13 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, struct usb_device *dev = chip->dev; struct uac_format_type_i_continuous_descriptor *fmt; unsigned int num_channels = 0, chconfig = 0; + struct usb_host_interface *ctrl_intf; struct audioformat *fp; int clock = 0; u64 format; + ctrl_intf = snd_usb_find_ctrl_interface(chip, iface_no); + /* get audio formats */ if (protocol == UAC_VERSION_1) { struct uac1_as_header_descriptor *as = @@ -740,7 +743,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, format = le16_to_cpu(as->wFormatTag); /* remember the format value */ - iterm = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + iterm = snd_usb_find_input_terminal_descriptor(ctrl_intf, as->bTerminalLink, protocol); if (iterm) { @@ -776,7 +779,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, * lookup the terminal associated to this interface * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + input_term = snd_usb_find_input_terminal_descriptor(ctrl_intf, as->bTerminalLink, protocol); if (input_term) { @@ -786,7 +789,7 @@ snd_usb_get_audioformat_uac12(struct snd_usb_audio *chip, goto found_clock; } - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + output_term = snd_usb_find_output_terminal_descriptor(ctrl_intf, as->bTerminalLink, protocol); if (output_term) { @@ -870,6 +873,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, struct uac3_cluster_header_descriptor *cluster; struct uac3_as_header_descriptor *as = NULL; struct uac3_hc_descriptor_header hc_header; + struct usb_host_interface *ctrl_intf; struct snd_pcm_chmap_elem *chmap; struct snd_usb_power_domain *pd; unsigned char badd_profile; @@ -881,6 +885,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, int err; badd_profile = chip->badd_profile; + ctrl_intf = snd_usb_find_ctrl_interface(chip, iface_no); if (badd_profile >= UAC3_FUNCTION_SUBCLASS_GENERIC_IO) { unsigned int maxpacksize = @@ -966,7 +971,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, cluster_id, - snd_usb_ctrl_intf(chip), + snd_usb_ctrl_intf(ctrl_intf), &hc_header, sizeof(hc_header)); if (err < 0) return ERR_PTR(err); @@ -990,7 +995,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, UAC3_CS_REQ_HIGH_CAPABILITY_DESCRIPTOR, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, cluster_id, - snd_usb_ctrl_intf(chip), + snd_usb_ctrl_intf(ctrl_intf), cluster, wLength); if (err < 0) { kfree(cluster); @@ -1011,7 +1016,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, * lookup the terminal associated to this interface * to extract the clock */ - input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf, + input_term = snd_usb_find_input_terminal_descriptor(ctrl_intf, as->bTerminalLink, UAC_VERSION_3); if (input_term) { @@ -1019,7 +1024,7 @@ snd_usb_get_audioformat_uac3(struct snd_usb_audio *chip, goto found_clock; } - output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf, + output_term = snd_usb_find_output_terminal_descriptor(ctrl_intf, as->bTerminalLink, UAC_VERSION_3); if (output_term) { @@ -1068,7 +1073,7 @@ found_clock: UAC_VERSION_3, iface_no); - pd = snd_usb_find_power_domain(chip->ctrl_intf, + pd = snd_usb_find_power_domain(ctrl_intf, as->bTerminalLink); /* ok, let's parse further... */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 43d4029edab4..b0f042c99608 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,6 +21,15 @@ struct media_intf_devnode; #define MAX_CARD_INTERFACES 16 +/* + * Structure holding assosiation between Audio Control Interface + * and given Streaming or Midi Interface. + */ +struct snd_intf_to_ctrl { + u8 interface; + struct usb_host_interface *ctrl_intf; +}; + struct snd_usb_audio { int index; struct usb_device *dev; @@ -63,6 +72,9 @@ struct snd_usb_audio { struct usb_host_interface *ctrl_intf; /* the audio control interface */ struct media_device *media_dev; struct media_intf_devnode *ctl_intf_media_devnode; + + unsigned int num_intf_to_ctrl; + struct snd_intf_to_ctrl intf_to_ctrl[MAX_CARD_INTERFACES]; }; #define USB_AUDIO_IFACE_UNUSED ((void *)-1L) From 86297bb30ae094e14a3a6c62b870a2f301a180a2 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Tue, 6 Aug 2024 15:43:00 +0200 Subject: [PATCH 183/410] ASoC: cs43130: Constify struct reg_sequence and reg_sequences 'struct reg_sequence' and 'struct reg_sequences' are not modified in this drivers. Constifying these structures moves some data to a read-only section, so increase overall security. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o After: ===== text data bss dec hex filename 55562 6729 64 62355 f393 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET Link: https://patch.msgid.link/5b906a0cc9b7be15d0d6310069f54254a75ea767.1722951770.git.christophe.jaillet@wanadoo.fr Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/cs43130.c | 40 +++++++++++++++++++------------------- 1 file changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index cb4ca80f36d2..f8e2fb69ada2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1805,7 +1805,7 @@ static struct attribute *hpload_attrs[] = { }; ATTRIBUTE_GROUPS(hpload); -static struct reg_sequence hp_en_cal_seq[] = { +static const struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1820,7 +1820,7 @@ static struct reg_sequence hp_en_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_en_cal_seq2[] = { +static const struct reg_sequence hp_en_cal_seq2[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, @@ -1828,7 +1828,7 @@ static struct reg_sequence hp_en_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, }; -static struct reg_sequence hp_dis_cal_seq[] = { +static const struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_DXD1, 0x99}, {CS43130_DXD12, 0}, @@ -1836,12 +1836,12 @@ static struct reg_sequence hp_dis_cal_seq[] = { {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dis_cal_seq2[] = { +static const struct reg_sequence hp_dis_cal_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0}, }; -static struct reg_sequence hp_dc_ch_l_seq[] = { +static const struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1851,12 +1851,12 @@ static struct reg_sequence hp_dc_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_l_seq2[] = { +static const struct reg_sequence hp_dc_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x81}, }; -static struct reg_sequence hp_dc_ch_r_seq[] = { +static const struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1866,12 +1866,12 @@ static struct reg_sequence hp_dc_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_dc_ch_r_seq2[] = { +static const struct reg_sequence hp_dc_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x91}, }; -static struct reg_sequence hp_ac_ch_l_seq[] = { +static const struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x0A}, {CS43130_DXD17, 0x93}, @@ -1881,12 +1881,12 @@ static struct reg_sequence hp_ac_ch_l_seq[] = { {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_l_seq2[] = { +static const struct reg_sequence hp_ac_ch_l_seq2[] = { {CS43130_HP_LOAD_1, 0x80}, {CS43130_HP_LOAD_1, 0x82}, }; -static struct reg_sequence hp_ac_ch_r_seq[] = { +static const struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_DXD1, 0x99}, {CS43130_DXD19, 0x8A}, {CS43130_DXD17, 0x15}, @@ -1896,24 +1896,24 @@ static struct reg_sequence hp_ac_ch_r_seq[] = { {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_ac_ch_r_seq2[] = { +static const struct reg_sequence hp_ac_ch_r_seq2[] = { {CS43130_HP_LOAD_1, 0x90}, {CS43130_HP_LOAD_1, 0x92}, }; -static struct reg_sequence hp_cln_seq[] = { +static const struct reg_sequence hp_cln_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, {CS43130_HP_MEAS_LOAD_2, 0}, }; struct reg_sequences { - struct reg_sequence *seq; - int size; - unsigned int msk; + const struct reg_sequence *seq; + int size; + unsigned int msk; }; -static struct reg_sequences hpload_seq1[] = { +static const struct reg_sequences hpload_seq1[] = { { .seq = hp_en_cal_seq, .size = ARRAY_SIZE(hp_en_cal_seq), @@ -1951,7 +1951,7 @@ static struct reg_sequences hpload_seq1[] = { }, }; -static struct reg_sequences hpload_seq2[] = { +static const struct reg_sequences hpload_seq2[] = { { .seq = hp_en_cal_seq2, .size = ARRAY_SIZE(hp_en_cal_seq2), @@ -2041,7 +2041,7 @@ static int cs43130_update_hpload(unsigned int msk, int ac_idx, } static int cs43130_hpload_proc(struct cs43130_private *cs43130, - struct reg_sequence *seq, int seq_size, + const struct reg_sequence *seq, int seq_size, unsigned int rslt_msk, int ac_idx) { int ret; @@ -2122,7 +2122,7 @@ static void cs43130_imp_meas(struct work_struct *wk) int i, ret, ac_idx; struct cs43130_private *cs43130; struct snd_soc_component *component; - struct reg_sequences *hpload_seq; + const struct reg_sequences *hpload_seq; cs43130 = container_of(wk, struct cs43130_private, work); component = cs43130->component; From 6024b86b4a618b6973cf6fc5ed3fa21280e395b9 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 12 Aug 2024 15:34:22 +0530 Subject: [PATCH 184/410] ASoC: amd: acp: Convert comma to semicolon Replace a comma between expression statements by a semicolon. Signed-off-by: Vijendar Mukunda Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240812100429.2594745-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index f7e4af850309..08f368b3bbc8 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -657,9 +657,9 @@ static int mc_probe(struct platform_device *pdev) ctx->private = amd_ctx; card = &ctx->card; card->dev = &pdev->dev; - card->name = "amd-soundwire", - card->owner = THIS_MODULE, - card->late_probe = asoc_sdw_card_late_probe, + card->name = "amd-soundwire"; + card->owner = THIS_MODULE; + card->late_probe = asoc_sdw_card_late_probe; snd_soc_card_set_drvdata(card, ctx); From ab73c7c0e5800a44690023cfdfeac72d3b6b95e8 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Tue, 6 Aug 2024 15:52:24 +0200 Subject: [PATCH 185/410] ASoC: rt1318: Constify struct reg_sequence 'struct reg_sequence' is not modified in this driver. Constifying this structure moves some data to a read-only section, so increase overall security. While at it, remove rt1318_INIT_REG_LEN which is ununsed. On a x86_64, with allmodconfig: Before: ====== text data bss dec hex filename 22062 4859 32 26953 6949 sound/soc/codecs/rt1318.o After: ===== text data bss dec hex filename 24742 2171 32 26945 6941 sound/soc/codecs/rt1318.o Signed-off-by: Christophe JAILLET Link: https://patch.msgid.link/96561dd2962d4312eb0e68ab850027f44350d070.1722952334.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/rt1318.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1318.c b/sound/soc/codecs/rt1318.c index 83b29b441be9..e12b1e96a53a 100644 --- a/sound/soc/codecs/rt1318.c +++ b/sound/soc/codecs/rt1318.c @@ -30,7 +30,7 @@ #include "rt1318.h" -static struct reg_sequence init_list[] = { +static const struct reg_sequence init_list[] = { { 0x0000C000, 0x01}, { 0x0000F20D, 0x00}, { 0x0000F212, 0x3E}, @@ -254,7 +254,6 @@ static struct reg_sequence init_list[] = { { 0x0000C320, 0x20}, { 0x0000C203, 0x9C}, }; -#define rt1318_INIT_REG_LEN ARRAY_SIZE(init_list) static const struct reg_default rt1318_reg[] = { { 0xc000, 0x00 }, From c6f3abbbdc99930f831d5c76dac32ddbd9b88fa1 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Tue, 13 Aug 2024 13:38:50 +0530 Subject: [PATCH 186/410] ASoC: amd: acp: add legacy driver support for ACP7.1 based platforms Add acp pci driver and machine driver changes for ACP7.1 based platforms for legacy stack. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240813080850.3107409-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-legacy-common.c | 7 ++++--- sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-pci.c | 4 ++++ sound/soc/amd/acp/acp70.c | 12 ++++++++++-- sound/soc/amd/acp/amd.h | 1 + 6 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 04bd605fdce3..3cc083fac837 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -270,6 +270,7 @@ static int acp_power_on(struct acp_chip_info *chip) acp_pgfsm_ctrl_reg = ACP63_PGFSM_CONTROL; break; case ACP70_DEV: + case ACP71_DEV: acp_pgfsm_stat_reg = ACP70_PGFSM_STATUS; acp_pgfsm_ctrl_reg = ACP70_PGFSM_CONTROL; break; @@ -336,10 +337,9 @@ int acp_deinit(struct acp_chip_info *chip) if (ret) return ret; - if (chip->acp_rev != ACP70_DEV) + if (chip->acp_rev < ACP70_DEV) writel(0, chip->base + ACP_CONTROL); - - if (chip->acp_rev >= ACP70_DEV) + else writel(0x01, chip->base + ACP_ZSC_DSP_CTRL); return 0; } @@ -461,6 +461,7 @@ void check_acp_config(struct pci_dev *pci, struct acp_chip_info *chip) check_acp6x_config(chip); break; case ACP70_DEV: + case ACP71_DEV: pdm_addr = ACP70_PDM_ADDR; check_acp70_config(chip); break; diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index a36300a4ed8a..e9ff4815c12c 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1766,7 +1766,7 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) } else if (drv_data->platform == ACP63) { links[i].platforms = platform_acp63_component; links[i].num_platforms = ARRAY_SIZE(platform_acp63_component); - } else if (drv_data->platform == ACP70) { + } else if ((drv_data->platform == ACP70) || (drv_data->platform == ACP71)) { links[i].platforms = platform_acp70_component; links[i].num_platforms = ARRAY_SIZE(platform_acp70_component); } else { diff --git a/sound/soc/amd/acp/acp-mach.h b/sound/soc/amd/acp/acp-mach.h index a48546d8d407..93d9e3886b7e 100644 --- a/sound/soc/amd/acp/acp-mach.h +++ b/sound/soc/amd/acp/acp-mach.h @@ -56,6 +56,7 @@ enum platform_end_point { REMBRANDT, ACP63, ACP70, + ACP71, }; struct acp_mach_ops { diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index b0304b813cad..f7450a5bd103 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -95,6 +95,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id chip->name = "acp_asoc_acp70"; chip->acp_rev = ACP70_DEV; break; + case 0x71: + chip->name = "acp_asoc_acp70"; + chip->acp_rev = ACP71_DEV; + break; default: dev_err(dev, "Unsupported device revision:0x%x\n", pci->revision); ret = -EINVAL; diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index a2cbdcca4313..81636faa22cd 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -147,7 +147,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) return -ENODEV; } - if (chip->acp_rev != ACP70_DEV) { + switch (chip->acp_rev) { + case ACP70_DEV: + case ACP71_DEV: + break; + default: dev_err(&pdev->dev, "Un-supported ACP Revision %d\n", chip->acp_rev); return -ENODEV; } @@ -178,7 +182,11 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) adata->num_dai = ARRAY_SIZE(acp70_dai); adata->rsrc = &rsrc; adata->machines = snd_soc_acpi_amd_acp70_acp_machines; - adata->platform = ACP70; + if (chip->acp_rev == ACP70_DEV) + adata->platform = ACP70; + else + adata->platform = ACP71; + adata->flag = chip->flag; acp_machine_select(adata); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index c095a34a7229..4f0cd7dd04e5 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -22,6 +22,7 @@ #define ACP6X_DEV 6 #define ACP63_DEV 0x63 #define ACP70_DEV 0x70 +#define ACP71_DEV 0x71 #define DMIC_INSTANCE 0x00 #define I2S_SP_INSTANCE 0x01 From 8f712c12f34daaaa2e47ba07cf3b348d3a442986 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sun, 11 Aug 2024 21:51:41 +0800 Subject: [PATCH 187/410] ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781 Rename dai_driver name to unify the name between TAS2563 and TAS2781. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240811135144.178-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 313c17f76ca4..8c9efd9183ff 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -672,7 +672,7 @@ static const struct snd_soc_dai_ops tasdevice_dai_ops = { static struct snd_soc_dai_driver tasdevice_dai_driver[] = { { - .name = "tas2781_codec", + .name = "tasdev_codec", .id = 0, .playback = { .stream_name = "Playback", From 0c3ad39b791c2ecf718afcaca30e5ceafa939d5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Aug 2024 15:48:41 +0200 Subject: [PATCH 188/410] ALSA: usb-audio: Define macros for quirk table entries Many entries in the USB-audio quirk tables have relatively complex expressions. For improving the readability, introduce a few macros. Those are applied in the following patch. Link: https://patch.msgid.link/20240814134844.2726-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 77 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 77 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f13a8d63a019..9443ada51539 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -35,6 +35,83 @@ .bInterfaceClass = USB_CLASS_AUDIO, \ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL +/* Quirk .driver_info, followed by the definition of the quirk entry; + * put like QUIRK_DRIVER_INFO { ... } in each entry of the quirk table + */ +#define QUIRK_DRIVER_INFO \ + .driver_info = (unsigned long)&(const struct snd_usb_audio_quirk) + +/* + * Macros for quirk data entries + */ + +/* Quirk data entry for ignoring the interface */ +#define QUIRK_DATA_IGNORE(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_IGNORE_INTERFACE +/* Quirk data entry for a standard audio interface */ +#define QUIRK_DATA_STANDARD_AUDIO(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_AUDIO_STANDARD_INTERFACE +/* Quirk data entry for a standard MIDI interface */ +#define QUIRK_DATA_STANDARD_MIDI(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_MIDI_STANDARD_INTERFACE +/* Quirk data entry for a standard mixer interface */ +#define QUIRK_DATA_STANDARD_MIXER(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_AUDIO_STANDARD_MIXER + +/* Quirk data entry for Yamaha MIDI */ +#define QUIRK_DATA_MIDI_YAMAHA(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_MIDI_YAMAHA +/* Quirk data entry for Edirol UAxx */ +#define QUIRK_DATA_EDIROL_UAXX(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_AUDIO_EDIROL_UAXX +/* Quirk data entry for raw bytes interface */ +#define QUIRK_DATA_RAW_BYTES(_ifno) \ + .ifnum = (_ifno), .type = QUIRK_MIDI_RAW_BYTES + +/* Quirk composite array terminator */ +#define QUIRK_COMPOSITE_END { .ifnum = -1 } + +/* Quirk data entry for composite quirks; + * followed by the quirk array that is terminated with QUIRK_COMPOSITE_END + * e.g. QUIRK_DATA_COMPOSITE { { quirk1 }, { quirk2 },..., QUIRK_COMPOSITE_END } + */ +#define QUIRK_DATA_COMPOSITE \ + .ifnum = QUIRK_ANY_INTERFACE, \ + .type = QUIRK_COMPOSITE, \ + .data = &(const struct snd_usb_audio_quirk[]) + +/* Quirk data entry for a fixed audio endpoint; + * followed by audioformat definition + * e.g. QUIRK_DATA_AUDIOFORMAT(n) { .formats = xxx, ... } + */ +#define QUIRK_DATA_AUDIOFORMAT(_ifno) \ + .ifnum = (_ifno), \ + .type = QUIRK_AUDIO_FIXED_ENDPOINT, \ + .data = &(const struct audioformat) + +/* Quirk data entry for a fixed MIDI endpoint; + * followed by snd_usb_midi_endpoint_info definition + * e.g. QUIRK_DATA_MIDI_FIXED_ENDPOINT(n) { .out_cables = x, .in_cables = y } + */ +#define QUIRK_DATA_MIDI_FIXED_ENDPOINT(_ifno) \ + .ifnum = (_ifno), \ + .type = QUIRK_MIDI_FIXED_ENDPOINT, \ + .data = &(const struct snd_usb_midi_endpoint_info) +/* Quirk data entry for a MIDIMAN MIDI endpoint */ +#define QUIRK_DATA_MIDI_MIDIMAN(_ifno) \ + .ifnum = (_ifno), \ + .type = QUIRK_MIDI_MIDIMAN, \ + .data = &(const struct snd_usb_midi_endpoint_info) +/* Quirk data entry for a EMAGIC MIDI endpoint */ +#define QUIRK_DATA_MIDI_EMAGIC(_ifno) \ + .ifnum = (_ifno), \ + .type = QUIRK_MIDI_EMAGIC, \ + .data = &(const struct snd_usb_midi_endpoint_info) + +/* + * Here we go... the quirk table definition begins: + */ + /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), From d79e13f8e8abb5cd3a2a0f9fc9bc3fc750c5b06f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 14 Aug 2024 15:48:42 +0200 Subject: [PATCH 189/410] ALSA: usb-audio: Replace complex quirk lines with macros Apply the newly introduced macros for reduce the complex expressions and cast in the quirk table definitions. It results in a significant code reduction, too. There should be no functional changes. Link: https://patch.msgid.link/20240814134844.2726-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 2206 ++++++++++---------------------------- 1 file changed, 591 insertions(+), 1615 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 9443ada51539..cd273397bc59 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -115,7 +115,7 @@ /* FTDI devices */ { USB_DEVICE(0x0403, 0xb8d8), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "STARR LABS", */ /* .product_name = "Starr Labs MIDI USB device", */ .ifnum = 0, @@ -126,10 +126,8 @@ { /* Creative BT-D1 */ USB_DEVICE(0x041e, 0x0005), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, @@ -164,18 +162,11 @@ */ { USB_AUDIO_DEVICE(0x041e, 0x4095), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(2) }, { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .fmt_bits = 16, @@ -191,9 +182,7 @@ .rate_table = (unsigned int[]) { 48000 }, }, }, - { - .ifnum = -1 - }, + QUIRK_COMPOSITE_END }, }, }, @@ -205,31 +194,18 @@ */ { USB_DEVICE(0x0424, 0xb832), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Standard Microsystems Corp.", .product_name = "HP Wireless Audio", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { /* Mixer */ - { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE, - }, + { QUIRK_DATA_IGNORE(0) }, /* Playback */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, + { QUIRK_DATA_IGNORE(1) }, /* Capture */ - { - .ifnum = 2, - .type = QUIRK_IGNORE_INTERFACE, - }, + { QUIRK_DATA_IGNORE(2) }, /* HID Device, .ifnum = 3 */ - { - .ifnum = -1, - } + QUIRK_COMPOSITE_END } } }, @@ -252,20 +228,18 @@ #define YAMAHA_DEVICE(id, name) { \ USB_DEVICE(0x0499, id), \ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + QUIRK_DRIVER_INFO { \ .vendor_name = "Yamaha", \ .product_name = name, \ - .ifnum = QUIRK_ANY_INTERFACE, \ - .type = QUIRK_MIDI_YAMAHA \ + QUIRK_DATA_MIDI_YAMAHA(QUIRK_ANY_INTERFACE) \ } \ } #define YAMAHA_INTERFACE(id, intf, name) { \ USB_DEVICE_VENDOR_SPEC(0x0499, id), \ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \ + QUIRK_DRIVER_INFO { \ .vendor_name = "Yamaha", \ .product_name = name, \ - .ifnum = intf, \ - .type = QUIRK_MIDI_YAMAHA \ + QUIRK_DATA_MIDI_YAMAHA(intf) \ } \ } YAMAHA_DEVICE(0x1000, "UX256"), @@ -352,135 +326,67 @@ YAMAHA_DEVICE(0x105c, NULL), YAMAHA_DEVICE(0x105d, NULL), { USB_DEVICE(0x0499, 0x1503), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Yamaha", */ /* .product_name = "MOX6/MOX8", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_YAMAHA - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0499, 0x1507), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Yamaha", */ /* .product_name = "THR10", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_YAMAHA - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0499, 0x1509), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Yamaha", */ /* .product_name = "Steinberg UR22", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_YAMAHA - }, - { - .ifnum = 4, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + { QUIRK_DATA_IGNORE(4) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0499, 0x150a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Yamaha", */ /* .product_name = "THR5A", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_YAMAHA - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0499, 0x150c), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Yamaha", */ /* .product_name = "THR10C", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_YAMAHA - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_MIDI_YAMAHA(3) }, + QUIRK_COMPOSITE_END } } }, @@ -514,7 +420,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS, .idVendor = 0x0499, .bInterfaceClass = USB_CLASS_VENDOR_SPEC, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUTODETECT } @@ -525,16 +431,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ { USB_DEVICE(0x0582, 0x0000), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "UA-100", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 4, .iface = 0, @@ -549,9 +451,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, @@ -566,106 +466,66 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0007, .in_cables = 0x0007 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0002), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-4", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x000f, .in_cables = 0x000f } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0003), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "SC-8850", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x003f, .in_cables = 0x003f } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0004), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "U-8", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0005, .in_cables = 0x0005 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -673,152 +533,92 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Has ID 0x0099 when not in "Advanced Driver" mode. * The UM-2EX has only one input, but we cannot detect this. */ USB_DEVICE(0x0582, 0x0005), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-2", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0003, .in_cables = 0x0003 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0007), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "SC-8820", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0013, .in_cables = 0x0013 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0008), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "PC-300", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* has ID 0x009d when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0009), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-1", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x000b), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "SK-500", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0013, .in_cables = 0x0013 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -826,31 +626,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* thanks to Emiliano Grilli * for helping researching this data */ USB_DEVICE(0x0582, 0x000c), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "SC-D70", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0007, .in_cables = 0x0007 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -864,35 +652,23 @@ YAMAHA_DEVICE(0x7010, "UB99"), * the 96kHz sample rate. */ USB_DEVICE(0x0582, 0x0010), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-5", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x0013 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0012), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "XV-5050", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -901,12 +677,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0015 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0014), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-880", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x01ff, .in_cables = 0x01ff } @@ -915,74 +689,48 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0017 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0016), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "SD-90", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x000f, .in_cables = 0x000f } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* has ID 0x001c when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x001b), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "MMP-2", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* has ID 0x001e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x001d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "V-SYNTH", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -991,12 +739,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0024 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0023), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-550", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x003f, .in_cables = 0x003f } @@ -1009,20 +755,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), * and no MIDI. */ USB_DEVICE(0x0582, 0x0025), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-20", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, @@ -1037,9 +776,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 2, @@ -1054,28 +791,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(3) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* has ID 0x0028 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0027), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "SD-20", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0003, .in_cables = 0x0007 } @@ -1084,12 +815,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x002a when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0029), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "SD-80", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x000f, .in_cables = 0x000f } @@ -1102,39 +831,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), * but offers only 16-bit PCM and no MIDI. */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x002b), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-700", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 3, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_EDIROL_UAXX(1) }, + { QUIRK_DATA_EDIROL_UAXX(2) }, + { QUIRK_DATA_EDIROL_UAXX(3) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x002e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x002d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "XV-2020", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1143,12 +857,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0030 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x002f), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "VariOS", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0007, .in_cables = 0x0007 } @@ -1157,12 +869,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0034 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0033), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "PCR", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0003, .in_cables = 0x0007 } @@ -1174,12 +884,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * later revisions use IDs 0x0054 and 0x00a2. */ USB_DEVICE(0x0582, 0x0037), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "Digital Piano", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1192,39 +900,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), * and no MIDI. */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x003b), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "BOSS", .product_name = "GS-10", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + { QUIRK_DATA_STANDARD_MIDI(3) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x0041 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0040), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "GI-20", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1233,12 +926,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0043 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0042), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "RS-70", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1247,36 +938,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0049 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0047), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "EDIROL", */ /* .product_name = "UR-80", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { /* in the 96 kHz modes, only interface 1 is there */ - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x004a when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0048), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "EDIROL", */ /* .product_name = "UR-80", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0003, .in_cables = 0x0007 } @@ -1285,35 +964,23 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x004e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x004c), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "PCR-A", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x004f when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x004d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "PCR-A", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0003, .in_cables = 0x0007 } @@ -1325,76 +992,52 @@ YAMAHA_DEVICE(0x7010, "UB99"), * is standard compliant, but has only 16-bit PCM. */ USB_DEVICE(0x0582, 0x0050), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-3FX", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0052), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UM-1SX", - .ifnum = 0, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DATA_STANDARD_MIDI(0) } }, { USB_DEVICE(0x0582, 0x0060), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "EXR Series", - .ifnum = 0, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DATA_STANDARD_MIDI(0) } }, { /* has ID 0x0066 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0064), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "EDIROL", */ /* .product_name = "PCR-1", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x0067 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0065), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "EDIROL", */ /* .product_name = "PCR-1", */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0003 } @@ -1403,12 +1046,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x006e when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x006d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "FANTOM-X", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1421,39 +1062,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), * offers only 16-bit PCM at 44.1 kHz and no MIDI. */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0074), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-25", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_EDIROL_UAXX(0) }, + { QUIRK_DATA_EDIROL_UAXX(1) }, + { QUIRK_DATA_EDIROL_UAXX(2) }, + QUIRK_COMPOSITE_END } } }, { /* has ID 0x0076 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0075), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "BOSS", .product_name = "DR-880", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1462,12 +1088,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x007b when not in "Advanced Driver" mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x007a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", /* "RD" or "RD-700SX"? */ - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0003, .in_cables = 0x0003 } @@ -1476,12 +1100,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x0081 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Roland", .product_name = "G-70", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1490,12 +1112,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* has ID 0x008c when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x008b), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "PC-50", - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1507,56 +1127,31 @@ YAMAHA_DEVICE(0x7010, "UB99"), * is standard compliant, but has only 16-bit PCM and no MIDI. */ USB_DEVICE(0x0582, 0x00a3), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-4FX", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_EDIROL_UAXX(0) }, + { QUIRK_DATA_EDIROL_UAXX(1) }, + { QUIRK_DATA_EDIROL_UAXX(2) }, + QUIRK_COMPOSITE_END } } }, { /* Edirol M-16DX */ USB_DEVICE(0x0582, 0x00c4), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -1566,37 +1161,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), * offers only 16-bit PCM at 44.1 kHz and no MIDI. */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x00e6), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "EDIROL", .product_name = "UA-25EX", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UAXX - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_EDIROL_UAXX(0) }, + { QUIRK_DATA_EDIROL_UAXX(1) }, + { QUIRK_DATA_EDIROL_UAXX(2) }, + QUIRK_COMPOSITE_END } } }, { /* Edirol UM-3G */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = 0, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(0) { .out_cables = 0x0007, .in_cables = 0x0007 } @@ -1605,45 +1185,29 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* BOSS ME-25 */ USB_DEVICE(0x0582, 0x0113), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* only 44.1 kHz works at the moment */ USB_DEVICE(0x0582, 0x0120), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Roland", */ /* .product_name = "OCTO-CAPTURE", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 10, .iface = 0, @@ -1659,9 +1223,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 12, .iface = 1, @@ -1677,40 +1239,26 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } + { QUIRK_DATA_IGNORE(3) }, + { QUIRK_DATA_IGNORE(4) }, + QUIRK_COMPOSITE_END } } }, { /* only 44.1 kHz works at the moment */ USB_DEVICE(0x0582, 0x012f), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Roland", */ /* .product_name = "QUAD-CAPTURE", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 4, .iface = 0, @@ -1726,9 +1274,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 6, .iface = 1, @@ -1744,54 +1290,32 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } + { QUIRK_DATA_IGNORE(3) }, + { QUIRK_DATA_IGNORE(4) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0582, 0x0159), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Roland", */ /* .product_name = "UA-22", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(2) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -1799,19 +1323,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* UA101 and co are supported by another driver */ { USB_DEVICE(0x0582, 0x0044), /* UA-1000 high speed */ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = QUIRK_NODEV_INTERFACE }, }, { USB_DEVICE(0x0582, 0x007d), /* UA-101 high speed */ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = QUIRK_NODEV_INTERFACE }, }, { USB_DEVICE(0x0582, 0x008d), /* UA-101 full speed */ - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = QUIRK_NODEV_INTERFACE }, }, @@ -1822,7 +1346,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS, .idVendor = 0x0582, .bInterfaceClass = USB_CLASS_VENDOR_SPEC, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_AUTODETECT } @@ -1837,12 +1361,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * compliant USB MIDI ports for external MIDI and controls. */ USB_DEVICE_VENDOR_SPEC(0x06f8, 0xb000), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Hercules", .product_name = "DJ Console (WE)", - .ifnum = 4, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(4) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1852,12 +1374,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Midiman/M-Audio devices */ { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1002), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 2x2", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x0003, .in_cables = 0x0003 } @@ -1865,12 +1385,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1011), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 1x1", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1878,12 +1396,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1015), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "Keystation", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -1891,12 +1407,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1021), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 4x4", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x000f, .in_cables = 0x000f } @@ -1909,12 +1423,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Thanks to Olaf Giesbrecht */ USB_DEVICE_VER(0x0763, 0x1031, 0x0100, 0x0109), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 8x8", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x01ff, .in_cables = 0x01ff } @@ -1922,12 +1434,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1033), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 8x8", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x01ff, .in_cables = 0x01ff } @@ -1935,12 +1445,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x1041), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "MidiSport 2x4", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(QUIRK_ANY_INTERFACE) { .out_cables = 0x000f, .in_cables = 0x0003 } @@ -1948,76 +1456,41 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "Quattro", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { /* * Interfaces 0-2 are "Windows-compatible", 16-bit only, * and share endpoints with the other interfaces. * Ignore them. The other interfaces can do 24 bits, * but captured samples are big-endian (see usbaudio.c). */ + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, + { QUIRK_DATA_IGNORE(2) }, + { QUIRK_DATA_IGNORE(3) }, + { QUIRK_DATA_STANDARD_AUDIO(4) }, + { QUIRK_DATA_STANDARD_AUDIO(5) }, + { QUIRK_DATA_IGNORE(6) }, + { QUIRK_DATA_STANDARD_AUDIO(7) }, + { QUIRK_DATA_STANDARD_AUDIO(8) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 5, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 6, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 7, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 8, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 9, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(9) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2003), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "AudioPhile", - .ifnum = 6, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(6) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -2025,12 +1498,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2008), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "Ozone", - .ifnum = 3, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(3) { .out_cables = 0x0001, .in_cables = 0x0001 } @@ -2038,93 +1509,45 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x200d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "M-Audio", .product_name = "OmniStudio", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, + { QUIRK_DATA_IGNORE(2) }, + { QUIRK_DATA_IGNORE(3) }, + { QUIRK_DATA_STANDARD_AUDIO(4) }, + { QUIRK_DATA_STANDARD_AUDIO(5) }, + { QUIRK_DATA_IGNORE(6) }, + { QUIRK_DATA_STANDARD_AUDIO(7) }, + { QUIRK_DATA_STANDARD_AUDIO(8) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 5, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 6, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 7, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 8, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 9, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(9) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x0763, 0x2019), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "M-Audio", */ /* .product_name = "Ozone Academic", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_MIDIMAN, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(3) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -2134,21 +1557,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2030), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track C400", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(1) }, /* Playback */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, .iface = 2, @@ -2172,9 +1588,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Capture */ { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 3, @@ -2196,30 +1610,21 @@ YAMAHA_DEVICE(0x7010, "UB99"), .clock = 0x80, } }, - /* MIDI */ - { - .ifnum = -1 /* Interface = 4 */ - } + /* MIDI: Interface = 4*/ + QUIRK_COMPOSITE_END } } }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2031), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track C600", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(1) }, /* Playback */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 2, @@ -2243,9 +1648,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Capture */ { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, .iface = 3, @@ -2267,29 +1670,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), .clock = 0x80, } }, - /* MIDI */ - { - .ifnum = -1 /* Interface = 4 */ - } + /* MIDI: Interface = 4 */ + QUIRK_COMPOSITE_END } } }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2080), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 1, @@ -2311,9 +1705,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 2, @@ -2335,28 +1727,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, /* interface 3 (MIDI) is standard compliant */ - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { USB_DEVICE_VENDOR_SPEC(0x0763, 0x2081), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "M-Audio", */ /* .product_name = "Fast Track Ultra 8R", */ - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 1, @@ -2378,9 +1761,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 2, @@ -2402,9 +1783,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, /* interface 3 (MIDI) is standard compliant */ - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -2412,21 +1791,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Casio devices */ { USB_DEVICE(0x07cf, 0x6801), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Casio", .product_name = "PL-40R", - .ifnum = 0, - .type = QUIRK_MIDI_YAMAHA + QUIRK_DATA_MIDI_YAMAHA(0) } }, { /* this ID is used by several devices without a product ID */ USB_DEVICE(0x07cf, 0x6802), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Casio", .product_name = "Keyboard", - .ifnum = 0, - .type = QUIRK_MIDI_YAMAHA + QUIRK_DATA_MIDI_YAMAHA(0) } }, @@ -2439,23 +1816,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), .idVendor = 0x07fd, .idProduct = 0x0001, .bDeviceSubClass = 2, - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "MOTU", .product_name = "Fastlane", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_MIDI_RAW_BYTES - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_RAW_BYTES(0) }, + { QUIRK_DATA_IGNORE(1) }, + QUIRK_COMPOSITE_END } } }, @@ -2463,12 +1830,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Emagic devices */ { USB_DEVICE(0x086a, 0x0001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Emagic", .product_name = "Unitor8", - .ifnum = 2, - .type = QUIRK_MIDI_EMAGIC, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_EMAGIC(2) { .out_cables = 0x80ff, .in_cables = 0x80ff } @@ -2476,12 +1841,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x086a, 0x0002), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Emagic", /* .product_name = "AMT8", */ - .ifnum = 2, - .type = QUIRK_MIDI_EMAGIC, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_EMAGIC(2) { .out_cables = 0x80ff, .in_cables = 0x80ff } @@ -2489,12 +1852,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x086a, 0x0003), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Emagic", /* .product_name = "MT4", */ - .ifnum = 2, - .type = QUIRK_MIDI_EMAGIC, - .data = & (const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_EMAGIC(2) { .out_cables = 0x800f, .in_cables = 0x8003 } @@ -2504,38 +1865,35 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* KORG devices */ { USB_DEVICE_VENDOR_SPEC(0x0944, 0x0200), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "KORG, Inc.", /* .product_name = "PANDORA PX5D", */ - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE, + QUIRK_DATA_STANDARD_MIDI(3) } }, { USB_DEVICE_VENDOR_SPEC(0x0944, 0x0201), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "KORG, Inc.", /* .product_name = "ToneLab ST", */ - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE, + QUIRK_DATA_STANDARD_MIDI(3) } }, { USB_DEVICE_VENDOR_SPEC(0x0944, 0x0204), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "KORG, Inc.", /* .product_name = "ToneLab EX", */ - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE, + QUIRK_DATA_STANDARD_MIDI(3) } }, /* AKAI devices */ { USB_DEVICE(0x09e8, 0x0062), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "AKAI", .product_name = "MPD16", .ifnum = 0, @@ -2546,21 +1904,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* Akai MPC Element */ USB_DEVICE(0x09e8, 0x0021), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_STANDARD_MIDI(1) }, + QUIRK_COMPOSITE_END } } }, @@ -2569,66 +1917,36 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* Steinberg MI2 */ USB_DEVICE_VENDOR_SPEC(0x0a4e, 0x2040), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(3) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* Steinberg MI4 */ USB_DEVICE_VENDOR_SPEC(0x0a4e, 0x4040), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = & (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(3) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -2636,34 +1954,31 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* TerraTec devices */ { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "TerraTec", .product_name = "PHASE 26", - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DATA_STANDARD_MIDI(3) } }, { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0013), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "TerraTec", .product_name = "PHASE 26", - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DATA_STANDARD_MIDI(3) } }, { USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0014), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "TerraTec", .product_name = "PHASE 26", - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DATA_STANDARD_MIDI(3) } }, { USB_DEVICE(0x0ccd, 0x0035), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Miditech", .product_name = "Play'n Roll", .ifnum = 0, @@ -2678,7 +1993,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Novation EMS devices */ { USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Novation", .product_name = "ReMOTE Audio/XStation", .ifnum = 4, @@ -2687,7 +2002,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE_VENDOR_SPEC(0x1235, 0x0002), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Novation", .product_name = "Speedio", .ifnum = 3, @@ -2696,38 +2011,29 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { USB_DEVICE(0x1235, 0x000a), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Novation", */ /* .product_name = "Nocturn", */ - .ifnum = 0, - .type = QUIRK_MIDI_RAW_BYTES + QUIRK_DATA_RAW_BYTES(0) } }, { USB_DEVICE(0x1235, 0x000e), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { /* .vendor_name = "Novation", */ /* .product_name = "Launchpad", */ - .ifnum = 0, - .type = QUIRK_MIDI_RAW_BYTES + QUIRK_DATA_RAW_BYTES(0) } }, { USB_DEVICE(0x1235, 0x0010), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Focusrite", .product_name = "Saffire 6 USB", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2754,9 +2060,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, @@ -2778,28 +2082,19 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = 1, - .type = QUIRK_MIDI_RAW_BYTES - }, - { - .ifnum = -1 - } + { QUIRK_DATA_RAW_BYTES(1) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE(0x1235, 0x0018), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Novation", .product_name = "Twitch", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = & (const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -2818,19 +2113,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = 1, - .type = QUIRK_MIDI_RAW_BYTES - }, - { - .ifnum = -1 - } + { QUIRK_DATA_RAW_BYTES(1) }, + QUIRK_COMPOSITE_END } } }, { USB_DEVICE_VENDOR_SPEC(0x1235, 0x4661), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Novation", .product_name = "ReMOTE25", .ifnum = 0, @@ -2842,25 +2132,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* VirusTI Desktop */ USB_DEVICE_VENDOR_SPEC(0x133e, 0x0815), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(3) { .out_cables = 0x0003, .in_cables = 0x0003 } }, - { - .ifnum = 4, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = -1 - } + { QUIRK_DATA_IGNORE(4) }, + QUIRK_COMPOSITE_END } } }, @@ -2888,7 +2169,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* QinHeng devices */ { USB_DEVICE(0x1a86, 0x752d), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "QinHeng", .product_name = "CH345", .ifnum = 1, @@ -2902,7 +2183,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Miditech devices */ { USB_DEVICE(0x4752, 0x0011), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Miditech", .product_name = "Midistart-2", .ifnum = 0, @@ -2914,7 +2195,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* this ID used by both Miditech MidiStudio-2 and CME UF-x */ USB_DEVICE(0x7104, 0x2202), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .ifnum = 0, .type = QUIRK_MIDI_CME } @@ -2924,20 +2205,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), { /* Thanks to Clemens Ladisch */ USB_DEVICE(0x0dba, 0x1000), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Digidesign", .product_name = "MBox", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]){ + QUIRK_DATA_COMPOSITE{ + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, @@ -2958,9 +2232,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 1, @@ -2981,9 +2253,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -2991,24 +2261,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* DIGIDESIGN MBOX 2 */ { USB_DEVICE(0x0dba, 0x3000), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Digidesign", .product_name = "Mbox 2", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 2, @@ -3026,15 +2286,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + { QUIRK_DATA_IGNORE(3) }, { - .ifnum = 3, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 4, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { - .formats = SNDRV_PCM_FMTBIT_S24_3BE, + QUIRK_DATA_AUDIOFORMAT(4) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 2, .iface = 4, .altsetting = 2, @@ -3051,14 +2306,9 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, + { QUIRK_DATA_IGNORE(5) }, { - .ifnum = 5, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 6, - .type = QUIRK_MIDI_MIDIMAN, - .data = &(const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_MIDIMAN(6) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, @@ -3066,33 +2316,21 @@ YAMAHA_DEVICE(0x7010, "UB99"), .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, /* DIGIDESIGN MBOX 3 */ { USB_DEVICE(0x0dba, 0x5000), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Digidesign", .product_name = "Mbox 3", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_IGNORE(1) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .fmt_bits = 24, .channels = 4, @@ -3119,9 +2357,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .fmt_bits = 24, .channels = 4, @@ -3145,36 +2381,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 4, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = &(const struct snd_usb_midi_endpoint_info) { + QUIRK_DATA_MIDI_FIXED_ENDPOINT(4) { .out_cables = 0x0001, .in_cables = 0x0001 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "TASCAM", .product_name = "US122 MKII", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 1, @@ -3195,9 +2420,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3205,20 +2428,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Denon DN-X1600 */ { USB_AUDIO_DEVICE(0x154e, 0x500e), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Denon", .product_name = "DN-X1600", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]){ + QUIRK_DATA_COMPOSITE{ + { QUIRK_DATA_IGNORE(0) }, { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 1, @@ -3239,9 +2455,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 2, @@ -3261,13 +2475,8 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = 4, - .type = QUIRK_MIDI_STANDARD_INTERFACE, - }, - { - .ifnum = -1 - } + { QUIRK_DATA_STANDARD_MIDI(4) }, + QUIRK_COMPOSITE_END } } }, @@ -3276,17 +2485,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), { USB_DEVICE(0x045e, 0x0283), .bInterfaceClass = USB_CLASS_PER_INTERFACE, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Microsoft", .product_name = "XboxLive Headset/Xbox Communicator", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { { /* playback */ - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 1, .iface = 0, @@ -3302,9 +2507,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, { /* capture */ - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 1, .iface = 1, @@ -3318,9 +2521,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_max = 16000 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3329,18 +2530,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), { USB_DEVICE(0x200c, 0x100b), .bInterfaceClass = USB_CLASS_PER_INTERFACE, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 1, @@ -3359,9 +2553,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3374,28 +2566,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), * enabled in create_standard_audio_quirk(). */ USB_DEVICE(0x1686, 0x00dd), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - /* Playback */ - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE, - }, - { - /* Capture */ - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE, - }, - { - /* Midi */ - .ifnum = 3, - .type = QUIRK_MIDI_STANDARD_INTERFACE - }, - { - .ifnum = -1 - }, + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(1) }, /* Playback */ + { QUIRK_DATA_STANDARD_AUDIO(2) }, /* Capture */ + { QUIRK_DATA_STANDARD_MIDI(3) }, /* Midi */ + QUIRK_COMPOSITE_END } } }, @@ -3409,18 +2585,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING, - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_MIDI_STANDARD_INTERFACE + QUIRK_DRIVER_INFO { + QUIRK_DATA_STANDARD_MIDI(QUIRK_ANY_INTERFACE) } }, /* Rane SL-1 */ { USB_DEVICE(0x13e5, 0x0001), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_AUDIO_STANDARD_INTERFACE + QUIRK_DRIVER_INFO { + QUIRK_DATA_STANDARD_AUDIO(QUIRK_ANY_INTERFACE) } }, @@ -3436,24 +2610,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), * and only the 48 kHz sample rate works for the playback interface. */ USB_DEVICE(0x0a12, 0x1243), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - /* Capture */ - { - .ifnum = 1, - .type = QUIRK_IGNORE_INTERFACE, - }, + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, + { QUIRK_DATA_IGNORE(1) }, /* Capture */ /* Playback */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 2, @@ -3472,9 +2635,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - }, + QUIRK_COMPOSITE_END } } }, @@ -3487,19 +2648,12 @@ YAMAHA_DEVICE(0x7010, "UB99"), * even on windows. */ USB_DEVICE(0x19b5, 0x0021), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, /* Playback */ { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 1, @@ -3518,29 +2672,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - }, + QUIRK_COMPOSITE_END } } }, /* MOTU Microbook II */ { USB_DEVICE_VENDOR_SPEC(0x07fd, 0x0004), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "MOTU", .product_name = "MicroBookII", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(0) }, { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 6, .iface = 0, @@ -3561,9 +2706,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3BE, .channels = 8, .iface = 0, @@ -3584,9 +2727,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3598,14 +2739,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * The feedback for the output is the input. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0023), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 12, .iface = 0, @@ -3622,9 +2759,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 10, .iface = 0, @@ -3642,9 +2777,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3687,14 +2820,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, // outputs .iface = 0, @@ -3711,9 +2840,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, // inputs .iface = 0, @@ -3731,9 +2858,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 48000 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3744,14 +2869,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * The feedback for the output is the dummy input. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000e), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -3768,9 +2889,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 2, .iface = 0, @@ -3788,9 +2907,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3801,14 +2918,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * PCM is 6 channels out & 4 channels in @ 44.1 fixed */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000d), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, //Master, Headphones & Booth .iface = 0, @@ -3825,9 +2938,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, //2x RCA inputs (CH1 & CH2) .iface = 0, @@ -3845,9 +2956,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3859,14 +2968,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * The Feedback for the output is the input */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001e), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 4, .iface = 0, @@ -3883,9 +2988,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, .iface = 0, @@ -3903,9 +3006,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3916,14 +3017,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * 10 channels playback & 12 channels capture @ 44.1/48/96kHz S24LE */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000a), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 10, .iface = 0, @@ -3944,9 +3041,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 12, .iface = 0, @@ -3968,9 +3063,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -3982,14 +3075,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * The Feedback for the output is the input */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0029), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, .iface = 0, @@ -4006,9 +3095,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 6, .iface = 0, @@ -4026,9 +3113,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4046,20 +3131,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ { USB_AUDIO_DEVICE(0x534d, 0x0021), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "MacroSilicon", .product_name = "MS210x", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(2) }, { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 3, @@ -4074,9 +3152,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_max = 48000, } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4094,20 +3170,13 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ { USB_AUDIO_DEVICE(0x534d, 0x2109), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "MacroSilicon", .product_name = "MS2109", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_MIXER(2) }, { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_MIXER, - }, - { - .ifnum = 3, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(3) { .formats = SNDRV_PCM_FMTBIT_S16_LE, .channels = 2, .iface = 3, @@ -4122,9 +3191,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_max = 48000, } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4134,14 +3201,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * 8 channels playback & 8 channels capture @ 44.1/48/96kHz S24LE */ USB_DEVICE_VENDOR_SPEC(0x08e4, 0x017f), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 0, @@ -4160,9 +3223,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 0, @@ -4182,9 +3243,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100, 48000, 96000 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4194,14 +3253,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * 10 channels playback & 12 channels capture @ 48kHz S24LE */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 10, .iface = 0, @@ -4220,9 +3275,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 12, .iface = 0, @@ -4240,9 +3293,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 48000 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4254,14 +3305,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Capture on EP 0x86 */ USB_DEVICE_VENDOR_SPEC(0x08e4, 0x0163), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 0, @@ -4281,9 +3328,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, .iface = 0, @@ -4303,9 +3348,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100, 48000, 96000 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4316,14 +3359,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), * and 8 channels in @ 48 fixed (endpoint 0x82). */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0013), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, // outputs .iface = 0, @@ -4340,9 +3379,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - .ifnum = 0, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, .channels = 8, // inputs .iface = 0, @@ -4360,9 +3397,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 48000 } } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4373,28 +3408,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), */ USB_DEVICE(0x1395, 0x0300), .bInterfaceClass = USB_CLASS_PER_INTERFACE, - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { // Communication - { - .ifnum = 3, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, + { QUIRK_DATA_STANDARD_AUDIO(3) }, // Recording - { - .ifnum = 4, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, + { QUIRK_DATA_STANDARD_AUDIO(4) }, // Main - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, - { - .ifnum = -1 - } + { QUIRK_DATA_STANDARD_AUDIO(1) }, + QUIRK_COMPOSITE_END } } }, @@ -4403,21 +3425,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Fiero SC-01 (firmware v1.0.0 @ 48 kHz) */ USB_DEVICE(0x2b53, 0x0023), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Fiero", .product_name = "SC-01", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, /* Playback */ { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4437,9 +3452,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Capture */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4458,9 +3471,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .clock = 0x29 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4469,21 +3480,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Fiero SC-01 (firmware v1.0.0 @ 96 kHz) */ USB_DEVICE(0x2b53, 0x0024), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Fiero", .product_name = "SC-01", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, /* Playback */ { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4503,9 +3507,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Capture */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4524,9 +3526,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .clock = 0x29 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4535,21 +3535,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Fiero SC-01 (firmware v1.1.0) */ USB_DEVICE(0x2b53, 0x0031), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Fiero", .product_name = "SC-01", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = &(const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_STANDARD_INTERFACE - }, + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_STANDARD_AUDIO(0) }, /* Playback */ { - .ifnum = 1, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(1) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4570,9 +3563,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), }, /* Capture */ { - .ifnum = 2, - .type = QUIRK_AUDIO_FIXED_ENDPOINT, - .data = &(const struct audioformat) { + QUIRK_DATA_AUDIOFORMAT(2) { .formats = SNDRV_PCM_FMTBIT_S32_LE, .channels = 2, .fmt_bits = 24, @@ -4592,9 +3583,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .clock = 0x29 } }, - { - .ifnum = -1 - } + QUIRK_COMPOSITE_END } } }, @@ -4603,27 +3592,14 @@ YAMAHA_DEVICE(0x7010, "UB99"), * For the standard mode, Mythware XA001AU has ID ffad:a001 */ USB_DEVICE_VENDOR_SPEC(0xffad, 0xa001), - .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + QUIRK_DRIVER_INFO { .vendor_name = "Mythware", .product_name = "XA001AU", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_IGNORE_INTERFACE, - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_STANDARD_INTERFACE, - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_STANDARD_INTERFACE, - }, - { - .ifnum = -1 - } + QUIRK_DATA_COMPOSITE { + { QUIRK_DATA_IGNORE(0) }, + { QUIRK_DATA_STANDARD_AUDIO(1) }, + { QUIRK_DATA_STANDARD_AUDIO(2) }, + QUIRK_COMPOSITE_END } } }, From 73abd9698960c5d097559432cac13bb1f0aaf3df Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:38 -0300 Subject: [PATCH 190/410] ASoC: fsl_audmix: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-1-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_audmix.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index 1671a3037c60..f3a24758aedb 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -512,7 +512,6 @@ static void fsl_audmix_remove(struct platform_device *pdev) platform_device_unregister(priv->pdev); } -#ifdef CONFIG_PM static int fsl_audmix_runtime_resume(struct device *dev) { struct fsl_audmix *priv = dev_get_drvdata(dev); @@ -540,14 +539,11 @@ static int fsl_audmix_runtime_suspend(struct device *dev) return 0; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_audmix_pm = { - SET_RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, - fsl_audmix_runtime_resume, - NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + RUNTIME_PM_OPS(fsl_audmix_runtime_suspend, fsl_audmix_runtime_resume, + NULL) + SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static struct platform_driver fsl_audmix_driver = { @@ -556,7 +552,7 @@ static struct platform_driver fsl_audmix_driver = { .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, - .pm = &fsl_audmix_pm, + .pm = pm_ptr(&fsl_audmix_pm), }, }; module_platform_driver(fsl_audmix_driver); From b7e4dd8da05a9de18471868a914d609522d04fdd Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:39 -0300 Subject: [PATCH 191/410] ASoC: fsl_mqs: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-2-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_mqs.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c95b84a54dc4..df160834c81b 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -265,7 +265,6 @@ static void fsl_mqs_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); } -#ifdef CONFIG_PM static int fsl_mqs_runtime_resume(struct device *dev) { struct fsl_mqs *mqs_priv = dev_get_drvdata(dev); @@ -299,14 +298,10 @@ static int fsl_mqs_runtime_suspend(struct device *dev) return 0; } -#endif static const struct dev_pm_ops fsl_mqs_pm_ops = { - SET_RUNTIME_PM_OPS(fsl_mqs_runtime_suspend, - fsl_mqs_runtime_resume, - NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) + RUNTIME_PM_OPS(fsl_mqs_runtime_suspend, fsl_mqs_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) }; static const struct fsl_mqs_soc_data fsl_mqs_imx8qm_data = { @@ -390,7 +385,7 @@ static struct platform_driver fsl_mqs_driver = { .driver = { .name = "fsl-mqs", .of_match_table = fsl_mqs_dt_ids, - .pm = &fsl_mqs_pm_ops, + .pm = pm_ptr(&fsl_mqs_pm_ops), }, }; From bbc0798c402acb6799a8332fb0a49c05f4a5a414 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:40 -0300 Subject: [PATCH 192/410] ASoC: fsl_rpmsg: Switch to RUNTIME_PM_OPS() Replace SET_RUNTIME_PM_OPS() with its modern RUNTIME_PM_OPS() alternative. The combined usage of pm_ptr() and RUNTIME_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-3-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_rpmsg.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 467d6bc9f956..be46fbfd487a 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -286,7 +286,6 @@ static void fsl_rpmsg_remove(struct platform_device *pdev) platform_device_unregister(rpmsg->card_pdev); } -#ifdef CONFIG_PM static int fsl_rpmsg_runtime_resume(struct device *dev) { struct fsl_rpmsg *rpmsg = dev_get_drvdata(dev); @@ -321,12 +320,10 @@ static int fsl_rpmsg_runtime_suspend(struct device *dev) return 0; } -#endif static const struct dev_pm_ops fsl_rpmsg_pm_ops = { - SET_RUNTIME_PM_OPS(fsl_rpmsg_runtime_suspend, - fsl_rpmsg_runtime_resume, - NULL) + RUNTIME_PM_OPS(fsl_rpmsg_runtime_suspend, fsl_rpmsg_runtime_resume, + NULL) }; static struct platform_driver fsl_rpmsg_driver = { @@ -334,7 +331,7 @@ static struct platform_driver fsl_rpmsg_driver = { .remove_new = fsl_rpmsg_remove, .driver = { .name = "fsl_rpmsg", - .pm = &fsl_rpmsg_pm_ops, + .pm = pm_ptr(&fsl_rpmsg_pm_ops), .of_match_table = fsl_rpmsg_ids, }, }; From 01661bb9560def44eff0c79ebf14ec7ce0fdffab Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:41 -0300 Subject: [PATCH 193/410] ASoC: fsl_spdif: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-4-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_spdif.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a63121c888e0..eace399cb064 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1667,7 +1667,6 @@ static void fsl_spdif_remove(struct platform_device *pdev) pm_runtime_disable(&pdev->dev); } -#ifdef CONFIG_PM static int fsl_spdif_runtime_suspend(struct device *dev) { struct fsl_spdif_priv *spdif_priv = dev_get_drvdata(dev); @@ -1739,13 +1738,11 @@ disable_core_clk: return ret; } -#endif /* CONFIG_PM */ static const struct dev_pm_ops fsl_spdif_pm = { - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) - SET_RUNTIME_PM_OPS(fsl_spdif_runtime_suspend, fsl_spdif_runtime_resume, - NULL) + SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) + RUNTIME_PM_OPS(fsl_spdif_runtime_suspend, fsl_spdif_runtime_resume, + NULL) }; static const struct of_device_id fsl_spdif_dt_ids[] = { @@ -1763,7 +1760,7 @@ static struct platform_driver fsl_spdif_driver = { .driver = { .name = "fsl-spdif-dai", .of_match_table = fsl_spdif_dt_ids, - .pm = &fsl_spdif_pm, + .pm = pm_ptr(&fsl_spdif_pm), }, .probe = fsl_spdif_probe, .remove_new = fsl_spdif_remove, From 8ffb2fe2e92c98e5c69e9af9656baed7d298059a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:42 -0300 Subject: [PATCH 194/410] ASoC: fsl_ssi: Switch to SYSTEM_SLEEP_PM_OPS Replace SET_SYSTEM_SLEEP_PM_OPS() with its modern SYSTEM_SLEEP_PM_OPS() alternative. The combined usage of pm_sleep_ptr() and SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM_SLEEP ifdefery from the suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-5-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_ssi.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4ca3a16f7ac0..c4c1d9c44056 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1693,7 +1693,6 @@ static void fsl_ssi_remove(struct platform_device *pdev) } } -#ifdef CONFIG_PM_SLEEP static int fsl_ssi_suspend(struct device *dev) { struct fsl_ssi *ssi = dev_get_drvdata(dev); @@ -1723,17 +1722,16 @@ static int fsl_ssi_resume(struct device *dev) return regcache_sync(regs); } -#endif /* CONFIG_PM_SLEEP */ static const struct dev_pm_ops fsl_ssi_pm = { - SET_SYSTEM_SLEEP_PM_OPS(fsl_ssi_suspend, fsl_ssi_resume) + SYSTEM_SLEEP_PM_OPS(fsl_ssi_suspend, fsl_ssi_resume) }; static struct platform_driver fsl_ssi_driver = { .driver = { .name = "fsl-ssi-dai", .of_match_table = fsl_ssi_ids, - .pm = &fsl_ssi_pm, + .pm = pm_sleep_ptr(&fsl_ssi_pm), }, .probe = fsl_ssi_probe, .remove_new = fsl_ssi_remove, From c504885a351b57c26c077992dde6904b8bbbdcd4 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:43 -0300 Subject: [PATCH 195/410] ASoC: imx-audmux: Switch to SYSTEM_SLEEP_PM_OPS Replace SET_SYSTEM_SLEEP_PM_OPS() with its modern SYSTEM_SLEEP_PM_OPS() alternative. The combined usage of pm_sleep_ptr() and SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM_SLEEP ifdefery from the suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-6-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-audmux.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 747ab2f1aae3..f97ae14dc452 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -320,7 +320,6 @@ static void imx_audmux_remove(struct platform_device *pdev) audmux_debugfs_remove(); } -#ifdef CONFIG_PM_SLEEP static int imx_audmux_suspend(struct device *dev) { int i; @@ -348,10 +347,9 @@ static int imx_audmux_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ static const struct dev_pm_ops imx_audmux_pm = { - SET_SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume) + SYSTEM_SLEEP_PM_OPS(imx_audmux_suspend, imx_audmux_resume) }; static struct platform_driver imx_audmux_driver = { @@ -359,7 +357,7 @@ static struct platform_driver imx_audmux_driver = { .remove_new = imx_audmux_remove, .driver = { .name = DRIVER_NAME, - .pm = &imx_audmux_pm, + .pm = pm_sleep_ptr(&imx_audmux_pm), .of_match_table = imx_audmux_dt_ids, } }; From bcbbf713061c41f696f47baa4bdcb789a59d4f21 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 8 Aug 2024 15:49:44 -0300 Subject: [PATCH 196/410] ASoC: imx-pcm-rpmsg: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Replace SET_RUNTIME_PM_OPS()/SET SYSTEM_SLEEP_PM_OPS() with their modern RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() alternatives. The combined usage of pm_ptr() and RUNTIME_PM_OPS/SYSTEM_SLEEP_PM_OPS() allows the compiler to evaluate if the runtime suspend/resume() functions are used at build time or are simply dead code. This allows removing the CONFIG_PM ifdefery from the runtime suspend/resume() functions. Signed-off-by: Fabio Estevam Link: https://patch.msgid.link/20240808184944.267686-7-festevam@gmail.com Signed-off-by: Mark Brown --- sound/soc/fsl/imx-pcm-rpmsg.c | 14 ++++---------- 1 file changed, 4 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index b0944a07ab47..22156f99dcee 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -753,7 +753,6 @@ static void imx_rpmsg_pcm_remove(struct platform_device *pdev) destroy_workqueue(info->rpmsg_wq); } -#ifdef CONFIG_PM static int imx_rpmsg_pcm_runtime_resume(struct device *dev) { struct rpmsg_info *info = dev_get_drvdata(dev); @@ -771,9 +770,7 @@ static int imx_rpmsg_pcm_runtime_suspend(struct device *dev) return 0; } -#endif -#ifdef CONFIG_PM_SLEEP static int imx_rpmsg_pcm_suspend(struct device *dev) { struct rpmsg_info *info = dev_get_drvdata(dev); @@ -809,14 +806,11 @@ static int imx_rpmsg_pcm_resume(struct device *dev) return 0; } -#endif /* CONFIG_PM_SLEEP */ static const struct dev_pm_ops imx_rpmsg_pcm_pm_ops = { - SET_RUNTIME_PM_OPS(imx_rpmsg_pcm_runtime_suspend, - imx_rpmsg_pcm_runtime_resume, - NULL) - SET_SYSTEM_SLEEP_PM_OPS(imx_rpmsg_pcm_suspend, - imx_rpmsg_pcm_resume) + RUNTIME_PM_OPS(imx_rpmsg_pcm_runtime_suspend, + imx_rpmsg_pcm_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(imx_rpmsg_pcm_suspend, imx_rpmsg_pcm_resume) }; static const struct platform_device_id imx_rpmsg_pcm_id_table[] = { @@ -832,7 +826,7 @@ static struct platform_driver imx_pcm_rpmsg_driver = { .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, - .pm = &imx_rpmsg_pcm_pm_ops, + .pm = pm_ptr(&imx_rpmsg_pcm_pm_ops), }, }; module_platform_driver(imx_pcm_rpmsg_driver); From c8c3d9f8e3ffc3e095159fdfda37038df74efdc5 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 30 Jul 2024 01:30:51 +0000 Subject: [PATCH 197/410] ASoC: soc-pcm: remove snd_soc_dpcm_stream_{lock/unlock}_irq() soc-pcm.c has snd_soc_dpcm_stream_{lock/unlock}_irq() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/875xsnll85.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 25 +++++++------------------ 1 file changed, 7 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1edcb8d6f6ee..f79abcf36105 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,21 +49,9 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -static inline void snd_soc_dpcm_stream_lock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_lock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) -static inline void snd_soc_dpcm_stream_unlock_irq(struct snd_soc_pcm_runtime *rtd, - int stream) -{ - snd_pcm_stream_unlock_irq(snd_soc_dpcm_get_substream(rtd, stream)); -} - #define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) @@ -260,14 +248,14 @@ static void dpcm_set_fe_update_state(struct snd_soc_pcm_runtime *fe, struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); if (state == SND_SOC_DPCM_UPDATE_NO && fe->dpcm[stream].trigger_pending) { dpcm_fe_dai_do_trigger(substream, fe->dpcm[stream].trigger_pending - 1); fe->dpcm[stream].trigger_pending = 0; } fe->dpcm[stream].runtime_update = state; - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); } static void dpcm_set_be_update_state(struct snd_soc_pcm_runtime *be, @@ -1272,10 +1260,10 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, dpcm->be = be; dpcm->fe = fe; dpcm->state = SND_SOC_DPCM_LINK_STATE_NEW; - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(fe_substream); list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(fe_substream); dev_dbg(fe->dev, "connected new DPCM %s path %s %s %s\n", snd_pcm_direction_name(stream), fe->dai_link->name, @@ -1320,11 +1308,12 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; + struct snd_pcm_substream *substream = snd_soc_dpcm_get_substream(fe, stream); LIST_HEAD(deleted_dpcms); snd_soc_dpcm_mutex_assert_held(fe); - snd_soc_dpcm_stream_lock_irq(fe, stream); + snd_pcm_stream_lock_irq(substream); for_each_dpcm_be_safe(fe, stream, dpcm, d) { dev_dbg(fe->dev, "ASoC: BE %s disconnect check for %s\n", snd_pcm_direction_name(stream), @@ -1343,7 +1332,7 @@ void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) list_del(&dpcm->list_be); list_move(&dpcm->list_fe, &deleted_dpcms); } - snd_soc_dpcm_stream_unlock_irq(fe, stream); + snd_pcm_stream_unlock_irq(substream); while (!list_empty(&deleted_dpcms)) { dpcm = list_first_entry(&deleted_dpcms, struct snd_soc_dpcm, From 60b5c173f5542adf020f5235db7fe5e5fa4ae0d8 Mon Sep 17 00:00:00 2001 From: tangbin Date: Sat, 13 Jul 2024 11:34:28 -0400 Subject: [PATCH 198/410] ASoC: loongson: Remove useless variable definitions In the function loongson_pcm_trigger and loongson_pcm_open, the 'ret' is useless, so remove it to simplify code. Signed-off-by: tangbin Link: https://patch.msgid.link/20240713153428.44858-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_dma.c | 15 +++++---------- 1 file changed, 5 insertions(+), 10 deletions(-) diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c index 4fcc2868160b..0238f88bc674 100644 --- a/sound/soc/loongson/loongson_dma.c +++ b/sound/soc/loongson/loongson_dma.c @@ -95,7 +95,6 @@ static int loongson_pcm_trigger(struct snd_soc_component *component, struct device *dev = substream->pcm->card->dev; void __iomem *order_reg = prtd->dma_data->order_addr; u64 val; - int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -129,7 +128,7 @@ static int loongson_pcm_trigger(struct snd_soc_component *component, return -EINVAL; } - return ret; + return 0; } static int loongson_pcm_hw_params(struct snd_soc_component *component, @@ -230,7 +229,6 @@ static int loongson_pcm_open(struct snd_soc_component *component, struct snd_card *card = substream->pcm->card; struct loongson_runtime_data *prtd; struct loongson_dma_data *dma_data; - int ret; /* * For mysterious reasons (and despite what the manual says) @@ -252,20 +250,17 @@ static int loongson_pcm_open(struct snd_soc_component *component, prtd->dma_desc_arr = dma_alloc_coherent(card->dev, PAGE_SIZE, &prtd->dma_desc_arr_phy, GFP_KERNEL); - if (!prtd->dma_desc_arr) { - ret = -ENOMEM; + if (!prtd->dma_desc_arr) goto desc_err; - } + prtd->dma_desc_arr_size = PAGE_SIZE / sizeof(*prtd->dma_desc_arr); prtd->dma_pos_desc = dma_alloc_coherent(card->dev, sizeof(*prtd->dma_pos_desc), &prtd->dma_pos_desc_phy, GFP_KERNEL); - if (!prtd->dma_pos_desc) { - ret = -ENOMEM; + if (!prtd->dma_pos_desc) goto pos_err; - } dma_data = snd_soc_dai_get_dma_data(snd_soc_rtd_to_cpu(rtd, 0), substream); prtd->dma_data = dma_data; @@ -279,7 +274,7 @@ pos_err: desc_err: kfree(prtd); - return ret; + return -ENOMEM; } static int loongson_pcm_close(struct snd_soc_component *component, From 0a9173541b3f8cde8ad923eebd2e157650e13f35 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 15 Aug 2024 12:21:35 +0800 Subject: [PATCH 199/410] ASoc: tas2781: Remove unnecessary line feed and space Remove unnecessary line feed for tasdevice_dsp_create_ctrls, and remove two unnecessary spaces in tas2563_digital_gain_get and tas2563_digital_gain_put. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240815042138.1997-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8c9efd9183ff..9a32e0504857 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -157,7 +157,7 @@ static int tas2563_digital_gain_get( mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); goto out; @@ -203,7 +203,7 @@ static int tas2563_digital_gain_put( vol = clamp(vol, 0, max); mutex_lock(&tas_dev->codec_lock); /* Read the primary device */ - ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); + ret = tasdevice_dev_bulk_read(tas_dev, 0, reg, data, 4); if (ret) { dev_err(tas_dev->dev, "%s, get AMP vol error\n", __func__); rc = -1; @@ -423,8 +423,7 @@ static int tasdevice_configuration_put( return ret; } -static int tasdevice_dsp_create_ctrls( - struct tasdevice_priv *tas_priv) +static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; char *prog_name, *conf_name; From d32cf9fbcb613a8f43e2211e964ccc8bb7271dcf Mon Sep 17 00:00:00 2001 From: Zhang Zekun Date: Fri, 16 Aug 2024 10:18:26 +0800 Subject: [PATCH 200/410] ALSA: aoa: Use helper function for_each_child_of_node() for_each_child_of_node can help to iterate through the device_node, and we don't need to use while loop. No functional change with this conversion. Signed-off-by: Zhang Zekun Link: https://patch.msgid.link/20240816021826.65190-1-zhangzekun11@huawei.com Signed-off-by: Takashi Iwai --- sound/aoa/soundbus/i2sbus/core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 5431d2c49421..ce84288168e4 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -335,7 +335,7 @@ static int i2sbus_add_dev(struct macio_dev *macio, static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) { - struct device_node *np = NULL; + struct device_node *np; int got = 0, err; struct i2sbus_control *control = NULL; @@ -347,7 +347,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) return -ENODEV; } - while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + for_each_child_of_node(dev->ofdev.dev.of_node, np) { if (of_device_is_compatible(np, "i2sbus") || of_device_is_compatible(np, "i2s-modem")) { got += i2sbus_add_dev(dev, control, np); From c8a3231ae6d02b8f0cf08dc88f763f505dc35257 Mon Sep 17 00:00:00 2001 From: Yue Haibing Date: Fri, 16 Aug 2024 18:02:09 +0800 Subject: [PATCH 201/410] ALSA: oss: Remove unused declarations These functions is never implemented and used. Signed-off-by: Yue Haibing Link: https://patch.msgid.link/20240816100209.879043-1-yuehaibing@huawei.com Signed-off-by: Takashi Iwai --- sound/core/oss/pcm_plugin.h | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 55035dbf23f0..7b76cf64157e 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -74,7 +74,6 @@ int snd_pcm_plugin_build(struct snd_pcm_substream *handle, size_t extra, struct snd_pcm_plugin **ret); int snd_pcm_plugin_free(struct snd_pcm_plugin *plugin); -int snd_pcm_plugin_clear(struct snd_pcm_plugin **first); int snd_pcm_plug_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t frames); snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size); snd_pcm_sframes_t snd_pcm_plug_slave_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t clt_size); @@ -139,8 +138,6 @@ int snd_pcm_area_copy(const struct snd_pcm_channel_area *src_channel, size_t dst_offset, size_t samples, snd_pcm_format_t format); -void *snd_pcm_plug_buf_alloc(struct snd_pcm_substream *plug, snd_pcm_uframes_t size); -void snd_pcm_plug_buf_unlock(struct snd_pcm_substream *plug, void *ptr); #else static inline snd_pcm_sframes_t snd_pcm_plug_client_size(struct snd_pcm_substream *handle, snd_pcm_uframes_t drv_size) { return drv_size; } From d08ea4193a72c5e3090240872ff7ed60a70716e6 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 15 Aug 2024 17:53:20 +0100 Subject: [PATCH 202/410] ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 we seems to have ended up with duplicate clocks for frame-sync on sm8250, it has both va and fsgen which are exactly same things. Remove the redundant va clock and make it align with other SoCs. Codec driver does not even handle va clock, so remove this from the bindings and examples to avoid any confusion. Signed-off-by: Srinivas Kandagatla Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240815165320.18836-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 ++----------------- 1 file changed, 2 insertions(+), 20 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 06b5f7be3608..6f5644a89feb 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -64,6 +64,7 @@ allOf: compatible: enum: - qcom,sc7280-lpass-wsa-macro + - qcom,sm8250-lpass-wsa-macro - qcom,sm8450-lpass-wsa-macro - qcom,sc8280xp-lpass-wsa-macro then: @@ -79,24 +80,6 @@ allOf: - const: dcodec - const: fsgen - - if: - properties: - compatible: - enum: - - qcom,sm8250-lpass-wsa-macro - then: - properties: - clocks: - minItems: 6 - clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: va - - const: fsgen - - if: properties: compatible: @@ -130,8 +113,7 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, - <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; clock-output-names = "mclk"; }; From b0b228bb8d546edd23e2467ed7669d4ce62ae763 Mon Sep 17 00:00:00 2001 From: Yue Haibing Date: Sat, 17 Aug 2024 17:33:34 +0800 Subject: [PATCH 203/410] ALSA: seq: Remove unused declarations These functions are never implemented and used. Signed-off-by: Yue Haibing Link: https://patch.msgid.link/20240817093334.1120002-1-yuehaibing@huawei.com Signed-off-by: Takashi Iwai --- include/sound/seq_kernel.h | 4 ---- sound/core/seq/oss/seq_oss_device.h | 4 ---- sound/core/seq/seq_queue.h | 1 - sound/core/seq/seq_timer.h | 2 -- 4 files changed, 11 deletions(-) diff --git a/include/sound/seq_kernel.h b/include/sound/seq_kernel.h index c8621671fa70..00c32eed2124 100644 --- a/include/sound/seq_kernel.h +++ b/include/sound/seq_kernel.h @@ -86,10 +86,6 @@ static inline size_t snd_seq_event_packet_size(struct snd_seq_event *ev) /* interface for OSS emulation */ int snd_seq_set_queue_tempo(int client, struct snd_seq_queue_tempo *tempo); -/* port callback routines */ -void snd_port_init_callback(struct snd_seq_port_callback *p); -struct snd_seq_port_callback *snd_port_alloc_callback(void); - /* port attach/detach */ int snd_seq_event_port_attach(int client, struct snd_seq_port_callback *pcbp, int cap, int type, int midi_channels, int midi_voices, char *portname); diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index f0e964b19af7..98dd20b42976 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -116,10 +116,6 @@ __poll_t snd_seq_oss_poll(struct seq_oss_devinfo *dp, struct file *file, poll_ta void snd_seq_oss_reset(struct seq_oss_devinfo *dp); -/* */ -void snd_seq_oss_process_queue(struct seq_oss_devinfo *dp, abstime_t time); - - /* proc interface */ void snd_seq_oss_system_info_read(struct snd_info_buffer *buf); void snd_seq_oss_midi_info_read(struct snd_info_buffer *buf); diff --git a/sound/core/seq/seq_queue.h b/sound/core/seq/seq_queue.h index c69105dc1a10..74cc31aacdac 100644 --- a/sound/core/seq/seq_queue.h +++ b/sound/core/seq/seq_queue.h @@ -84,7 +84,6 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop); int snd_seq_queue_check_access(int queueid, int client); int snd_seq_queue_timer_set_tempo(int queueid, int client, struct snd_seq_queue_tempo *info); int snd_seq_queue_set_owner(int queueid, int client, int locked); -int snd_seq_queue_set_locked(int queueid, int client, int locked); int snd_seq_queue_timer_open(int queueid); int snd_seq_queue_timer_close(int queueid); int snd_seq_queue_use(int queueid, int client, int use); diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 3b906064bde1..c8803216a3a4 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -109,8 +109,6 @@ static inline void snd_seq_inc_time_nsec(snd_seq_real_time_t *tm, unsigned long struct snd_seq_queue; int snd_seq_timer_open(struct snd_seq_queue *q); int snd_seq_timer_close(struct snd_seq_queue *q); -int snd_seq_timer_midi_open(struct snd_seq_queue *q); -int snd_seq_timer_midi_close(struct snd_seq_queue *q); void snd_seq_timer_defaults(struct snd_seq_timer *tmr); void snd_seq_timer_reset(struct snd_seq_timer *tmr); int snd_seq_timer_stop(struct snd_seq_timer *tmr); From ff6615efa87482600af14c1a1c4b5da4e32f9a82 Mon Sep 17 00:00:00 2001 From: Yue Haibing Date: Sat, 17 Aug 2024 17:35:27 +0800 Subject: [PATCH 204/410] ALSA: trident: Remove unused declarations Commit 8bb8b453cb45 ("[ALSA] trident - clean up obsolete synth codes") remove synth functions but leave declarations. And Commit e5723b41abe5 ("[ALSA] Remove sequencer instrument layer") left snd_trident_attach_synthesizer(). Signed-off-by: Yue Haibing Link: https://patch.msgid.link/20240817093527.1120240-1-yuehaibing@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/trident/trident.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/trident/trident.h b/sound/pci/trident/trident.h index 9768a7fc2349..ed2d4eecc704 100644 --- a/sound/pci/trident/trident.h +++ b/sound/pci/trident/trident.h @@ -406,7 +406,6 @@ int snd_trident_create_gameport(struct snd_trident *trident); int snd_trident_pcm(struct snd_trident *trident, int device); int snd_trident_foldback_pcm(struct snd_trident *trident, int device); int snd_trident_spdif_pcm(struct snd_trident *trident, int device); -int snd_trident_attach_synthesizer(struct snd_trident * trident); struct snd_trident_voice *snd_trident_alloc_voice(struct snd_trident * trident, int type, int client, int port); void snd_trident_free_voice(struct snd_trident * trident, struct snd_trident_voice *voice); @@ -419,9 +418,5 @@ extern const struct dev_pm_ops snd_trident_pm; struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident, struct snd_pcm_substream *substream); int snd_trident_free_pages(struct snd_trident *trident, struct snd_util_memblk *blk); -struct snd_util_memblk *snd_trident_synth_alloc(struct snd_trident *trident, unsigned int size); -int snd_trident_synth_free(struct snd_trident *trident, struct snd_util_memblk *blk); -int snd_trident_synth_copy_from_user(struct snd_trident *trident, struct snd_util_memblk *blk, - int offset, const char __user *data, int size); #endif /* __SOUND_TRIDENT_H */ From 48f1434a4632c7da1a6a94e159512ebddbe13392 Mon Sep 17 00:00:00 2001 From: Yuntao Liu Date: Thu, 15 Aug 2024 09:13:12 +0000 Subject: [PATCH 205/410] ALSA: hda: cs35l41: fix module autoloading Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from spi_device_id table. Fixes: 7b2f3eb492da ("ALSA: hda: cs35l41: Add support for CS35L41 in HDA systems") Signed-off-by: Yuntao Liu Link: https://patch.msgid.link/20240815091312.757139-1-liuyuntao12@huawei.com Signed-off-by: Takashi Iwai --- sound/pci/hda/cs35l41_hda_spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/cs35l41_hda_spi.c b/sound/pci/hda/cs35l41_hda_spi.c index b76c0dfd5fef..f8c356ad0d34 100644 --- a/sound/pci/hda/cs35l41_hda_spi.c +++ b/sound/pci/hda/cs35l41_hda_spi.c @@ -38,6 +38,7 @@ static const struct spi_device_id cs35l41_hda_spi_id[] = { { "cs35l41-hda", 0 }, {} }; +MODULE_DEVICE_TABLE(spi, cs35l41_hda_spi_id); static const struct acpi_device_id cs35l41_acpi_hda_match[] = { { "CSC3551", 0 }, From e949df0b021cdd503c4e72db12b1b4154a82d698 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Tue, 13 Aug 2024 13:06:58 +0100 Subject: [PATCH 206/410] ALSA: aloop: Allow using global timers Allow using global timers as a timer source when card id is equal to -1 in the timer_source parameter. Signed-off-by: Ivan Orlov Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240813120701.171743-2-ivan.orlov0322@gmail.com --- sound/drivers/aloop.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 3650d47b7d58..eb8a68a06c4d 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -1129,6 +1129,8 @@ static int loopback_parse_timer_id(const char *str, } } } + if (card_idx == -1) + tid->dev_class = SNDRV_TIMER_CLASS_GLOBAL; if (!err && tid) { tid->card = card_idx; tid->device = dev; From 8fad71b6771afc1f12f1cdf616ae0fca628802e9 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Tue, 13 Aug 2024 13:06:59 +0100 Subject: [PATCH 207/410] Docs/sound: Add documentation for userspace-driven ALSA timers Add the documentation which describes the new userspace-driven timers API introduced in this patch series. The documentation contains: - Description of userspace-driven ALSA timers, what they are for - Description of the timers API - Example of how the timers can be created and triggered - How the timers can be used as a timer sources for snd-aloop module Suggested-by: Axel Holzinger Signed-off-by: Ivan Orlov Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240813120701.171743-3-ivan.orlov0322@gmail.com --- Documentation/sound/index.rst | 1 + Documentation/sound/utimers.rst | 126 ++++++++++++++++++++++++++++++++ 2 files changed, 127 insertions(+) create mode 100644 Documentation/sound/utimers.rst diff --git a/Documentation/sound/index.rst b/Documentation/sound/index.rst index 7e67e12730d3..c437f2a4bc85 100644 --- a/Documentation/sound/index.rst +++ b/Documentation/sound/index.rst @@ -13,6 +13,7 @@ Sound Subsystem Documentation alsa-configuration hd-audio/index cards/index + utimers .. only:: subproject and html diff --git a/Documentation/sound/utimers.rst b/Documentation/sound/utimers.rst new file mode 100644 index 000000000000..ec21567d3f72 --- /dev/null +++ b/Documentation/sound/utimers.rst @@ -0,0 +1,126 @@ +.. SPDX-License-Identifier: GPL-2.0 + +======================= +Userspace-driven timers +======================= + +:Author: Ivan Orlov + +Preface +======= + +This document describes the userspace-driven timers: virtual ALSA timers +which could be created and controlled by userspace applications using +IOCTL calls. Such timers could be useful when synchronizing audio +stream with timer sources which we don't have ALSA timers exported for +(e.g. PTP clocks), and when synchronizing the audio stream going through +two virtual sound devices using ``snd-aloop`` (for instance, when +we have a network application sending frames to one snd-aloop device, +and another sound application listening on the other end of snd-aloop). + +Enabling userspace-driven timers +================================ + +The userspace-driven timers could be enabled in the kernel using the +``CONFIG_SND_UTIMER`` configuration option. It depends on the +``CONFIG_SND_TIMER`` option, so it also should be enabled. + +Userspace-driven timers API +=========================== + +Userspace application can create a userspace-driven ALSA timer by +executing the ``SNDRV_TIMER_IOCTL_CREATE`` ioctl call on the +``/dev/snd/timer`` device file descriptor. The ``snd_timer_uinfo`` +structure should be passed as an ioctl argument: + +:: + + struct snd_timer_uinfo { + __u64 resolution; + int fd; + unsigned int id; + unsigned char reserved[16]; + } + +The ``resolution`` field sets the desired resolution in nanoseconds for +the virtual timer. ``resolution`` field simply provides an information +about the virtual timer, but does not affect the timing itself. ``id`` +field gets overwritten by the ioctl, and the identifier you get in this +field after the call can be used as a timer subdevice number when +passing the timer to ``snd-aloop`` kernel module or other userspace +applications. There could be up to 128 userspace-driven timers in the +system at one moment of time, thus the id value ranges from 0 to 127. + +Besides from overwriting the ``snd_timer_uinfo`` struct, ioctl stores +a timer file descriptor, which can be used to trigger the timer, in the +``fd`` field of the ``snd_timer_uinfo`` struct. Allocation of a file +descriptor for the timer guarantees that the timer can only be triggered +by the process which created it. The timer then can be triggered with +``SNDRV_TIMER_IOCTL_TRIGGER`` ioctl call on the timer file descriptor. + +So, the example code for creating and triggering the timer would be: + +:: + + static struct snd_timer_uinfo utimer_info = { + /* Timer is going to tick (presumably) every 1000000 ns */ + .resolution = 1000000ULL, + .id = -1, + }; + + int timer_device_fd = open("/dev/snd/timer", O_RDWR | O_CLOEXEC); + + if (ioctl(timer_device_fd, SNDRV_TIMER_IOCTL_CREATE, &utimer_info)) { + perror("Failed to create the timer"); + return -1; + } + + ... + + /* + * Now we want to trigger the timer. Callbacks of all of the + * timer instances binded to this timer will be executed after + * this call. + */ + ioctl(utimer_info.fd, SNDRV_TIMER_IOCTL_TRIGGER, NULL); + + ... + + /* Now, destroy the timer */ + close(timer_info.fd); + + +More detailed example of creating and ticking the timer could be found +in the utimer ALSA selftest. + +Userspace-driven timers and snd-aloop +------------------------------------- + +Userspace-driven timers could be easily used with ``snd-aloop`` module +when synchronizing two sound applications on both ends of the virtual +sound loopback. For instance, if one of the applications receives sound +frames from network and sends them to snd-aloop pcm device, and another +application listens for frames on the other snd-aloop pcm device, it +makes sense that the ALSA middle layer should initiate a data +transaction when the new period of data is received through network, but +not when the certain amount of jiffies elapses. Userspace-driven ALSA +timers could be used to achieve this. + +To use userspace-driven ALSA timer as a timer source of snd-aloop, pass +the following string as the snd-aloop ``timer_source`` parameter: + +:: + + # modprobe snd-aloop timer_source="-1.4." + +Where ``utimer_id`` is the id of the timer you created with +``SNDRV_TIMER_IOCTL_CREATE``, and ``4`` is the number of +userspace-driven timers device (``SNDRV_TIMER_GLOBAL_UDRIVEN``). + +``resolution`` for the userspace-driven ALSA timer used with snd-aloop +should be calculated as ``1000000000ULL / frame_rate * period_size`` as +the timer is going to tick every time a new period of frames is ready. + +After that, each time you trigger the timer with +``SNDRV_TIMER_IOCTL_TRIGGER`` the new period of data will be transferred +from one snd-aloop device to another. From 37745918e0e7575bc40f38da93a99b9fa6406224 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Tue, 13 Aug 2024 13:07:00 +0100 Subject: [PATCH 208/410] ALSA: timer: Introduce virtual userspace-driven timers Implement two ioctl calls in order to support virtual userspace-driven ALSA timers. The first ioctl is SNDRV_TIMER_IOCTL_CREATE, which gets the snd_timer_uinfo struct as a parameter and puts a file descriptor of a virtual timer into the `fd` field of the snd_timer_unfo structure. It also updates the `id` field of the snd_timer_uinfo struct, which provides a unique identifier for the timer (basically, the subdevice number which can be used when creating timer instances). This patch also introduces a tiny id allocator for the userspace-driven timers, which guarantees that we don't have more than 128 of them in the system. Another ioctl is SNDRV_TIMER_IOCTL_TRIGGER, which allows us to trigger the virtual timer (and calls snd_timer_interrupt for the timer under the hood), causing all of the timer instances binded to this timer to execute their callbacks. The maximum amount of ticks available for the timer is 1 for the sake of simplicity of the userspace API. 'start', 'stop', 'open' and 'close' callbacks for the userspace-driven timers are empty since we don't really do any hardware initialization here. Suggested-by: Axel Holzinger Signed-off-by: Ivan Orlov Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240813120701.171743-4-ivan.orlov0322@gmail.com --- include/uapi/sound/asound.h | 17 ++- sound/core/Kconfig | 10 ++ sound/core/timer.c | 225 ++++++++++++++++++++++++++++++++++++ 3 files changed, 251 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 8bf7e8a0eb6f..4cd513215bcd 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -869,7 +869,7 @@ struct snd_ump_block_info { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 7) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 8) enum { SNDRV_TIMER_CLASS_NONE = -1, @@ -894,6 +894,7 @@ enum { #define SNDRV_TIMER_GLOBAL_RTC 1 /* unused */ #define SNDRV_TIMER_GLOBAL_HPET 2 #define SNDRV_TIMER_GLOBAL_HRTIMER 3 +#define SNDRV_TIMER_GLOBAL_UDRIVEN 4 /* info flags */ #define SNDRV_TIMER_FLG_SLAVE (1<<0) /* cannot be controlled */ @@ -974,6 +975,18 @@ struct snd_timer_status { }; #endif +/* + * This structure describes the userspace-driven timer. Such timers are purely virtual, + * and can only be triggered from software (for instance, by userspace application). + */ +struct snd_timer_uinfo { + /* To pretend being a normal timer, we need to know the resolution in ns. */ + __u64 resolution; + int fd; + unsigned int id; + unsigned char reserved[16]; +}; + #define SNDRV_TIMER_IOCTL_PVERSION _IOR('T', 0x00, int) #define SNDRV_TIMER_IOCTL_NEXT_DEVICE _IOWR('T', 0x01, struct snd_timer_id) #define SNDRV_TIMER_IOCTL_TREAD_OLD _IOW('T', 0x02, int) @@ -990,6 +1003,8 @@ struct snd_timer_status { #define SNDRV_TIMER_IOCTL_CONTINUE _IO('T', 0xa2) #define SNDRV_TIMER_IOCTL_PAUSE _IO('T', 0xa3) #define SNDRV_TIMER_IOCTL_TREAD64 _IOW('T', 0xa4, int) +#define SNDRV_TIMER_IOCTL_CREATE _IOWR('T', 0xa5, struct snd_timer_uinfo) +#define SNDRV_TIMER_IOCTL_TRIGGER _IO('T', 0xa6) #if __BITS_PER_LONG == 64 #define SNDRV_TIMER_IOCTL_TREAD SNDRV_TIMER_IOCTL_TREAD_OLD diff --git a/sound/core/Kconfig b/sound/core/Kconfig index f0ba87d4e504..2c5b9f964703 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -242,6 +242,16 @@ config SND_JACK_INJECTION_DEBUG Say Y if you are debugging via jack injection interface. If unsure select "N". +config SND_UTIMER + bool "Enable support for userspace-controlled virtual timers" + depends on SND_TIMER + help + Say Y to enable the support of userspace-controlled timers. These + timers are purely virtual, and they are supposed to be triggered + from userspace. They could be quite useful when synchronizing the + sound timing with userspace applications (for instance, when sending + data through snd-aloop). + config SND_VMASTER bool diff --git a/sound/core/timer.c b/sound/core/timer.c index 71a07c1662f5..7610f102360d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -13,6 +13,8 @@ #include #include #include +#include +#include #include #include #include @@ -109,6 +111,16 @@ struct snd_timer_status64 { unsigned char reserved[64]; /* reserved */ }; +#ifdef CONFIG_SND_UTIMER +#define SNDRV_UTIMERS_MAX_COUNT 128 +/* Internal data structure for keeping the state of the userspace-driven timer */ +struct snd_utimer { + char *name; + struct snd_timer *timer; + unsigned int id; +}; +#endif + #define SNDRV_TIMER_IOCTL_STATUS64 _IOR('T', 0x14, struct snd_timer_status64) /* list of timers */ @@ -2009,6 +2021,217 @@ enum { SNDRV_TIMER_IOCTL_PAUSE_OLD = _IO('T', 0x23), }; +#ifdef CONFIG_SND_UTIMER +/* + * Since userspace-driven timers are passed to userspace, we need to have an identifier + * which will allow us to use them (basically, the subdevice number of udriven timer). + */ +static DEFINE_IDA(snd_utimer_ids); + +static void snd_utimer_put_id(struct snd_utimer *utimer) +{ + int timer_id = utimer->id; + + snd_BUG_ON(timer_id < 0 || timer_id >= SNDRV_UTIMERS_MAX_COUNT); + ida_free(&snd_utimer_ids, timer_id); +} + +static int snd_utimer_take_id(void) +{ + return ida_alloc_max(&snd_utimer_ids, SNDRV_UTIMERS_MAX_COUNT - 1, GFP_KERNEL); +} + +static void snd_utimer_free(struct snd_utimer *utimer) +{ + snd_timer_free(utimer->timer); + snd_utimer_put_id(utimer); + kfree(utimer->name); + kfree(utimer); +} + +static int snd_utimer_release(struct inode *inode, struct file *file) +{ + struct snd_utimer *utimer = (struct snd_utimer *)file->private_data; + + snd_utimer_free(utimer); + return 0; +} + +static int snd_utimer_trigger(struct file *file) +{ + struct snd_utimer *utimer = (struct snd_utimer *)file->private_data; + + snd_timer_interrupt(utimer->timer, utimer->timer->sticks); + return 0; +} + +static long snd_utimer_ioctl(struct file *file, unsigned int ioctl, unsigned long arg) +{ + switch (ioctl) { + case SNDRV_TIMER_IOCTL_TRIGGER: + return snd_utimer_trigger(file); + } + + return -ENOTTY; +} + +static const struct file_operations snd_utimer_fops = { + .llseek = noop_llseek, + .release = snd_utimer_release, + .unlocked_ioctl = snd_utimer_ioctl, +}; + +static int snd_utimer_start(struct snd_timer *t) +{ + return 0; +} + +static int snd_utimer_stop(struct snd_timer *t) +{ + return 0; +} + +static int snd_utimer_open(struct snd_timer *t) +{ + return 0; +} + +static int snd_utimer_close(struct snd_timer *t) +{ + return 0; +} + +static const struct snd_timer_hardware timer_hw = { + .flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_WORK, + .open = snd_utimer_open, + .close = snd_utimer_close, + .start = snd_utimer_start, + .stop = snd_utimer_stop, +}; + +static int snd_utimer_create(struct snd_timer_uinfo *utimer_info, + struct snd_utimer **r_utimer) +{ + struct snd_utimer *utimer; + struct snd_timer *timer; + struct snd_timer_id tid; + int utimer_id; + int err = 0; + + if (!utimer_info || utimer_info->resolution == 0) + return -EINVAL; + + utimer = kzalloc(sizeof(*utimer), GFP_KERNEL); + if (!utimer) + return -ENOMEM; + + /* We hold the ioctl lock here so we won't get a race condition when allocating id */ + utimer_id = snd_utimer_take_id(); + if (utimer_id < 0) { + err = utimer_id; + goto err_take_id; + } + + utimer->name = kasprintf(GFP_KERNEL, "snd-utimer%d", utimer_id); + if (!utimer->name) { + err = -ENOMEM; + goto err_get_name; + } + + utimer->id = utimer_id; + + tid.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION; + tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL; + tid.card = -1; + tid.device = SNDRV_TIMER_GLOBAL_UDRIVEN; + tid.subdevice = utimer_id; + + err = snd_timer_new(NULL, utimer->name, &tid, &timer); + if (err < 0) { + pr_err("Can't create userspace-driven timer\n"); + goto err_timer_new; + } + + timer->module = THIS_MODULE; + timer->hw = timer_hw; + timer->hw.resolution = utimer_info->resolution; + timer->hw.ticks = 1; + timer->max_instances = MAX_SLAVE_INSTANCES; + + utimer->timer = timer; + + err = snd_timer_global_register(timer); + if (err < 0) { + pr_err("Can't register a userspace-driven timer\n"); + goto err_timer_reg; + } + + *r_utimer = utimer; + return 0; + +err_timer_reg: + snd_timer_free(timer); +err_timer_new: + kfree(utimer->name); +err_get_name: + snd_utimer_put_id(utimer); +err_take_id: + kfree(utimer); + + return err; +} + +static int snd_utimer_ioctl_create(struct file *file, + struct snd_timer_uinfo __user *_utimer_info) +{ + struct snd_utimer *utimer; + struct snd_timer_uinfo *utimer_info __free(kfree) = NULL; + int err, timer_fd; + + utimer_info = memdup_user(_utimer_info, sizeof(*utimer_info)); + if (IS_ERR(utimer_info)) + return PTR_ERR(no_free_ptr(utimer_info)); + + err = snd_utimer_create(utimer_info, &utimer); + if (err < 0) + return err; + + utimer_info->id = utimer->id; + + timer_fd = anon_inode_getfd(utimer->name, &snd_utimer_fops, utimer, O_RDWR | O_CLOEXEC); + if (timer_fd < 0) { + snd_utimer_free(utimer); + return timer_fd; + } + + utimer_info->fd = timer_fd; + + err = copy_to_user(_utimer_info, utimer_info, sizeof(*utimer_info)); + if (err) { + /* + * "Leak" the fd, as there is nothing we can do about it. + * It might have been closed already since anon_inode_getfd + * makes it available for userspace. + * + * We have to rely on the process exit path to do any + * necessary cleanup (e.g. releasing the file). + */ + return -EFAULT; + } + + return 0; +} + +#else + +static int snd_utimer_ioctl_create(struct file *file, + struct snd_timer_uinfo __user *_utimer_info) +{ + return -ENOTTY; +} + +#endif + static long __snd_timer_user_ioctl(struct file *file, unsigned int cmd, unsigned long arg, bool compat) { @@ -2053,6 +2276,8 @@ static long __snd_timer_user_ioctl(struct file *file, unsigned int cmd, case SNDRV_TIMER_IOCTL_PAUSE: case SNDRV_TIMER_IOCTL_PAUSE_OLD: return snd_timer_user_pause(file); + case SNDRV_TIMER_IOCTL_CREATE: + return snd_utimer_ioctl_create(file, argp); } return -ENOTTY; } From 1026392d10af7b35361c7ec4e14569d88c0c33e4 Mon Sep 17 00:00:00 2001 From: Ivan Orlov Date: Tue, 13 Aug 2024 13:07:01 +0100 Subject: [PATCH 209/410] selftests: ALSA: Cover userspace-driven timers with test Add a test for the new functionality of userspace-driven timers and the tool which allows us to count timer ticks in a certain time period. The test: 1. Creates a userspace-driven timer with ioctl to /dev/snd/timer 2. Starts the `global-timer` application to count the ticks of the timer from step 1. 3. Asynchronously triggers the timer multiple times with some interval 4. Compares the amount of caught ticks with the amount of trigger calls. Since we can't include and in one file due to overlapping declarations, I have to split the test into two applications: one of them counts the amount of timer ticks in the defined time period, and another one is the actual test which creates the timer, triggers it periodically and starts the first app to count the amount of ticks in a separate thread. Besides from testing the functionality itself, the test represents a sample application showing userspace-driven ALSA timers API. Also, the timer test includes a test case which tries to create a timer with invalid resolution (=0), and NULL as a timer info structure. Signed-off-by: Ivan Orlov Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240813120701.171743-5-ivan.orlov0322@gmail.com --- tools/testing/selftests/alsa/Makefile | 4 +- tools/testing/selftests/alsa/global-timer.c | 87 +++++++++++ tools/testing/selftests/alsa/utimer-test.c | 164 ++++++++++++++++++++ 3 files changed, 253 insertions(+), 2 deletions(-) create mode 100644 tools/testing/selftests/alsa/global-timer.c create mode 100644 tools/testing/selftests/alsa/utimer-test.c diff --git a/tools/testing/selftests/alsa/Makefile b/tools/testing/selftests/alsa/Makefile index c1ce39874e2b..25be68025290 100644 --- a/tools/testing/selftests/alsa/Makefile +++ b/tools/testing/selftests/alsa/Makefile @@ -12,9 +12,9 @@ LDLIBS+=-lpthread OVERRIDE_TARGETS = 1 -TEST_GEN_PROGS := mixer-test pcm-test test-pcmtest-driver +TEST_GEN_PROGS := mixer-test pcm-test test-pcmtest-driver utimer-test -TEST_GEN_PROGS_EXTENDED := libatest.so +TEST_GEN_PROGS_EXTENDED := libatest.so global-timer TEST_FILES := conf.d pcm-test.conf diff --git a/tools/testing/selftests/alsa/global-timer.c b/tools/testing/selftests/alsa/global-timer.c new file mode 100644 index 000000000000..c15ec0ba851a --- /dev/null +++ b/tools/testing/selftests/alsa/global-timer.c @@ -0,0 +1,87 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * This tool is used by the utimer test, and it allows us to + * count the ticks of a global timer in a certain time frame + * (which is set by `timeout` parameter). + * + * Author: Ivan Orlov + */ +#include +#include +#include +#include + +static int ticked; +static void async_callback(snd_async_handler_t *ahandler) +{ + ticked++; +} + +static char timer_name[64]; +static void bind_to_timer(int device, int subdevice, int timeout) +{ + snd_timer_t *handle; + snd_timer_params_t *params; + snd_async_handler_t *ahandler; + + time_t end; + + sprintf(timer_name, "hw:CLASS=%d,SCLASS=%d,DEV=%d,SUBDEV=%d", + SND_TIMER_CLASS_GLOBAL, SND_TIMER_SCLASS_NONE, + device, subdevice); + + snd_timer_params_alloca(¶ms); + + if (snd_timer_open(&handle, timer_name, SND_TIMER_OPEN_NONBLOCK) < 0) { + perror("Can't open the timer"); + exit(EXIT_FAILURE); + } + + snd_timer_params_set_auto_start(params, 1); + snd_timer_params_set_ticks(params, 1); + if (snd_timer_params(handle, params) < 0) { + perror("Can't set timer params"); + exit(EXIT_FAILURE); + } + + if (snd_async_add_timer_handler(&ahandler, handle, async_callback, NULL) < 0) { + perror("Can't create a handler"); + exit(EXIT_FAILURE); + } + end = time(NULL) + timeout; + if (snd_timer_start(handle) < 0) { + perror("Failed to start the timer"); + exit(EXIT_FAILURE); + } + printf("Timer has started\n"); + while (time(NULL) <= end) { + /* + * Waiting for the timeout to elapse. Can't use sleep here, as it gets + * constantly interrupted by the signal from the timer (SIGIO) + */ + } + snd_timer_stop(handle); + snd_timer_close(handle); +} + +int main(int argc, char *argv[]) +{ + int device, subdevice, timeout; + + if (argc < 4) { + perror("Usage: %s "); + return EXIT_FAILURE; + } + + setlinebuf(stdout); + + device = atoi(argv[1]); + subdevice = atoi(argv[2]); + timeout = atoi(argv[3]); + + bind_to_timer(device, subdevice, timeout); + + printf("Total ticks count: %d\n", ticked); + + return EXIT_SUCCESS; +} diff --git a/tools/testing/selftests/alsa/utimer-test.c b/tools/testing/selftests/alsa/utimer-test.c new file mode 100644 index 000000000000..32ee3ce57721 --- /dev/null +++ b/tools/testing/selftests/alsa/utimer-test.c @@ -0,0 +1,164 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * This test covers the functionality of userspace-driven ALSA timers. Such timers + * are purely virtual (so they don't directly depend on the hardware), and they could be + * created and triggered by userspace applications. + * + * Author: Ivan Orlov + */ +#include "../kselftest_harness.h" +#include +#include +#include +#include +#include +#include +#include +#include + +#define FRAME_RATE 8000 +#define PERIOD_SIZE 4410 +#define UTIMER_DEFAULT_ID -1 +#define UTIMER_DEFAULT_FD -1 +#define NANO 1000000000ULL +#define TICKS_COUNT 10 +#define TICKS_RECORDING_DELTA 5 +#define TIMER_OUTPUT_BUF_LEN 1024 +#define TIMER_FREQ_SEC 1 +#define RESULT_PREFIX_LEN strlen("Total ticks count: ") + +enum timer_app_event { + TIMER_APP_STARTED, + TIMER_APP_RESULT, + TIMER_NO_EVENT, +}; + +FIXTURE(timer_f) { + struct snd_timer_uinfo *utimer_info; +}; + +FIXTURE_SETUP(timer_f) { + int timer_dev_fd; + + if (geteuid()) + SKIP(return, "This test needs root to run!"); + + self->utimer_info = calloc(1, sizeof(*self->utimer_info)); + ASSERT_NE(NULL, self->utimer_info); + + /* Resolution is the time the period of frames takes in nanoseconds */ + self->utimer_info->resolution = (NANO / FRAME_RATE * PERIOD_SIZE); + + timer_dev_fd = open("/dev/snd/timer", O_RDONLY); + ASSERT_GE(timer_dev_fd, 0); + + ASSERT_EQ(ioctl(timer_dev_fd, SNDRV_TIMER_IOCTL_CREATE, self->utimer_info), 0); + ASSERT_GE(self->utimer_info->fd, 0); + + close(timer_dev_fd); +} + +FIXTURE_TEARDOWN(timer_f) { + close(self->utimer_info->fd); + free(self->utimer_info); +} + +static void *ticking_func(void *data) +{ + int i; + int *fd = (int *)data; + + for (i = 0; i < TICKS_COUNT; i++) { + /* Well, trigger the timer! */ + ioctl(*fd, SNDRV_TIMER_IOCTL_TRIGGER, NULL); + sleep(TIMER_FREQ_SEC); + } + + return NULL; +} + +static enum timer_app_event parse_timer_output(const char *s) +{ + if (strstr(s, "Timer has started")) + return TIMER_APP_STARTED; + if (strstr(s, "Total ticks count")) + return TIMER_APP_RESULT; + + return TIMER_NO_EVENT; +} + +static int parse_timer_result(const char *s) +{ + char *end; + long d; + + d = strtol(s + RESULT_PREFIX_LEN, &end, 10); + if (end == s + RESULT_PREFIX_LEN) + return -1; + + return d; +} + +/* + * This test triggers the timer and counts ticks at the same time. The amount + * of the timer trigger calls should be equal to the amount of ticks received. + */ +TEST_F(timer_f, utimer) { + char command[64]; + pthread_t ticking_thread; + int total_ticks = 0; + FILE *rfp; + char *buf = malloc(TIMER_OUTPUT_BUF_LEN); + + ASSERT_NE(buf, NULL); + + /* The timeout should be the ticks interval * count of ticks + some delta */ + sprintf(command, "./global-timer %d %d %d", SNDRV_TIMER_GLOBAL_UDRIVEN, + self->utimer_info->id, TICKS_COUNT * TIMER_FREQ_SEC + TICKS_RECORDING_DELTA); + + rfp = popen(command, "r"); + while (fgets(buf, TIMER_OUTPUT_BUF_LEN, rfp)) { + buf[TIMER_OUTPUT_BUF_LEN - 1] = 0; + switch (parse_timer_output(buf)) { + case TIMER_APP_STARTED: + /* global-timer waits for timer to trigger, so start the ticking thread */ + pthread_create(&ticking_thread, NULL, ticking_func, + &self->utimer_info->fd); + break; + case TIMER_APP_RESULT: + total_ticks = parse_timer_result(buf); + break; + case TIMER_NO_EVENT: + break; + } + } + pthread_join(ticking_thread, NULL); + ASSERT_EQ(total_ticks, TICKS_COUNT); + pclose(rfp); +} + +TEST(wrong_timers_test) { + int timer_dev_fd; + int utimer_fd; + size_t i; + struct snd_timer_uinfo wrong_timer = { + .resolution = 0, + .id = UTIMER_DEFAULT_ID, + .fd = UTIMER_DEFAULT_FD, + }; + + timer_dev_fd = open("/dev/snd/timer", O_RDONLY); + ASSERT_GE(timer_dev_fd, 0); + + utimer_fd = ioctl(timer_dev_fd, SNDRV_TIMER_IOCTL_CREATE, &wrong_timer); + ASSERT_LT(utimer_fd, 0); + /* Check that id was not updated */ + ASSERT_EQ(wrong_timer.id, UTIMER_DEFAULT_ID); + + /* Test the NULL as an argument is processed correctly */ + ASSERT_LT(ioctl(timer_dev_fd, SNDRV_TIMER_IOCTL_CREATE, NULL), 0); + + close(timer_dev_fd); +} + +TEST_HARNESS_MAIN From 3531df81dca2f42accbc11821c5fa28e4104efd5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 19 Aug 2024 10:47:49 +0200 Subject: [PATCH 210/410] ALSA: seq: Drop superfluous filter argument of get_event_dest_client() All callers of get_event_dest_clienter() pass 0 to the filter argument, and it means that the check there is utterly redundant. Drop the superfluous filter argument and its check as a code cleanup. Link: https://patch.msgid.link/20240819084757.11902-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 19 +++++++------------ 1 file changed, 7 insertions(+), 12 deletions(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 6be548baa6df..a9b455018592 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -70,7 +70,7 @@ static int bounce_error_event(struct snd_seq_client *client, int err, int atomic, int hop); static int snd_seq_deliver_single_event(struct snd_seq_client *client, struct snd_seq_event *event, - int filter, int atomic, int hop); + int atomic, int hop); #if IS_ENABLED(CONFIG_SND_SEQ_UMP) static void free_ump_info(struct snd_seq_client *client); @@ -525,10 +525,8 @@ static int check_port_perm(struct snd_seq_client_port *port, unsigned int flags) /* * check if the destination client is available, and return the pointer - * if filter is non-zero, client filter bitmap is tested. */ -static struct snd_seq_client *get_event_dest_client(struct snd_seq_event *event, - int filter) +static struct snd_seq_client *get_event_dest_client(struct snd_seq_event *event) { struct snd_seq_client *dest; @@ -543,8 +541,6 @@ static struct snd_seq_client *get_event_dest_client(struct snd_seq_event *event, if ((dest->filter & SNDRV_SEQ_FILTER_USE_EVENT) && ! test_bit(event->type, dest->event_filter)) goto __not_avail; - if (filter && !(dest->filter & filter)) - goto __not_avail; return dest; /* ok - accessible */ __not_avail: @@ -588,7 +584,7 @@ static int bounce_error_event(struct snd_seq_client *client, bounce_ev.data.quote.origin = event->dest; bounce_ev.data.quote.event = event; bounce_ev.data.quote.value = -err; /* use positive value */ - result = snd_seq_deliver_single_event(NULL, &bounce_ev, 0, atomic, hop + 1); + result = snd_seq_deliver_single_event(NULL, &bounce_ev, atomic, hop + 1); if (result < 0) { client->event_lost++; return result; @@ -655,7 +651,7 @@ int __snd_seq_deliver_single_event(struct snd_seq_client *dest, */ static int snd_seq_deliver_single_event(struct snd_seq_client *client, struct snd_seq_event *event, - int filter, int atomic, int hop) + int atomic, int hop) { struct snd_seq_client *dest = NULL; struct snd_seq_client_port *dest_port = NULL; @@ -664,7 +660,7 @@ static int snd_seq_deliver_single_event(struct snd_seq_client *client, direct = snd_seq_ev_is_direct(event); - dest = get_event_dest_client(event, filter); + dest = get_event_dest_client(event); if (dest == NULL) goto __skip; dest_port = snd_seq_port_use_ptr(dest, event->dest.port); @@ -744,8 +740,7 @@ static int __deliver_to_subscribers(struct snd_seq_client *client, /* convert time according to flag with subscription */ update_timestamp_of_queue(event, subs->info.queue, subs->info.flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL); - err = snd_seq_deliver_single_event(client, event, - 0, atomic, hop); + err = snd_seq_deliver_single_event(client, event, atomic, hop); if (err < 0) { /* save first error that occurs and continue */ if (!result) @@ -818,7 +813,7 @@ static int snd_seq_deliver_event(struct snd_seq_client *client, struct snd_seq_e event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS) result = deliver_to_subscribers(client, event, atomic, hop); else - result = snd_seq_deliver_single_event(client, event, 0, atomic, hop); + result = snd_seq_deliver_single_event(client, event, atomic, hop); return result; } From b27404b2bbf951a11500525dea15681cc970226c Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Aug 2024 08:55:46 +0800 Subject: [PATCH 211/410] ALSA/ASoC/SoundWire: Intel: use single definition for SDW_INTEL_MAX_LINKS MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The definitions are currently duplicated in intel-sdw-acpi.c and sof_sdw.c. Move the definition to the sdw_intel.h header, and change the prefix to make it Intel-specific. No functionality change in this patch. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Reviewed-by: Takashi Iwai Link: https://patch.msgid.link/20240819005548.5867-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/linux/soundwire/sdw_intel.h | 5 +++++ sound/hda/intel-sdw-acpi.c | 5 ++--- sound/soc/intel/boards/sof_sdw.c | 4 +++- sound/soc/intel/boards/sof_sdw_common.h | 4 +--- sound/soc/intel/boards/sof_sdw_hdmi.c | 2 ++ 5 files changed, 13 insertions(+), 7 deletions(-) diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index d537587b4499..87d82ea9a13a 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -447,4 +447,9 @@ extern const struct sdw_intel_hw_ops sdw_intel_lnl_hw_ops; #define SDW_INTEL_DEV_NUM_IDA_MIN 6 +/* + * Max number of links supported in hardware + */ +#define SDW_INTEL_MAX_LINKS 4 + #endif diff --git a/sound/hda/intel-sdw-acpi.c b/sound/hda/intel-sdw-acpi.c index f3b2a610df23..04d6b6beabca 100644 --- a/sound/hda/intel-sdw-acpi.c +++ b/sound/hda/intel-sdw-acpi.c @@ -17,7 +17,6 @@ #include #define SDW_LINK_TYPE 4 /* from Intel ACPI documentation */ -#define SDW_MAX_LINKS 4 static int ctrl_link_mask; module_param_named(sdw_link_mask, ctrl_link_mask, int, 0444); @@ -87,9 +86,9 @@ sdw_intel_scan_controller(struct sdw_intel_acpi_info *info) } /* Check count is within bounds */ - if (count > SDW_MAX_LINKS) { + if (count > SDW_INTEL_MAX_LINKS) { dev_err(&adev->dev, "Link count %d exceeds max %d\n", - count, SDW_MAX_LINKS); + count, SDW_INTEL_MAX_LINKS); return -EINVAL; } diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf..b1fb6fabd3b7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -5,12 +5,14 @@ * sof_sdw - ASOC Machine driver for Intel SoundWire platforms */ +#include #include #include #include #include #include #include +#include #include #include "sof_sdw_common.h" #include "../../codecs/rt711.h" @@ -883,7 +885,7 @@ static int create_sdw_dailinks(struct snd_soc_card *card, struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; int ret, i; - for (i = 0; i < SDW_MAX_LINKS; i++) + for (i = 0; i < SDW_INTEL_MAX_LINKS; i++) intel_ctx->sdw_pin_index[i] = SOC_SDW_INTEL_BIDIR_PDI_BASE; /* generate DAI links by each sdw link */ diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 02f3eebd019d..c10d2711b730 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -19,8 +19,6 @@ #define SOC_SDW_MAX_CPU_DAIS 16 #define SOC_SDW_INTEL_BIDIR_PDI_BASE 2 -#define SDW_MAX_LINKS 4 - /* 8 combinations with 4 links + unused group 0 */ #define SDW_MAX_GROUPS 9 @@ -57,7 +55,7 @@ enum { struct intel_mc_ctx { struct sof_hdmi_private hdmi; /* To store SDW Pin index for each SoundWire link */ - unsigned int sdw_pin_index[SDW_MAX_LINKS]; + unsigned int sdw_pin_index[SDW_INTEL_MAX_LINKS]; }; extern unsigned long sof_sdw_quirk; diff --git a/sound/soc/intel/boards/sof_sdw_hdmi.c b/sound/soc/intel/boards/sof_sdw_hdmi.c index 4084119a9a5f..f92867deb029 100644 --- a/sound/soc/intel/boards/sof_sdw_hdmi.c +++ b/sound/soc/intel/boards/sof_sdw_hdmi.c @@ -5,10 +5,12 @@ * sof_sdw_hdmi - Helpers to handle HDMI from generic machine driver */ +#include #include #include #include #include +#include #include #include #include From d2234596be2192d9c1ae34f8c7534531191bc433 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Aug 2024 08:55:47 +0800 Subject: [PATCH 212/410] soundwire: intel: add probe-time check on link id MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit In older platforms, the number of links was constant and hard-coded to 4. Newer platforms can have varying number of links, so we need to add a probe-time check to make sure the ACPI-reported information with _DSD properties is aligned with hardware capabilities reported in the SoundWire LCAP register. Acked-by: Vinod Koul Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Acked-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://patch.msgid.link/20240819005548.5867-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- drivers/soundwire/intel.h | 7 +++++++ drivers/soundwire/intel_ace2x.c | 20 ++++++++++++++++++++ drivers/soundwire/intel_auxdevice.c | 14 ++++++++++++++ include/linux/soundwire/sdw_intel.h | 3 +++ 4 files changed, 44 insertions(+) diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 68838e843b54..1db4d9d3a3ba 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -222,6 +222,13 @@ static inline bool sdw_intel_sync_check_cmdsync_unlocked(struct sdw_intel *sdw) return false; } +static inline int sdw_intel_get_link_count(struct sdw_intel *sdw) +{ + if (SDW_INTEL_CHECK_OPS(sdw, get_link_count)) + return SDW_INTEL_OPS(sdw, get_link_count)(sdw); + return 4; /* default on older generations */ +} + /* common bus management */ int intel_start_bus(struct sdw_intel *sdw); int intel_start_bus_after_reset(struct sdw_intel *sdw); diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index 781fe0aefa68..fff312c6968d 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -706,10 +706,30 @@ static void intel_program_sdi(struct sdw_intel *sdw, int dev_num) __func__, sdw->instance, dev_num); } +static int intel_get_link_count(struct sdw_intel *sdw) +{ + int ret; + + ret = hdac_bus_eml_get_count(sdw->link_res->hbus, true, AZX_REG_ML_LEPTR_ID_SDW); + if (!ret) { + dev_err(sdw->cdns.dev, "%s: could not retrieve link count\n", __func__); + return -ENODEV; + } + + if (ret > SDW_INTEL_MAX_LINKS) { + dev_err(sdw->cdns.dev, "%s: link count %d exceed max %d\n", __func__, ret, SDW_INTEL_MAX_LINKS); + return -EINVAL; + } + + return ret; +} + const struct sdw_intel_hw_ops sdw_intel_lnl_hw_ops = { .debugfs_init = intel_ace2x_debugfs_init, .debugfs_exit = intel_ace2x_debugfs_exit, + .get_link_count = intel_get_link_count, + .register_dai = intel_register_dai, .check_clock_stop = intel_check_clock_stop, diff --git a/drivers/soundwire/intel_auxdevice.c b/drivers/soundwire/intel_auxdevice.c index 8807e01cbf7c..d110f2b587d5 100644 --- a/drivers/soundwire/intel_auxdevice.c +++ b/drivers/soundwire/intel_auxdevice.c @@ -317,6 +317,20 @@ static int intel_link_probe(struct auxiliary_device *auxdev, bus->link_id = auxdev->id; bus->clk_stop_timeout = 1; + /* + * paranoia check: make sure ACPI-reported number of links is aligned with + * hardware capabilities. + */ + ret = sdw_intel_get_link_count(sdw); + if (ret < 0) { + dev_err(dev, "%s: sdw_intel_get_link_count failed: %d\n", __func__, ret); + return ret; + } + if (ret <= sdw->instance) { + dev_err(dev, "%s: invalid link id %d, link count %d\n", __func__, auxdev->id, ret); + return -EINVAL; + } + sdw_cdns_probe(cdns); /* Set ops */ diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index 87d82ea9a13a..cb8e7396b4db 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -388,6 +388,7 @@ struct sdw_intel; /* struct intel_sdw_hw_ops - SoundWire ops for Intel platforms. * @debugfs_init: initialize all debugfs capabilities * @debugfs_exit: close and cleanup debugfs capabilities + * @get_link_count: fetch link count from hardware registers * @register_dai: read all PDI information and register DAIs * @check_clock_stop: throw error message if clock is not stopped. * @start_bus: normal start @@ -412,6 +413,8 @@ struct sdw_intel_hw_ops { void (*debugfs_init)(struct sdw_intel *sdw); void (*debugfs_exit)(struct sdw_intel *sdw); + int (*get_link_count)(struct sdw_intel *sdw); + int (*register_dai)(struct sdw_intel *sdw); void (*check_clock_stop)(struct sdw_intel *sdw); From 1f3662838a05f1ab6af89a417f6f252d91d0806b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 19 Aug 2024 08:55:48 +0800 Subject: [PATCH 213/410] soundwire: intel: increase maximum number of links MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Intel platforms have enabled 4 links since the beginning, newer platforms now have 5 links. Update the definition accordingly. This patch will have no effect on older platforms where the number of links was hard-coded. A follow-up patch will add a dynamic check that the ACPI-reported information is aligned with hardware capabilities on newer platforms. Acked-by: Vinod Koul Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Acked-by: Mark Brown Reviewed-by: Takashi Iwai Link: https://patch.msgid.link/20240819005548.5867-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/linux/soundwire/sdw_intel.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h index cb8e7396b4db..37ae69365fe2 100644 --- a/include/linux/soundwire/sdw_intel.h +++ b/include/linux/soundwire/sdw_intel.h @@ -453,6 +453,6 @@ extern const struct sdw_intel_hw_ops sdw_intel_lnl_hw_ops; /* * Max number of links supported in hardware */ -#define SDW_INTEL_MAX_LINKS 4 +#define SDW_INTEL_MAX_LINKS 5 #endif From e486feb7b8ec04ec7cd53476acc9e18afd4a6a7d Mon Sep 17 00:00:00 2001 From: Frank Li Date: Wed, 14 Aug 2024 13:44:20 -0400 Subject: [PATCH 214/410] ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml Convert binding doc tlv320aic31xx.txt to yaml format. Additional change: - add i2c node in example. - replace MICBIAS_OFF with MICBIAS_2_0v in example because MICBIAS_OFF have been defined in header file. - add ref to dai-common.yaml. - add #sound-dai-cells. Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a30000/codec@18: failed to match any schema with compatible: ['ti,tlv320dac3100'] Signed-off-by: Frank Li Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240814174422.4026100-1-Frank.Li@nxp.com Signed-off-by: Mark Brown --- .../bindings/sound/ti,tlv320dac3100.yaml | 127 ++++++++++++++++++ .../bindings/sound/tlv320aic31xx.txt | 77 ----------- 2 files changed, 127 insertions(+), 77 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml new file mode 100644 index 000000000000..85e937e34962 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml @@ -0,0 +1,127 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320dac3100.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tlv320aic31xx Codec module + +maintainers: + - Shenghao Ding + +description: | + CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + + CODEC input pins: + * MIC1LP, devices with ADC + * MIC1RP, devices with ADC + * MIC1LM, devices with ADC + * AIN1, devices without ADC + * AIN2, devices without ADC + + The pins can be used in referring sound node's audio-routing property. + +properties: + compatible: + enum: + - ti,tlv320aic310x # - Generic TLV320AIC31xx with mono speaker amp + - ti,tlv320aic311x # - Generic TLV320AIC31xx with stereo speaker amp + - ti,tlv320aic3100 # - TLV320AIC3100 (mono speaker amp, no MiniDSP) + - ti,tlv320aic3110 # - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + - ti,tlv320aic3120 # - TLV320AIC3120 (mono speaker amp, MiniDSP) + - ti,tlv320aic3111 # - TLV320AIC3111 (stereo speaker amp, MiniDSP) + - ti,tlv320dac3100 # - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) + - ti,tlv320dac3101 # - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + HPVDD-supply: true + + SPRVDD-supply: true + + SPLVDD-supply: true + + AVDD-supply: true + + IOVDD-supply: true + + DVDD-supply: true + + reset-gpios: + description: GPIO specification for the active low RESET input. + + ai31xx-micbias-vg: + $ref: /schemas/types.yaml#/definitions/uint32 + default: 1 + enum: [1, 2, 3] + description: | + MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + + ai31xx-ocmv: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1, 2, 3] + description: | + output common-mode voltage setting + 0 - 1.35V, + 1 - 1.5V, + 2 - 1.65V, + 3 - 1.8V + + gpio-reset: + description: gpio pin number used for codec reset + deprecated: true + + +required: + - compatible + - reg + - HPVDD-supply + - SPRVDD-supply + - SPLVDD-supply + - AVDD-supply + - IOVDD-supply + - DVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include + #include + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + sound@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = ; + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt deleted file mode 100644 index bbad98d5b986..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ /dev/null @@ -1,77 +0,0 @@ -Texas Instruments - tlv320aic31xx Codec module - -The tlv320aic31xx serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp - "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp - "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) - "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) - "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) - "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - "ti,tlv320dac3100" - TLV320DAC3100 (no ADC, mono speaker amp, no MiniDSP) - "ti,tlv320dac3101" - TLV320DAC3101 (no ADC, stereo speaker amp, no MiniDSP) - -- reg - - I2C slave address -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai31xx-micbias-vg - MicBias Voltage setting - 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V - 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V - 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD - If this node is not mentioned or if the value is unknown, then - micbias is set to 2.0V. -- ai31xx-ocmv - output common-mode voltage setting - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * HPL - * HPR - * SPL, devices with stereo speaker amp - * SPR, devices with stereo speaker amp - * SPK, devices with mono speaker amp - * MICBIAS - -CODEC input pins: - * MIC1LP, devices with ADC - * MIC1RP, devices with ADC - * MIC1LM, devices with ADC - * AIN1, devices without ADC - * AIN2, devices without ADC - -The pins can be used in referring sound node's audio-routing property. - -Example: -#include -#include - -tlv320aic31xx: tlv320aic31xx@18 { - compatible = "ti,tlv320aic311x"; - reg = <0x18>; - - ai31xx-micbias-vg = ; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - HPVDD-supply = <®ulator>; - SPRVDD-supply = <®ulator>; - SPLVDD-supply = <®ulator>; - AVDD-supply = <®ulator>; - IOVDD-supply = <®ulator>; - DVDD-supply = <®ulator>; -}; From ec7bccd770b628a465458194aeac4408cdcc5ccd Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:16 +0200 Subject: [PATCH 215/410] ALSA: hda: Move SST device entries to AVS The avs-driver succeeds the skylake-driver. It suppots all configurations of its predecessor and more. Reflect that in the existing selection table. Signed-off-by: Cezary Rojewski Reviewed-by: Takashi Iwai Acked-by: Andy Shevchenko Link: https://patch.msgid.link/20240814083929.1217319-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/hda/intel-dsp-config.c | 111 ++++++++++++++++------------------- 1 file changed, 50 insertions(+), 61 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 913880b09065..f018bd779862 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -56,10 +56,59 @@ static const struct config_entry config_table[] = { }, #endif /* - * Apollolake (Broxton-P) + * Skylake, Kabylake, Apollolake * the legacy HDAudio driver is used except on Up Squared (SOF) and * Chromebooks (SST), as well as devices based on the ES8336 codec */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_AVS) + { + .flags = FLAG_SST, + .device = PCI_DEVICE_ID_INTEL_HDA_SKL_LP, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC, + .device = PCI_DEVICE_ID_INTEL_HDA_SKL_LP, + }, + { + .flags = FLAG_SST, + .device = PCI_DEVICE_ID_INTEL_HDA_KBL_LP, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, + { + .flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC, + .device = PCI_DEVICE_ID_INTEL_HDA_KBL_LP, + }, + { + .flags = FLAG_SST, + .device = PCI_DEVICE_ID_INTEL_HDA_APL, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) { .flags = FLAG_SOF, @@ -81,66 +130,6 @@ static const struct config_entry config_table[] = { .codec_hid = &essx_83x6, }, #endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) - { - .flags = FLAG_SST, - .device = PCI_DEVICE_ID_INTEL_HDA_APL, - .dmi_table = (const struct dmi_system_id []) { - { - .ident = "Google Chromebooks", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), - } - }, - {} - } - }, -#endif -/* - * Skylake and Kabylake use legacy HDAudio driver except for Google - * Chromebooks (SST) - */ - -/* Sunrise Point-LP */ -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKL) - { - .flags = FLAG_SST, - .device = PCI_DEVICE_ID_INTEL_HDA_SKL_LP, - .dmi_table = (const struct dmi_system_id []) { - { - .ident = "Google Chromebooks", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), - } - }, - {} - } - }, - { - .flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC, - .device = PCI_DEVICE_ID_INTEL_HDA_SKL_LP, - }, -#endif -/* Kabylake-LP */ -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL) - { - .flags = FLAG_SST, - .device = PCI_DEVICE_ID_INTEL_HDA_KBL_LP, - .dmi_table = (const struct dmi_system_id []) { - { - .ident = "Google Chromebooks", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "Google"), - } - }, - {} - } - }, - { - .flags = FLAG_SST | FLAG_SST_ONLY_IF_DMIC, - .device = PCI_DEVICE_ID_INTEL_HDA_KBL_LP, - }, -#endif /* * Geminilake uses legacy HDAudio driver except for Google From cd5c4dd97f35cd7cf772fc8c12738bd971cbb58e Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:17 +0200 Subject: [PATCH 216/410] ASoC: Intel: Drop skl_machine_pdata usage Preparation step in the skylake-driver removal process. Signed-off-by: Cezary Rojewski Acked-by: Andy Shevchenko Link: https://patch.msgid.link/20240814083929.1217319-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 6 ------ sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 1 - sound/soc/intel/common/soc-acpi-intel-hda-match.c | 6 ------ sound/soc/intel/common/soc-acpi-intel-icl-match.c | 6 ------ sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 11 ----------- sound/soc/intel/common/soc-acpi-intel-skl-match.c | 5 ----- 6 files changed, 35 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c index 3df89e4511da..8bbb1052faf2 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cnl-match.c @@ -8,7 +8,6 @@ #include #include -#include "../skylake/skl.h" #include "soc-acpi-intel-sdw-mockup-match.h" static const struct snd_soc_acpi_codecs essx_83x6 = { @@ -16,16 +15,11 @@ static const struct snd_soc_acpi_codecs essx_83x6 = { .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; -static struct skl_machine_pdata cnl_pdata = { - .use_tplg_pcm = true, -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_machines[] = { { .id = "INT34C2", .drv_name = "cnl_rt274", .fw_filename = "intel/dsp_fw_cnl.bin", - .pdata = &cnl_pdata, .sof_tplg_filename = "sof-cnl-rt274.tplg", }, { diff --git a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c index 84639c41a268..78255d56b08c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-ehl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-ehl-match.c @@ -8,7 +8,6 @@ #include #include -#include "../skylake/skl.h" struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[] = { { diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c index 2017fd0d676f..007ccd8a60e5 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hda-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c @@ -8,11 +8,6 @@ #include #include -#include "../skylake/skl.h" - -static struct skl_machine_pdata hda_pdata = { - .use_tplg_pcm = true, -}; struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { { @@ -28,7 +23,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { * can be used if we need a more complicated machine driver * combining HDA+other device (e.g. DMIC). */ - .pdata = &hda_pdata, }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c index 39875d67dcd1..6ce75fbb842e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c @@ -8,23 +8,17 @@ #include #include -#include "../skylake/skl.h" static const struct snd_soc_acpi_codecs essx_83x6 = { .num_codecs = 3, .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; -static struct skl_machine_pdata icl_pdata = { - .use_tplg_pcm = true, -}; - struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_machines[] = { { .id = "INT34C2", .drv_name = "icl_rt274", .fw_filename = "intel/dsp_fw_icl.bin", - .pdata = &icl_pdata, .sof_tplg_filename = "sof-icl-rt274.tplg", }, { diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index 4e817f559d38..d4c158d8441b 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -8,9 +8,6 @@ #include #include -#include "../skylake/skl.h" - -static struct skl_machine_pdata skl_dmic_data; static const struct snd_soc_acpi_codecs kbl_codecs = { .num_codecs = 1, @@ -54,7 +51,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data, }, { .id = "MX98357A", @@ -62,7 +58,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_codecs, - .pdata = &skl_dmic_data, }, { .id = "MX98927", @@ -70,7 +65,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_5663_5514_codecs, - .pdata = &skl_dmic_data, }, { .id = "MX98927", @@ -78,7 +72,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_poppy_codecs, - .pdata = &skl_dmic_data, }, { .id = "10EC5663", @@ -91,7 +84,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, - .pdata = &skl_dmic_data, }, { .id = "DLGS7219", @@ -99,7 +91,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98927_codecs, - .pdata = &skl_dmic_data }, { .id = "10EC5660", @@ -117,13 +108,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, - .pdata = &skl_dmic_data }, { .id = "MX98373", .drv_name = "kbl_max98373", .fw_filename = "intel/dsp_fw_kbl.bin", - .pdata = &skl_dmic_data }, {}, }; diff --git a/sound/soc/intel/common/soc-acpi-intel-skl-match.c b/sound/soc/intel/common/soc-acpi-intel-skl-match.c index 75302e956742..ee6463202918 100644 --- a/sound/soc/intel/common/soc-acpi-intel-skl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-skl-match.c @@ -8,9 +8,6 @@ #include #include -#include "../skylake/skl.h" - -static struct skl_machine_pdata skl_dmic_data; static const struct snd_soc_acpi_codecs skl_codecs = { .num_codecs = 1, @@ -29,7 +26,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { .fw_filename = "intel/dsp_fw_release.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data, }, { .id = "MX98357A", @@ -37,7 +33,6 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_skl_machines[] = { .fw_filename = "intel/dsp_fw_release.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &skl_codecs, - .pdata = &skl_dmic_data, }, {}, }; From 4d61ed7609d8b8fef71f78db6a8ac3221a46ea17 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:18 +0200 Subject: [PATCH 217/410] ASoC: Intel: Remove bxt_rt298 board driver The driver has no users. Succeeded by: - avs_rt298 (./intel/avs/boards/rt298.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-4-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/bxt_rt298.c | 670 ----------------------------- 3 files changed, 686 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_rt298.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f1faa5dd2a4f..490c8d5737fb 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -319,20 +319,6 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -config SND_SOC_INTEL_BXT_RT298_MACH - tristate "Broxton with RT298 I2S mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_RT298 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_HDA_DSP_COMMON - help - This adds support for ASoC machine driver for Broxton platforms - with RT286 I2S audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - endif ## SND_SOC_INTEL_APL if SND_SOC_SOF_APOLLOLAKE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index dc6fe110f279..9ba79715fb86 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -4,7 +4,6 @@ snd-soc-sst-bdw-rt5650-mach-y := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-y := bdw-rt5677.o snd-soc-bdw-rt286-y := bdw_rt286.o snd-soc-sst-bxt-da7219_max98357a-y := bxt_da7219_max98357a.o -snd-soc-sst-bxt-rt298-y := bxt_rt298.o snd-soc-sst-sof-pcm512x-y := sof_pcm512x.o snd-soc-sst-sof-wm8804-y := sof_wm8804.o snd-soc-sst-bytcr-rt5640-y := bytcr_rt5640.o @@ -52,7 +51,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MACH) += snd-soc-sof_da7219.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-hsw-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o -obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-bdw-rt286.o diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c deleted file mode 100644 index dce6a2086f2a..000000000000 --- a/sound/soc/intel/boards/bxt_rt298.c +++ /dev/null @@ -1,670 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Broxton-P I2S Machine Driver - * - * Copyright (C) 2014-2016, Intel Corporation - * - * Modified from: - * Intel Skylake I2S Machine driver - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/hdac_hdmi.h" -#include "../../codecs/rt298.h" -#include "hda_dsp_common.h" - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack broxton_headset; -static struct snd_soc_jack broxton_hdmi[3]; - -struct bxt_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct bxt_rt286_private { - struct list_head hdmi_pcm_list; - bool common_hdmi_codec_drv; -}; - -enum { - BXT_DPCM_AUDIO_PB = 0, - BXT_DPCM_AUDIO_CP, - BXT_DPCM_AUDIO_REF_CP, - BXT_DPCM_AUDIO_DMIC_CP, - BXT_DPCM_AUDIO_HDMI1_PB, - BXT_DPCM_AUDIO_HDMI2_PB, - BXT_DPCM_AUDIO_HDMI3_PB, -}; - -static struct snd_soc_jack_pin broxton_headset_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static const struct snd_kcontrol_new broxton_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Mic Jack"), -}; - -static const struct snd_soc_dapm_widget broxton_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC2", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), -}; - -static const struct snd_soc_dapm_route broxton_rt298_map[] = { - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack detect */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - - /* digital mics */ - {"DMIC1 Pin", NULL, "DMIC2"}, - {"DMic", NULL, "SoC DMIC"}, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI2", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "AIF1 Playback", NULL, "ssp5 Tx"}, - { "ssp5 Tx", NULL, "codec0_out"}, - { "ssp5 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp5 Rx" }, - { "ssp5 Rx", NULL, "AIF1 Capture" }, - - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "Capture" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, -}; - -static const struct snd_soc_dapm_route geminilake_rt298_map[] = { - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack detect */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - - /* digital mics */ - {"DMIC1 Pin", NULL, "DMIC2"}, - {"DMic", NULL, "SoC DMIC"}, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI2", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "AIF1 Playback", NULL, "ssp2 Tx"}, - { "ssp2 Tx", NULL, "codec0_out"}, - { "ssp2 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp2 Rx" }, - { "ssp2 Rx", NULL, "AIF1 Capture" }, - - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "Capture" }, - - { "dmic_voice", NULL, "DMIC16k Rx" }, - { "DMIC16k Rx", NULL, "Capture" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, -}; - -static int broxton_rt298_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static int broxton_rt298_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - int ret = 0; - - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &broxton_headset, - broxton_headset_pins, ARRAY_SIZE(broxton_headset_pins)); - - if (ret) - return ret; - - snd_soc_component_set_jack(component, &broxton_headset, NULL); - - snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - - return 0; -} - -static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct bxt_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = BXT_DPCM_AUDIO_HDMI1_PB + dai->id; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP5 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int broxton_rt298_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT298_SCLK_S_PLL, - 19200000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - return ret; -} - -static const struct snd_soc_ops broxton_rt298_ops = { - .hw_params = broxton_rt298_hw_params, -}; - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - chan->min = chan->max = 4; - - return 0; -} - -static const unsigned int channels_dmic[] = { - 1, 2, 3, 4, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static int broxton_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = 4; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_dmic_channels); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops broxton_dmic_ops = { - .startup = broxton_dmic_startup, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int bxt_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * on this platform for PCM device we support: - * 48Khz - * stereo - * 16-bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops broxton_rt286_fe_ops = { - .startup = bxt_fe_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp5_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP5 Pin"))); -SND_SOC_DAILINK_DEF(ssp5_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", - "rt298-aif1"))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); - -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", - "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(dmic16k, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:0e.0"))); - -/* broxton digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link broxton_rt298_dais[] = { - /* Front End DAI links */ - [BXT_DPCM_AUDIO_PB] = - { - .name = "Bxt Audio Port", - .stream_name = "Audio", - .nonatomic = 1, - .dynamic = 1, - .init = broxton_rt298_fe_init, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &broxton_rt286_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [BXT_DPCM_AUDIO_CP] = - { - .name = "Bxt Audio Capture Port", - .stream_name = "Audio Record", - .nonatomic = 1, - .dynamic = 1, - .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &broxton_rt286_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [BXT_DPCM_AUDIO_REF_CP] = - { - .name = "Bxt Audio Reference cap", - .stream_name = "refcap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [BXT_DPCM_AUDIO_DMIC_CP] = - { - .name = "Bxt Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &broxton_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI1_PB] = - { - .name = "Bxt HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI2_PB] = - { - .name = "Bxt HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI3_PB] = - { - .name = "Bxt HDMI Port3", - .stream_name = "Hdmi3", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - /* Back End DAI links */ - { - /* SSP5 - Codec */ - .name = "SSP5-Codec", - .id = 0, - .no_pcm = 1, - .init = broxton_rt298_codec_init, - .dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broxton_ssp5_fixup, - .ops = &broxton_rt298_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp5_pin, ssp5_codec, platform), - }, - { - .name = "dmic01", - .id = 1, - .be_hw_params_fixup = broxton_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "dmic16k", - .id = 2, - .be_hw_params_fixup = broxton_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int bxt_card_late_probe(struct snd_soc_card *card) -{ - struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(card); - struct bxt_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - if (ctx->common_hdmi_codec_drv) { - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, - head); - component = pcm->codec_dai->component; - return hda_dsp_hdmi_build_controls(card, component); - } - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &broxton_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &broxton_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - - -/* broxton audio machine driver for SPT + RT298S */ -static struct snd_soc_card broxton_rt298 = { - .name = "broxton-rt298", - .owner = THIS_MODULE, - .dai_link = broxton_rt298_dais, - .num_links = ARRAY_SIZE(broxton_rt298_dais), - .controls = broxton_controls, - .num_controls = ARRAY_SIZE(broxton_controls), - .dapm_widgets = broxton_widgets, - .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), - .dapm_routes = broxton_rt298_map, - .num_dapm_routes = ARRAY_SIZE(broxton_rt298_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = bxt_card_late_probe, - -}; - -static struct snd_soc_card geminilake_rt298 = { - .name = "geminilake-rt298", - .owner = THIS_MODULE, - .dai_link = broxton_rt298_dais, - .num_links = ARRAY_SIZE(broxton_rt298_dais), - .controls = broxton_controls, - .num_controls = ARRAY_SIZE(broxton_controls), - .dapm_widgets = broxton_widgets, - .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), - .dapm_routes = geminilake_rt298_map, - .num_dapm_routes = ARRAY_SIZE(geminilake_rt298_map), - .fully_routed = true, - .late_probe = bxt_card_late_probe, -}; - -static int broxton_audio_probe(struct platform_device *pdev) -{ - struct bxt_rt286_private *ctx; - struct snd_soc_card *card = - (struct snd_soc_card *)pdev->id_entry->driver_data; - struct snd_soc_acpi_mach *mach; - const char *platform_name; - int ret; - int i; - - for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) { - if (card->dai_link[i].codecs->name && - !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", - I2C_NAME_SIZE)) { - if (!strncmp(card->name, "broxton-rt298", - PLATFORM_NAME_SIZE)) { - card->dai_link[i].name = "SSP5-Codec"; - card->dai_link[i].cpus->dai_name = "SSP5 Pin"; - } else if (!strncmp(card->name, "geminilake-rt298", - PLATFORM_NAME_SIZE)) { - card->dai_link[i].name = "SSP2-Codec"; - card->dai_link[i].cpus->dai_name = "SSP2 Pin"; - } - } - } - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - card->dev = &pdev->dev; - snd_soc_card_set_drvdata(card, ctx); - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - platform_name = mach->mach_params.platform; - - ret = snd_soc_fixup_dai_links_platform_name(card, - platform_name); - if (ret) - return ret; - - ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; - - return devm_snd_soc_register_card(&pdev->dev, card); -} - -static const struct platform_device_id bxt_board_ids[] = { - { .name = "bxt_alc298s_i2s", .driver_data = - (unsigned long)&broxton_rt298 }, - { .name = "glk_alc298s_i2s", .driver_data = - (unsigned long)&geminilake_rt298 }, - {} -}; -MODULE_DEVICE_TABLE(platform, bxt_board_ids); - -static struct platform_driver broxton_audio = { - .probe = broxton_audio_probe, - .driver = { - .name = "bxt_alc298s_i2s", - .pm = &snd_soc_pm_ops, - }, - .id_table = bxt_board_ids, -}; -module_platform_driver(broxton_audio) - -/* Module information */ -MODULE_AUTHOR("Ramesh Babu "); -MODULE_AUTHOR("Senthilnathan Veppur "); -MODULE_DESCRIPTION("Intel SST Audio for Broxton"); -MODULE_LICENSE("GPL v2"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); From fa07502e01569b39265008bac783a1202a7560e5 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:19 +0200 Subject: [PATCH 218/410] ASoC: Intel: Remove bxt_da7219_max98357a board driver The driver has no users. Succeeded by: - avs_da7219 (./intel/avs/boards/da7219.c) - avs_max98357a (./intel/avs/boards/max98357a.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 16 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ------------------ 3 files changed, 738 deletions(-) delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 490c8d5737fb..b91e091d2348 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -305,22 +305,6 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON -if SND_SOC_INTEL_APL - -config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH - tristate "Broxton with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SND_HDA_CODEC_HDMI - select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC - help - This adds support for ASoC machine driver for Broxton-P platforms - with DA7219 + MAX98357A I2S audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - -endif ## SND_SOC_INTEL_APL - if SND_SOC_SOF_APOLLOLAKE config SND_SOC_INTEL_SOF_WM8804_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9ba79715fb86..778a4ea957be 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -3,7 +3,6 @@ snd-soc-hsw-rt5640-y := hsw_rt5640.o snd-soc-sst-bdw-rt5650-mach-y := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-y := bdw-rt5677.o snd-soc-bdw-rt286-y := bdw_rt286.o -snd-soc-sst-bxt-da7219_max98357a-y := bxt_da7219_max98357a.o snd-soc-sst-sof-pcm512x-y := sof_pcm512x.o snd-soc-sst-sof-wm8804-y := sof_wm8804.o snd-soc-sst-bytcr-rt5640-y := bytcr_rt5640.o @@ -50,7 +49,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SOF_ES8336_MACH) += snd-soc-sof_es8336.o obj-$(CONFIG_SND_SOC_INTEL_SOF_NAU8825_MACH) += snd-soc-sof_nau8825.o obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MACH) += snd-soc-sof_da7219.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-hsw-rt5640.o -obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SOF_PCM512x_MACH) += snd-soc-sst-sof-pcm512x.o obj-$(CONFIG_SND_SOC_INTEL_SOF_WM8804_MACH) += snd-soc-sst-sof-wm8804.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-bdw-rt286.o diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c deleted file mode 100644 index e1082bfe5ca9..000000000000 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ /dev/null @@ -1,720 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Broxton-P I2S Machine Driver - * - * Copyright (C) 2016, Intel Corporation - * - * Modified from: - * Intel Skylake I2S Machine driver - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/hdac_hdmi.h" -#include "../../codecs/da7219.h" -#include "../common/soc-intel-quirks.h" -#include "hda_dsp_common.h" - -#define BXT_DIALOG_CODEC_DAI "da7219-hifi" -#define BXT_MAXIM_CODEC_DAI "HiFi" -#define DUAL_CHANNEL 2 -#define QUAD_CHANNEL 4 - -static struct snd_soc_jack broxton_headset; -static struct snd_soc_jack broxton_hdmi[3]; - -struct bxt_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct bxt_card_private { - struct list_head hdmi_pcm_list; - bool common_hdmi_codec_drv; -}; - -enum { - BXT_DPCM_AUDIO_PB = 0, - BXT_DPCM_AUDIO_CP, - BXT_DPCM_AUDIO_HS_PB, - BXT_DPCM_AUDIO_REF_CP, - BXT_DPCM_AUDIO_DMIC_CP, - BXT_DPCM_AUDIO_HDMI1_PB, - BXT_DPCM_AUDIO_HDMI2_PB, - BXT_DPCM_AUDIO_HDMI3_PB, -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - int ret = 0; - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - - codec_dai = snd_soc_card_get_codec_dai(card, BXT_DIALOG_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop PLL: %d\n", ret); - } else if(SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304); - if (ret) - dev_err(card->dev, "failed to start PLL: %d\n", ret); - } - - return ret; -} - -static const struct snd_kcontrol_new broxton_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Line Out"), - SOC_DAPM_PIN_SWITCH("Spk"), -}; - -static const struct snd_soc_dapm_widget broxton_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_POST_PMD|SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_SPK("Spk", NULL), -}; - -static const struct snd_soc_dapm_route audio_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - {"Headphone Jack", NULL, "HPL"}, - {"Headphone Jack", NULL, "HPR"}, - - /* other jacks */ - {"MIC", NULL, "Headset Mic"}, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI2", NULL, "hif7-0 Output"}, - - {"hifi3", NULL, "iDisp3 Tx"}, - {"iDisp3 Tx", NULL, "iDisp3_out"}, - {"hifi2", NULL, "iDisp2 Tx"}, - {"iDisp2 Tx", NULL, "iDisp2_out"}, - {"hifi1", NULL, "iDisp1 Tx"}, - {"iDisp1 Tx", NULL, "iDisp1_out"}, - - /* DMIC */ - {"dmic01_hifi", NULL, "DMIC01 Rx"}, - {"DMIC01 Rx", NULL, "DMIC AIF"}, - - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, - { "Line Out", NULL, "Platform Clock" }, - - /* speaker */ - {"Spk", NULL, "Speaker"}, - - {"HiFi Playback", NULL, "ssp5 Tx"}, - {"ssp5 Tx", NULL, "codec0_out"}, - - {"Playback", NULL, "ssp1 Tx"}, - {"ssp1 Tx", NULL, "codec1_out"}, - - {"codec0_in", NULL, "ssp1 Rx"}, - {"ssp1 Rx", NULL, "Capture"}, -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Line Out", - .mask = SND_JACK_LINEOUT, - }, -}; - -static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = DUAL_CHANNEL; - - /* set SSP to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int broxton_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - int clk_freq; - - /* Configure sysclk for codec */ - clk_freq = 19200000; - - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, clk_freq, - SND_SOC_CLOCK_IN); - - if (ret) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &broxton_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(broxton_headset.jack, SND_JACK_BTN_3, - KEY_VOICECOMMAND); - - snd_soc_component_set_jack(component, &broxton_headset, NULL); - - snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - - return ret; -} - -static int broxton_hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct bxt_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct bxt_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = BXT_DPCM_AUDIO_HDMI1_PB + dai->id; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int broxton_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - DUAL_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static const unsigned int channels_quad[] = { - QUAD_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels_quad = { - .count = ARRAY_SIZE(channels_quad), - .list = channels_quad, - .mask = 0, -}; - -static int bxt_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops broxton_da7219_fe_ops = { - .startup = bxt_fe_startup, -}; - -static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - if (params_channels(params) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static int broxton_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels_quad); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops broxton_dmic_ops = { - .startup = broxton_dmic_startup, -}; - -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int broxton_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -}; - -static const struct snd_soc_ops broxton_refcap_ops = { - .startup = broxton_refcap_startup, -}; - -/* broxton digital audio interface glue - connects codec <--> CPU */ -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(system2, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin2"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - - /* Back End DAI */ -SND_SOC_DAILINK_DEF(ssp5_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP5 Pin"))); -SND_SOC_DAILINK_DEF(ssp5_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", - BXT_MAXIM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", - BXT_DIALOG_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); - -SND_SOC_DAILINK_DEF(dmic16k_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:0e.0"))); - -static struct snd_soc_dai_link broxton_dais[] = { - /* Front End DAI links */ - [BXT_DPCM_AUDIO_PB] = - { - .name = "Bxt Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = broxton_da7219_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &broxton_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [BXT_DPCM_AUDIO_CP] = - { - .name = "Bxt Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &broxton_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [BXT_DPCM_AUDIO_HS_PB] = { - .name = "Bxt Audio Headset Playback", - .stream_name = "Headset Playback", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &broxton_da7219_fe_ops, - SND_SOC_DAILINK_REG(system2, dummy, platform), - }, - [BXT_DPCM_AUDIO_REF_CP] = - { - .name = "Bxt Audio Reference cap", - .stream_name = "Refcap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &broxton_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [BXT_DPCM_AUDIO_DMIC_CP] = - { - .name = "Bxt Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &broxton_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI1_PB] = - { - .name = "Bxt HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI2_PB] = - { - .name = "Bxt HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [BXT_DPCM_AUDIO_HDMI3_PB] = - { - .name = "Bxt HDMI Port3", - .stream_name = "Hdmi3", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - /* Back End DAI links */ - { - /* SSP5 - Codec */ - .name = "SSP5-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broxton_ssp_fixup, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(ssp5_pin, ssp5_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = broxton_da7219_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = broxton_ssp_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .ignore_suspend = 1, - .be_hw_params_fixup = broxton_dmic_fixup, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = broxton_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, - { - .name = "dmic16k", - .id = 6, - .be_hw_params_fixup = broxton_dmic_fixup, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k_pin, dmic_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int bxt_card_late_probe(struct snd_soc_card *card) -{ - struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); - struct bxt_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - if (ctx->common_hdmi_codec_drv) { - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct bxt_hdmi_pcm, - head); - component = pcm->codec_dai->component; - return hda_dsp_hdmi_build_controls(card, component); - } - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &broxton_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &broxton_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* broxton audio machine driver for SPT + da7219 */ -static struct snd_soc_card broxton_audio_card = { - .name = "bxtda7219max", - .owner = THIS_MODULE, - .dai_link = broxton_dais, - .num_links = ARRAY_SIZE(broxton_dais), - .controls = broxton_controls, - .num_controls = ARRAY_SIZE(broxton_controls), - .dapm_widgets = broxton_widgets, - .num_dapm_widgets = ARRAY_SIZE(broxton_widgets), - .dapm_routes = audio_map, - .num_dapm_routes = ARRAY_SIZE(audio_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = bxt_card_late_probe, -}; - -static int broxton_audio_probe(struct platform_device *pdev) -{ - struct bxt_card_private *ctx; - struct snd_soc_acpi_mach *mach; - const char *platform_name; - int ret; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - broxton_audio_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&broxton_audio_card, ctx); - - /* override platform name, if required */ - mach = pdev->dev.platform_data; - platform_name = mach->mach_params.platform; - - ret = snd_soc_fixup_dai_links_platform_name(&broxton_audio_card, - platform_name); - if (ret) - return ret; - - ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; - - return devm_snd_soc_register_card(&pdev->dev, &broxton_audio_card); -} - -static const struct platform_device_id bxt_board_ids[] = { - { .name = "bxt_da7219_mx98357a" }, - { } -}; -MODULE_DEVICE_TABLE(platform, bxt_board_ids); - -static struct platform_driver broxton_audio = { - .probe = broxton_audio_probe, - .driver = { - .name = "bxt_da7219_max98357a", - .pm = &snd_soc_pm_ops, - }, - .id_table = bxt_board_ids, -}; -module_platform_driver(broxton_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); -MODULE_AUTHOR("Sathyanarayana Nujella "); -MODULE_AUTHOR("Rohit Ainapure "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_AUTHOR("Conrad Cooke "); -MODULE_AUTHOR("Naveen Manohar "); -MODULE_AUTHOR("Mac Chiang "); -MODULE_AUTHOR("Brent Lu "); -MODULE_LICENSE("GPL v2"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); From a08b5fde945ef8b427d2d29515a806fe571d7639 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:20 +0200 Subject: [PATCH 219/410] ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver The driver has no users. Succeeded by: - avs_rt5514 (./intel/avs/boards/rt5514.c) - avs_rt5663 (./intel/avs/boards/rt5663.c) - avs_max98927 (./intel/avs/boards/max98927.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 17 - sound/soc/intel/boards/Makefile | 2 - .../intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ------------------ 3 files changed, 888 deletions(-) delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index b91e091d2348..e82605e734a4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -338,23 +338,6 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_SKYLAKE_SSP_CLK - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 778a4ea957be..8270836b347e 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -24,7 +24,6 @@ snd-soc-sof_da7219-y := sof_da7219.o snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o snd-soc-kbl_rt5663_max98927-y := kbl_rt5663_max98927.o -snd-soc-kbl_rt5663_rt5514_max98927-y := kbl_rt5663_rt5514_max98927.o snd-soc-kbl_rt5660-y := kbl_rt5660.o snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o @@ -68,7 +67,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec. obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o -obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt5663_rt5514_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c deleted file mode 100644 index a32ce8f972f3..000000000000 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ /dev/null @@ -1,869 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Kabylake I2S Machine Driver with MAXIM98927 - * RT5514 and RT5663 Codecs - * - * Copyright (C) 2017, Intel Corporation - * - * Modified from: - * Intel Kabylake I2S Machine driver supporting MAXIM98927 and - * RT5663 codecs - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/rt5514.h" -#include "../../codecs/rt5663.h" -#include "../../codecs/hdac_hdmi.h" -#include -#include -#include - -#define KBL_REALTEK_CODEC_DAI "rt5663-aif" -#define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" -#define KBL_MAXIM_CODEC_DAI "max98927-aif1" -#define MAXIM_DEV0_NAME "i2c-MX98927:00" -#define MAXIM_DEV1_NAME "i2c-MX98927:01" -#define RT5514_DEV_NAME "i2c-10EC5514:00" -#define RT5663_DEV_NAME "i2c-10EC5663:00" -#define RT5514_AIF1_BCLK_FREQ (48000 * 8 * 16) -#define RT5514_AIF1_SYSCLK_FREQ 12288000 -#define NAME_SIZE 32 - -#define DMIC_CH(p) p->list[p->count-1] - - -static struct snd_soc_card kabylake_audio_card; -static const struct snd_pcm_hw_constraint_list *dmic_constraints; - -struct kbl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct kbl_codec_private { - struct snd_soc_jack kabylake_headset; - struct list_head hdmi_pcm_list; - struct snd_soc_jack kabylake_hdmi[2]; - struct clk *mclk; - struct clk *sclk; -}; - -enum { - KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, - KBL_DPCM_AUDIO_HS_PB, - KBL_DPCM_AUDIO_ECHO_REF_CP, - KBL_DPCM_AUDIO_DMIC_CP, - KBL_DPCM_AUDIO_RT5514_DSP, - KBL_DPCM_AUDIO_HDMI1_PB, - KBL_DPCM_AUDIO_HDMI2_PB, -}; - -static const struct snd_kcontrol_new kabylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - SOC_DAPM_PIN_SWITCH("DMIC"), -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); - int ret = 0; - - /* - * MCLK/SCLK need to be ON early for a successful synchronization of - * codec internal clock. And the clocks are turned off during - * POST_PMD after the stream is stopped. - */ - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); - if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); - return ret; - } - - ret = clk_prepare_enable(priv->mclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); - return ret; - } - - /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); - if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); - return ret; - } - - ret = clk_prepare_enable(priv->sclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); - } - break; - case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); - break; - default: - return 0; - } - - return 0; -} - -static const struct snd_soc_dapm_widget kabylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), - -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route kabylake_map[] = { - /* Headphones */ - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* speaker */ - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, - - /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, - { "IN1P", NULL, "Headset Mic" }, - { "IN1N", NULL, "Headset Mic" }, - - /* CODEC BE connections */ - { "Left HiFi Playback", NULL, "ssp0 Tx" }, - { "Right HiFi Playback", NULL, "ssp0 Tx" }, - { "ssp0 Tx", NULL, "spk_out" }, - - { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "hs_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "AIF Capture" }, - - { "codec1_in", NULL, "ssp0 Rx" }, - { "ssp0 Rx", NULL, "AIF1 Capture" }, - - /* IV feedback path */ - { "codec0_fb_in", NULL, "ssp0 Rx"}, - { "ssp0 Rx", NULL, "Left HiFi Capture" }, - { "ssp0 Rx", NULL, "Right HiFi Capture" }, - - /* DMIC */ - { "DMIC1L", NULL, "DMIC" }, - { "DMIC1R", NULL, "DMIC" }, - { "DMIC2L", NULL, "DMIC" }, - { "DMIC2R", NULL, "DMIC" }, - - { "hifi2", NULL, "iDisp2 Tx" }, - { "iDisp2 Tx", NULL, "iDisp2_out" }, - { "hifi1", NULL, "iDisp1 Tx" }, - { "iDisp1 Tx", NULL, "iDisp1_out" }, -}; - -static struct snd_soc_codec_conf max98927_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME), - .name_prefix = "Right", - }, - { - .dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME), - .name_prefix = "Left", - }, -}; - - -static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - int ret; - - dapm = snd_soc_component_get_dapm(component); - ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - if (ret) - dev_err(rtd->dev, "Ref Cap -Ignore suspend failed = %d\n", ret); - - return ret; -} - -static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack; - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(&kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - - snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); - - ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "DMIC"); - if (ret) - dev_err(rtd->dev, "DMIC - Ignore suspend failed = %d\n", ret); - - return ret; -} - -static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct kbl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = device; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); -} - -static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int kbl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops kabylake_rt5663_fe_ops = { - .startup = kbl_fe_startup, -}; - -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; - - /* - * The following loop will be called only for playback stream - * In this platform, there is only one playback device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { - rtd_dpcm = dpcm; - break; - } - - /* - * This following loop will be called only for capture stream - * In this platform, there is only one capture device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { - rtd_dpcm = dpcm; - break; - } - - if (!rtd_dpcm) - return -EINVAL; - - /* - * The above 2 loops are mutually exclusive based on the stream direction, - * thus rtd_dpcm variable will never be overwritten - */ - - /* - * The ADSP will convert the FE rate to 48k, stereo, 24 bit - */ - if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } else if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio DMIC cap")) { - if (params_channels(params) == 2 || - DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); - - return 0; -} - -static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->component, - RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, - RT5663_CLK_SEL_I2S1_ASRC); - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops kabylake_rt5663_ops = { - .hw_params = kabylake_rt5663_hw_params, -}; - -static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - int ret = 0, j; - - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, RT5514_DEV_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "set sysclk err: %d\n", ret); - return ret; - } - } - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "DEV0 TDM slot err:%d\n", ret); - return ret; - } - } - - if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "DEV1 TDM slot err:%d\n", ret); - return ret; - } - } - } - return ret; -} - -static const struct snd_soc_ops kabylake_ssp0_ops = { - .hw_params = kabylake_ssp0_hw_params, -}; - -static const unsigned int channels_dmic[] = { - 4, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static const unsigned int dmic_2ch[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { - .count = ARRAY_SIZE(dmic_2ch), - .list = dmic_2ch, - .mask = 0, -}; - -static int kabylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = DMIC_CH(dmic_constraints); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dmic_constraints); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops kabylake_dmic_ops = { - .startup = kabylake_dmic_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(system2, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin2"))); - -SND_SOC_DAILINK_DEF(echoref, - DAILINK_COMP_ARRAY(COMP_CPU("Echoref Pin"))); - -SND_SOC_DAILINK_DEF(spi_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("spi-PRP0001:00"))); -SND_SOC_DAILINK_DEF(spi_platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("spi-PRP0001:00"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY( - /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI), - /* dmic */ COMP_CODEC(RT5514_DEV_NAME, KBL_REALTEK_DMIC_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC(RT5663_DEV_NAME, KBL_REALTEK_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = kabylake_rt5663_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_HS_PB] = { - .name = "Kbl Audio Headset Playback", - .stream_name = "Headset Audio", - .dpcm_playback = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(system2, dummy, platform), - }, - [KBL_DPCM_AUDIO_ECHO_REF_CP] = { - .name = "Kbl Audio Echo Reference cap", - .stream_name = "Echoreference Capture", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - SND_SOC_DAILINK_REG(echoref, dummy, platform), - }, - [KBL_DPCM_AUDIO_RT5514_DSP] = { - .name = "rt5514 dsp", - .stream_name = "Wake on Voice", - SND_SOC_DAILINK_REG(spi_cpu, dummy, spi_platform), - }, - [KBL_DPCM_AUDIO_DMIC_CP] = { - .name = "Kbl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &kabylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - /* Back End DAI links */ - /* single Back end dai for both max speakers and dmic */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ops = &kabylake_ssp0_ops, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = kabylake_rt5663_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .ops = &kabylake_rt5663_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, -}; - -static int kabylake_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) -{ - struct snd_soc_component *component = dapm->component; - struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); - int ret = 0; - - if (!component || strcmp(component->name, RT5514_DEV_NAME)) - return 0; - - if (IS_ERR(priv->mclk)) - return 0; - - /* - * It's required to control mclk directly in the set_bias_level - * function for rt5514 codec or the recording function could - * break. - */ - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_ON) { - if (!__clk_is_enabled(priv->mclk)) - return 0; - dev_dbg(card->dev, "Disable mclk"); - clk_disable_unprepare(priv->mclk); - } else { - dev_dbg(card->dev, "Enable mclk"); - ret = clk_set_rate(priv->mclk, 24000000); - if (ret) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); - return ret; - } - - ret = clk_prepare_enable(priv->mclk); - if (ret) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", - ret); - - /* mclk is already enabled in FW */ - ret = 0; - } - } - break; - default: - break; - } - - return ret; -} - -static int kabylake_card_late_probe(struct snd_soc_card *card) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); - struct kbl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP,pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &ctx->kabylake_hdmi[i]); - - if (err) - return err; - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &ctx->kabylake_hdmi[i]); - if (err < 0) - return err; - i++; - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* - * kabylake audio machine driver for MAX98927 + RT5514 + RT5663 - */ -static struct snd_soc_card kabylake_audio_card = { - .name = "kbl-r5514-5663-max", - .owner = THIS_MODULE, - .dai_link = kabylake_dais, - .num_links = ARRAY_SIZE(kabylake_dais), - .set_bias_level = kabylake_set_bias_level, - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98927_codec_conf, - .num_configs = ARRAY_SIZE(max98927_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static int kabylake_audio_probe(struct platform_device *pdev) -{ - struct kbl_codec_private *ctx; - struct snd_soc_acpi_mach *mach; - int ret; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - kabylake_audio_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&kabylake_audio_card, ctx); - - mach = pdev->dev.platform_data; - if (mach) - dmic_constraints = mach->mach_params.dmic_num == 2 ? - &constraints_dmic_2ch : &constraints_dmic_channels; - - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } - - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } - - return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); -} - -static const struct platform_device_id kbl_board_ids[] = { - { .name = "kbl_r5514_5663_max" }, - { } -}; -MODULE_DEVICE_TABLE(platform, kbl_board_ids); - -static struct platform_driver kabylake_audio = { - .probe = kabylake_audio_probe, - .driver = { - .name = "kbl_r5514_5663_max", - .pm = &snd_soc_pm_ops, - }, - .id_table = kbl_board_ids, -}; - -module_platform_driver(kabylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-RT5663 RT5514 & MAX98927"); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); From 1af24289751253e58850ba572c584f7e6b1caa87 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:21 +0200 Subject: [PATCH 220/410] ASoC: Intel: Remove kbl_rt5663_max98927 board driver The driver has no users. Succeeded by: - avs_rt5663 (./intel/avs/boards/rt5663.c) - avs_max98927 (./intel/avs/boards/max98927.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-7-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 15 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ------------------ 3 files changed, 1090 deletions(-) delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index e82605e734a4..7d536c0633a0 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -323,21 +323,6 @@ endif ## SND_SOC_SOF_APOLLOLAKE if SND_SOC_INTEL_KBL -config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH - tristate "KBL with RT5663 and MAX98927 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_RT5663 - select SND_SOC_MAX98927 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_SKYLAKE_SSP_CLK - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 8270836b347e..ff8f627e108c 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -23,7 +23,6 @@ snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o -snd-soc-kbl_rt5663_max98927-y := kbl_rt5663_max98927.o snd-soc-kbl_rt5660-y := kbl_rt5660.o snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o @@ -66,7 +65,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o -obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH) += snd-soc-kbl_rt5663_max98927.o obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c deleted file mode 100644 index 9da89436a917..000000000000 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ /dev/null @@ -1,1073 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Kabylake I2S Machine Driver with MAXIM98927 - * and RT5663 Codecs - * - * Copyright (C) 2017, Intel Corporation - * - * Modified from: - * Intel Skylake I2S Machine driver - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/rt5663.h" -#include "../../codecs/hdac_hdmi.h" -#include -#include -#include - -#define KBL_REALTEK_CODEC_DAI "rt5663-aif" -#define KBL_MAXIM_CODEC_DAI "max98927-aif1" -#define DMIC_CH(p) p->list[p->count-1] -#define MAXIM_DEV0_NAME "i2c-MX98927:00" -#define MAXIM_DEV1_NAME "i2c-MX98927:01" - -static struct snd_soc_card *kabylake_audio_card; -static const struct snd_pcm_hw_constraint_list *dmic_constraints; -static struct snd_soc_jack skylake_hdmi[3]; - -struct kbl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct kbl_rt5663_private { - struct snd_soc_jack kabylake_headset; - struct list_head hdmi_pcm_list; - struct clk *mclk; - struct clk *sclk; -}; - -enum { - KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, - KBL_DPCM_AUDIO_HS_PB, - KBL_DPCM_AUDIO_ECHO_REF_CP, - KBL_DPCM_AUDIO_REF_CP, - KBL_DPCM_AUDIO_DMIC_CP, - KBL_DPCM_AUDIO_HDMI1_PB, - KBL_DPCM_AUDIO_HDMI2_PB, - KBL_DPCM_AUDIO_HDMI3_PB, -}; - -static const struct snd_kcontrol_new kabylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct kbl_rt5663_private *priv = snd_soc_card_get_drvdata(card); - int ret = 0; - - /* - * MCLK/SCLK need to be ON early for a successful synchronization of - * codec internal clock. And the clocks are turned off during - * POST_PMD after the stream is stopped. - */ - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - /* Enable MCLK */ - ret = clk_set_rate(priv->mclk, 24000000); - if (ret < 0) { - dev_err(card->dev, "Can't set rate for mclk, err: %d\n", - ret); - return ret; - } - - ret = clk_prepare_enable(priv->mclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); - return ret; - } - - /* Enable SCLK */ - ret = clk_set_rate(priv->sclk, 3072000); - if (ret < 0) { - dev_err(card->dev, "Can't set rate for sclk, err: %d\n", - ret); - clk_disable_unprepare(priv->mclk); - return ret; - } - - ret = clk_prepare_enable(priv->sclk); - if (ret < 0) { - dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); - clk_disable_unprepare(priv->mclk); - } - break; - case SND_SOC_DAPM_POST_PMD: - clk_disable_unprepare(priv->mclk); - clk_disable_unprepare(priv->sclk); - break; - default: - return 0; - } - - return 0; -} - -static const struct snd_soc_dapm_widget kabylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static const struct snd_soc_dapm_route kabylake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* speaker */ - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, - - /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, - { "IN1P", NULL, "Headset Mic" }, - { "IN1N", NULL, "Headset Mic" }, - { "DMic", NULL, "SoC DMIC" }, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI3", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "Left HiFi Playback", NULL, "ssp0 Tx" }, - { "Right HiFi Playback", NULL, "ssp0 Tx" }, - { "ssp0 Tx", NULL, "spk_out" }, - - { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "hs_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "AIF Capture" }, - - /* IV feedback path */ - { "codec0_fb_in", NULL, "ssp0 Rx"}, - { "ssp0 Rx", NULL, "Left HiFi Capture" }, - { "ssp0 Rx", NULL, "Right HiFi Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, -}; - -enum { - KBL_DPCM_AUDIO_5663_PB = 0, - KBL_DPCM_AUDIO_5663_CP, - KBL_DPCM_AUDIO_5663_HDMI1_PB, - KBL_DPCM_AUDIO_5663_HDMI2_PB, -}; - -static const struct snd_kcontrol_new kabylake_5663_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), -}; - -static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route kabylake_5663_map[] = { - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* other jacks */ - { "Headset Mic", NULL, "Platform Clock" }, - { "IN1P", NULL, "Headset Mic" }, - { "IN1N", NULL, "Headset Mic" }, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI3", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "AIF Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "AIF Capture" }, - - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, -}; - -static struct snd_soc_codec_conf max98927_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME), - .name_prefix = "Right", - }, - { - .dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME), - .name_prefix = "Left", - }, -}; - -static int kabylake_rt5663_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - if (ret) { - dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); - return ret; - } - - return ret; -} - -static int kabylake_rt5663_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack; - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, - &ctx->kabylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - - snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); - - return ret; -} - -static int kabylake_rt5663_max98927_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - ret = kabylake_rt5663_codec_init(rtd); - if (ret) - return ret; - - ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - if (ret) { - dev_err(rtd->dev, "SoC DMIC ignore suspend failed %d\n", ret); - return ret; - } - - return ret; -} - -static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) -{ - struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct kbl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = device; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); -} - -static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); -} - -static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); -} - -static int kabylake_5663_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_5663_HDMI1_PB); -} - -static int kabylake_5663_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_5663_HDMI2_PB); -} - -static unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int kbl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops kabylake_rt5663_fe_ops = { - .startup = kbl_fe_startup, -}; - -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; - - /* - * The following loop will be called only for playback stream - * In this platform, there is only one playback device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { - rtd_dpcm = dpcm; - break; - } - - /* - * This following loop will be called only for capture stream - * In this platform, there is only one capture device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { - rtd_dpcm = dpcm; - break; - } - - if (!rtd_dpcm) - return -EINVAL; - - /* - * The above 2 loops are mutually exclusive based on the stream direction, - * thus rtd_dpcm variable will never be overwritten - */ - - /* - * The ADSP will convert the FE rate to 48k, stereo, 24 bit - */ - if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); - - return 0; -} - -static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - /* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */ - rt5663_sel_asrc_clk_src(codec_dai->component, - RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER, - RT5663_CLK_SEL_I2S1_ASRC); - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops kabylake_rt5663_ops = { - .hw_params = kabylake_rt5663_hw_params, -}; - -static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - int ret = 0, j; - - for_each_rtd_codec_dais(rtd, j, codec_dai) { - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { - /* - * Use channel 4 and 5 for the first amp - */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - } - if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { - /* - * Use channel 6 and 7 for the second amp - */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); - if (ret < 0) { - dev_err(rtd->dev, "set TDM slot err:%d\n", ret); - return ret; - } - } - } - return ret; -} - -static const struct snd_soc_ops kabylake_ssp0_ops = { - .hw_params = kabylake_ssp0_hw_params, -}; - -static unsigned int channels_dmic[] = { - 2, 4, -}; - -static struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static const unsigned int dmic_2ch[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { - .count = ARRAY_SIZE(dmic_2ch), - .list = dmic_2ch, - .mask = 0, -}; - -static int kabylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = DMIC_CH(dmic_constraints); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dmic_constraints); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops kabylake_dmic_ops = { - .startup = kabylake_dmic_startup, -}; - -static unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int kabylake_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -} - -static const struct snd_soc_ops skylake_refcap_ops = { - .startup = kabylake_refcap_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(system2, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin2"))); - -SND_SOC_DAILINK_DEF(echoref, - DAILINK_COMP_ARRAY(COMP_CPU("Echoref Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY( - /* Left */ COMP_CODEC(MAXIM_DEV0_NAME, KBL_MAXIM_CODEC_DAI), - /* Right */ COMP_CODEC(MAXIM_DEV1_NAME, KBL_MAXIM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5663:00", - KBL_REALTEK_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic01_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = kabylake_rt5663_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_HS_PB] = { - .name = "Kbl Audio Headset Playback", - .stream_name = "Headset Audio", - .dpcm_playback = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(system2, dummy, platform), - }, - [KBL_DPCM_AUDIO_ECHO_REF_CP] = { - .name = "Kbl Audio Echo Reference cap", - .stream_name = "Echoreference Capture", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - SND_SOC_DAILINK_REG(echoref, dummy, platform), - }, - [KBL_DPCM_AUDIO_REF_CP] = { - .name = "Kbl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [KBL_DPCM_AUDIO_DMIC_CP] = { - .name = "Kbl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &kabylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Kbl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .dpcm_playback = 1, - .ops = &kabylake_ssp0_ops, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = kabylake_rt5663_max98927_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .ops = &kabylake_rt5663_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .be_hw_params_fixup = kabylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic01_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = kabylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -static struct snd_soc_dai_link kabylake_5663_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_5663_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_5663_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_rt5663_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_5663_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_5663_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 0, - .no_pcm = 1, - .init = kabylake_rt5663_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .ops = &kabylake_rt5663_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "iDisp1", - .id = 1, - .dpcm_playback = 1, - .init = kabylake_5663_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 2, - .init = kabylake_5663_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int kabylake_card_late_probe(struct snd_soc_card *card) -{ - struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(card); - struct kbl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* kabylake audio machine driver for SPT + RT5663 */ -static struct snd_soc_card kabylake_audio_card_rt5663_m98927 = { - .name = "kblrt5663max", - .owner = THIS_MODULE, - .dai_link = kabylake_dais, - .num_links = ARRAY_SIZE(kabylake_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98927_codec_conf, - .num_configs = ARRAY_SIZE(max98927_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -/* kabylake audio machine driver for RT5663 */ -static struct snd_soc_card kabylake_audio_card_rt5663 = { - .name = "kblrt5663", - .owner = THIS_MODULE, - .dai_link = kabylake_5663_dais, - .num_links = ARRAY_SIZE(kabylake_5663_dais), - .controls = kabylake_5663_controls, - .num_controls = ARRAY_SIZE(kabylake_5663_controls), - .dapm_widgets = kabylake_5663_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_5663_widgets), - .dapm_routes = kabylake_5663_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_5663_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static int kabylake_audio_probe(struct platform_device *pdev) -{ - struct kbl_rt5663_private *ctx; - struct snd_soc_acpi_mach *mach; - int ret; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - kabylake_audio_card = - (struct snd_soc_card *)pdev->id_entry->driver_data; - - kabylake_audio_card->dev = &pdev->dev; - snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - - mach = pdev->dev.platform_data; - if (mach) - dmic_constraints = mach->mach_params.dmic_num == 2 ? - &constraints_dmic_2ch : &constraints_dmic_channels; - - ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } - - ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } - - return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); -} - -static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_rt5663", - .driver_data = (kernel_ulong_t)&kabylake_audio_card_rt5663, - }, - { - .name = "kbl_rt5663_m98927", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_rt5663_m98927, - }, - { } -}; -MODULE_DEVICE_TABLE(platform, kbl_board_ids); - -static struct platform_driver kabylake_audio = { - .probe = kabylake_audio_probe, - .driver = { - .name = "kbl_rt5663_m98927", - .pm = &snd_soc_pm_ops, - }, - .id_table = kbl_board_ids, -}; - -module_platform_driver(kabylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-RT5663 & MAX98927 in I2S mode"); -MODULE_AUTHOR("Naveen M "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_LICENSE("GPL v2"); From 1a40ef882fee37006243ebf0b4848c7811672fe2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:22 +0200 Subject: [PATCH 221/410] ASoC: Intel: Remove kbl_rt5660 board driver The driver has no users. Succeeded by: - avs_rt5660 (./intel/avs/boards/rt5660.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 12 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/kbl_rt5660.c | 567 ---------------------------- 3 files changed, 581 deletions(-) delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7d536c0633a0..3125a802adfa 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -348,18 +348,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH Say Y if you have such a device. If unsure select "N". -config SND_SOC_INTEL_KBL_RT5660_MACH - tristate "KBL with RT5660 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on GPIOLIB || COMPILE_TEST - select SND_SOC_RT5660 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5660 I2S audio codec. - Say Y if you have such a device. - endif ## SND_SOC_INTEL_KBL if SND_SOC_SOF_GEMINILAKE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ff8f627e108c..1b49088f8a7a 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -23,7 +23,6 @@ snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o -snd-soc-kbl_rt5660-y := kbl_rt5660.o snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o @@ -65,7 +64,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o -obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5660_MACH) += snd-soc-kbl_rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c deleted file mode 100644 index 66885cb36f24..000000000000 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ /dev/null @@ -1,567 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2018-19 Canonical Corporation. - -/* - * Intel Kabylake I2S Machine Driver with RT5660 Codec - * - * Modified from: - * Intel Kabylake I2S Machine driver supporting MAXIM98357a and - * DA7219 codecs - * Also referred to: - * Intel Broadwell I2S Machine driver supporting RT5677 codec - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include - -#include "../../codecs/hdac_hdmi.h" -#include "../../codecs/rt5660.h" - -#define KBL_RT5660_CODEC_DAI "rt5660-aif1" -#define DUAL_CHANNEL 2 - -static struct snd_soc_card *kabylake_audio_card; -static struct snd_soc_jack skylake_hdmi[3]; -static struct snd_soc_jack lineout_jack; -static struct snd_soc_jack mic_jack; - -struct kbl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct kbl_codec_private { - struct gpio_desc *gpio_lo_mute; - struct list_head hdmi_pcm_list; -}; - -enum { - KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, - KBL_DPCM_AUDIO_HDMI1_PB, - KBL_DPCM_AUDIO_HDMI2_PB, - KBL_DPCM_AUDIO_HDMI3_PB, -}; - -#define GPIO_LINEOUT_MUTE_INDEX 0 -#define GPIO_LINEOUT_DET_INDEX 3 -#define GPIO_LINEIN_DET_INDEX 4 - -static const struct acpi_gpio_params lineout_mute_gpio = { GPIO_LINEOUT_MUTE_INDEX, 0, true }; -static const struct acpi_gpio_params lineout_det_gpio = { GPIO_LINEOUT_DET_INDEX, 0, false }; -static const struct acpi_gpio_params mic_det_gpio = { GPIO_LINEIN_DET_INDEX, 0, false }; - - -static const struct acpi_gpio_mapping acpi_rt5660_gpios[] = { - { "lineout-mute-gpios", &lineout_mute_gpio, 1 }, - { "lineout-det-gpios", &lineout_det_gpio, 1 }, - { "mic-det-gpios", &mic_det_gpio, 1 }, - { NULL }, -}; - -static struct snd_soc_jack_pin lineout_jack_pin = { - .pin = "Line Out", - .mask = SND_JACK_LINEOUT, -}; - -static struct snd_soc_jack_pin mic_jack_pin = { - .pin = "Line In", - .mask = SND_JACK_MICROPHONE, -}; - -static struct snd_soc_jack_gpio lineout_jack_gpio = { - .name = "lineout-det", - .report = SND_JACK_LINEOUT, - .debounce_time = 200, -}; - -static struct snd_soc_jack_gpio mic_jack_gpio = { - .name = "mic-det", - .report = SND_JACK_MICROPHONE, - .debounce_time = 200, -}; - -static int kabylake_5660_event_lineout(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct kbl_codec_private *priv = snd_soc_card_get_drvdata(dapm->card); - - gpiod_set_value_cansleep(priv->gpio_lo_mute, - !(SND_SOC_DAPM_EVENT_ON(event))); - - return 0; -} - -static const struct snd_kcontrol_new kabylake_rt5660_controls[] = { - SOC_DAPM_PIN_SWITCH("Line In"), - SOC_DAPM_PIN_SWITCH("Line Out"), -}; - -static const struct snd_soc_dapm_widget kabylake_rt5660_widgets[] = { - SND_SOC_DAPM_MIC("Line In", NULL), - SND_SOC_DAPM_LINE("Line Out", kabylake_5660_event_lineout), -}; - -static const struct snd_soc_dapm_route kabylake_rt5660_map[] = { - /* other jacks */ - {"IN1P", NULL, "Line In"}, - {"IN2P", NULL, "Line In"}, - {"Line Out", NULL, "LOUTR"}, - {"Line Out", NULL, "LOUTL"}, - - /* CODEC BE connections */ - { "AIF1 Playback", NULL, "ssp0 Tx"}, - { "ssp0 Tx", NULL, "codec0_out"}, - - { "codec0_in", NULL, "ssp0 Rx" }, - { "ssp0 Rx", NULL, "AIF1 Capture" }, - - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, -}; - -static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = DUAL_CHANNEL; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int kabylake_rt5660_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - - ret = devm_acpi_dev_add_driver_gpios(component->dev, acpi_rt5660_gpios); - if (ret) - dev_warn(component->dev, "Failed to add driver gpios\n"); - - /* Request rt5660 GPIO for lineout mute control, return if fails */ - ctx->gpio_lo_mute = gpiod_get(component->dev, "lineout-mute", - GPIOD_OUT_HIGH); - if (IS_ERR(ctx->gpio_lo_mute)) { - dev_err(component->dev, "Can't find GPIO_MUTE# gpio\n"); - return PTR_ERR(ctx->gpio_lo_mute); - } - - /* Create and initialize headphone jack, this jack is not mandatory, don't return if fails */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Lineout Jack", - SND_JACK_LINEOUT, &lineout_jack, - &lineout_jack_pin, 1); - if (ret) - dev_warn(component->dev, "Can't create Lineout jack\n"); - else { - lineout_jack_gpio.gpiod_dev = component->dev; - ret = snd_soc_jack_add_gpios(&lineout_jack, 1, - &lineout_jack_gpio); - if (ret) - dev_warn(component->dev, "Can't add Lineout jack gpio\n"); - } - - /* Create and initialize mic jack, this jack is not mandatory, don't return if fails */ - ret = snd_soc_card_jack_new_pins(rtd->card, "Mic Jack", - SND_JACK_MICROPHONE, &mic_jack, - &mic_jack_pin, 1); - if (ret) - dev_warn(component->dev, "Can't create mic jack\n"); - else { - mic_jack_gpio.gpiod_dev = component->dev; - ret = snd_soc_jack_add_gpios(&mic_jack, 1, &mic_jack_gpio); - if (ret) - dev_warn(component->dev, "Can't add mic jack gpio\n"); - } - - /* Here we enable some dapms in advance to reduce the pop noise for recording via line-in */ - snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); - snd_soc_dapm_force_enable_pin(dapm, "BST1"); - snd_soc_dapm_force_enable_pin(dapm, "BST2"); - - return 0; -} - -static void kabylake_rt5660_codec_exit(struct snd_soc_pcm_runtime *rtd) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - - /* - * The .exit() can be reached without going through the .init() - * so explicitly test if the gpiod is valid - */ - if (!IS_ERR_OR_NULL(ctx->gpio_lo_mute)) - gpiod_put(ctx->gpio_lo_mute); -} - -static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct kbl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = device; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); -} - -static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); -} - -static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); -} - -static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - RT5660_SCLK_S_PLL1, params_rate(params) * 512, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, 0, - RT5660_PLL1_S_BCLK, - params_rate(params) * 50, - params_rate(params) * 512); - if (ret < 0) - dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops kabylake_rt5660_ops = { - .hw_params = kabylake_rt5660_hw_params, -}; - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - DUAL_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int kbl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops kabylake_rt5660_fe_ops = { - .startup = kbl_fe_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC3277:00", KBL_RT5660_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* kabylake digital audio interface glue - connects rt5660 codec <--> CPU */ -static struct snd_soc_dai_link kabylake_rt5660_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_rt5660_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_rt5660_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Kbl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .init = kabylake_rt5660_codec_init, - .exit = kabylake_rt5660_codec_exit, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp0_fixup, - .ops = &kabylake_rt5660_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "iDisp1", - .id = 1, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 2, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 3, - .init = kabylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - - -#define NAME_SIZE 32 -static int kabylake_card_late_probe(struct snd_soc_card *card) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); - struct kbl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* kabylake audio machine driver for rt5660 */ -static struct snd_soc_card kabylake_audio_card_rt5660 = { - .name = "kblrt5660", - .owner = THIS_MODULE, - .dai_link = kabylake_rt5660_dais, - .num_links = ARRAY_SIZE(kabylake_rt5660_dais), - .controls = kabylake_rt5660_controls, - .num_controls = ARRAY_SIZE(kabylake_rt5660_controls), - .dapm_widgets = kabylake_rt5660_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_rt5660_widgets), - .dapm_routes = kabylake_rt5660_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_rt5660_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static int kabylake_audio_probe(struct platform_device *pdev) -{ - struct kbl_codec_private *ctx; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - kabylake_audio_card = - (struct snd_soc_card *)pdev->id_entry->driver_data; - - kabylake_audio_card->dev = &pdev->dev; - snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); -} - -static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_rt5660", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_rt5660, - }, - { } -}; -MODULE_DEVICE_TABLE(platform, kbl_board_ids); - -static struct platform_driver kabylake_audio = { - .probe = kabylake_audio_probe, - .driver = { - .name = "kbl_rt5660", - .pm = &snd_soc_pm_ops, - }, - .id_table = kbl_board_ids, -}; - -module_platform_driver(kabylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-RT5660 in I2S mode"); -MODULE_AUTHOR("Hui Wang "); -MODULE_LICENSE("GPL v2"); From 1daa8dce04619f39d4d8ee43ae2a0cec9ab31897 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:23 +0200 Subject: [PATCH 222/410] ASoC: Intel: Remove kbl_da7219_max98927 board driver The driver has no users. Succeeded by: - avs_da7219 (./intel/avs/boards/da7219.c) - avs_max98927 (./intel/avs/boards/max98927.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 15 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------------------ 3 files changed, 1192 deletions(-) delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3125a802adfa..a330ee665acf 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -333,21 +333,6 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH create an alsa sound card for DA7219 + MAX98357A I2S audio codec. Say Y if you have such a device. -config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH - tristate "KBL with DA7219 and MAX98927 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_DA7219 - select SND_SOC_MAX98927 - select SND_SOC_MAX98373_I2C - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for DA7219 + MAX98927 I2S audio codec. - Say Y if you have such a device. - If unsure select "N". - endif ## SND_SOC_INTEL_KBL if SND_SOC_SOF_GEMINILAKE diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 1b49088f8a7a..c79625d59696 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,7 +22,6 @@ snd-soc-sof_es8336-y := sof_es8336.o snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o -snd-soc-kbl_da7219_max98927-y := kbl_da7219_max98927.o snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o @@ -63,7 +62,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o -obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH) += snd-soc-kbl_da7219_max98927.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c deleted file mode 100644 index 02ed77a07e23..000000000000 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ /dev/null @@ -1,1175 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2018 Intel Corporation. - -/* - * Intel Kabylake I2S Machine Driver with MAX98927, MAX98373 & DA7219 Codecs - * - * Modified from: - * Intel Kabylake I2S Machine driver supporting MAX98927 and - * RT5663 codecs - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/da7219.h" -#include "../../codecs/hdac_hdmi.h" - -#define KBL_DIALOG_CODEC_DAI "da7219-hifi" -#define MAX98927_CODEC_DAI "max98927-aif1" -#define MAX98927_DEV0_NAME "i2c-MX98927:00" -#define MAX98927_DEV1_NAME "i2c-MX98927:01" - -#define MAX98373_CODEC_DAI "max98373-aif1" -#define MAX98373_DEV0_NAME "i2c-MX98373:00" -#define MAX98373_DEV1_NAME "i2c-MX98373:01" - - -#define DUAL_CHANNEL 2 -#define QUAD_CHANNEL 4 -#define NAME_SIZE 32 - -static struct snd_soc_card *kabylake_audio_card; -static struct snd_soc_jack kabylake_hdmi[3]; - -struct kbl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct kbl_codec_private { - struct snd_soc_jack kabylake_headset; - struct list_head hdmi_pcm_list; -}; - -enum { - KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_ECHO_REF_CP, - KBL_DPCM_AUDIO_REF_CP, - KBL_DPCM_AUDIO_DMIC_CP, - KBL_DPCM_AUDIO_HDMI1_PB, - KBL_DPCM_AUDIO_HDMI2_PB, - KBL_DPCM_AUDIO_HDMI3_PB, - KBL_DPCM_AUDIO_HS_PB, - KBL_DPCM_AUDIO_CP, -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(card->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop PLL: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, - 0, DA7219_PLL_FREQ_OUT_98304); - if (ret) - dev_err(card->dev, "failed to start PLL: %d\n", ret); - } - - return ret; -} - -static const struct snd_kcontrol_new kabylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - SOC_DAPM_PIN_SWITCH("Line Out"), -}; - -static const struct snd_soc_dapm_widget kabylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Line Out", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route kabylake_map[] = { - /* speaker */ - { "Left Spk", NULL, "Left BE_OUT" }, - { "Right Spk", NULL, "Right BE_OUT" }, - - /* other jacks */ - { "DMic", NULL, "SoC DMIC" }, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI3", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "Left HiFi Playback", NULL, "ssp0 Tx" }, - { "Right HiFi Playback", NULL, "ssp0 Tx" }, - { "ssp0 Tx", NULL, "spk_out" }, - - /* IV feedback path */ - { "codec0_fb_in", NULL, "ssp0 Rx"}, - { "ssp0 Rx", NULL, "Left HiFi Capture" }, - { "ssp0 Rx", NULL, "Right HiFi Capture" }, - - /* AEC capture path */ - { "echo_ref_out", NULL, "ssp0 Rx" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi1", NULL, "iDisp1 Tx" }, - { "iDisp1 Tx", NULL, "iDisp1_out" }, - { "hifi2", NULL, "iDisp2 Tx" }, - { "iDisp2 Tx", NULL, "iDisp2_out" }, - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, -}; - -static const struct snd_soc_dapm_route kabylake_ssp1_map[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - /* other jacks */ - { "MIC", NULL, "Headset Mic" }, - - /* CODEC BE connections */ - { "Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "hs_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "Capture" }, - - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, - { "Line Out", NULL, "Platform Clock" }, -}; - -static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *runtime = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - int ret, j; - - for_each_rtd_codec_dais(runtime, j, codec_dai) { - - if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); - if (ret < 0) { - dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); - return ret; - } - } - if (!strcmp(codec_dai->component->name, MAX98927_DEV1_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); - if (ret < 0) { - dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); - return ret; - } - } - if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0x30, 3, 8, 16); - if (ret < 0) { - dev_err(runtime->dev, - "DEV0 TDM slot err:%d\n", ret); - return ret; - } - } - if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { - ret = snd_soc_dai_set_tdm_slot(codec_dai, - 0xC0, 3, 8, 16); - if (ret < 0) { - dev_err(runtime->dev, - "DEV1 TDM slot err:%d\n", ret); - return ret; - } - } - } - - return 0; -} - -static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai; - int j, ret; - - for_each_rtd_codec_dais(rtd, j, codec_dai) { - const char *name = codec_dai->component->name; - struct snd_soc_component *component = codec_dai->component; - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); - char pin_name[20]; - - if (strcmp(name, MAX98927_DEV0_NAME) && - strcmp(name, MAX98927_DEV1_NAME) && - strcmp(name, MAX98373_DEV0_NAME) && - strcmp(name, MAX98373_DEV1_NAME)) - continue; - - snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", - codec_dai->component->name_prefix); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = snd_soc_dapm_enable_pin(dapm, pin_name); - if (ret) { - dev_err(rtd->dev, "failed to enable %s: %d\n", - pin_name, ret); - return ret; - } - snd_soc_dapm_sync(dapm); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - ret = snd_soc_dapm_disable_pin(dapm, pin_name); - if (ret) { - dev_err(rtd->dev, "failed to disable %s: %d\n", - pin_name, ret); - return ret; - } - snd_soc_dapm_sync(dapm); - break; - } - } - - return 0; -} - -static const struct snd_soc_ops kabylake_ssp0_ops = { - .hw_params = kabylake_ssp0_hw_params, - .trigger = kabylake_ssp0_trigger, -}; - -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL; - - /* - * The following loop will be called only for playback stream - * In this platform, there is only one playback device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) { - rtd_dpcm = dpcm; - break; - } - - /* - * This following loop will be called only for capture stream - * In this platform, there is only one capture device on every SSP - */ - for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) { - rtd_dpcm = dpcm; - break; - } - - if (!rtd_dpcm) - return -EINVAL; - - /* - * The above 2 loops are mutually exclusive based on the stream direction, - * thus rtd_dpcm variable will never be overwritten - */ - - /* - * The ADSP will convert the FE rate to 48k, stereo, 24 bit - */ - if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") || - !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) { - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - } - - /* - * The speaker on the SSP0 supports S16_LE and not S24_LE. - * thus changing the mask here - */ - if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec")) - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); - - return 0; -} - -static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_jack *jack; - struct snd_soc_card *card = rtd->card; - int ret; - - - ret = snd_soc_dapm_add_routes(&card->dapm, - kabylake_ssp1_map, - ARRAY_SIZE(kabylake_ssp1_map)); - - if (ret) - return ret; - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - jack = &ctx->kabylake_headset; - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); - - snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); - - return 0; -} - -static int kabylake_dmic_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - if (ret) - dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); - - return ret; -} - -static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct kbl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = device; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); -} - -static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); -} - -static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); -} - -static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - DUAL_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static unsigned int channels_quad[] = { - QUAD_CHANNEL, -}; - -static struct snd_pcm_hw_constraint_list constraints_channels_quad = { - .count = ARRAY_SIZE(channels_quad), - .list = channels_quad, - .mask = 0, -}; - -static int kbl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops kabylake_da7219_fe_ops = { - .startup = kbl_fe_startup, -}; - -static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* - * set BE channel constraint as user FE channels - */ - - if (params_channels(params) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static int kabylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels_quad); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops kabylake_dmic_ops = { - .startup = kabylake_dmic_startup, -}; - -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int kabylake_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -} - - -static const struct snd_soc_ops skylake_refcap_ops = { - .startup = kabylake_refcap_startup, -}; - -static struct snd_soc_codec_conf max98927_codec_conf[] = { - - { - .dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME), - .name_prefix = "Right", - }, - - { - .dlc = COMP_CODEC_CONF(MAX98927_DEV1_NAME), - .name_prefix = "Left", - }, -}; - -static struct snd_soc_codec_conf max98373_codec_conf[] = { - - { - .dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME), - .name_prefix = "Right", - }, - - { - .dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME), - .name_prefix = "Left", - }, -}; - -static struct snd_soc_dai_link_component max98373_ssp0_codec_components[] = { - { /* Left */ - .name = MAX98373_DEV0_NAME, - .dai_name = MAX98373_CODEC_DAI, - }, - - { /* For Right */ - .name = MAX98373_DEV1_NAME, - .dai_name = MAX98373_CODEC_DAI, - }, - -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(echoref, - DAILINK_COMP_ARRAY(COMP_CPU("Echoref Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(system2, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin2"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY( - /* Left */ COMP_CODEC(MAX98927_DEV0_NAME, MAX98927_CODEC_DAI), - /* For Right */ COMP_CODEC(MAX98927_DEV1_NAME, MAX98927_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", - KBL_DIALOG_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = kabylake_da7219_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_ECHO_REF_CP] = { - .name = "Kbl Audio Echo Reference cap", - .stream_name = "Echoreference Capture", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - SND_SOC_DAILINK_REG(echoref, dummy, platform), - }, - [KBL_DPCM_AUDIO_REF_CP] = { - .name = "Kbl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [KBL_DPCM_AUDIO_DMIC_CP] = { - .name = "Kbl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &kabylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Kbl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - [KBL_DPCM_AUDIO_HS_PB] = { - .name = "Kbl Audio Headset Playback", - .stream_name = "Headset Audio", - .dpcm_playback = 1, - .nonatomic = 1, - .dynamic = 1, - .init = kabylake_da7219_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system2, dummy, platform), - }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .ops = &kabylake_ssp0_ops, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = kabylake_da7219_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .init = kabylake_dmic_init, - .be_hw_params_fixup = kabylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = kabylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -/* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = kabylake_da7219_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_ECHO_REF_CP] = { - .name = "Kbl Audio Echo Reference cap", - .stream_name = "Echoreference Capture", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - SND_SOC_DAILINK_REG(echoref, dummy, platform), - }, - [KBL_DPCM_AUDIO_REF_CP] = { - .name = "Kbl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [KBL_DPCM_AUDIO_DMIC_CP] = { - .name = "Kbl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &kabylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Kbl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_DSP_B | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .dpcm_playback = 1, - .dpcm_capture = 1, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .ops = &kabylake_ssp0_ops, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec), - }, - { - .name = "dmic01", - .id = 1, - .init = kabylake_dmic_init, - .be_hw_params_fixup = kabylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 2, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 3, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 4, - .init = kabylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -static int kabylake_card_late_probe(struct snd_soc_card *card) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); - struct kbl_hdmi_pcm *pcm; - struct snd_soc_dapm_context *dapm = &card->dapm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &kabylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &kabylake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - if (!component) - return -EINVAL; - - - err = hdac_hdmi_jack_port_init(component, &card->dapm); - - if (err < 0) - return err; - - err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); - if (err) { - dev_err(card->dev, "failed to disable Left Spk: %d\n", err); - return err; - } - - err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); - if (err) { - dev_err(card->dev, "failed to disable Right Spk: %d\n", err); - return err; - } - - return snd_soc_dapm_sync(dapm); -} - -/* kabylake audio machine driver for SPT + DA7219 */ -static struct snd_soc_card kbl_audio_card_da7219_m98927 = { - .name = "kblda7219m98927", - .owner = THIS_MODULE, - .dai_link = kabylake_dais, - .num_links = ARRAY_SIZE(kabylake_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98927_codec_conf, - .num_configs = ARRAY_SIZE(max98927_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -/* kabylake audio machine driver for Maxim98927 */ -static struct snd_soc_card kbl_audio_card_max98927 = { - .name = "kblmax98927", - .owner = THIS_MODULE, - .dai_link = kabylake_max98_927_373_dais, - .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98927_codec_conf, - .num_configs = ARRAY_SIZE(max98927_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static struct snd_soc_card kbl_audio_card_da7219_m98373 = { - .name = "kblda7219m98373", - .owner = THIS_MODULE, - .dai_link = kabylake_dais, - .num_links = ARRAY_SIZE(kabylake_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98373_codec_conf, - .num_configs = ARRAY_SIZE(max98373_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static struct snd_soc_card kbl_audio_card_max98373 = { - .name = "kblmax98373", - .owner = THIS_MODULE, - .dai_link = kabylake_max98_927_373_dais, - .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .codec_conf = max98373_codec_conf, - .num_configs = ARRAY_SIZE(max98373_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static int kabylake_audio_probe(struct platform_device *pdev) -{ - struct kbl_codec_private *ctx; - struct snd_soc_dai_link *kbl_dai_link; - struct snd_soc_dai_link_component **codecs; - int i; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - kabylake_audio_card = - (struct snd_soc_card *)pdev->id_entry->driver_data; - - kbl_dai_link = kabylake_audio_card->dai_link; - - /* Update codecs for SSP0 with max98373 codec info */ - if (!strcmp(pdev->name, "kbl_da7219_max98373") || - (!strcmp(pdev->name, "kbl_max98373"))) { - for (i = 0; i < kabylake_audio_card->num_links; ++i) { - if (strcmp(kbl_dai_link[i].name, "SSP0-Codec")) - continue; - - codecs = &(kbl_dai_link[i].codecs); - *codecs = max98373_ssp0_codec_components; - kbl_dai_link[i].num_codecs = - ARRAY_SIZE(max98373_ssp0_codec_components); - break; - } - } - kabylake_audio_card->dev = &pdev->dev; - snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - - return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); -} - -static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_da7219_max98927", - .driver_data = - (kernel_ulong_t)&kbl_audio_card_da7219_m98927, - }, - { - .name = "kbl_max98927", - .driver_data = - (kernel_ulong_t)&kbl_audio_card_max98927, - }, - { - .name = "kbl_da7219_max98373", - .driver_data = - (kernel_ulong_t)&kbl_audio_card_da7219_m98373, - }, - { - .name = "kbl_max98373", - .driver_data = - (kernel_ulong_t)&kbl_audio_card_max98373, - }, - { } -}; -MODULE_DEVICE_TABLE(platform, kbl_board_ids); - -static struct platform_driver kabylake_audio = { - .probe = kabylake_audio_probe, - .driver = { - .name = "kbl_da7219_max98_927_373", - .pm = &snd_soc_pm_ops, - }, - .id_table = kbl_board_ids, -}; - -module_platform_driver(kabylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927/MAX98373 & DA7219"); -MODULE_AUTHOR("Mac Chiang "); -MODULE_LICENSE("GPL v2"); From 15d6966580f3e40fe2f4ecfcde2edd69cc5508e9 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:24 +0200 Subject: [PATCH 223/410] ASoC: Intel: Remove kbl_da7219_max98357a board driver The driver has no users. Succeeded by: - avs_da7219 (./intel/avs/boards/da7219.c) - avs_max98357a (./intel/avs/boards/max98357a.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ------------------ 3 files changed, 704 deletions(-) delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index a330ee665acf..33f169bc1db0 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -321,20 +321,6 @@ config SND_SOC_INTEL_SOF_WM8804_MACH endif ## SND_SOC_SOF_APOLLOLAKE -if SND_SOC_INTEL_KBL - -config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH - tristate "KBL with DA7219 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_INTEL_DA7219_MAX98357A_GENERIC - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for DA7219 + MAX98357A I2S audio codec. - Say Y if you have such a device. - -endif ## SND_SOC_INTEL_KBL - if SND_SOC_SOF_GEMINILAKE config SND_SOC_INTEL_GLK_DA7219_MAX98357A_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index c79625d59696..4019ddb5588c 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,7 +21,6 @@ snd-soc-sof_cs42l42-y := sof_cs42l42.o snd-soc-sof_es8336-y := sof_es8336.o snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o -snd-soc-kbl_da7219_max98357a-y := kbl_da7219_max98357a.o snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o @@ -61,7 +60,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_CX2072X_MACH) += snd-soc-sst-byt-cht-cx2072x. obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o -obj-$(CONFIG_SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH) += snd-soc-kbl_da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c deleted file mode 100644 index 154f6a74ed15..000000000000 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ /dev/null @@ -1,688 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2017-18 Intel Corporation. - -/* - * Intel Kabylake I2S Machine Driver with MAX98357A & DA7219 Codecs - * - * Modified from: - * Intel Kabylake I2S Machine driver supporting MAXIM98927 and - * RT5663 codecs - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/da7219.h" -#include "../../codecs/hdac_hdmi.h" - -#define KBL_DIALOG_CODEC_DAI "da7219-hifi" -#define KBL_MAXIM_CODEC_DAI "HiFi" -#define MAXIM_DEV0_NAME "MX98357A:00" -#define DUAL_CHANNEL 2 -#define QUAD_CHANNEL 4 - -static struct snd_soc_card *kabylake_audio_card; -static struct snd_soc_jack skylake_hdmi[3]; - -struct kbl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct kbl_codec_private { - struct snd_soc_jack kabylake_headset; - struct list_head hdmi_pcm_list; -}; - -enum { - KBL_DPCM_AUDIO_PB = 0, - KBL_DPCM_AUDIO_CP, - KBL_DPCM_AUDIO_REF_CP, - KBL_DPCM_AUDIO_DMIC_CP, - KBL_DPCM_AUDIO_HDMI1_PB, - KBL_DPCM_AUDIO_HDMI2_PB, - KBL_DPCM_AUDIO_HDMI3_PB, -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret = 0; - - codec_dai = snd_soc_card_get_codec_dai(card, KBL_DIALOG_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_OFF(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, - DA7219_SYSCLK_MCLK, 0, 0); - if (ret) - dev_err(card->dev, "failed to stop PLL: %d\n", ret); - } else if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM, - 0, DA7219_PLL_FREQ_OUT_98304); - if (ret) - dev_err(card->dev, "failed to start PLL: %d\n", ret); - } - - return ret; -} - -static const struct snd_kcontrol_new kabylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Line Out"), -}; - -static const struct snd_soc_dapm_widget kabylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Line Out", - .mask = SND_JACK_LINEOUT, - }, -}; - -static const struct snd_soc_dapm_route kabylake_map[] = { - { "Headphone Jack", NULL, "HPL" }, - { "Headphone Jack", NULL, "HPR" }, - - /* speaker */ - { "Spk", NULL, "Speaker" }, - - /* other jacks */ - { "MIC", NULL, "Headset Mic" }, - { "DMic", NULL, "SoC DMIC" }, - - {"HDMI1", NULL, "hif5-0 Output"}, - {"HDMI2", NULL, "hif6-0 Output"}, - {"HDMI3", NULL, "hif7-0 Output"}, - - /* CODEC BE connections */ - { "HiFi Playback", NULL, "ssp0 Tx" }, - { "ssp0 Tx", NULL, "codec0_out" }, - - { "Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi1", NULL, "iDisp1 Tx" }, - { "iDisp1 Tx", NULL, "iDisp1_out" }, - { "hifi2", NULL, "iDisp2 Tx" }, - { "iDisp2 Tx", NULL, "iDisp2_out" }, - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, - { "Line Out", NULL, "Platform Clock" }, -}; - -static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = DUAL_CHANNEL; - - /* set SSP to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct snd_soc_jack *jack; - int ret; - - /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24576000, - SND_SOC_CLOCK_IN); - if (ret) { - dev_err(rtd->dev, "can't set codec sysclk configuration\n"); - return ret; - } - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(kabylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, - &ctx->kabylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); - return ret; - } - - jack = &ctx->kabylake_headset; - - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); - snd_soc_component_set_jack(component, &ctx->kabylake_headset, NULL); - - ret = snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - if (ret) - dev_err(rtd->dev, "SoC DMIC - Ignore suspend failed %d\n", ret); - - return ret; -} - -static int kabylake_hdmi_init(struct snd_soc_pcm_runtime *rtd, int device) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct kbl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = device; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int kabylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI1_PB); -} - -static int kabylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI2_PB); -} - -static int kabylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - return kabylake_hdmi_init(rtd, KBL_DPCM_AUDIO_HDMI3_PB); -} - -static int kabylake_da7219_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - DUAL_CHANNEL, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static unsigned int channels_quad[] = { - QUAD_CHANNEL, -}; - -static struct snd_pcm_hw_constraint_list constraints_channels_quad = { - .count = ARRAY_SIZE(channels_quad), - .list = channels_quad, - .mask = 0, -}; - -static int kbl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = DUAL_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops kabylake_da7219_fe_ops = { - .startup = kbl_fe_startup, -}; - -static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - /* - * set BE channel constraint as user FE channels - */ - - if (params_channels(params) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static int kabylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels_quad); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops kabylake_dmic_ops = { - .startup = kabylake_dmic_startup, -}; - -static unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int kabylake_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -} - -static const struct snd_soc_ops skylake_refcap_ops = { - .startup = kabylake_refcap_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC(MAXIM_DEV0_NAME, - KBL_MAXIM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", - KBL_DIALOG_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", - "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_dais[] = { - /* Front End DAI links */ - [KBL_DPCM_AUDIO_PB] = { - .name = "Kbl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = kabylake_da7219_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_CP] = { - .name = "Kbl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &kabylake_da7219_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [KBL_DPCM_AUDIO_REF_CP] = { - .name = "Kbl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [KBL_DPCM_AUDIO_DMIC_CP] = { - .name = "Kbl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &kabylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Kbl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Kbl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [KBL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Kbl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = kabylake_da7219_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = kabylake_ssp_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .be_hw_params_fixup = kabylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = kabylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = kabylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = kabylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int kabylake_card_late_probe(struct snd_soc_card *card) -{ - struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); - struct kbl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* kabylake audio machine driver for SPT + DA7219 */ -static struct snd_soc_card kabylake_audio_card_da7219_m98357a = { - .name = "kblda7219max", - .owner = THIS_MODULE, - .dai_link = kabylake_dais, - .num_links = ARRAY_SIZE(kabylake_dais), - .controls = kabylake_controls, - .num_controls = ARRAY_SIZE(kabylake_controls), - .dapm_widgets = kabylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), - .dapm_routes = kabylake_map, - .num_dapm_routes = ARRAY_SIZE(kabylake_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = kabylake_card_late_probe, -}; - -static int kabylake_audio_probe(struct platform_device *pdev) -{ - struct kbl_codec_private *ctx; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - kabylake_audio_card = - (struct snd_soc_card *)pdev->id_entry->driver_data; - - kabylake_audio_card->dev = &pdev->dev; - snd_soc_card_set_drvdata(kabylake_audio_card, ctx); - return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); -} - -static const struct platform_device_id kbl_board_ids[] = { - { - .name = "kbl_da7219_mx98357a", - .driver_data = - (kernel_ulong_t)&kabylake_audio_card_da7219_m98357a, - }, - { } -}; -MODULE_DEVICE_TABLE(platform, kbl_board_ids); - -static struct platform_driver kabylake_audio = { - .probe = kabylake_audio_probe, - .driver = { - .name = "kbl_da7219_max98357a", - .pm = &snd_soc_pm_ops, - }, - .id_table = kbl_board_ids, -}; - -module_platform_driver(kabylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-DA7219 & MAX98357A in I2S mode"); -MODULE_AUTHOR("Naveen Manohar "); -MODULE_LICENSE("GPL v2"); From 51d8e9b20db840e78e0d1ff585cf4c8eb4e091b0 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:25 +0200 Subject: [PATCH 224/410] ASoC: Intel: Remove skl_rt286 board driver The driver has no users. Succeeded by: - avs_rt286 (./intel/avs/boards/rt286.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 13 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/skl_rt286.c | 568 ----------------------------- 3 files changed, 583 deletions(-) delete mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 33f169bc1db0..8e4f39f71531 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -254,19 +254,6 @@ endif ## SND_SST_ATOM_HIFI2_PLATFORM if SND_SOC_INTEL_SKL -config SND_SOC_INTEL_SKL_RT286_MACH - tristate "SKL with RT286 I2S mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_RT286 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC machine driver for Skylake platforms - with RT286 I2S audio codec. - Say Y or m if you have such a device. - If unsure select "N". - config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH tristate "SKL with NAU88L25 and SSM4567 in I2S Mode" depends on I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 4019ddb5588c..ecb2c3e5f2b5 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,7 +21,6 @@ snd-soc-sof_cs42l42-y := sof_cs42l42.o snd-soc-sof_es8336-y := sof_es8336.o snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o -snd-soc-skl_rt286-y := skl_rt286.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o @@ -60,7 +59,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_CX2072X_MACH) += snd-soc-sst-byt-cht-cx2072x. obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o -obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c deleted file mode 100644 index 3ea03f814403..000000000000 --- a/sound/soc/intel/boards/skl_rt286.c +++ /dev/null @@ -1,568 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Skylake I2S Machine Driver - * - * Copyright (C) 2014-2015, Intel Corporation - * - * Modified from: - * Intel Broadwell Wildcatpoint SST Audio - * - * Copyright (C) 2013, Intel Corporation - */ - -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/rt286.h" -#include "../../codecs/hdac_hdmi.h" - -static struct snd_soc_jack skylake_headset; -static struct snd_soc_jack skylake_hdmi[3]; - -struct skl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct skl_rt286_private { - struct list_head hdmi_pcm_list; -}; - -enum { - SKL_DPCM_AUDIO_PB = 0, - SKL_DPCM_AUDIO_DB_PB, - SKL_DPCM_AUDIO_CP, - SKL_DPCM_AUDIO_REF_CP, - SKL_DPCM_AUDIO_DMIC_CP, - SKL_DPCM_AUDIO_HDMI1_PB, - SKL_DPCM_AUDIO_HDMI2_PB, - SKL_DPCM_AUDIO_HDMI3_PB, -}; - -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin skylake_headset_pins[] = { - { - .pin = "Mic Jack", - .mask = SND_JACK_MICROPHONE, - }, - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, -}; - -static const struct snd_kcontrol_new skylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Speaker"), - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Mic Jack"), -}; - -static const struct snd_soc_dapm_widget skylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), - SND_SOC_DAPM_MIC("DMIC2", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("HDMI1", NULL), - SND_SOC_DAPM_SPK("HDMI2", NULL), - SND_SOC_DAPM_SPK("HDMI3", NULL), -}; - -static const struct snd_soc_dapm_route skylake_rt286_map[] = { - /* speaker */ - {"Speaker", NULL, "SPOR"}, - {"Speaker", NULL, "SPOL"}, - - /* HP jack connectors - unknown if we have jack deteck */ - {"Headphone Jack", NULL, "HPO Pin"}, - - /* other jacks */ - {"MIC1", NULL, "Mic Jack"}, - - /* digital mics */ - {"DMIC1 Pin", NULL, "DMIC2"}, - {"DMic", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - { "AIF1 Playback", NULL, "ssp0 Tx"}, - { "ssp0 Tx", NULL, "codec0_out"}, - { "ssp0 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp0 Rx" }, - { "codec1_in", NULL, "ssp0 Rx" }, - { "ssp0 Rx", NULL, "AIF1 Capture" }, - - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, - -}; - -static int skylake_rt286_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - int ret; - - ret = snd_soc_card_jack_new_pins(rtd->card, "Headset", - SND_JACK_HEADSET | SND_JACK_BTN_0, - &skylake_headset, - skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); - - if (ret) - return ret; - - snd_soc_component_set_jack(component, &skylake_headset, NULL); - - snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - - return 0; -} - -static int skylake_hdmi_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI1_PB + dai->id; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int skl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * on this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops skylake_rt286_fe_ops = { - .startup = skl_fe_startup, -}; - -static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The output is 48KHz, stereo, 16bits */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static int skylake_rt286_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, - SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops skylake_rt286_ops = { - .hw_params = skylake_rt286_hw_params, -}; - -static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - if (params_channels(params) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static const unsigned int channels_dmic[] = { - 2, 4, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static int skylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = 4; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_dmic_channels); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops skylake_dmic_ops = { - .startup = skylake_dmic_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(deepbuffer, - DAILINK_COMP_ARRAY(COMP_CPU("Deepbuffer Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1"))); - -SND_SOC_DAILINK_DEF(dmic01_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* skylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link skylake_rt286_dais[] = { - /* Front End DAI links */ - [SKL_DPCM_AUDIO_PB] = { - .name = "Skl Audio Port", - .stream_name = "Audio", - .nonatomic = 1, - .dynamic = 1, - .init = skylake_rt286_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_POST - }, - .dpcm_playback = 1, - .ops = &skylake_rt286_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_DB_PB] = { - .name = "Skl Deepbuffer Port", - .stream_name = "Deep Buffer Audio", - .nonatomic = 1, - .dynamic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_POST - }, - .dpcm_playback = 1, - .ops = &skylake_rt286_fe_ops, - SND_SOC_DAILINK_REG(deepbuffer, dummy, platform), - }, - [SKL_DPCM_AUDIO_CP] = { - .name = "Skl Audio Capture Port", - .stream_name = "Audio Record", - .nonatomic = 1, - .dynamic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_POST - }, - .dpcm_capture = 1, - .ops = &skylake_rt286_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_REF_CP] = { - .name = "Skl Audio Reference cap", - .stream_name = "refcap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [SKL_DPCM_AUDIO_DMIC_CP] = { - .name = "Skl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Skl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Skl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Skl HDMI Port3", - .stream_name = "Hdmi3", - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .init = skylake_rt286_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp0_fixup, - .ops = &skylake_rt286_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - .name = "dmic01", - .id = 1, - .be_hw_params_fixup = skylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic01_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 2, - .init = skylake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 3, - .init = skylake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 4, - .init = skylake_hdmi_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int skylake_card_late_probe(struct snd_soc_card *card) -{ - struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card); - struct skl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* skylake audio machine driver for SPT + RT286S */ -static struct snd_soc_card skylake_rt286 = { - .name = "skylake-rt286", - .owner = THIS_MODULE, - .dai_link = skylake_rt286_dais, - .num_links = ARRAY_SIZE(skylake_rt286_dais), - .controls = skylake_controls, - .num_controls = ARRAY_SIZE(skylake_controls), - .dapm_widgets = skylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), - .dapm_routes = skylake_rt286_map, - .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = skylake_card_late_probe, -}; - -static int skylake_audio_probe(struct platform_device *pdev) -{ - struct skl_rt286_private *ctx; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - skylake_rt286.dev = &pdev->dev; - snd_soc_card_set_drvdata(&skylake_rt286, ctx); - - return devm_snd_soc_register_card(&pdev->dev, &skylake_rt286); -} - -static const struct platform_device_id skl_board_ids[] = { - { .name = "skl_alc286s_i2s" }, - { .name = "kbl_alc286s_i2s" }, - { } -}; -MODULE_DEVICE_TABLE(platform, skl_board_ids); - -static struct platform_driver skylake_audio = { - .probe = skylake_audio_probe, - .driver = { - .name = "skl_alc286s_i2s", - .pm = &snd_soc_pm_ops, - }, - .id_table = skl_board_ids, - -}; - -module_platform_driver(skylake_audio) - -/* Module information */ -MODULE_AUTHOR("Omair Mohammed Abdullah "); -MODULE_DESCRIPTION("Intel SST Audio for Skylake"); -MODULE_LICENSE("GPL v2"); From 4dbf2f9a725d1370d67f9a3bce2f33e913b57e52 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:26 +0200 Subject: [PATCH 225/410] ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver The driver has no users. Succeeded by: - avs_nau8825 (./intel/avs/boards/nau8825.c) - avs_ssm4567 (./intel/avs/boards/ssm4567.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-12-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 14 - sound/soc/intel/boards/Makefile | 2 - sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ------------------ 3 files changed, 767 deletions(-) delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 8e4f39f71531..24abd1969de3 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -254,20 +254,6 @@ endif ## SND_SST_ATOM_HIFI2_PLATFORM if SND_SOC_INTEL_SKL -config SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH - tristate "SKL with NAU88L25 and SSM4567 in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_NAU8825 - select SND_SOC_SSM4567 - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for NAU88L25 + SSM4567. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH tristate "SKL with NAU88L25 and MAX98357A in I2S Mode" depends on I2C && ACPI diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index ecb2c3e5f2b5..b14650b304d9 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -23,7 +23,6 @@ snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o -snd-soc-skl_nau88l25_ssm4567-y := skl_nau88l25_ssm4567.o snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ @@ -60,7 +59,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o -obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o obj-$(CONFIG_SND_SOC_INTEL_EHL_RT5660_MACH) += snd-soc-ehl-rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c deleted file mode 100644 index d53bf3516c0d..000000000000 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ /dev/null @@ -1,751 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Skylake I2S Machine Driver for NAU88L25+SSM4567 - * - * Copyright (C) 2015, Intel Corporation - * - * Modified from: - * Intel Skylake I2S Machine Driver for NAU88L25 and SSM4567 - * - * Copyright (C) 2015, Intel Corporation - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/nau8825.h" -#include "../../codecs/hdac_hdmi.h" - -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" -#define SKL_SSM_CODEC_DAI "ssm4567-hifi" -#define DMIC_CH(p) p->list[p->count-1] - -static struct snd_soc_jack skylake_headset; -static struct snd_soc_card skylake_audio_card; -static const struct snd_pcm_hw_constraint_list *dmic_constraints; -static struct snd_soc_jack skylake_hdmi[3]; - -struct skl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct skl_nau88125_private { - struct list_head hdmi_pcm_list; -}; -enum { - SKL_DPCM_AUDIO_PB = 0, - SKL_DPCM_AUDIO_CP, - SKL_DPCM_AUDIO_REF_CP, - SKL_DPCM_AUDIO_DMIC_CP, - SKL_DPCM_AUDIO_HDMI1_PB, - SKL_DPCM_AUDIO_HDMI2_PB, - SKL_DPCM_AUDIO_HDMI3_PB, -}; - -static const struct snd_kcontrol_new skylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Left Speaker"), - SOC_DAPM_PIN_SWITCH("Right Speaker"), -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } - return ret; -} - -static const struct snd_soc_dapm_widget skylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Left Speaker", NULL), - SND_SOC_DAPM_SPK("Right Speaker", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("DP1", NULL), - SND_SOC_DAPM_SPK("DP2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route skylake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - {"Headphone Jack", NULL, "HPOL"}, - {"Headphone Jack", NULL, "HPOR"}, - - /* speaker */ - {"Left Speaker", NULL, "Left OUT"}, - {"Right Speaker", NULL, "Right OUT"}, - - /* other jacks */ - {"MIC", NULL, "Headset Mic"}, - {"DMic", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - { "Left Playback", NULL, "ssp0 Tx"}, - { "Right Playback", NULL, "ssp0 Tx"}, - { "ssp0 Tx", NULL, "codec0_out"}, - - /* IV feedback path */ - { "codec0_lp_in", NULL, "ssp0 Rx"}, - { "ssp0 Rx", NULL, "Left Capture Sense" }, - { "ssp0 Rx", NULL, "Right Capture Sense" }, - - { "Playback", NULL, "ssp1 Tx"}, - { "ssp1 Tx", NULL, "codec1_out"}, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, - - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, -}; - -static struct snd_soc_codec_conf ssm4567_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF("i2c-INT343B:00"), - .name_prefix = "Left", - }, - { - .dlc = COMP_CODEC_CONF("i2c-INT343B:01"), - .name_prefix = "Right", - }, -}; - -static int skylake_ssm4567_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - - /* Slot 1 for left */ - ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_codec(rtd, 0), 0x01, 0x01, 2, 48); - if (ret < 0) - return ret; - - /* Slot 2 for right */ - ret = snd_soc_dai_set_tdm_slot(snd_soc_rtd_to_codec(rtd, 1), 0x02, 0x02, 2, 48); - if (ret < 0) - return ret; - - return ret; -} - -static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - - /* - * 4 buttons here map to the google Reference headset - * The use of these buttons can be decided by the user space. - */ - ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - nau8825_enable_jack_detect(component, &skylake_headset); - - snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - - return ret; -} - -static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - - -static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int skl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * on this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops skylake_nau8825_fe_ops = { - .startup = skl_fe_startup, -}; - -static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - return 0; -} - -static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; - -static const unsigned int channels_dmic[] = { - 2, 4, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static const unsigned int dmic_2ch[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { - .count = ARRAY_SIZE(dmic_2ch), - .list = dmic_2ch, - .mask = 0, -}; - -static int skylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = DMIC_CH(dmic_constraints); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dmic_constraints); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops skylake_dmic_ops = { - .startup = skylake_dmic_startup, -}; - -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int skylake_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -} - -static const struct snd_soc_ops skylake_refcap_ops = { - .startup = skylake_refcap_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY( - /* Left */ COMP_CODEC("i2c-INT343B:00", SKL_SSM_CODEC_DAI), - /* Right */ COMP_CODEC("i2c-INT343B:01", SKL_SSM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10508825:00", SKL_NUVOTON_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic01_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* skylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link skylake_dais[] = { - /* Front End DAI links */ - [SKL_DPCM_AUDIO_PB] = { - .name = "Skl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = skylake_nau8825_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &skylake_nau8825_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_CP] = { - .name = "Skl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &skylake_nau8825_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_REF_CP] = { - .name = "Skl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [SKL_DPCM_AUDIO_DMIC_CP] = { - .name = "Skl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Skl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Skl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Skl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_DSP_A | - SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .init = skylake_ssm4567_codec_init, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = skylake_nau8825_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .ops = &skylake_nau8825_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .ignore_suspend = 1, - .be_hw_params_fixup = skylake_dmic_fixup, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic01_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = skylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = skylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = skylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int skylake_card_late_probe(struct snd_soc_card *card) -{ - struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card); - struct skl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, - &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* skylake audio machine driver for SPT + NAU88L25 */ -static struct snd_soc_card skylake_audio_card = { - .name = "sklnau8825adi", - .owner = THIS_MODULE, - .dai_link = skylake_dais, - .num_links = ARRAY_SIZE(skylake_dais), - .controls = skylake_controls, - .num_controls = ARRAY_SIZE(skylake_controls), - .dapm_widgets = skylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), - .dapm_routes = skylake_map, - .num_dapm_routes = ARRAY_SIZE(skylake_map), - .codec_conf = ssm4567_codec_conf, - .num_configs = ARRAY_SIZE(ssm4567_codec_conf), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = skylake_card_late_probe, -}; - -static int skylake_audio_probe(struct platform_device *pdev) -{ - struct skl_nau88125_private *ctx; - struct snd_soc_acpi_mach *mach; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - skylake_audio_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - - mach = pdev->dev.platform_data; - if (mach) - dmic_constraints = mach->mach_params.dmic_num == 2 ? - &constraints_dmic_2ch : &constraints_dmic_channels; - - return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); -} - -static const struct platform_device_id skl_board_ids[] = { - { .name = "skl_n88l25_s4567" }, - { .name = "kbl_n88l25_s4567" }, - { } -}; -MODULE_DEVICE_TABLE(platform, skl_board_ids); - -static struct platform_driver skylake_audio = { - .probe = skylake_audio_probe, - .driver = { - .name = "skl_n88l25_s4567", - .pm = &snd_soc_pm_ops, - }, - .id_table = skl_board_ids, -}; - -module_platform_driver(skylake_audio) - -/* Module information */ -MODULE_AUTHOR("Conrad Cooke "); -MODULE_AUTHOR("Harsha Priya "); -MODULE_AUTHOR("Naveen M "); -MODULE_AUTHOR("Sathya Prakash M R "); -MODULE_AUTHOR("Yong Zhi "); -MODULE_DESCRIPTION("Intel Audio Machine driver for SKL with NAU88L25 and SSM4567 in I2S Mode"); -MODULE_LICENSE("GPL v2"); From 6de8dddc56b0577df996212b634f82f6f1fb013c Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:27 +0200 Subject: [PATCH 226/410] ASoC: Intel: Remove skl_nau88l25_max98357a board driver The driver has no users. Succeeded by: - avs_nau8825 (./intel/avs/boards/nau8825.c) - avs_max98357a (./intel/avs/boards/max98357a.c) Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-13-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 18 - sound/soc/intel/boards/Makefile | 2 - .../soc/intel/boards/skl_nau88l25_max98357a.c | 704 ------------------ 3 files changed, 724 deletions(-) delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24abd1969de3..1c0b1ace3f6f 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -252,24 +252,6 @@ config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH endif ## SND_SST_ATOM_HIFI2_PLATFORM -if SND_SOC_INTEL_SKL - -config SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH - tristate "SKL with NAU88L25 and MAX98357A in I2S Mode" - depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST - select SND_SOC_NAU8825 - select SND_SOC_MAX98357A - select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for NAU88L25 + MAX98357A. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". - -endif ## SND_SOC_INTEL_SKL - config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC tristate select SND_SOC_DA7219 diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index b14650b304d9..7adff070a7df 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -22,7 +22,6 @@ snd-soc-sof_es8336-y := sof_es8336.o snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o -snd-skl_nau88l25_max98357a-y := skl_nau88l25_max98357a.o snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ @@ -58,7 +57,6 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_CX2072X_MACH) += snd-soc-sst-byt-cht-cx2072x. obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_ES8316_MACH) += snd-soc-sst-byt-cht-es8316.o obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH) += snd-soc-sst-byt-cht-nocodec.o -obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o obj-$(CONFIG_SND_SOC_INTEL_EHL_RT5660_MACH) += snd-soc-ehl-rt5660.o obj-$(CONFIG_SND_SOC_INTEL_SOUNDWIRE_SOF_MACH) += snd-soc-sof-sdw.o diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c deleted file mode 100644 index 91fe9834aab4..000000000000 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ /dev/null @@ -1,704 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * Intel Skylake I2S Machine Driver with MAXIM98357A - * and NAU88L25 - * - * Copyright (C) 2015, Intel Corporation - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include "../../codecs/nau8825.h" -#include "../../codecs/hdac_hdmi.h" - -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" -#define SKL_MAXIM_CODEC_DAI "HiFi" -#define DMIC_CH(p) p->list[p->count-1] - -static struct snd_soc_jack skylake_headset; -static struct snd_soc_card skylake_audio_card; -static const struct snd_pcm_hw_constraint_list *dmic_constraints; -static struct snd_soc_jack skylake_hdmi[3]; - -struct skl_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; - int device; -}; - -struct skl_nau8825_private { - struct list_head hdmi_pcm_list; -}; - -enum { - SKL_DPCM_AUDIO_PB = 0, - SKL_DPCM_AUDIO_CP, - SKL_DPCM_AUDIO_REF_CP, - SKL_DPCM_AUDIO_DMIC_CP, - SKL_DPCM_AUDIO_HDMI1_PB, - SKL_DPCM_AUDIO_HDMI2_PB, - SKL_DPCM_AUDIO_HDMI3_PB, -}; - -static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); - return -EIO; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } - - return ret; -} - -static const struct snd_kcontrol_new skylake_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), -}; - -static const struct snd_soc_dapm_widget skylake_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), - SND_SOC_DAPM_SPK("DP1", NULL), - SND_SOC_DAPM_SPK("DP2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, - platform_clock_control, SND_SOC_DAPM_PRE_PMU | - SND_SOC_DAPM_POST_PMD), -}; - -static struct snd_soc_jack_pin jack_pins[] = { - { - .pin = "Headphone Jack", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - -static const struct snd_soc_dapm_route skylake_map[] = { - /* HP jack connectors - unknown if we have jack detection */ - { "Headphone Jack", NULL, "HPOL" }, - { "Headphone Jack", NULL, "HPOR" }, - - /* speaker */ - { "Spk", NULL, "Speaker" }, - - /* other jacks */ - { "MIC", NULL, "Headset Mic" }, - { "DMic", NULL, "SoC DMIC" }, - - /* CODEC BE connections */ - { "HiFi Playback", NULL, "ssp0 Tx" }, - { "ssp0 Tx", NULL, "codec0_out" }, - - { "Playback", NULL, "ssp1 Tx" }, - { "ssp1 Tx", NULL, "codec1_out" }, - - { "codec0_in", NULL, "ssp1 Rx" }, - { "ssp1 Rx", NULL, "Capture" }, - - /* DMIC */ - { "dmic01_hifi", NULL, "DMIC01 Rx" }, - { "DMIC01 Rx", NULL, "DMIC AIF" }, - - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, - - { "Headphone Jack", NULL, "Platform Clock" }, - { "Headset Mic", NULL, "Platform Clock" }, -}; - -static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *rate = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_RATE); - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); - - /* The ADSP will convert the FE rate to 48k, stereo */ - rate->min = rate->max = 48000; - chan->min = chan->max = 2; - - /* set SSP0 to 24 bit */ - snd_mask_none(fmt); - snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); - - return 0; -} - -static int skylake_nau8825_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - int ret; - struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; - - /* - * Headset buttons map to the google Reference headset. - * These can be configured by userspace. - */ - ret = snd_soc_card_jack_new_pins(&skylake_audio_card, "Headset Jack", - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3, &skylake_headset, - jack_pins, - ARRAY_SIZE(jack_pins)); - if (ret) { - dev_err(rtd->dev, "Headset Jack creation failed %d\n", ret); - return ret; - } - - nau8825_enable_jack_detect(component, &skylake_headset); - - snd_soc_dapm_ignore_suspend(&rtd->card->dapm, "SoC DMIC"); - - return ret; -} - -static int skylake_hdmi1_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI1_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int skylake_hdmi2_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI2_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int skylake_hdmi3_init(struct snd_soc_pcm_runtime *rtd) -{ - struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(rtd->card); - struct snd_soc_dai *dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_hdmi_pcm *pcm; - - pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - - pcm->device = SKL_DPCM_AUDIO_HDMI3_PB; - pcm->codec_dai = dai; - - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); - - return 0; -} - -static int skylake_nau8825_fe_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dapm_context *dapm; - struct snd_soc_component *component = snd_soc_rtd_to_cpu(rtd, 0)->component; - - dapm = snd_soc_component_get_dapm(component); - snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); - - return 0; -} - -static const unsigned int rates[] = { - 48000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static const unsigned int channels[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_channels = { - .count = ARRAY_SIZE(channels), - .list = channels, - .mask = 0, -}; - -static int skl_fe_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - /* - * On this platform for PCM device we support, - * 48Khz - * stereo - * 16 bit audio - */ - - runtime->hw.channels_max = 2; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); - - snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); - - return 0; -} - -static const struct snd_soc_ops skylake_nau8825_fe_ops = { - .startup = skl_fe_startup, -}; - -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); - - return ret; -} - -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, -}; - -static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, - struct snd_pcm_hw_params *params) -{ - struct snd_interval *chan = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - - if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2) - chan->min = chan->max = 2; - else - chan->min = chan->max = 4; - - return 0; -} - -static const unsigned int channels_dmic[] = { - 2, 4, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { - .count = ARRAY_SIZE(channels_dmic), - .list = channels_dmic, - .mask = 0, -}; - -static const unsigned int dmic_2ch[] = { - 2, -}; - -static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { - .count = ARRAY_SIZE(dmic_2ch), - .list = dmic_2ch, - .mask = 0, -}; - -static int skylake_dmic_startup(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - - runtime->hw.channels_max = DMIC_CH(dmic_constraints); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - dmic_constraints); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); -} - -static const struct snd_soc_ops skylake_dmic_ops = { - .startup = skylake_dmic_startup, -}; - -static const unsigned int rates_16000[] = { - 16000, -}; - -static const struct snd_pcm_hw_constraint_list constraints_16000 = { - .count = ARRAY_SIZE(rates_16000), - .list = rates_16000, -}; - -static const unsigned int ch_mono[] = { - 1, -}; - -static const struct snd_pcm_hw_constraint_list constraints_refcap = { - .count = ARRAY_SIZE(ch_mono), - .list = ch_mono, -}; - -static int skylake_refcap_startup(struct snd_pcm_substream *substream) -{ - substream->runtime->hw.channels_max = 1; - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - &constraints_refcap); - - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_16000); -} - -static const struct snd_soc_ops skylake_refcap_ops = { - .startup = skylake_refcap_startup, -}; - -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(system, - DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); - -SND_SOC_DAILINK_DEF(reference, - DAILINK_COMP_ARRAY(COMP_CPU("Reference Pin"))); - -SND_SOC_DAILINK_DEF(dmic, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC Pin"))); - -SND_SOC_DAILINK_DEF(hdmi1, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI1 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi2, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI2 Pin"))); - -SND_SOC_DAILINK_DEF(hdmi3, - DAILINK_COMP_ARRAY(COMP_CPU("HDMI3 Pin"))); - -SND_SOC_DAILINK_DEF(ssp0_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin"))); -SND_SOC_DAILINK_DEF(ssp0_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", SKL_MAXIM_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(ssp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin"))); -SND_SOC_DAILINK_DEF(ssp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10508825:00", - SKL_NUVOTON_CODEC_DAI))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(idisp1_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_pin, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* skylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link skylake_dais[] = { - /* Front End DAI links */ - [SKL_DPCM_AUDIO_PB] = { - .name = "Skl Audio Port", - .stream_name = "Audio", - .dynamic = 1, - .nonatomic = 1, - .init = skylake_nau8825_fe_init, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .ops = &skylake_nau8825_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_CP] = { - .name = "Skl Audio Capture Port", - .stream_name = "Audio Record", - .dynamic = 1, - .nonatomic = 1, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_capture = 1, - .ops = &skylake_nau8825_fe_ops, - SND_SOC_DAILINK_REG(system, dummy, platform), - }, - [SKL_DPCM_AUDIO_REF_CP] = { - .name = "Skl Audio Reference cap", - .stream_name = "Wake on Voice", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_refcap_ops, - SND_SOC_DAILINK_REG(reference, dummy, platform), - }, - [SKL_DPCM_AUDIO_DMIC_CP] = { - .name = "Skl Audio DMIC cap", - .stream_name = "dmiccap", - .init = NULL, - .dpcm_capture = 1, - .nonatomic = 1, - .dynamic = 1, - .ops = &skylake_dmic_ops, - SND_SOC_DAILINK_REG(dmic, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI1_PB] = { - .name = "Skl HDMI Port1", - .stream_name = "Hdmi1", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi1, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI2_PB] = { - .name = "Skl HDMI Port2", - .stream_name = "Hdmi2", - .dpcm_playback = 1, - .init = NULL, - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi2, dummy, platform), - }, - [SKL_DPCM_AUDIO_HDMI3_PB] = { - .name = "Skl HDMI Port3", - .stream_name = "Hdmi3", - .trigger = { - SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, - .dpcm_playback = 1, - .init = NULL, - .nonatomic = 1, - .dynamic = 1, - SND_SOC_DAILINK_REG(hdmi3, dummy, platform), - }, - - /* Back End DAI links */ - { - /* SSP0 - Codec */ - .name = "SSP0-Codec", - .id = 0, - .no_pcm = 1, - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .dpcm_playback = 1, - SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), - }, - { - /* SSP1 - Codec */ - .name = "SSP1-Codec", - .id = 1, - .no_pcm = 1, - .init = skylake_nau8825_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBC_CFC, - .ignore_pmdown_time = 1, - .be_hw_params_fixup = skylake_ssp_fixup, - .ops = &skylake_nau8825_ops, - .dpcm_playback = 1, - .dpcm_capture = 1, - SND_SOC_DAILINK_REG(ssp1_pin, ssp1_codec, platform), - }, - { - .name = "dmic01", - .id = 2, - .be_hw_params_fixup = skylake_dmic_fixup, - .ignore_suspend = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "iDisp1", - .id = 3, - .dpcm_playback = 1, - .init = skylake_hdmi1_init, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 4, - .init = skylake_hdmi2_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 5, - .init = skylake_hdmi3_init, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform), - }, -}; - -#define NAME_SIZE 32 -static int skylake_card_late_probe(struct snd_soc_card *card) -{ - struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card); - struct skl_hdmi_pcm *pcm; - struct snd_soc_component *component = NULL; - int err, i = 0; - char jack_name[NAME_SIZE]; - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, - &skylake_hdmi[i]); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &skylake_hdmi[i]); - if (err < 0) - return err; - - i++; - } - - if (!component) - return -EINVAL; - - return hdac_hdmi_jack_port_init(component, &card->dapm); -} - -/* skylake audio machine driver for SPT + NAU88L25 */ -static struct snd_soc_card skylake_audio_card = { - .name = "sklnau8825max", - .owner = THIS_MODULE, - .dai_link = skylake_dais, - .num_links = ARRAY_SIZE(skylake_dais), - .controls = skylake_controls, - .num_controls = ARRAY_SIZE(skylake_controls), - .dapm_widgets = skylake_widgets, - .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), - .dapm_routes = skylake_map, - .num_dapm_routes = ARRAY_SIZE(skylake_map), - .fully_routed = true, - .disable_route_checks = true, - .late_probe = skylake_card_late_probe, -}; - -static int skylake_audio_probe(struct platform_device *pdev) -{ - struct skl_nau8825_private *ctx; - struct snd_soc_acpi_mach *mach; - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) - return -ENOMEM; - - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - - skylake_audio_card.dev = &pdev->dev; - snd_soc_card_set_drvdata(&skylake_audio_card, ctx); - - mach = pdev->dev.platform_data; - if (mach) - dmic_constraints = mach->mach_params.dmic_num == 2 ? - &constraints_dmic_2ch : &constraints_dmic_channels; - - return devm_snd_soc_register_card(&pdev->dev, &skylake_audio_card); -} - -static const struct platform_device_id skl_board_ids[] = { - { .name = "skl_n88l25_m98357a" }, - { .name = "kbl_n88l25_m98357a" }, - { } -}; -MODULE_DEVICE_TABLE(platform, skl_board_ids); - -static struct platform_driver skylake_audio = { - .probe = skylake_audio_probe, - .driver = { - .name = "skl_n88l25_m98357a", - .pm = &snd_soc_pm_ops, - }, - .id_table = skl_board_ids, -}; - -module_platform_driver(skylake_audio) - -/* Module information */ -MODULE_DESCRIPTION("Audio Machine driver-NAU88L25 & MAX98357A in I2S mode"); -MODULE_AUTHOR("Rohit Ainapure Date: Wed, 14 Aug 2024 10:39:28 +0200 Subject: [PATCH 227/410] ASoC: Intel: Remove skylake driver The avs-driver found in sound/soc/intel/avs is a direct replacement to the existing skylake-driver. It covers all features supported by it and more and aligns with the recommended flows and requirements based on Windows driver equivalent. For the official kernel tree the deprecation begun with v6.0. Most skylake-drivers users moved to avs- or SOF-driver when AudioDSP capabilities are available on the platform or to snd-hda-intel (sound/pci/hda) when such capabilities are not. For the supported trees the deprecation begun with v5.4 with v5.15 being the first where the skylake-driver is disabled entirely. All machine board drivers that consume the DSP driver have their replacements present within sound/soc/intel/avs/boards/ directory. Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-14-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/boards/Kconfig | 4 +- sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 ---- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 ---- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 --------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 ---------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 --- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 --- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ------- sound/soc/intel/skylake/skl-sst-ipc.h | 169 -- sound/soc/intel/skylake/skl-sst-utils.c | 425 --- sound/soc/intel/skylake/skl-sst.c | 599 ---- sound/soc/intel/skylake/skl-topology.c | 3605 ----------------------- sound/soc/intel/skylake/skl-topology.h | 524 ---- sound/soc/intel/skylake/skl.c | 1177 -------- sound/soc/intel/skylake/skl.h | 207 -- 27 files changed, 2 insertions(+), 14821 deletions(-) delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 38b61dfd1487..a32fb0a8d7d7 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -67,126 +67,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI Baytrail/Cherrytrail. If you want to enable SOF on Baytrail/Cherrytrail, you need to deselect this option first. -config SND_SOC_INTEL_SKYLAKE - tristate "All Skylake/SST Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKL - select SND_SOC_INTEL_APL - select SND_SOC_INTEL_KBL - select SND_SOC_INTEL_GLK - select SND_SOC_INTEL_CNL - select SND_SOC_INTEL_CFL - help - This is a backwards-compatible option to select all devices - supported by the Intel SST/Skylake driver. This option is no - longer recommended and will be deprecated when the SOF - driver is introduced. Distributions should explicitly - select which platform uses this driver. - -config SND_SOC_INTEL_SKL - tristate "Skylake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel Skylake platform with the DSP enabled - in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_APL - tristate "Broxton/ApolloLake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel Broxton/ApolloLake platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_KBL - tristate "Kabylake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel Kabylake platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_GLK - tristate "GeminiLake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel GeminiLake platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_CNL - tristate "CannonLake/WhiskyLake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel CNL/WHL platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_CFL - tristate "CoffeeLake Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel CoffeeLake platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_CML_H - tristate "CometLake-H Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel CometLake-H platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_CML_LP - tristate "CometLake-LP Platforms" - depends on PCI && ACPI - depends on COMMON_CLK - select SND_SOC_INTEL_SKYLAKE_FAMILY - help - If you have a Intel CometLake-LP platform with the DSP - enabled in the BIOS then enable this option by saying Y or m. - -config SND_SOC_INTEL_SKYLAKE_FAMILY - tristate - select SND_SOC_INTEL_SKYLAKE_COMMON - -if SND_SOC_INTEL_SKYLAKE_FAMILY - -config SND_SOC_INTEL_SKYLAKE_SSP_CLK - tristate - -config SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC - bool "HDAudio codec support" - help - If you have Intel Skylake or Kabylake with HDAudio codec - and DMIC present then enable this option by saying Y. - -config SND_SOC_INTEL_SKYLAKE_COMMON - tristate - select SND_HDA_EXT_CORE - select SND_HDA_DSP_LOADER - select SND_SOC_TOPOLOGY - select SND_SOC_INTEL_SST - select SND_SOC_HDAC_HDA - select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_DSP_CONFIG - help - If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ - GeminiLake or CannonLake platform with the DSP enabled in the BIOS - then enable this option by saying Y or m. - -endif ## SND_SOC_INTEL_SKYLAKE_FAMILY - endif ## SND_SOC_INTEL_SST_TOPLEVEL if SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index d44b2652c707..8ecc7047d700 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -5,7 +5,6 @@ obj-$(CONFIG_SND_SOC) += common/ # Platform Support obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += atom/ obj-$(CONFIG_SND_SOC_INTEL_CATPT) += catpt/ -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += skylake/ obj-$(CONFIG_SND_SOC_INTEL_KEEMBAY) += keembay/ obj-$(CONFIG_SND_SOC_INTEL_AVS) += avs/ diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 1c0b1ace3f6f..e53fa4be5e28 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -300,7 +300,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH endif ## SND_SOC_SOF_GEMINILAKE -if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC +if SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "Skylake+ with HDA Codecs" @@ -316,7 +316,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH Say Y or m if you have such a device. This is a recommended option. If unsure select "N". -endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC +endif ## SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile deleted file mode 100644 index ad9be6168428..000000000000 --- a/sound/soc/intel/skylake/Makefile +++ /dev/null @@ -1,15 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0-only -snd-soc-skl-y := skl.o skl-pcm.o skl-nhlt.o skl-messages.o skl-topology.o \ - skl-sst-ipc.o skl-sst-dsp.o cnl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o cnl-sst.o skl-sst-utils.o - -ifdef CONFIG_DEBUG_FS - snd-soc-skl-y += skl-debug.o -endif - -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_COMMON) += snd-soc-skl.o - -#Skylake Clock device support -snd-soc-skl-ssp-clk-y := skl-ssp-clk.o - -obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE_SSP_CLK) += snd-soc-skl-ssp-clk.o diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c deleted file mode 100644 index fd4fdcb95224..000000000000 --- a/sound/soc/intel/skylake/bxt-sst.c +++ /dev/null @@ -1,629 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * bxt-sst.c - DSP library functions for BXT platform - * - * Copyright (C) 2015-16 Intel Corp - * Author:Rafal Redzimski - * Jeeja KP - */ - -#include -#include -#include -#include - -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "skl.h" - -#define BXT_BASEFW_TIMEOUT 3000 -#define BXT_ROM_INIT_TIMEOUT 70 -#define BXT_IPC_PURGE_FW 0x01004000 - -#define BXT_ROM_INIT 0x5 -#define BXT_ADSP_SRAM0_BASE 0x80000 - -/* Firmware status window */ -#define BXT_ADSP_FW_STATUS BXT_ADSP_SRAM0_BASE -#define BXT_ADSP_ERROR_CODE (BXT_ADSP_FW_STATUS + 0x4) - -#define BXT_ADSP_SRAM1_BASE 0xA0000 - -#define BXT_INSTANCE_ID 0 -#define BXT_BASE_FW_MODULE_ID 0 - -#define BXT_ADSP_FW_BIN_HDR_OFFSET 0x2000 - -/* Delay before scheduling D0i3 entry */ -#define BXT_D0I3_DELAY 5000 - -static unsigned int bxt_get_errorcode(struct sst_dsp *ctx) -{ - return sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE); -} - -static int -bxt_load_library(struct sst_dsp *ctx, struct skl_lib_info *linfo, int lib_count) -{ - struct snd_dma_buffer dmab; - struct skl_dev *skl = ctx->thread_context; - struct firmware stripped_fw; - int ret = 0, i, dma_id, stream_tag; - - /* library indices start from 1 to N. 0 represents base FW */ - for (i = 1; i < lib_count; i++) { - ret = skl_prepare_lib_load(skl, &skl->lib_info[i], &stripped_fw, - BXT_ADSP_FW_BIN_HDR_OFFSET, i); - if (ret < 0) - goto load_library_failed; - - stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, - stripped_fw.size, &dmab); - if (stream_tag <= 0) { - dev_err(ctx->dev, "Lib prepare DMA err: %x\n", - stream_tag); - ret = stream_tag; - goto load_library_failed; - } - - dma_id = stream_tag - 1; - memcpy(dmab.area, stripped_fw.data, stripped_fw.size); - - ctx->dsp_ops.trigger(ctx->dev, true, stream_tag); - ret = skl_sst_ipc_load_library(&skl->ipc, dma_id, i, true); - if (ret < 0) - dev_err(ctx->dev, "IPC Load Lib for %s fail: %d\n", - linfo[i].name, ret); - - ctx->dsp_ops.trigger(ctx->dev, false, stream_tag); - ctx->dsp_ops.cleanup(ctx->dev, &dmab, stream_tag); - } - - return ret; - -load_library_failed: - skl_release_library(linfo, lib_count); - return ret; -} - -/* - * First boot sequence has some extra steps. Core 0 waits for power - * status on core 1, so power up core 1 also momentarily, keep it in - * reset/stall and then turn it off - */ -static int sst_bxt_prepare_fw(struct sst_dsp *ctx, - const void *fwdata, u32 fwsize) -{ - int stream_tag, ret; - - stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab); - if (stream_tag <= 0) { - dev_err(ctx->dev, "Failed to prepare DMA FW loading err: %x\n", - stream_tag); - return stream_tag; - } - - ctx->dsp_ops.stream_tag = stream_tag; - memcpy(ctx->dmab.area, fwdata, fwsize); - - /* Step 1: Power up core 0 and core1 */ - ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK | - SKL_DSP_CORE_MASK(1)); - if (ret < 0) { - dev_err(ctx->dev, "dsp core0/1 power up failed\n"); - goto base_fw_load_failed; - } - - /* Step 2: Purge FW request */ - sst_dsp_shim_write(ctx, SKL_ADSP_REG_HIPCI, SKL_ADSP_REG_HIPCI_BUSY | - (BXT_IPC_PURGE_FW | ((stream_tag - 1) << 9))); - - /* Step 3: Unset core0 reset state & unstall/run core0 */ - ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret); - ret = -EIO; - goto base_fw_load_failed; - } - - /* Step 4: Wait for DONE Bit */ - ret = sst_dsp_register_poll(ctx, SKL_ADSP_REG_HIPCIE, - SKL_ADSP_REG_HIPCIE_DONE, - SKL_ADSP_REG_HIPCIE_DONE, - BXT_INIT_TIMEOUT, "HIPCIE Done"); - if (ret < 0) { - dev_err(ctx->dev, "Timeout for Purge Request%d\n", ret); - goto base_fw_load_failed; - } - - /* Step 5: power down core1 */ - ret = skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); - if (ret < 0) { - dev_err(ctx->dev, "dsp core1 power down failed\n"); - goto base_fw_load_failed; - } - - /* Step 6: Enable Interrupt */ - skl_ipc_int_enable(ctx); - skl_ipc_op_int_enable(ctx); - - /* Step 7: Wait for ROM init */ - ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK, - SKL_FW_INIT, BXT_ROM_INIT_TIMEOUT, "ROM Load"); - if (ret < 0) { - dev_err(ctx->dev, "Timeout for ROM init, ret:%d\n", ret); - goto base_fw_load_failed; - } - - return ret; - -base_fw_load_failed: - ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); - skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); - skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - return ret; -} - -static int sst_transfer_fw_host_dma(struct sst_dsp *ctx) -{ - int ret; - - ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag); - ret = sst_dsp_register_poll(ctx, BXT_ADSP_FW_STATUS, SKL_FW_STS_MASK, - BXT_ROM_INIT, BXT_BASEFW_TIMEOUT, "Firmware boot"); - - ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag); - ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag); - - return ret; -} - -static int bxt_load_base_firmware(struct sst_dsp *ctx) -{ - struct firmware stripped_fw; - struct skl_dev *skl = ctx->thread_context; - int ret, i; - - if (ctx->fw == NULL) { - ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); - if (ret < 0) { - dev_err(ctx->dev, "Request firmware failed %d\n", ret); - return ret; - } - } - - /* prase uuids on first boot */ - if (skl->is_first_boot) { - ret = snd_skl_parse_uuids(ctx, ctx->fw, BXT_ADSP_FW_BIN_HDR_OFFSET, 0); - if (ret < 0) - goto sst_load_base_firmware_failed; - } - - stripped_fw.data = ctx->fw->data; - stripped_fw.size = ctx->fw->size; - skl_dsp_strip_extended_manifest(&stripped_fw); - - - for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) { - ret = sst_bxt_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); - if (ret == 0) - break; - } - - if (ret < 0) { - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", - sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), - sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - - dev_err(ctx->dev, "Core En/ROM load fail:%d\n", ret); - goto sst_load_base_firmware_failed; - } - - ret = sst_transfer_fw_host_dma(ctx); - if (ret < 0) { - dev_err(ctx->dev, "Transfer firmware failed %d\n", ret); - dev_info(ctx->dev, "Error code=0x%x: FW status=0x%x\n", - sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), - sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - - skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - } else { - dev_dbg(ctx->dev, "Firmware download successful\n"); - ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - if (ret == 0) { - dev_err(ctx->dev, "DSP boot fail, FW Ready timeout\n"); - skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - ret = -EIO; - } else { - ret = 0; - skl->fw_loaded = true; - } - } - - return ret; - -sst_load_base_firmware_failed: - release_firmware(ctx->fw); - ctx->fw = NULL; - return ret; -} - -/* - * Decide the D0i3 state that can be targeted based on the usecase - * ref counts and DSP state - * - * Decision Matrix: (X= dont care; state = target state) - * - * DSP state != SKL_DSP_RUNNING ; state = no d0i3 - * - * DSP state == SKL_DSP_RUNNING , the following matrix applies - * non_d0i3 >0; streaming =X; non_streaming =X; state = no d0i3 - * non_d0i3 =X; streaming =0; non_streaming =0; state = no d0i3 - * non_d0i3 =0; streaming >0; non_streaming =X; state = streaming d0i3 - * non_d0i3 =0; streaming =0; non_streaming =X; state = non-streaming d0i3 - */ -static int bxt_d0i3_target_state(struct sst_dsp *ctx) -{ - struct skl_dev *skl = ctx->thread_context; - struct skl_d0i3_data *d0i3 = &skl->d0i3; - - if (skl->cores.state[SKL_DSP_CORE0_ID] != SKL_DSP_RUNNING) - return SKL_DSP_D0I3_NONE; - - if (d0i3->non_d0i3) - return SKL_DSP_D0I3_NONE; - else if (d0i3->streaming) - return SKL_DSP_D0I3_STREAMING; - else if (d0i3->non_streaming) - return SKL_DSP_D0I3_NON_STREAMING; - else - return SKL_DSP_D0I3_NONE; -} - -static void bxt_set_dsp_D0i3(struct work_struct *work) -{ - int ret; - struct skl_ipc_d0ix_msg msg; - struct skl_dev *skl = container_of(work, - struct skl_dev, d0i3.work.work); - struct sst_dsp *ctx = skl->dsp; - struct skl_d0i3_data *d0i3 = &skl->d0i3; - int target_state; - - dev_dbg(ctx->dev, "In %s:\n", __func__); - - /* D0i3 entry allowed only if core 0 alone is running */ - if (skl_dsp_get_enabled_cores(ctx) != SKL_DSP_CORE0_MASK) { - dev_warn(ctx->dev, - "D0i3 allowed when only core0 running:Exit\n"); - return; - } - - target_state = bxt_d0i3_target_state(ctx); - if (target_state == SKL_DSP_D0I3_NONE) - return; - - msg.instance_id = 0; - msg.module_id = 0; - msg.wake = 1; - msg.streaming = 0; - if (target_state == SKL_DSP_D0I3_STREAMING) - msg.streaming = 1; - - ret = skl_ipc_set_d0ix(&skl->ipc, &msg); - - if (ret < 0) { - dev_err(ctx->dev, "Failed to set DSP to D0i3 state\n"); - return; - } - - /* Set Vendor specific register D0I3C.I3 to enable D0i3*/ - if (skl->update_d0i3c) - skl->update_d0i3c(skl->dev, true); - - d0i3->state = target_state; - skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING_D0I3; -} - -static int bxt_schedule_dsp_D0i3(struct sst_dsp *ctx) -{ - struct skl_dev *skl = ctx->thread_context; - struct skl_d0i3_data *d0i3 = &skl->d0i3; - - /* Schedule D0i3 only if the usecase ref counts are appropriate */ - if (bxt_d0i3_target_state(ctx) != SKL_DSP_D0I3_NONE) { - - dev_dbg(ctx->dev, "%s: Schedule D0i3\n", __func__); - - schedule_delayed_work(&d0i3->work, - msecs_to_jiffies(BXT_D0I3_DELAY)); - } - - return 0; -} - -static int bxt_set_dsp_D0i0(struct sst_dsp *ctx) -{ - int ret; - struct skl_ipc_d0ix_msg msg; - struct skl_dev *skl = ctx->thread_context; - - dev_dbg(ctx->dev, "In %s:\n", __func__); - - /* First Cancel any pending attempt to put DSP to D0i3 */ - cancel_delayed_work_sync(&skl->d0i3.work); - - /* If DSP is currently in D0i3, bring it to D0i0 */ - if (skl->cores.state[SKL_DSP_CORE0_ID] != SKL_DSP_RUNNING_D0I3) - return 0; - - dev_dbg(ctx->dev, "Set DSP to D0i0\n"); - - msg.instance_id = 0; - msg.module_id = 0; - msg.streaming = 0; - msg.wake = 0; - - if (skl->d0i3.state == SKL_DSP_D0I3_STREAMING) - msg.streaming = 1; - - /* Clear Vendor specific register D0I3C.I3 to disable D0i3*/ - if (skl->update_d0i3c) - skl->update_d0i3c(skl->dev, false); - - ret = skl_ipc_set_d0ix(&skl->ipc, &msg); - if (ret < 0) { - dev_err(ctx->dev, "Failed to set DSP to D0i0\n"); - return ret; - } - - skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING; - skl->d0i3.state = SKL_DSP_D0I3_NONE; - - return 0; -} - -static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) -{ - struct skl_dev *skl = ctx->thread_context; - int ret; - struct skl_ipc_dxstate_info dx; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - - if (skl->fw_loaded == false) { - skl->boot_complete = false; - ret = bxt_load_base_firmware(ctx); - if (ret < 0) { - dev_err(ctx->dev, "reload fw failed: %d\n", ret); - return ret; - } - - if (skl->lib_count > 1) { - ret = bxt_load_library(ctx, skl->lib_info, - skl->lib_count); - if (ret < 0) { - dev_err(ctx->dev, "reload libs failed: %d\n", ret); - return ret; - } - } - skl->cores.state[core_id] = SKL_DSP_RUNNING; - return ret; - } - - /* If core 0 is being turned on, turn on core 1 as well */ - if (core_id == SKL_DSP_CORE0_ID) - ret = skl_dsp_core_power_up(ctx, core_mask | - SKL_DSP_CORE_MASK(1)); - else - ret = skl_dsp_core_power_up(ctx, core_mask); - - if (ret < 0) - goto err; - - if (core_id == SKL_DSP_CORE0_ID) { - - /* - * Enable interrupt after SPA is set and before - * DSP is unstalled - */ - skl_ipc_int_enable(ctx); - skl_ipc_op_int_enable(ctx); - skl->boot_complete = false; - } - - ret = skl_dsp_start_core(ctx, core_mask); - if (ret < 0) - goto err; - - if (core_id == SKL_DSP_CORE0_ID) { - ret = wait_event_timeout(skl->boot_wait, - skl->boot_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - - /* If core 1 was turned on for booting core 0, turn it off */ - skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); - if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", - sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), - sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; - } - } - - /* Tell FW if additional core in now On */ - - if (core_id != SKL_DSP_CORE0_ID) { - dx.core_mask = core_mask; - dx.dx_mask = core_mask; - - ret = skl_ipc_set_dx(&skl->ipc, BXT_INSTANCE_ID, - BXT_BASE_FW_MODULE_ID, &dx); - if (ret < 0) { - dev_err(ctx->dev, "IPC set_dx for core %d fail: %d\n", - core_id, ret); - goto err; - } - } - - skl->cores.state[core_id] = SKL_DSP_RUNNING; - return 0; -err: - if (core_id == SKL_DSP_CORE0_ID) - core_mask |= SKL_DSP_CORE_MASK(1); - skl_dsp_disable_core(ctx, core_mask); - - return ret; -} - -static int bxt_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id) -{ - int ret; - struct skl_ipc_dxstate_info dx; - struct skl_dev *skl = ctx->thread_context; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - - dx.core_mask = core_mask; - dx.dx_mask = SKL_IPC_D3_MASK; - - dev_dbg(ctx->dev, "core mask=%x dx_mask=%x\n", - dx.core_mask, dx.dx_mask); - - ret = skl_ipc_set_dx(&skl->ipc, BXT_INSTANCE_ID, - BXT_BASE_FW_MODULE_ID, &dx); - if (ret < 0) { - dev_err(ctx->dev, - "Failed to set DSP to D3:core id = %d;Continue reset\n", - core_id); - /* - * In case of D3 failure, re-download the firmware, so set - * fw_loaded to false. - */ - skl->fw_loaded = false; - } - - if (core_id == SKL_DSP_CORE0_ID) { - /* disable Interrupt */ - skl_ipc_op_int_disable(ctx); - skl_ipc_int_disable(ctx); - } - ret = skl_dsp_disable_core(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "Failed to disable core %d\n", ret); - return ret; - } - skl->cores.state[core_id] = SKL_DSP_RESET; - return 0; -} - -static const struct skl_dsp_fw_ops bxt_fw_ops = { - .set_state_D0 = bxt_set_dsp_D0, - .set_state_D3 = bxt_set_dsp_D3, - .set_state_D0i3 = bxt_schedule_dsp_D0i3, - .set_state_D0i0 = bxt_set_dsp_D0i0, - .load_fw = bxt_load_base_firmware, - .get_fw_errcode = bxt_get_errorcode, - .load_library = bxt_load_library, -}; - -static struct sst_ops skl_ops = { - .irq_handler = skl_dsp_sst_interrupt, - .write = sst_shim32_write, - .read = sst_shim32_read, - .free = skl_dsp_free, -}; - -static struct sst_dsp_device skl_dev = { - .thread = skl_dsp_irq_thread_handler, - .ops = &skl_ops, -}; - -int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp) -{ - struct skl_dev *skl; - struct sst_dsp *sst; - int ret; - - ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); - if (ret < 0) { - dev_err(dev, "%s: no device\n", __func__); - return ret; - } - - skl = *dsp; - sst = skl->dsp; - sst->fw_ops = bxt_fw_ops; - sst->addr.lpe = mmio_base; - sst->addr.shim = mmio_base; - sst->addr.sram0_base = BXT_ADSP_SRAM0_BASE; - sst->addr.sram1_base = BXT_ADSP_SRAM1_BASE; - sst->addr.w0_stat_sz = SKL_ADSP_W0_STAT_SZ; - sst->addr.w0_up_sz = SKL_ADSP_W0_UP_SZ; - - sst_dsp_mailbox_init(sst, (BXT_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), - SKL_ADSP_W0_UP_SZ, BXT_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); - - ret = skl_ipc_init(dev, skl); - if (ret) { - skl_dsp_free(sst); - return ret; - } - - /* set the D0i3 check */ - skl->ipc.ops.check_dsp_lp_on = skl_ipc_check_D0i0; - - skl->boot_complete = false; - init_waitqueue_head(&skl->boot_wait); - INIT_DELAYED_WORK(&skl->d0i3.work, bxt_set_dsp_D0i3); - skl->d0i3.state = SKL_DSP_D0I3_NONE; - - return skl_dsp_acquire_irq(sst); -} -EXPORT_SYMBOL_GPL(bxt_sst_dsp_init); - -int bxt_sst_init_fw(struct device *dev, struct skl_dev *skl) -{ - int ret; - struct sst_dsp *sst = skl->dsp; - - ret = sst->fw_ops.load_fw(sst); - if (ret < 0) { - dev_err(dev, "Load base fw failed: %x\n", ret); - return ret; - } - - skl_dsp_init_core_state(sst); - - if (skl->lib_count > 1) { - ret = sst->fw_ops.load_library(sst, skl->lib_info, - skl->lib_count); - if (ret < 0) { - dev_err(dev, "Load Library failed : %x\n", ret); - return ret; - } - } - skl->is_first_boot = false; - - return 0; -} -EXPORT_SYMBOL_GPL(bxt_sst_init_fw); - -void bxt_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl) -{ - - skl_release_library(skl->lib_info, skl->lib_count); - if (skl->dsp->fw) - release_firmware(skl->dsp->fw); - skl_freeup_uuid_list(skl); - skl_ipc_free(&skl->ipc); - skl->dsp->ops->free(skl->dsp); -} -EXPORT_SYMBOL_GPL(bxt_sst_dsp_cleanup); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Broxton IPC driver"); diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.c b/sound/soc/intel/skylake/cnl-sst-dsp.c deleted file mode 100644 index 3ef1b194add1..000000000000 --- a/sound/soc/intel/skylake/cnl-sst-dsp.c +++ /dev/null @@ -1,266 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * cnl-sst-dsp.c - CNL SST library generic function - * - * Copyright (C) 2016-17, Intel Corporation. - * Author: Guneshwor Singh - * - * Modified from: - * SKL SST library generic function - * Copyright (C) 2014-15, Intel Corporation. - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ -#include -#include "../common/sst-dsp.h" -#include "../common/sst-ipc.h" -#include "../common/sst-dsp-priv.h" -#include "cnl-sst-dsp.h" - -/* various timeout values */ -#define CNL_DSP_PU_TO 50 -#define CNL_DSP_PD_TO 50 -#define CNL_DSP_RESET_TO 50 - -static int -cnl_dsp_core_set_reset_state(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, - CNL_ADSP_REG_ADSPCS, CNL_ADSPCS_CRST(core_mask), - CNL_ADSPCS_CRST(core_mask)); - - /* poll with timeout to check if operation successful */ - return sst_dsp_register_poll(ctx, - CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CRST(core_mask), - CNL_ADSPCS_CRST(core_mask), - CNL_DSP_RESET_TO, - "Set reset"); -} - -static int -cnl_dsp_core_unset_reset_state(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CRST(core_mask), 0); - - /* poll with timeout to check if operation successful */ - return sst_dsp_register_poll(ctx, - CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CRST(core_mask), - 0, - CNL_DSP_RESET_TO, - "Unset reset"); -} - -static bool is_cnl_dsp_core_enable(struct sst_dsp *ctx, unsigned int core_mask) -{ - int val; - bool is_enable; - - val = sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPCS); - - is_enable = (val & CNL_ADSPCS_CPA(core_mask)) && - (val & CNL_ADSPCS_SPA(core_mask)) && - !(val & CNL_ADSPCS_CRST(core_mask)) && - !(val & CNL_ADSPCS_CSTALL(core_mask)); - - dev_dbg(ctx->dev, "DSP core(s) enabled? %d: core_mask %#x\n", - is_enable, core_mask); - - return is_enable; -} - -static int cnl_dsp_reset_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* stall core */ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CSTALL(core_mask), - CNL_ADSPCS_CSTALL(core_mask)); - - /* set reset state */ - return cnl_dsp_core_set_reset_state(ctx, core_mask); -} - -static int cnl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* unset reset state */ - ret = cnl_dsp_core_unset_reset_state(ctx, core_mask); - if (ret < 0) - return ret; - - /* run core */ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CSTALL(core_mask), 0); - - if (!is_cnl_dsp_core_enable(ctx, core_mask)) { - cnl_dsp_reset_core(ctx, core_mask); - dev_err(ctx->dev, "DSP core mask %#x enable failed\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -static int cnl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_SPA(core_mask), - CNL_ADSPCS_SPA(core_mask)); - - /* poll with timeout to check if operation successful */ - return sst_dsp_register_poll(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CPA(core_mask), - CNL_ADSPCS_CPA(core_mask), - CNL_DSP_PU_TO, - "Power up"); -} - -static int cnl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_SPA(core_mask), 0); - - /* poll with timeout to check if operation successful */ - return sst_dsp_register_poll(ctx, - CNL_ADSP_REG_ADSPCS, - CNL_ADSPCS_CPA(core_mask), - 0, - CNL_DSP_PD_TO, - "Power down"); -} - -int cnl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* power up */ - ret = cnl_dsp_core_power_up(ctx, core_mask); - if (ret < 0) { - dev_dbg(ctx->dev, "DSP core mask %#x power up failed", - core_mask); - return ret; - } - - return cnl_dsp_start_core(ctx, core_mask); -} - -int cnl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - ret = cnl_dsp_reset_core(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "DSP core mask %#x reset failed\n", - core_mask); - return ret; - } - - /* power down core*/ - ret = cnl_dsp_core_power_down(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "DSP core mask %#x power down failed\n", - core_mask); - return ret; - } - - if (is_cnl_dsp_core_enable(ctx, core_mask)) { - dev_err(ctx->dev, "DSP core mask %#x disable failed\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -irqreturn_t cnl_dsp_sst_interrupt(int irq, void *dev_id) -{ - struct sst_dsp *ctx = dev_id; - u32 val; - irqreturn_t ret = IRQ_NONE; - - spin_lock(&ctx->spinlock); - - val = sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPIS); - ctx->intr_status = val; - - if (val == 0xffffffff) { - spin_unlock(&ctx->spinlock); - return IRQ_NONE; - } - - if (val & CNL_ADSPIS_IPC) { - cnl_ipc_int_disable(ctx); - ret = IRQ_WAKE_THREAD; - } - - spin_unlock(&ctx->spinlock); - - return ret; -} - -void cnl_dsp_free(struct sst_dsp *dsp) -{ - cnl_ipc_int_disable(dsp); - - free_irq(dsp->irq, dsp); - cnl_ipc_op_int_disable(dsp); - cnl_dsp_disable_core(dsp, SKL_DSP_CORE0_MASK); -} -EXPORT_SYMBOL_GPL(cnl_dsp_free); - -void cnl_ipc_int_enable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_ADSPIC, - CNL_ADSPIC_IPC, CNL_ADSPIC_IPC); -} - -void cnl_ipc_int_disable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits_unlocked(ctx, CNL_ADSP_REG_ADSPIC, - CNL_ADSPIC_IPC, 0); -} - -void cnl_ipc_op_int_enable(struct sst_dsp *ctx) -{ - /* enable IPC DONE interrupt */ - sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_DONE, - CNL_ADSP_REG_HIPCCTL_DONE); - - /* enable IPC BUSY interrupt */ - sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_BUSY, - CNL_ADSP_REG_HIPCCTL_BUSY); -} - -void cnl_ipc_op_int_disable(struct sst_dsp *ctx) -{ - /* disable IPC DONE interrupt */ - sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_DONE, 0); - - /* disable IPC BUSY interrupt */ - sst_dsp_shim_update_bits(ctx, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_BUSY, 0); -} - -bool cnl_ipc_int_status(struct sst_dsp *ctx) -{ - return sst_dsp_shim_read_unlocked(ctx, CNL_ADSP_REG_ADSPIS) & - CNL_ADSPIS_IPC; -} - -void cnl_ipc_free(struct sst_generic_ipc *ipc) -{ - cnl_ipc_op_int_disable(ipc->dsp); - sst_ipc_fini(ipc); -} diff --git a/sound/soc/intel/skylake/cnl-sst-dsp.h b/sound/soc/intel/skylake/cnl-sst-dsp.h deleted file mode 100644 index d3cf4bd1a070..000000000000 --- a/sound/soc/intel/skylake/cnl-sst-dsp.h +++ /dev/null @@ -1,103 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Cannonlake SST DSP Support - * - * Copyright (C) 2016-17, Intel Corporation. - */ - -#ifndef __CNL_SST_DSP_H__ -#define __CNL_SST_DSP_H__ - -struct sst_dsp; -struct sst_dsp_device; -struct sst_generic_ipc; - -/* Intel HD Audio General DSP Registers */ -#define CNL_ADSP_GEN_BASE 0x0 -#define CNL_ADSP_REG_ADSPCS (CNL_ADSP_GEN_BASE + 0x04) -#define CNL_ADSP_REG_ADSPIC (CNL_ADSP_GEN_BASE + 0x08) -#define CNL_ADSP_REG_ADSPIS (CNL_ADSP_GEN_BASE + 0x0c) - -/* Intel HD Audio Inter-Processor Communication Registers */ -#define CNL_ADSP_IPC_BASE 0xc0 -#define CNL_ADSP_REG_HIPCTDR (CNL_ADSP_IPC_BASE + 0x00) -#define CNL_ADSP_REG_HIPCTDA (CNL_ADSP_IPC_BASE + 0x04) -#define CNL_ADSP_REG_HIPCTDD (CNL_ADSP_IPC_BASE + 0x08) -#define CNL_ADSP_REG_HIPCIDR (CNL_ADSP_IPC_BASE + 0x10) -#define CNL_ADSP_REG_HIPCIDA (CNL_ADSP_IPC_BASE + 0x14) -#define CNL_ADSP_REG_HIPCIDD (CNL_ADSP_IPC_BASE + 0x18) -#define CNL_ADSP_REG_HIPCCTL (CNL_ADSP_IPC_BASE + 0x28) - -/* HIPCTDR */ -#define CNL_ADSP_REG_HIPCTDR_BUSY BIT(31) - -/* HIPCTDA */ -#define CNL_ADSP_REG_HIPCTDA_DONE BIT(31) - -/* HIPCIDR */ -#define CNL_ADSP_REG_HIPCIDR_BUSY BIT(31) - -/* HIPCIDA */ -#define CNL_ADSP_REG_HIPCIDA_DONE BIT(31) - -/* CNL HIPCCTL */ -#define CNL_ADSP_REG_HIPCCTL_DONE BIT(1) -#define CNL_ADSP_REG_HIPCCTL_BUSY BIT(0) - -/* CNL HIPCT */ -#define CNL_ADSP_REG_HIPCT_BUSY BIT(31) - -/* Intel HD Audio SRAM Window 1 */ -#define CNL_ADSP_SRAM1_BASE 0xa0000 - -#define CNL_ADSP_MMIO_LEN 0x10000 - -#define CNL_ADSP_W0_STAT_SZ 0x1000 - -#define CNL_ADSP_W0_UP_SZ 0x1000 - -#define CNL_ADSP_W1_SZ 0x1000 - -#define CNL_FW_STS_MASK 0xf - -#define CNL_ADSPIC_IPC 0x1 -#define CNL_ADSPIS_IPC 0x1 - -#define CNL_DSP_CORES 4 -#define CNL_DSP_CORES_MASK ((1 << CNL_DSP_CORES) - 1) - -/* core reset - asserted high */ -#define CNL_ADSPCS_CRST_SHIFT 0 -#define CNL_ADSPCS_CRST(x) (x << CNL_ADSPCS_CRST_SHIFT) - -/* core run/stall - when set to 1 core is stalled */ -#define CNL_ADSPCS_CSTALL_SHIFT 8 -#define CNL_ADSPCS_CSTALL(x) (x << CNL_ADSPCS_CSTALL_SHIFT) - -/* set power active - when set to 1 turn core on */ -#define CNL_ADSPCS_SPA_SHIFT 16 -#define CNL_ADSPCS_SPA(x) (x << CNL_ADSPCS_SPA_SHIFT) - -/* current power active - power status of cores, set by hardware */ -#define CNL_ADSPCS_CPA_SHIFT 24 -#define CNL_ADSPCS_CPA(x) (x << CNL_ADSPCS_CPA_SHIFT) - -int cnl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask); -int cnl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask); -irqreturn_t cnl_dsp_sst_interrupt(int irq, void *dev_id); -void cnl_dsp_free(struct sst_dsp *dsp); - -void cnl_ipc_int_enable(struct sst_dsp *ctx); -void cnl_ipc_int_disable(struct sst_dsp *ctx); -void cnl_ipc_op_int_enable(struct sst_dsp *ctx); -void cnl_ipc_op_int_disable(struct sst_dsp *ctx); -bool cnl_ipc_int_status(struct sst_dsp *ctx); -void cnl_ipc_free(struct sst_generic_ipc *ipc); - -int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp); -int cnl_sst_init_fw(struct device *dev, struct skl_dev *skl); -void cnl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl); - -#endif /*__CNL_SST_DSP_H__*/ diff --git a/sound/soc/intel/skylake/cnl-sst.c b/sound/soc/intel/skylake/cnl-sst.c deleted file mode 100644 index 1275c149acc0..000000000000 --- a/sound/soc/intel/skylake/cnl-sst.c +++ /dev/null @@ -1,508 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * cnl-sst.c - DSP library functions for CNL platform - * - * Copyright (C) 2016-17, Intel Corporation. - * - * Author: Guneshwor Singh - * - * Modified from: - * HDA DSP library functions for SKL platform - * Copyright (C) 2014-15, Intel Corporation. - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include - -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "../common/sst-ipc.h" -#include "cnl-sst-dsp.h" -#include "skl.h" - -#define CNL_FW_ROM_INIT 0x1 -#define CNL_FW_INIT 0x5 -#define CNL_IPC_PURGE 0x01004000 -#define CNL_INIT_TIMEOUT 300 -#define CNL_BASEFW_TIMEOUT 3000 - -#define CNL_ADSP_SRAM0_BASE 0x80000 - -/* Firmware status window */ -#define CNL_ADSP_FW_STATUS CNL_ADSP_SRAM0_BASE -#define CNL_ADSP_ERROR_CODE (CNL_ADSP_FW_STATUS + 0x4) - -#define CNL_INSTANCE_ID 0 -#define CNL_BASE_FW_MODULE_ID 0 -#define CNL_ADSP_FW_HDR_OFFSET 0x2000 -#define CNL_ROM_CTRL_DMA_ID 0x9 - -static int cnl_prepare_fw(struct sst_dsp *ctx, const void *fwdata, u32 fwsize) -{ - - int ret, stream_tag; - - stream_tag = ctx->dsp_ops.prepare(ctx->dev, 0x40, fwsize, &ctx->dmab); - if (stream_tag <= 0) { - dev_err(ctx->dev, "dma prepare failed: 0%#x\n", stream_tag); - return stream_tag; - } - - ctx->dsp_ops.stream_tag = stream_tag; - memcpy(ctx->dmab.area, fwdata, fwsize); - - ret = skl_dsp_core_power_up(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "dsp core0 power up failed\n"); - ret = -EIO; - goto base_fw_load_failed; - } - - /* purge FW request */ - sst_dsp_shim_write(ctx, CNL_ADSP_REG_HIPCIDR, - CNL_ADSP_REG_HIPCIDR_BUSY | (CNL_IPC_PURGE | - ((stream_tag - 1) << CNL_ROM_CTRL_DMA_ID))); - - ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "Start dsp core failed ret: %d\n", ret); - ret = -EIO; - goto base_fw_load_failed; - } - - ret = sst_dsp_register_poll(ctx, CNL_ADSP_REG_HIPCIDA, - CNL_ADSP_REG_HIPCIDA_DONE, - CNL_ADSP_REG_HIPCIDA_DONE, - BXT_INIT_TIMEOUT, "HIPCIDA Done"); - if (ret < 0) { - dev_err(ctx->dev, "timeout for purge request: %d\n", ret); - goto base_fw_load_failed; - } - - /* enable interrupt */ - cnl_ipc_int_enable(ctx); - cnl_ipc_op_int_enable(ctx); - - ret = sst_dsp_register_poll(ctx, CNL_ADSP_FW_STATUS, CNL_FW_STS_MASK, - CNL_FW_ROM_INIT, CNL_INIT_TIMEOUT, - "rom load"); - if (ret < 0) { - dev_err(ctx->dev, "rom init timeout, ret: %d\n", ret); - goto base_fw_load_failed; - } - - return 0; - -base_fw_load_failed: - ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, stream_tag); - cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - - return ret; -} - -static int sst_transfer_fw_host_dma(struct sst_dsp *ctx) -{ - int ret; - - ctx->dsp_ops.trigger(ctx->dev, true, ctx->dsp_ops.stream_tag); - ret = sst_dsp_register_poll(ctx, CNL_ADSP_FW_STATUS, CNL_FW_STS_MASK, - CNL_FW_INIT, CNL_BASEFW_TIMEOUT, - "firmware boot"); - - ctx->dsp_ops.trigger(ctx->dev, false, ctx->dsp_ops.stream_tag); - ctx->dsp_ops.cleanup(ctx->dev, &ctx->dmab, ctx->dsp_ops.stream_tag); - - return ret; -} - -static int cnl_load_base_firmware(struct sst_dsp *ctx) -{ - struct firmware stripped_fw; - struct skl_dev *cnl = ctx->thread_context; - int ret, i; - - if (!ctx->fw) { - ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); - if (ret < 0) { - dev_err(ctx->dev, "request firmware failed: %d\n", ret); - goto cnl_load_base_firmware_failed; - } - } - - /* parse uuids if first boot */ - if (cnl->is_first_boot) { - ret = snd_skl_parse_uuids(ctx, ctx->fw, - CNL_ADSP_FW_HDR_OFFSET, 0); - if (ret < 0) - goto cnl_load_base_firmware_failed; - } - - stripped_fw.data = ctx->fw->data; - stripped_fw.size = ctx->fw->size; - skl_dsp_strip_extended_manifest(&stripped_fw); - - for (i = 0; i < BXT_FW_ROM_INIT_RETRY; i++) { - ret = cnl_prepare_fw(ctx, stripped_fw.data, stripped_fw.size); - if (!ret) - break; - dev_dbg(ctx->dev, "prepare firmware failed: %d\n", ret); - } - - if (ret < 0) - goto cnl_load_base_firmware_failed; - - ret = sst_transfer_fw_host_dma(ctx); - if (ret < 0) { - dev_err(ctx->dev, "transfer firmware failed: %d\n", ret); - cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - goto cnl_load_base_firmware_failed; - } - - ret = wait_event_timeout(cnl->boot_wait, cnl->boot_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - if (ret == 0) { - dev_err(ctx->dev, "FW ready timed-out\n"); - cnl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - ret = -EIO; - goto cnl_load_base_firmware_failed; - } - - cnl->fw_loaded = true; - - return 0; - -cnl_load_base_firmware_failed: - dev_err(ctx->dev, "firmware load failed: %d\n", ret); - release_firmware(ctx->fw); - ctx->fw = NULL; - - return ret; -} - -static int cnl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) -{ - struct skl_dev *cnl = ctx->thread_context; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - struct skl_ipc_dxstate_info dx; - int ret; - - if (!cnl->fw_loaded) { - cnl->boot_complete = false; - ret = cnl_load_base_firmware(ctx); - if (ret < 0) { - dev_err(ctx->dev, "fw reload failed: %d\n", ret); - return ret; - } - - cnl->cores.state[core_id] = SKL_DSP_RUNNING; - return ret; - } - - ret = cnl_dsp_enable_core(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "enable dsp core %d failed: %d\n", - core_id, ret); - goto err; - } - - if (core_id == SKL_DSP_CORE0_ID) { - /* enable interrupt */ - cnl_ipc_int_enable(ctx); - cnl_ipc_op_int_enable(ctx); - cnl->boot_complete = false; - - ret = wait_event_timeout(cnl->boot_wait, cnl->boot_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - if (ret == 0) { - dev_err(ctx->dev, - "dsp boot timeout, status=%#x error=%#x\n", - sst_dsp_shim_read(ctx, CNL_ADSP_FW_STATUS), - sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE)); - ret = -ETIMEDOUT; - goto err; - } - } else { - dx.core_mask = core_mask; - dx.dx_mask = core_mask; - - ret = skl_ipc_set_dx(&cnl->ipc, CNL_INSTANCE_ID, - CNL_BASE_FW_MODULE_ID, &dx); - if (ret < 0) { - dev_err(ctx->dev, "set_dx failed, core: %d ret: %d\n", - core_id, ret); - goto err; - } - } - cnl->cores.state[core_id] = SKL_DSP_RUNNING; - - return 0; -err: - cnl_dsp_disable_core(ctx, core_mask); - - return ret; -} - -static int cnl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id) -{ - struct skl_dev *cnl = ctx->thread_context; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - struct skl_ipc_dxstate_info dx; - int ret; - - dx.core_mask = core_mask; - dx.dx_mask = SKL_IPC_D3_MASK; - - ret = skl_ipc_set_dx(&cnl->ipc, CNL_INSTANCE_ID, - CNL_BASE_FW_MODULE_ID, &dx); - if (ret < 0) { - dev_err(ctx->dev, - "dsp core %d to d3 failed; continue reset\n", - core_id); - cnl->fw_loaded = false; - } - - /* disable interrupts if core 0 */ - if (core_id == SKL_DSP_CORE0_ID) { - skl_ipc_op_int_disable(ctx); - skl_ipc_int_disable(ctx); - } - - ret = cnl_dsp_disable_core(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "disable dsp core %d failed: %d\n", - core_id, ret); - return ret; - } - - cnl->cores.state[core_id] = SKL_DSP_RESET; - - return ret; -} - -static unsigned int cnl_get_errno(struct sst_dsp *ctx) -{ - return sst_dsp_shim_read(ctx, CNL_ADSP_ERROR_CODE); -} - -static const struct skl_dsp_fw_ops cnl_fw_ops = { - .set_state_D0 = cnl_set_dsp_D0, - .set_state_D3 = cnl_set_dsp_D3, - .load_fw = cnl_load_base_firmware, - .get_fw_errcode = cnl_get_errno, -}; - -static struct sst_ops cnl_ops = { - .irq_handler = cnl_dsp_sst_interrupt, - .write = sst_shim32_write, - .read = sst_shim32_read, - .free = cnl_dsp_free, -}; - -#define CNL_IPC_GLB_NOTIFY_RSP_SHIFT 29 -#define CNL_IPC_GLB_NOTIFY_RSP_MASK 0x1 -#define CNL_IPC_GLB_NOTIFY_RSP_TYPE(x) (((x) >> CNL_IPC_GLB_NOTIFY_RSP_SHIFT) \ - & CNL_IPC_GLB_NOTIFY_RSP_MASK) - -static irqreturn_t cnl_dsp_irq_thread_handler(int irq, void *context) -{ - struct sst_dsp *dsp = context; - struct skl_dev *cnl = dsp->thread_context; - struct sst_generic_ipc *ipc = &cnl->ipc; - struct skl_ipc_header header = {0}; - u32 hipcida, hipctdr, hipctdd; - int ipc_irq = 0; - - /* here we handle ipc interrupts only */ - if (!(dsp->intr_status & CNL_ADSPIS_IPC)) - return IRQ_NONE; - - hipcida = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCIDA); - hipctdr = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCTDR); - hipctdd = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCTDD); - - /* reply message from dsp */ - if (hipcida & CNL_ADSP_REG_HIPCIDA_DONE) { - sst_dsp_shim_update_bits(dsp, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_DONE, 0); - - /* clear done bit - tell dsp operation is complete */ - sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCIDA, - CNL_ADSP_REG_HIPCIDA_DONE, CNL_ADSP_REG_HIPCIDA_DONE); - - ipc_irq = 1; - - /* unmask done interrupt */ - sst_dsp_shim_update_bits(dsp, CNL_ADSP_REG_HIPCCTL, - CNL_ADSP_REG_HIPCCTL_DONE, CNL_ADSP_REG_HIPCCTL_DONE); - } - - /* new message from dsp */ - if (hipctdr & CNL_ADSP_REG_HIPCTDR_BUSY) { - header.primary = hipctdr; - header.extension = hipctdd; - dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x", - header.primary); - dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x", - header.extension); - - if (CNL_IPC_GLB_NOTIFY_RSP_TYPE(header.primary)) { - /* Handle Immediate reply from DSP Core */ - skl_ipc_process_reply(ipc, header); - } else { - dev_dbg(dsp->dev, "IPC irq: Notification from firmware\n"); - skl_ipc_process_notification(ipc, header); - } - /* clear busy interrupt */ - sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCTDR, - CNL_ADSP_REG_HIPCTDR_BUSY, CNL_ADSP_REG_HIPCTDR_BUSY); - - /* set done bit to ack dsp */ - sst_dsp_shim_update_bits_forced(dsp, CNL_ADSP_REG_HIPCTDA, - CNL_ADSP_REG_HIPCTDA_DONE, CNL_ADSP_REG_HIPCTDA_DONE); - ipc_irq = 1; - } - - if (ipc_irq == 0) - return IRQ_NONE; - - cnl_ipc_int_enable(dsp); - - /* continue to send any remaining messages */ - schedule_work(&ipc->kwork); - - return IRQ_HANDLED; -} - -static struct sst_dsp_device cnl_dev = { - .thread = cnl_dsp_irq_thread_handler, - .ops = &cnl_ops, -}; - -static void cnl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg) -{ - struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->tx.header); - - if (msg->tx.size) - sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size); - sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDD, - header->extension); - sst_dsp_shim_write_unlocked(ipc->dsp, CNL_ADSP_REG_HIPCIDR, - header->primary | CNL_ADSP_REG_HIPCIDR_BUSY); -} - -static bool cnl_ipc_is_dsp_busy(struct sst_dsp *dsp) -{ - u32 hipcidr; - - hipcidr = sst_dsp_shim_read_unlocked(dsp, CNL_ADSP_REG_HIPCIDR); - - return (hipcidr & CNL_ADSP_REG_HIPCIDR_BUSY); -} - -static int cnl_ipc_init(struct device *dev, struct skl_dev *cnl) -{ - struct sst_generic_ipc *ipc; - int err; - - ipc = &cnl->ipc; - ipc->dsp = cnl->dsp; - ipc->dev = dev; - - ipc->tx_data_max_size = CNL_ADSP_W1_SZ; - ipc->rx_data_max_size = CNL_ADSP_W0_UP_SZ; - - err = sst_ipc_init(ipc); - if (err) - return err; - - /* - * overriding tx_msg and is_dsp_busy since - * ipc registers are different for cnl - */ - ipc->ops.tx_msg = cnl_ipc_tx_msg; - ipc->ops.tx_data_copy = skl_ipc_tx_data_copy; - ipc->ops.is_dsp_busy = cnl_ipc_is_dsp_busy; - - return 0; -} - -int cnl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp) -{ - struct skl_dev *cnl; - struct sst_dsp *sst; - int ret; - - ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &cnl_dev); - if (ret < 0) { - dev_err(dev, "%s: no device\n", __func__); - return ret; - } - - cnl = *dsp; - sst = cnl->dsp; - sst->fw_ops = cnl_fw_ops; - sst->addr.lpe = mmio_base; - sst->addr.shim = mmio_base; - sst->addr.sram0_base = CNL_ADSP_SRAM0_BASE; - sst->addr.sram1_base = CNL_ADSP_SRAM1_BASE; - sst->addr.w0_stat_sz = CNL_ADSP_W0_STAT_SZ; - sst->addr.w0_up_sz = CNL_ADSP_W0_UP_SZ; - - sst_dsp_mailbox_init(sst, (CNL_ADSP_SRAM0_BASE + CNL_ADSP_W0_STAT_SZ), - CNL_ADSP_W0_UP_SZ, CNL_ADSP_SRAM1_BASE, - CNL_ADSP_W1_SZ); - - ret = cnl_ipc_init(dev, cnl); - if (ret) { - skl_dsp_free(sst); - return ret; - } - - cnl->boot_complete = false; - init_waitqueue_head(&cnl->boot_wait); - - return skl_dsp_acquire_irq(sst); -} -EXPORT_SYMBOL_GPL(cnl_sst_dsp_init); - -int cnl_sst_init_fw(struct device *dev, struct skl_dev *skl) -{ - int ret; - struct sst_dsp *sst = skl->dsp; - - ret = skl->dsp->fw_ops.load_fw(sst); - if (ret < 0) { - dev_err(dev, "load base fw failed: %d", ret); - return ret; - } - - skl_dsp_init_core_state(sst); - - skl->is_first_boot = false; - - return 0; -} -EXPORT_SYMBOL_GPL(cnl_sst_init_fw); - -void cnl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl) -{ - if (skl->dsp->fw) - release_firmware(skl->dsp->fw); - - skl_freeup_uuid_list(skl); - cnl_ipc_free(&skl->ipc); - - skl->dsp->ops->free(skl->dsp); -} -EXPORT_SYMBOL_GPL(cnl_sst_dsp_cleanup); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Cannonlake IPC driver"); diff --git a/sound/soc/intel/skylake/skl-debug.c b/sound/soc/intel/skylake/skl-debug.c deleted file mode 100644 index a15aa2ffa681..000000000000 --- a/sound/soc/intel/skylake/skl-debug.c +++ /dev/null @@ -1,248 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-debug.c - Debugfs for skl driver - * - * Copyright (C) 2016-17 Intel Corp - */ - -#include -#include -#include -#include "skl.h" -#include "skl-sst-dsp.h" -#include "skl-sst-ipc.h" -#include "skl-topology.h" -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" - -#define MOD_BUF PAGE_SIZE -#define FW_REG_BUF PAGE_SIZE -#define FW_REG_SIZE 0x60 - -struct skl_debug { - struct skl_dev *skl; - struct device *dev; - - struct dentry *fs; - struct dentry *modules; - u8 fw_read_buff[FW_REG_BUF]; -}; - -static ssize_t skl_print_pins(struct skl_module_pin *m_pin, char *buf, - int max_pin, ssize_t size, bool direction) -{ - int i; - ssize_t ret = 0; - - for (i = 0; i < max_pin; i++) { - ret += scnprintf(buf + size, MOD_BUF - size, - "%s %d\n\tModule %d\n\tInstance %d\n\t" - "In-used %s\n\tType %s\n" - "\tState %d\n\tIndex %d\n", - direction ? "Input Pin:" : "Output Pin:", - i, m_pin[i].id.module_id, - m_pin[i].id.instance_id, - m_pin[i].in_use ? "Used" : "Unused", - m_pin[i].is_dynamic ? "Dynamic" : "Static", - m_pin[i].pin_state, i); - size += ret; - } - return ret; -} - -static ssize_t skl_print_fmt(struct skl_module_fmt *fmt, char *buf, - ssize_t size, bool direction) -{ - return scnprintf(buf + size, MOD_BUF - size, - "%s\n\tCh %d\n\tFreq %d\n\tBit depth %d\n\t" - "Valid bit depth %d\n\tCh config %#x\n\tInterleaving %d\n\t" - "Sample Type %d\n\tCh Map %#x\n", - direction ? "Input Format:" : "Output Format:", - fmt->channels, fmt->s_freq, fmt->bit_depth, - fmt->valid_bit_depth, fmt->ch_cfg, - fmt->interleaving_style, fmt->sample_type, - fmt->ch_map); -} - -static ssize_t module_read(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - struct skl_module_cfg *mconfig = file->private_data; - struct skl_module *module = mconfig->module; - struct skl_module_res *res = &module->resources[mconfig->res_idx]; - char *buf; - ssize_t ret; - - buf = kzalloc(MOD_BUF, GFP_KERNEL); - if (!buf) - return -ENOMEM; - - ret = scnprintf(buf, MOD_BUF, "Module:\n\tUUID %pUL\n\tModule id %d\n" - "\tInstance id %d\n\tPvt_id %d\n", mconfig->guid, - mconfig->id.module_id, mconfig->id.instance_id, - mconfig->id.pvt_id); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Resources:\n\tCPC %#x\n\tIBS %#x\n\tOBS %#x\t\n", - res->cpc, res->ibs, res->obs); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Module data:\n\tCore %d\n\tIn queue %d\n\t" - "Out queue %d\n\tType %s\n", - mconfig->core_id, mconfig->max_in_queue, - mconfig->max_out_queue, - mconfig->is_loadable ? "loadable" : "inbuilt"); - - ret += skl_print_fmt(mconfig->in_fmt, buf, ret, true); - ret += skl_print_fmt(mconfig->out_fmt, buf, ret, false); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Fixup:\n\tParams %#x\n\tConverter %#x\n", - mconfig->params_fixup, mconfig->converter); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Module Gateway:\n\tType %#x\n\tVbus %#x\n\tHW conn %#x\n\tSlot %#x\n", - mconfig->dev_type, mconfig->vbus_id, - mconfig->hw_conn_type, mconfig->time_slot); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Pipeline:\n\tID %d\n\tPriority %d\n\tConn Type %d\n\t" - "Pages %#x\n", mconfig->pipe->ppl_id, - mconfig->pipe->pipe_priority, mconfig->pipe->conn_type, - mconfig->pipe->memory_pages); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "\tParams:\n\t\tHost DMA %d\n\t\tLink DMA %d\n", - mconfig->pipe->p_params->host_dma_id, - mconfig->pipe->p_params->link_dma_id); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "\tPCM params:\n\t\tCh %d\n\t\tFreq %d\n\t\tFormat %d\n", - mconfig->pipe->p_params->ch, - mconfig->pipe->p_params->s_freq, - mconfig->pipe->p_params->s_fmt); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "\tLink %#x\n\tStream %#x\n", - mconfig->pipe->p_params->linktype, - mconfig->pipe->p_params->stream); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "\tState %d\n\tPassthru %s\n", - mconfig->pipe->state, - mconfig->pipe->passthru ? "true" : "false"); - - ret += skl_print_pins(mconfig->m_in_pin, buf, - mconfig->max_in_queue, ret, true); - ret += skl_print_pins(mconfig->m_out_pin, buf, - mconfig->max_out_queue, ret, false); - - ret += scnprintf(buf + ret, MOD_BUF - ret, - "Other:\n\tDomain %d\n\tHomogeneous Input %s\n\t" - "Homogeneous Output %s\n\tIn Queue Mask %d\n\t" - "Out Queue Mask %d\n\tDMA ID %d\n\tMem Pages %d\n\t" - "Module Type %d\n\tModule State %d\n", - mconfig->domain, - mconfig->homogenous_inputs ? "true" : "false", - mconfig->homogenous_outputs ? "true" : "false", - mconfig->in_queue_mask, mconfig->out_queue_mask, - mconfig->dma_id, mconfig->mem_pages, mconfig->m_state, - mconfig->m_type); - - ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret); - - kfree(buf); - return ret; -} - -static const struct file_operations mcfg_fops = { - .open = simple_open, - .read = module_read, - .llseek = default_llseek, -}; - - -void skl_debug_init_module(struct skl_debug *d, - struct snd_soc_dapm_widget *w, - struct skl_module_cfg *mconfig) -{ - debugfs_create_file(w->name, 0444, d->modules, mconfig, - &mcfg_fops); -} - -static ssize_t fw_softreg_read(struct file *file, char __user *user_buf, - size_t count, loff_t *ppos) -{ - struct skl_debug *d = file->private_data; - struct sst_dsp *sst = d->skl->dsp; - size_t w0_stat_sz = sst->addr.w0_stat_sz; - void __iomem *in_base = sst->mailbox.in_base; - void __iomem *fw_reg_addr; - unsigned int offset; - char *tmp; - ssize_t ret = 0; - - tmp = kzalloc(FW_REG_BUF, GFP_KERNEL); - if (!tmp) - return -ENOMEM; - - fw_reg_addr = in_base - w0_stat_sz; - memset(d->fw_read_buff, 0, FW_REG_BUF); - - if (w0_stat_sz > 0) - __ioread32_copy(d->fw_read_buff, fw_reg_addr, w0_stat_sz >> 2); - - for (offset = 0; offset < FW_REG_SIZE; offset += 16) { - ret += scnprintf(tmp + ret, FW_REG_BUF - ret, "%#.4x: ", offset); - hex_dump_to_buffer(d->fw_read_buff + offset, 16, 16, 4, - tmp + ret, FW_REG_BUF - ret, 0); - ret += strlen(tmp + ret); - - /* print newline for each offset */ - if (FW_REG_BUF - ret > 0) - tmp[ret++] = '\n'; - } - - ret = simple_read_from_buffer(user_buf, count, ppos, tmp, ret); - kfree(tmp); - - return ret; -} - -static const struct file_operations soft_regs_ctrl_fops = { - .open = simple_open, - .read = fw_softreg_read, - .llseek = default_llseek, -}; - -struct skl_debug *skl_debugfs_init(struct skl_dev *skl) -{ - struct skl_debug *d; - - d = devm_kzalloc(&skl->pci->dev, sizeof(*d), GFP_KERNEL); - if (!d) - return NULL; - - /* create the debugfs dir with platform component's debugfs as parent */ - d->fs = debugfs_create_dir("dsp", skl->component->debugfs_root); - - d->skl = skl; - d->dev = &skl->pci->dev; - - /* now create the module dir */ - d->modules = debugfs_create_dir("modules", d->fs); - - debugfs_create_file("fw_soft_regs_rd", 0444, d->fs, d, - &soft_regs_ctrl_fops); - - return d; -} - -void skl_debugfs_exit(struct skl_dev *skl) -{ - struct skl_debug *d = skl->debugfs; - - debugfs_remove_recursive(d->fs); - - d = NULL; -} diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h deleted file mode 100644 index dfce91e11be1..000000000000 --- a/sound/soc/intel/skylake/skl-i2s.h +++ /dev/null @@ -1,87 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * skl-i2s.h - i2s blob mapping - * - * Copyright (C) 2017 Intel Corp - * Author: Subhransu S. Prusty < subhransu.s.prusty@intel.com> - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#ifndef __SOUND_SOC_SKL_I2S_H -#define __SOUND_SOC_SKL_I2S_H - -#define SKL_I2S_MAX_TIME_SLOTS 8 -#define SKL_MCLK_DIV_CLK_SRC_MASK GENMASK(17, 16) - -#define SKL_MNDSS_DIV_CLK_SRC_MASK GENMASK(21, 20) -#define SKL_SHIFT(x) (ffs(x) - 1) -#define SKL_MCLK_DIV_RATIO_MASK GENMASK(11, 0) - -#define is_legacy_blob(x) (x.signature != 0xEE) -#define ext_to_legacy_blob(i2s_config_blob_ext) \ - ((struct skl_i2s_config_blob_legacy *) i2s_config_blob_ext) - -#define get_clk_src(mclk, mask) \ - ((mclk.mdivctrl & mask) >> SKL_SHIFT(mask)) -struct skl_i2s_config { - u32 ssc0; - u32 ssc1; - u32 sscto; - u32 sspsp; - u32 sstsa; - u32 ssrsa; - u32 ssc2; - u32 sspsp2; - u32 ssc3; - u32 ssioc; -} __packed; - -struct skl_i2s_config_mclk { - u32 mdivctrl; - u32 mdivr; -}; - -struct skl_i2s_config_mclk_ext { - u32 mdivctrl; - u32 mdivr_count; - u32 mdivr[]; -} __packed; - -struct skl_i2s_config_blob_signature { - u32 minor_ver : 8; - u32 major_ver : 8; - u32 resvdz : 8; - u32 signature : 8; -} __packed; - -struct skl_i2s_config_blob_header { - struct skl_i2s_config_blob_signature sig; - u32 size; -}; - -/** - * struct skl_i2s_config_blob_legacy - Structure defines I2S Gateway - * configuration legacy blob - * - * @gtw_attr: Gateway attribute for the I2S Gateway - * @tdm_ts_group: TDM slot mapping against channels in the Gateway. - * @i2s_cfg: I2S HW registers - * @mclk: MCLK clock source and divider values - */ -struct skl_i2s_config_blob_legacy { - u32 gtw_attr; - u32 tdm_ts_group[SKL_I2S_MAX_TIME_SLOTS]; - struct skl_i2s_config i2s_cfg; - struct skl_i2s_config_mclk mclk; -}; - -struct skl_i2s_config_blob_ext { - u32 gtw_attr; - struct skl_i2s_config_blob_header hdr; - u32 tdm_ts_group[SKL_I2S_MAX_TIME_SLOTS]; - struct skl_i2s_config i2s_cfg; - struct skl_i2s_config_mclk_ext mclk; -} __packed; -#endif /* __SOUND_SOC_SKL_I2S_H */ diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c deleted file mode 100644 index fc2eb04da172..000000000000 --- a/sound/soc/intel/skylake/skl-messages.c +++ /dev/null @@ -1,1419 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-message.c - HDA DSP interface for FW registration, Pipe and Module - * configurations - * - * Copyright (C) 2015 Intel Corp - * Author:Rafal Redzimski - * Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include "skl-sst-dsp.h" -#include "cnl-sst-dsp.h" -#include "skl-sst-ipc.h" -#include "skl.h" -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "skl-topology.h" - -static int skl_alloc_dma_buf(struct device *dev, - struct snd_dma_buffer *dmab, size_t size) -{ - return snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, dmab); -} - -static int skl_free_dma_buf(struct device *dev, struct snd_dma_buffer *dmab) -{ - snd_dma_free_pages(dmab); - return 0; -} - -#define SKL_ASTATE_PARAM_ID 4 - -void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data) -{ - struct skl_ipc_large_config_msg msg = {0}; - - msg.large_param_id = SKL_ASTATE_PARAM_ID; - msg.param_data_size = (cnt * sizeof(struct skl_astate_param) + - sizeof(cnt)); - - skl_ipc_set_large_config(&skl->ipc, &msg, data); -} - -static int skl_dsp_setup_spib(struct device *dev, unsigned int size, - int stream_tag, int enable) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - struct hdac_stream *stream = snd_hdac_get_stream(bus, - SNDRV_PCM_STREAM_PLAYBACK, stream_tag); - - if (!stream) - return -EINVAL; - - /* enable/disable SPIB for this hdac stream */ - snd_hdac_stream_spbcap_enable(bus, enable, stream->index); - - /* set the spib value */ - snd_hdac_stream_set_spib(bus, stream, size); - - return 0; -} - -static int skl_dsp_prepare(struct device *dev, unsigned int format, - unsigned int size, struct snd_dma_buffer *dmab) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - struct hdac_ext_stream *estream; - struct hdac_stream *stream; - struct snd_pcm_substream substream; - int ret; - - if (!bus) - return -ENODEV; - - memset(&substream, 0, sizeof(substream)); - substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - - estream = snd_hdac_ext_stream_assign(bus, &substream, - HDAC_EXT_STREAM_TYPE_HOST); - if (!estream) - return -ENODEV; - - stream = hdac_stream(estream); - - /* assign decouple host dma channel */ - ret = snd_hdac_dsp_prepare(stream, format, size, dmab); - if (ret < 0) - return ret; - - skl_dsp_setup_spib(dev, size, stream->stream_tag, true); - - return stream->stream_tag; -} - -static int skl_dsp_trigger(struct device *dev, bool start, int stream_tag) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - struct hdac_stream *stream; - - if (!bus) - return -ENODEV; - - stream = snd_hdac_get_stream(bus, - SNDRV_PCM_STREAM_PLAYBACK, stream_tag); - if (!stream) - return -EINVAL; - - snd_hdac_dsp_trigger(stream, start); - - return 0; -} - -static int skl_dsp_cleanup(struct device *dev, - struct snd_dma_buffer *dmab, int stream_tag) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - struct hdac_stream *stream; - struct hdac_ext_stream *estream; - - if (!bus) - return -ENODEV; - - stream = snd_hdac_get_stream(bus, - SNDRV_PCM_STREAM_PLAYBACK, stream_tag); - if (!stream) - return -EINVAL; - - estream = stream_to_hdac_ext_stream(stream); - skl_dsp_setup_spib(dev, 0, stream_tag, false); - snd_hdac_ext_stream_release(estream, HDAC_EXT_STREAM_TYPE_HOST); - - snd_hdac_dsp_cleanup(stream, dmab); - - return 0; -} - -static struct skl_dsp_loader_ops skl_get_loader_ops(void) -{ - struct skl_dsp_loader_ops loader_ops; - - memset(&loader_ops, 0, sizeof(struct skl_dsp_loader_ops)); - - loader_ops.alloc_dma_buf = skl_alloc_dma_buf; - loader_ops.free_dma_buf = skl_free_dma_buf; - - return loader_ops; -}; - -static struct skl_dsp_loader_ops bxt_get_loader_ops(void) -{ - struct skl_dsp_loader_ops loader_ops; - - memset(&loader_ops, 0, sizeof(loader_ops)); - - loader_ops.alloc_dma_buf = skl_alloc_dma_buf; - loader_ops.free_dma_buf = skl_free_dma_buf; - loader_ops.prepare = skl_dsp_prepare; - loader_ops.trigger = skl_dsp_trigger; - loader_ops.cleanup = skl_dsp_cleanup; - - return loader_ops; -}; - -static const struct skl_dsp_ops dsp_ops[] = { - { - .id = PCI_DEVICE_ID_INTEL_HDA_SKL_LP, - .num_cores = 2, - .loader_ops = skl_get_loader_ops, - .init = skl_sst_dsp_init, - .init_fw = skl_sst_init_fw, - .cleanup = skl_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_KBL_LP, - .num_cores = 2, - .loader_ops = skl_get_loader_ops, - .init = skl_sst_dsp_init, - .init_fw = skl_sst_init_fw, - .cleanup = skl_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_APL, - .num_cores = 2, - .loader_ops = bxt_get_loader_ops, - .init = bxt_sst_dsp_init, - .init_fw = bxt_sst_init_fw, - .cleanup = bxt_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_GML, - .num_cores = 2, - .loader_ops = bxt_get_loader_ops, - .init = bxt_sst_dsp_init, - .init_fw = bxt_sst_init_fw, - .cleanup = bxt_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_CNL_LP, - .num_cores = 4, - .loader_ops = bxt_get_loader_ops, - .init = cnl_sst_dsp_init, - .init_fw = cnl_sst_init_fw, - .cleanup = cnl_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_CNL_H, - .num_cores = 4, - .loader_ops = bxt_get_loader_ops, - .init = cnl_sst_dsp_init, - .init_fw = cnl_sst_init_fw, - .cleanup = cnl_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_CML_LP, - .num_cores = 4, - .loader_ops = bxt_get_loader_ops, - .init = cnl_sst_dsp_init, - .init_fw = cnl_sst_init_fw, - .cleanup = cnl_sst_dsp_cleanup - }, - { - .id = PCI_DEVICE_ID_INTEL_HDA_CML_H, - .num_cores = 4, - .loader_ops = bxt_get_loader_ops, - .init = cnl_sst_dsp_init, - .init_fw = cnl_sst_init_fw, - .cleanup = cnl_sst_dsp_cleanup - }, -}; - -const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(dsp_ops); i++) { - if (dsp_ops[i].id == pci_id) - return &dsp_ops[i]; - } - - return NULL; -} - -int skl_init_dsp(struct skl_dev *skl) -{ - void __iomem *mmio_base; - struct hdac_bus *bus = skl_to_bus(skl); - struct skl_dsp_loader_ops loader_ops; - int irq = bus->irq; - const struct skl_dsp_ops *ops; - struct skl_dsp_cores *cores; - int ret; - - /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(bus, true); - snd_hdac_ext_bus_ppcap_int_enable(bus, true); - - /* read the BAR of the ADSP MMIO */ - mmio_base = pci_ioremap_bar(skl->pci, 4); - if (mmio_base == NULL) { - dev_err(bus->dev, "ioremap error\n"); - return -ENXIO; - } - - ops = skl_get_dsp_ops(skl->pci->device); - if (!ops) { - ret = -EIO; - goto unmap_mmio; - } - - loader_ops = ops->loader_ops(); - ret = ops->init(bus->dev, mmio_base, irq, - skl->fw_name, loader_ops, - &skl); - - if (ret < 0) - goto unmap_mmio; - - skl->dsp_ops = ops; - cores = &skl->cores; - cores->count = ops->num_cores; - - cores->state = kcalloc(cores->count, sizeof(*cores->state), GFP_KERNEL); - if (!cores->state) { - ret = -ENOMEM; - goto unmap_mmio; - } - - cores->usage_count = kcalloc(cores->count, sizeof(*cores->usage_count), - GFP_KERNEL); - if (!cores->usage_count) { - ret = -ENOMEM; - goto free_core_state; - } - - dev_dbg(bus->dev, "dsp registration status=%d\n", ret); - - return 0; - -free_core_state: - kfree(cores->state); - -unmap_mmio: - iounmap(mmio_base); - - return ret; -} - -int skl_free_dsp(struct skl_dev *skl) -{ - struct hdac_bus *bus = skl_to_bus(skl); - - /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(bus, false); - - skl->dsp_ops->cleanup(bus->dev, skl); - - kfree(skl->cores.state); - kfree(skl->cores.usage_count); - - if (skl->dsp->addr.lpe) - iounmap(skl->dsp->addr.lpe); - - return 0; -} - -/* - * In the case of "suspend_active" i.e, the Audio IP being active - * during system suspend, immediately excecute any pending D0i3 work - * before suspending. This is needed for the IP to work in low power - * mode during system suspend. In the case of normal suspend, cancel - * any pending D0i3 work. - */ -int skl_suspend_late_dsp(struct skl_dev *skl) -{ - struct delayed_work *dwork; - - if (!skl) - return 0; - - dwork = &skl->d0i3.work; - - if (dwork->work.func) { - if (skl->supend_active) - flush_delayed_work(dwork); - else - cancel_delayed_work_sync(dwork); - } - - return 0; -} - -int skl_suspend_dsp(struct skl_dev *skl) -{ - struct hdac_bus *bus = skl_to_bus(skl); - int ret; - - /* if ppcap is not supported return 0 */ - if (!bus->ppcap) - return 0; - - ret = skl_dsp_sleep(skl->dsp); - if (ret < 0) - return ret; - - /* disable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_int_enable(bus, false); - snd_hdac_ext_bus_ppcap_enable(bus, false); - - return 0; -} - -int skl_resume_dsp(struct skl_dev *skl) -{ - struct hdac_bus *bus = skl_to_bus(skl); - int ret; - - /* if ppcap is not supported return 0 */ - if (!bus->ppcap) - return 0; - - /* enable ppcap interrupt */ - snd_hdac_ext_bus_ppcap_enable(bus, true); - snd_hdac_ext_bus_ppcap_int_enable(bus, true); - - /* check if DSP 1st boot is done */ - if (skl->is_first_boot) - return 0; - - /* - * Disable dynamic clock and power gating during firmware - * and library download - */ - skl->enable_miscbdcge(skl->dev, false); - skl->clock_power_gating(skl->dev, false); - - ret = skl_dsp_wake(skl->dsp); - skl->enable_miscbdcge(skl->dev, true); - skl->clock_power_gating(skl->dev, true); - if (ret < 0) - return ret; - - if (skl->cfg.astate_cfg != NULL) { - skl_dsp_set_astate_cfg(skl, skl->cfg.astate_cfg->count, - skl->cfg.astate_cfg); - } - return ret; -} - -enum skl_bitdepth skl_get_bit_depth(int params) -{ - switch (params) { - case 8: - return SKL_DEPTH_8BIT; - - case 16: - return SKL_DEPTH_16BIT; - - case 24: - return SKL_DEPTH_24BIT; - - case 32: - return SKL_DEPTH_32BIT; - - default: - return SKL_DEPTH_INVALID; - - } -} - -/* - * Each module in DSP expects a base module configuration, which consists of - * PCM format information, which we calculate in driver and resource values - * which are read from widget information passed through topology binary - * This is send when we create a module with INIT_INSTANCE IPC msg - */ -static void skl_set_base_module_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_base_cfg *base_cfg) -{ - struct skl_module *module = mconfig->module; - struct skl_module_res *res = &module->resources[mconfig->res_idx]; - struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx]; - struct skl_module_fmt *format = &fmt->inputs[0].fmt; - - base_cfg->audio_fmt.number_of_channels = format->channels; - - base_cfg->audio_fmt.s_freq = format->s_freq; - base_cfg->audio_fmt.bit_depth = format->bit_depth; - base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; - base_cfg->audio_fmt.ch_cfg = format->ch_cfg; - base_cfg->audio_fmt.sample_type = format->sample_type; - - dev_dbg(skl->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", - format->bit_depth, format->valid_bit_depth, - format->ch_cfg); - - base_cfg->audio_fmt.channel_map = format->ch_map; - - base_cfg->audio_fmt.interleaving = format->interleaving_style; - - base_cfg->cpc = res->cpc; - base_cfg->ibs = res->ibs; - base_cfg->obs = res->obs; - base_cfg->is_pages = res->is_pages; -} - -static void fill_pin_params(struct skl_audio_data_format *pin_fmt, - struct skl_module_fmt *format) -{ - pin_fmt->number_of_channels = format->channels; - pin_fmt->s_freq = format->s_freq; - pin_fmt->bit_depth = format->bit_depth; - pin_fmt->valid_bit_depth = format->valid_bit_depth; - pin_fmt->ch_cfg = format->ch_cfg; - pin_fmt->sample_type = format->sample_type; - pin_fmt->channel_map = format->ch_map; - pin_fmt->interleaving = format->interleaving_style; -} - -/* - * Any module configuration begins with a base module configuration but - * can be followed by a generic extension containing audio format for all - * module's pins that are in use. - */ -static void skl_set_base_ext_module_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_base_cfg_ext *base_cfg_ext) -{ - struct skl_module *module = mconfig->module; - struct skl_module_pin_resources *pin_res; - struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx]; - struct skl_module_res *res = &module->resources[mconfig->res_idx]; - struct skl_module_fmt *format; - struct skl_pin_format *pin_fmt; - char *params; - int i; - - base_cfg_ext->nr_input_pins = res->nr_input_pins; - base_cfg_ext->nr_output_pins = res->nr_output_pins; - base_cfg_ext->priv_param_length = - mconfig->formats_config[SKL_PARAM_INIT].caps_size; - - for (i = 0; i < res->nr_input_pins; i++) { - pin_res = &res->input[i]; - pin_fmt = &base_cfg_ext->pins_fmt[i]; - - pin_fmt->pin_idx = pin_res->pin_index; - pin_fmt->buf_size = pin_res->buf_size; - - format = &fmt->inputs[pin_res->pin_index].fmt; - fill_pin_params(&pin_fmt->audio_fmt, format); - } - - for (i = 0; i < res->nr_output_pins; i++) { - pin_res = &res->output[i]; - pin_fmt = &base_cfg_ext->pins_fmt[res->nr_input_pins + i]; - - pin_fmt->pin_idx = pin_res->pin_index; - pin_fmt->buf_size = pin_res->buf_size; - - format = &fmt->outputs[pin_res->pin_index].fmt; - fill_pin_params(&pin_fmt->audio_fmt, format); - } - - if (!base_cfg_ext->priv_param_length) - return; - - params = (char *)base_cfg_ext + sizeof(struct skl_base_cfg_ext); - params += (base_cfg_ext->nr_input_pins + base_cfg_ext->nr_output_pins) * - sizeof(struct skl_pin_format); - - memcpy(params, mconfig->formats_config[SKL_PARAM_INIT].caps, - mconfig->formats_config[SKL_PARAM_INIT].caps_size); -} - -/* - * Copies copier capabilities into copier module and updates copier module - * config size. - */ -static void skl_copy_copier_caps(struct skl_module_cfg *mconfig, - struct skl_cpr_cfg *cpr_mconfig) -{ - if (mconfig->formats_config[SKL_PARAM_INIT].caps_size == 0) - return; - - memcpy(&cpr_mconfig->gtw_cfg.config_data, - mconfig->formats_config[SKL_PARAM_INIT].caps, - mconfig->formats_config[SKL_PARAM_INIT].caps_size); - - cpr_mconfig->gtw_cfg.config_length = - (mconfig->formats_config[SKL_PARAM_INIT].caps_size) / 4; -} - -#define SKL_NON_GATEWAY_CPR_NODE_ID 0xFFFFFFFF -/* - * Calculate the gatewat settings required for copier module, type of - * gateway and index of gateway to use - */ -static u32 skl_get_node_id(struct skl_dev *skl, - struct skl_module_cfg *mconfig) -{ - union skl_connector_node_id node_id = {0}; - union skl_ssp_dma_node ssp_node = {0}; - struct skl_pipe_params *params = mconfig->pipe->p_params; - - switch (mconfig->dev_type) { - case SKL_DEVICE_BT: - node_id.node.dma_type = - (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? - SKL_DMA_I2S_LINK_OUTPUT_CLASS : - SKL_DMA_I2S_LINK_INPUT_CLASS; - node_id.node.vindex = params->host_dma_id + - (mconfig->vbus_id << 3); - break; - - case SKL_DEVICE_I2S: - node_id.node.dma_type = - (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? - SKL_DMA_I2S_LINK_OUTPUT_CLASS : - SKL_DMA_I2S_LINK_INPUT_CLASS; - ssp_node.dma_node.time_slot_index = mconfig->time_slot; - ssp_node.dma_node.i2s_instance = mconfig->vbus_id; - node_id.node.vindex = ssp_node.val; - break; - - case SKL_DEVICE_DMIC: - node_id.node.dma_type = SKL_DMA_DMIC_LINK_INPUT_CLASS; - node_id.node.vindex = mconfig->vbus_id + - (mconfig->time_slot); - break; - - case SKL_DEVICE_HDALINK: - node_id.node.dma_type = - (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? - SKL_DMA_HDA_LINK_OUTPUT_CLASS : - SKL_DMA_HDA_LINK_INPUT_CLASS; - node_id.node.vindex = params->link_dma_id; - break; - - case SKL_DEVICE_HDAHOST: - node_id.node.dma_type = - (SKL_CONN_SOURCE == mconfig->hw_conn_type) ? - SKL_DMA_HDA_HOST_OUTPUT_CLASS : - SKL_DMA_HDA_HOST_INPUT_CLASS; - node_id.node.vindex = params->host_dma_id; - break; - - default: - node_id.val = 0xFFFFFFFF; - break; - } - - return node_id.val; -} - -static void skl_setup_cpr_gateway_cfg(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_cpr_cfg *cpr_mconfig) -{ - u32 dma_io_buf; - struct skl_module_res *res; - int res_idx = mconfig->res_idx; - - cpr_mconfig->gtw_cfg.node_id = skl_get_node_id(skl, mconfig); - - if (cpr_mconfig->gtw_cfg.node_id == SKL_NON_GATEWAY_CPR_NODE_ID) { - cpr_mconfig->cpr_feature_mask = 0; - return; - } - - if (skl->nr_modules) { - res = &mconfig->module->resources[mconfig->res_idx]; - cpr_mconfig->gtw_cfg.dma_buffer_size = res->dma_buffer_size; - goto skip_buf_size_calc; - } else { - res = &mconfig->module->resources[res_idx]; - } - - switch (mconfig->hw_conn_type) { - case SKL_CONN_SOURCE: - if (mconfig->dev_type == SKL_DEVICE_HDAHOST) - dma_io_buf = res->ibs; - else - dma_io_buf = res->obs; - break; - - case SKL_CONN_SINK: - if (mconfig->dev_type == SKL_DEVICE_HDAHOST) - dma_io_buf = res->obs; - else - dma_io_buf = res->ibs; - break; - - default: - dev_warn(skl->dev, "wrong connection type: %d\n", - mconfig->hw_conn_type); - return; - } - - cpr_mconfig->gtw_cfg.dma_buffer_size = - mconfig->dma_buffer_size * dma_io_buf; - - /* fallback to 2ms default value */ - if (!cpr_mconfig->gtw_cfg.dma_buffer_size) { - if (mconfig->hw_conn_type == SKL_CONN_SOURCE) - cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->obs; - else - cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * res->ibs; - } - -skip_buf_size_calc: - cpr_mconfig->cpr_feature_mask = 0; - cpr_mconfig->gtw_cfg.config_length = 0; - - skl_copy_copier_caps(mconfig, cpr_mconfig); -} - -#define DMA_CONTROL_ID 5 -#define DMA_I2S_BLOB_SIZE 21 - -int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps, - u32 caps_size, u32 node_id) -{ - struct skl_dma_control *dma_ctrl; - struct skl_ipc_large_config_msg msg = {0}; - int err = 0; - - - /* - * if blob size zero, then return - */ - if (caps_size == 0) - return 0; - - msg.large_param_id = DMA_CONTROL_ID; - msg.param_data_size = sizeof(struct skl_dma_control) + caps_size; - - dma_ctrl = kzalloc(msg.param_data_size, GFP_KERNEL); - if (dma_ctrl == NULL) - return -ENOMEM; - - dma_ctrl->node_id = node_id; - - /* - * NHLT blob may contain additional configs along with i2s blob. - * firmware expects only the i2s blob size as the config_length. - * So fix to i2s blob size. - * size in dwords. - */ - dma_ctrl->config_length = DMA_I2S_BLOB_SIZE; - - memcpy(dma_ctrl->config_data, caps, caps_size); - - err = skl_ipc_set_large_config(&skl->ipc, &msg, (u32 *)dma_ctrl); - - kfree(dma_ctrl); - return err; -} -EXPORT_SYMBOL_GPL(skl_dsp_set_dma_control); - -static void skl_setup_out_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_audio_data_format *out_fmt) -{ - struct skl_module *module = mconfig->module; - struct skl_module_iface *fmt = &module->formats[mconfig->fmt_idx]; - struct skl_module_fmt *format = &fmt->outputs[0].fmt; - - out_fmt->number_of_channels = (u8)format->channels; - out_fmt->s_freq = format->s_freq; - out_fmt->bit_depth = format->bit_depth; - out_fmt->valid_bit_depth = format->valid_bit_depth; - out_fmt->ch_cfg = format->ch_cfg; - - out_fmt->channel_map = format->ch_map; - out_fmt->interleaving = format->interleaving_style; - out_fmt->sample_type = format->sample_type; - - dev_dbg(skl->dev, "copier out format chan=%d fre=%d bitdepth=%d\n", - out_fmt->number_of_channels, format->s_freq, format->bit_depth); -} - -/* - * DSP needs SRC module for frequency conversion, SRC takes base module - * configuration and the target frequency as extra parameter passed as src - * config - */ -static void skl_set_src_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_src_module_cfg *src_mconfig) -{ - struct skl_module *module = mconfig->module; - struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx]; - struct skl_module_fmt *fmt = &iface->outputs[0].fmt; - - skl_set_base_module_format(skl, mconfig, - (struct skl_base_cfg *)src_mconfig); - - src_mconfig->src_cfg = fmt->s_freq; -} - -/* - * DSP needs updown module to do channel conversion. updown module take base - * module configuration and channel configuration - * It also take coefficients and now we have defaults applied here - */ -static void skl_set_updown_mixer_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_up_down_mixer_cfg *mixer_mconfig) -{ - struct skl_module *module = mconfig->module; - struct skl_module_iface *iface = &module->formats[mconfig->fmt_idx]; - struct skl_module_fmt *fmt = &iface->outputs[0].fmt; - - skl_set_base_module_format(skl, mconfig, - (struct skl_base_cfg *)mixer_mconfig); - mixer_mconfig->out_ch_cfg = fmt->ch_cfg; - mixer_mconfig->ch_map = fmt->ch_map; -} - -/* - * 'copier' is DSP internal module which copies data from Host DMA (HDA host - * dma) or link (hda link, SSP, PDM) - * Here we calculate the copier module parameters, like PCM format, output - * format, gateway settings - * copier_module_config is sent as input buffer with INIT_INSTANCE IPC msg - */ -static void skl_set_copier_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_cpr_cfg *cpr_mconfig) -{ - struct skl_audio_data_format *out_fmt = &cpr_mconfig->out_fmt; - struct skl_base_cfg *base_cfg = (struct skl_base_cfg *)cpr_mconfig; - - skl_set_base_module_format(skl, mconfig, base_cfg); - - skl_setup_out_format(skl, mconfig, out_fmt); - skl_setup_cpr_gateway_cfg(skl, mconfig, cpr_mconfig); -} - -/* - * Mic select module allows selecting one or many input channels, thus - * acting as a demux. - * - * Mic select module take base module configuration and out-format - * configuration - */ -static void skl_set_base_outfmt_format(struct skl_dev *skl, - struct skl_module_cfg *mconfig, - struct skl_base_outfmt_cfg *base_outfmt_mcfg) -{ - struct skl_audio_data_format *out_fmt = &base_outfmt_mcfg->out_fmt; - struct skl_base_cfg *base_cfg = - (struct skl_base_cfg *)base_outfmt_mcfg; - - skl_set_base_module_format(skl, mconfig, base_cfg); - skl_setup_out_format(skl, mconfig, out_fmt); -} - -static u16 skl_get_module_param_size(struct skl_dev *skl, - struct skl_module_cfg *mconfig) -{ - struct skl_module_res *res; - struct skl_module *module = mconfig->module; - u16 param_size; - - switch (mconfig->m_type) { - case SKL_MODULE_TYPE_COPIER: - param_size = sizeof(struct skl_cpr_cfg); - param_size += mconfig->formats_config[SKL_PARAM_INIT].caps_size; - return param_size; - - case SKL_MODULE_TYPE_SRCINT: - return sizeof(struct skl_src_module_cfg); - - case SKL_MODULE_TYPE_UPDWMIX: - return sizeof(struct skl_up_down_mixer_cfg); - - case SKL_MODULE_TYPE_BASE_OUTFMT: - case SKL_MODULE_TYPE_MIC_SELECT: - return sizeof(struct skl_base_outfmt_cfg); - - case SKL_MODULE_TYPE_MIXER: - case SKL_MODULE_TYPE_KPB: - return sizeof(struct skl_base_cfg); - - case SKL_MODULE_TYPE_ALGO: - default: - res = &module->resources[mconfig->res_idx]; - - param_size = sizeof(struct skl_base_cfg) + sizeof(struct skl_base_cfg_ext); - param_size += (res->nr_input_pins + res->nr_output_pins) * - sizeof(struct skl_pin_format); - param_size += mconfig->formats_config[SKL_PARAM_INIT].caps_size; - - return param_size; - } - - return 0; -} - -/* - * DSP firmware supports various modules like copier, SRC, updown etc. - * These modules required various parameters to be calculated and sent for - * the module initialization to DSP. By default a generic module needs only - * base module format configuration - */ - -static int skl_set_module_format(struct skl_dev *skl, - struct skl_module_cfg *module_config, - u16 *module_config_size, - void **param_data) -{ - u16 param_size; - - param_size = skl_get_module_param_size(skl, module_config); - - *param_data = kzalloc(param_size, GFP_KERNEL); - if (NULL == *param_data) - return -ENOMEM; - - *module_config_size = param_size; - - switch (module_config->m_type) { - case SKL_MODULE_TYPE_COPIER: - skl_set_copier_format(skl, module_config, *param_data); - break; - - case SKL_MODULE_TYPE_SRCINT: - skl_set_src_format(skl, module_config, *param_data); - break; - - case SKL_MODULE_TYPE_UPDWMIX: - skl_set_updown_mixer_format(skl, module_config, *param_data); - break; - - case SKL_MODULE_TYPE_BASE_OUTFMT: - case SKL_MODULE_TYPE_MIC_SELECT: - skl_set_base_outfmt_format(skl, module_config, *param_data); - break; - - case SKL_MODULE_TYPE_MIXER: - case SKL_MODULE_TYPE_KPB: - skl_set_base_module_format(skl, module_config, *param_data); - break; - - case SKL_MODULE_TYPE_ALGO: - default: - skl_set_base_module_format(skl, module_config, *param_data); - skl_set_base_ext_module_format(skl, module_config, - *param_data + - sizeof(struct skl_base_cfg)); - break; - } - - dev_dbg(skl->dev, "Module type=%d id=%d config size: %d bytes\n", - module_config->m_type, module_config->id.module_id, - param_size); - print_hex_dump_debug("Module params:", DUMP_PREFIX_OFFSET, 8, 4, - *param_data, param_size, false); - return 0; -} - -static int skl_get_queue_index(struct skl_module_pin *mpin, - struct skl_module_inst_id id, int max) -{ - int i; - - for (i = 0; i < max; i++) { - if (mpin[i].id.module_id == id.module_id && - mpin[i].id.instance_id == id.instance_id) - return i; - } - - return -EINVAL; -} - -/* - * Allocates queue for each module. - * if dynamic, the pin_index is allocated 0 to max_pin. - * In static, the pin_index is fixed based on module_id and instance id - */ -static int skl_alloc_queue(struct skl_module_pin *mpin, - struct skl_module_cfg *tgt_cfg, int max) -{ - int i; - struct skl_module_inst_id id = tgt_cfg->id; - /* - * if pin in dynamic, find first free pin - * otherwise find match module and instance id pin as topology will - * ensure a unique pin is assigned to this so no need to - * allocate/free - */ - for (i = 0; i < max; i++) { - if (mpin[i].is_dynamic) { - if (!mpin[i].in_use && - mpin[i].pin_state == SKL_PIN_UNBIND) { - - mpin[i].in_use = true; - mpin[i].id.module_id = id.module_id; - mpin[i].id.instance_id = id.instance_id; - mpin[i].id.pvt_id = id.pvt_id; - mpin[i].tgt_mcfg = tgt_cfg; - return i; - } - } else { - if (mpin[i].id.module_id == id.module_id && - mpin[i].id.instance_id == id.instance_id && - mpin[i].pin_state == SKL_PIN_UNBIND) { - - mpin[i].tgt_mcfg = tgt_cfg; - return i; - } - } - } - - return -EINVAL; -} - -static void skl_free_queue(struct skl_module_pin *mpin, int q_index) -{ - if (mpin[q_index].is_dynamic) { - mpin[q_index].in_use = false; - mpin[q_index].id.module_id = 0; - mpin[q_index].id.instance_id = 0; - mpin[q_index].id.pvt_id = 0; - } - mpin[q_index].pin_state = SKL_PIN_UNBIND; - mpin[q_index].tgt_mcfg = NULL; -} - -/* Module state will be set to unint, if all the out pin state is UNBIND */ - -static void skl_clear_module_state(struct skl_module_pin *mpin, int max, - struct skl_module_cfg *mcfg) -{ - int i; - bool found = false; - - for (i = 0; i < max; i++) { - if (mpin[i].pin_state == SKL_PIN_UNBIND) - continue; - found = true; - break; - } - - if (!found) - mcfg->m_state = SKL_MODULE_INIT_DONE; - return; -} - -/* - * A module needs to be instanataited in DSP. A mdoule is present in a - * collection of module referred as a PIPE. - * We first calculate the module format, based on module type and then - * invoke the DSP by sending IPC INIT_INSTANCE using ipc helper - */ -int skl_init_module(struct skl_dev *skl, - struct skl_module_cfg *mconfig) -{ - u16 module_config_size = 0; - void *param_data = NULL; - int ret; - struct skl_ipc_init_instance_msg msg; - - dev_dbg(skl->dev, "%s: module_id = %d instance=%d\n", __func__, - mconfig->id.module_id, mconfig->id.pvt_id); - - if (mconfig->pipe->state != SKL_PIPE_CREATED) { - dev_err(skl->dev, "Pipe not created state= %d pipe_id= %d\n", - mconfig->pipe->state, mconfig->pipe->ppl_id); - return -EIO; - } - - ret = skl_set_module_format(skl, mconfig, - &module_config_size, ¶m_data); - if (ret < 0) { - dev_err(skl->dev, "Failed to set module format ret=%d\n", ret); - return ret; - } - - msg.module_id = mconfig->id.module_id; - msg.instance_id = mconfig->id.pvt_id; - msg.ppl_instance_id = mconfig->pipe->ppl_id; - msg.param_data_size = module_config_size; - msg.core_id = mconfig->core_id; - msg.domain = mconfig->domain; - - ret = skl_ipc_init_instance(&skl->ipc, &msg, param_data); - if (ret < 0) { - dev_err(skl->dev, "Failed to init instance ret=%d\n", ret); - kfree(param_data); - return ret; - } - mconfig->m_state = SKL_MODULE_INIT_DONE; - kfree(param_data); - return ret; -} - -static void skl_dump_bind_info(struct skl_dev *skl, struct skl_module_cfg - *src_module, struct skl_module_cfg *dst_module) -{ - dev_dbg(skl->dev, "%s: src module_id = %d src_instance=%d\n", - __func__, src_module->id.module_id, src_module->id.pvt_id); - dev_dbg(skl->dev, "%s: dst_module=%d dst_instance=%d\n", __func__, - dst_module->id.module_id, dst_module->id.pvt_id); - - dev_dbg(skl->dev, "src_module state = %d dst module state = %d\n", - src_module->m_state, dst_module->m_state); -} - -/* - * On module freeup, we need to unbind the module with modules - * it is already bind. - * Find the pin allocated and unbind then using bind_unbind IPC - */ -int skl_unbind_modules(struct skl_dev *skl, - struct skl_module_cfg *src_mcfg, - struct skl_module_cfg *dst_mcfg) -{ - int ret; - struct skl_ipc_bind_unbind_msg msg; - struct skl_module_inst_id src_id = src_mcfg->id; - struct skl_module_inst_id dst_id = dst_mcfg->id; - int in_max = dst_mcfg->module->max_input_pins; - int out_max = src_mcfg->module->max_output_pins; - int src_index, dst_index, src_pin_state, dst_pin_state; - - skl_dump_bind_info(skl, src_mcfg, dst_mcfg); - - /* get src queue index */ - src_index = skl_get_queue_index(src_mcfg->m_out_pin, dst_id, out_max); - if (src_index < 0) - return 0; - - msg.src_queue = src_index; - - /* get dst queue index */ - dst_index = skl_get_queue_index(dst_mcfg->m_in_pin, src_id, in_max); - if (dst_index < 0) - return 0; - - msg.dst_queue = dst_index; - - src_pin_state = src_mcfg->m_out_pin[src_index].pin_state; - dst_pin_state = dst_mcfg->m_in_pin[dst_index].pin_state; - - if (src_pin_state != SKL_PIN_BIND_DONE || - dst_pin_state != SKL_PIN_BIND_DONE) - return 0; - - msg.module_id = src_mcfg->id.module_id; - msg.instance_id = src_mcfg->id.pvt_id; - msg.dst_module_id = dst_mcfg->id.module_id; - msg.dst_instance_id = dst_mcfg->id.pvt_id; - msg.bind = false; - - ret = skl_ipc_bind_unbind(&skl->ipc, &msg); - if (!ret) { - /* free queue only if unbind is success */ - skl_free_queue(src_mcfg->m_out_pin, src_index); - skl_free_queue(dst_mcfg->m_in_pin, dst_index); - - /* - * check only if src module bind state, bind is - * always from src -> sink - */ - skl_clear_module_state(src_mcfg->m_out_pin, out_max, src_mcfg); - } - - return ret; -} - -#define CPR_SINK_FMT_PARAM_ID 2 - -/* - * Once a module is instantiated it need to be 'bind' with other modules in - * the pipeline. For binding we need to find the module pins which are bind - * together - * This function finds the pins and then sends bund_unbind IPC message to - * DSP using IPC helper - */ -int skl_bind_modules(struct skl_dev *skl, - struct skl_module_cfg *src_mcfg, - struct skl_module_cfg *dst_mcfg) -{ - int ret = 0; - struct skl_ipc_bind_unbind_msg msg; - int in_max = dst_mcfg->module->max_input_pins; - int out_max = src_mcfg->module->max_output_pins; - int src_index, dst_index; - struct skl_module_fmt *format; - struct skl_cpr_pin_fmt pin_fmt; - struct skl_module *module; - struct skl_module_iface *fmt; - - skl_dump_bind_info(skl, src_mcfg, dst_mcfg); - - if (src_mcfg->m_state < SKL_MODULE_INIT_DONE || - dst_mcfg->m_state < SKL_MODULE_INIT_DONE) - return 0; - - src_index = skl_alloc_queue(src_mcfg->m_out_pin, dst_mcfg, out_max); - if (src_index < 0) - return -EINVAL; - - msg.src_queue = src_index; - dst_index = skl_alloc_queue(dst_mcfg->m_in_pin, src_mcfg, in_max); - if (dst_index < 0) { - skl_free_queue(src_mcfg->m_out_pin, src_index); - return -EINVAL; - } - - /* - * Copier module requires the separate large_config_set_ipc to - * configure the pins other than 0 - */ - if (src_mcfg->m_type == SKL_MODULE_TYPE_COPIER && src_index > 0) { - pin_fmt.sink_id = src_index; - module = src_mcfg->module; - fmt = &module->formats[src_mcfg->fmt_idx]; - - /* Input fmt is same as that of src module input cfg */ - format = &fmt->inputs[0].fmt; - fill_pin_params(&(pin_fmt.src_fmt), format); - - format = &fmt->outputs[src_index].fmt; - fill_pin_params(&(pin_fmt.dst_fmt), format); - ret = skl_set_module_params(skl, (void *)&pin_fmt, - sizeof(struct skl_cpr_pin_fmt), - CPR_SINK_FMT_PARAM_ID, src_mcfg); - - if (ret < 0) - goto out; - } - - msg.dst_queue = dst_index; - - dev_dbg(skl->dev, "src queue = %d dst queue =%d\n", - msg.src_queue, msg.dst_queue); - - msg.module_id = src_mcfg->id.module_id; - msg.instance_id = src_mcfg->id.pvt_id; - msg.dst_module_id = dst_mcfg->id.module_id; - msg.dst_instance_id = dst_mcfg->id.pvt_id; - msg.bind = true; - - ret = skl_ipc_bind_unbind(&skl->ipc, &msg); - - if (!ret) { - src_mcfg->m_state = SKL_MODULE_BIND_DONE; - src_mcfg->m_out_pin[src_index].pin_state = SKL_PIN_BIND_DONE; - dst_mcfg->m_in_pin[dst_index].pin_state = SKL_PIN_BIND_DONE; - return ret; - } -out: - /* error case , if IPC fails, clear the queue index */ - skl_free_queue(src_mcfg->m_out_pin, src_index); - skl_free_queue(dst_mcfg->m_in_pin, dst_index); - - return ret; -} - -static int skl_set_pipe_state(struct skl_dev *skl, struct skl_pipe *pipe, - enum skl_ipc_pipeline_state state) -{ - dev_dbg(skl->dev, "%s: pipe_state = %d\n", __func__, state); - - return skl_ipc_set_pipeline_state(&skl->ipc, pipe->ppl_id, state); -} - -/* - * A pipeline is a collection of modules. Before a module in instantiated a - * pipeline needs to be created for it. - * This function creates pipeline, by sending create pipeline IPC messages - * to FW - */ -int skl_create_pipeline(struct skl_dev *skl, struct skl_pipe *pipe) -{ - int ret; - - dev_dbg(skl->dev, "%s: pipe_id = %d\n", __func__, pipe->ppl_id); - - ret = skl_ipc_create_pipeline(&skl->ipc, pipe->memory_pages, - pipe->pipe_priority, pipe->ppl_id, - pipe->lp_mode); - if (ret < 0) { - dev_err(skl->dev, "Failed to create pipeline\n"); - return ret; - } - - pipe->state = SKL_PIPE_CREATED; - - return 0; -} - -/* - * A pipeline needs to be deleted on cleanup. If a pipeline is running, - * then pause it first. Before actual deletion, pipeline should enter - * reset state. Finish the procedure by sending delete pipeline IPC. - * DSP will stop the DMA engines and release resources - */ -int skl_delete_pipe(struct skl_dev *skl, struct skl_pipe *pipe) -{ - int ret; - - dev_dbg(skl->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id); - - /* If pipe was not created in FW, do not try to delete it */ - if (pipe->state < SKL_PIPE_CREATED) - return 0; - - /* If pipe is started, do stop the pipe in FW. */ - if (pipe->state >= SKL_PIPE_STARTED) { - ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED); - if (ret < 0) { - dev_err(skl->dev, "Failed to stop pipeline\n"); - return ret; - } - - pipe->state = SKL_PIPE_PAUSED; - } - - /* reset pipe state before deletion */ - ret = skl_set_pipe_state(skl, pipe, PPL_RESET); - if (ret < 0) { - dev_err(skl->dev, "Failed to reset pipe ret=%d\n", ret); - return ret; - } - - pipe->state = SKL_PIPE_RESET; - - ret = skl_ipc_delete_pipeline(&skl->ipc, pipe->ppl_id); - if (ret < 0) { - dev_err(skl->dev, "Failed to delete pipeline\n"); - return ret; - } - - pipe->state = SKL_PIPE_INVALID; - - return ret; -} - -/* - * A pipeline is also a scheduling entity in DSP which can be run, stopped - * For processing data the pipe need to be run by sending IPC set pipe state - * to DSP - */ -int skl_run_pipe(struct skl_dev *skl, struct skl_pipe *pipe) -{ - int ret; - - dev_dbg(skl->dev, "%s: pipe = %d\n", __func__, pipe->ppl_id); - - /* If pipe was not created in FW, do not try to pause or delete */ - if (pipe->state < SKL_PIPE_CREATED) - return 0; - - /* Pipe has to be paused before it is started */ - ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED); - if (ret < 0) { - dev_err(skl->dev, "Failed to pause pipe\n"); - return ret; - } - - pipe->state = SKL_PIPE_PAUSED; - - ret = skl_set_pipe_state(skl, pipe, PPL_RUNNING); - if (ret < 0) { - dev_err(skl->dev, "Failed to start pipe\n"); - return ret; - } - - pipe->state = SKL_PIPE_STARTED; - - return 0; -} - -/* - * Stop the pipeline by sending set pipe state IPC - * DSP doesnt implement stop so we always send pause message - */ -int skl_stop_pipe(struct skl_dev *skl, struct skl_pipe *pipe) -{ - int ret; - - dev_dbg(skl->dev, "In %s pipe=%d\n", __func__, pipe->ppl_id); - - /* If pipe was not created in FW, do not try to pause or delete */ - if (pipe->state < SKL_PIPE_PAUSED) - return 0; - - ret = skl_set_pipe_state(skl, pipe, PPL_PAUSED); - if (ret < 0) { - dev_dbg(skl->dev, "Failed to stop pipe\n"); - return ret; - } - - pipe->state = SKL_PIPE_PAUSED; - - return 0; -} - -/* - * Reset the pipeline by sending set pipe state IPC this will reset the DMA - * from the DSP side - */ -int skl_reset_pipe(struct skl_dev *skl, struct skl_pipe *pipe) -{ - int ret; - - /* If pipe was not created in FW, do not try to pause or delete */ - if (pipe->state < SKL_PIPE_PAUSED) - return 0; - - ret = skl_set_pipe_state(skl, pipe, PPL_RESET); - if (ret < 0) { - dev_dbg(skl->dev, "Failed to reset pipe ret=%d\n", ret); - return ret; - } - - pipe->state = SKL_PIPE_RESET; - - return 0; -} - -/* Algo parameter set helper function */ -int skl_set_module_params(struct skl_dev *skl, u32 *params, int size, - u32 param_id, struct skl_module_cfg *mcfg) -{ - struct skl_ipc_large_config_msg msg; - - msg.module_id = mcfg->id.module_id; - msg.instance_id = mcfg->id.pvt_id; - msg.param_data_size = size; - msg.large_param_id = param_id; - - return skl_ipc_set_large_config(&skl->ipc, &msg, params); -} - -int skl_get_module_params(struct skl_dev *skl, u32 *params, int size, - u32 param_id, struct skl_module_cfg *mcfg) -{ - struct skl_ipc_large_config_msg msg; - size_t bytes = size; - - msg.module_id = mcfg->id.module_id; - msg.instance_id = mcfg->id.pvt_id; - msg.param_data_size = size; - msg.large_param_id = param_id; - - return skl_ipc_get_large_config(&skl->ipc, &msg, ¶ms, &bytes); -} diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c deleted file mode 100644 index e617b4c335a4..000000000000 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ /dev/null @@ -1,269 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-nhlt.c - Intel SKL Platform NHLT parsing - * - * Copyright (C) 2015 Intel Corp - * Author: Sanjiv Kumar - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ -#include -#include -#include "skl.h" -#include "skl-i2s.h" - -static void skl_nhlt_trim_space(char *trim) -{ - char *s = trim; - int cnt; - int i; - - cnt = 0; - for (i = 0; s[i]; i++) { - if (!isspace(s[i])) - s[cnt++] = s[i]; - } - - s[cnt] = '\0'; -} - -int skl_nhlt_update_topology_bin(struct skl_dev *skl) -{ - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - struct hdac_bus *bus = skl_to_bus(skl); - struct device *dev = bus->dev; - - dev_dbg(dev, "oem_id %.6s, oem_table_id %.8s oem_revision %d\n", - nhlt->header.oem_id, nhlt->header.oem_table_id, - nhlt->header.oem_revision); - - snprintf(skl->tplg_name, sizeof(skl->tplg_name), "%x-%.6s-%.8s-%d%s", - skl->pci_id, nhlt->header.oem_id, nhlt->header.oem_table_id, - nhlt->header.oem_revision, "-tplg.bin"); - - skl_nhlt_trim_space(skl->tplg_name); - - return 0; -} - -static ssize_t platform_id_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - struct skl_dev *skl = bus_to_skl(bus); - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - char platform_id[32]; - - sprintf(platform_id, "%x-%.6s-%.8s-%d", skl->pci_id, - nhlt->header.oem_id, nhlt->header.oem_table_id, - nhlt->header.oem_revision); - - skl_nhlt_trim_space(platform_id); - return sysfs_emit(buf, "%s\n", platform_id); -} - -static DEVICE_ATTR_RO(platform_id); - -int skl_nhlt_create_sysfs(struct skl_dev *skl) -{ - struct device *dev = &skl->pci->dev; - - if (sysfs_create_file(&dev->kobj, &dev_attr_platform_id.attr)) - dev_warn(dev, "Error creating sysfs entry\n"); - - return 0; -} - -void skl_nhlt_remove_sysfs(struct skl_dev *skl) -{ - struct device *dev = &skl->pci->dev; - - if (skl->nhlt) - sysfs_remove_file(&dev->kobj, &dev_attr_platform_id.attr); -} - -/* - * Queries NHLT for all the fmt configuration for a particular endpoint and - * stores all possible rates supported in a rate table for the corresponding - * sclk/sclkfs. - */ -static void skl_get_ssp_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks, - struct nhlt_fmt *fmt, u8 id) -{ - struct skl_i2s_config_blob_ext *i2s_config_ext; - struct skl_i2s_config_blob_legacy *i2s_config; - struct skl_clk_parent_src *parent; - struct skl_ssp_clk *sclk, *sclkfs; - struct nhlt_fmt_cfg *fmt_cfg; - struct wav_fmt_ext *wav_fmt; - unsigned long rate; - int rate_index = 0; - u16 channels, bps; - u8 clk_src; - int i, j; - u32 fs; - - sclk = &ssp_clks[SKL_SCLK_OFS]; - sclkfs = &ssp_clks[SKL_SCLKFS_OFS]; - - if (fmt->fmt_count == 0) - return; - - fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; - for (i = 0; i < fmt->fmt_count; i++) { - struct nhlt_fmt_cfg *saved_fmt_cfg = fmt_cfg; - bool present = false; - - wav_fmt = &saved_fmt_cfg->fmt_ext; - - channels = wav_fmt->fmt.channels; - bps = wav_fmt->fmt.bits_per_sample; - fs = wav_fmt->fmt.samples_per_sec; - - /* - * In case of TDM configuration on a ssp, there can - * be more than one blob in which channel masks are - * different for each usecase for a specific rate and bps. - * But the sclk rate will be generated for the total - * number of channels used for that endpoint. - * - * So for the given fs and bps, choose blob which has - * the superset of all channels for that endpoint and - * derive the rate. - */ - for (j = i; j < fmt->fmt_count; j++) { - struct nhlt_fmt_cfg *tmp_fmt_cfg = fmt_cfg; - - wav_fmt = &tmp_fmt_cfg->fmt_ext; - if ((fs == wav_fmt->fmt.samples_per_sec) && - (bps == wav_fmt->fmt.bits_per_sample)) { - channels = max_t(u16, channels, - wav_fmt->fmt.channels); - saved_fmt_cfg = tmp_fmt_cfg; - } - /* Move to the next nhlt_fmt_cfg */ - tmp_fmt_cfg = (struct nhlt_fmt_cfg *)(tmp_fmt_cfg->config.caps + - tmp_fmt_cfg->config.size); - } - - rate = channels * bps * fs; - - /* check if the rate is added already to the given SSP's sclk */ - for (j = 0; (j < SKL_MAX_CLK_RATES) && - (sclk[id].rate_cfg[j].rate != 0); j++) { - if (sclk[id].rate_cfg[j].rate == rate) { - present = true; - break; - } - } - - /* Fill rate and parent for sclk/sclkfs */ - if (!present) { - struct nhlt_fmt_cfg *first_fmt_cfg; - - first_fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; - i2s_config_ext = (struct skl_i2s_config_blob_ext *) - first_fmt_cfg->config.caps; - - /* MCLK Divider Source Select */ - if (is_legacy_blob(i2s_config_ext->hdr.sig)) { - i2s_config = ext_to_legacy_blob(i2s_config_ext); - clk_src = get_clk_src(i2s_config->mclk, - SKL_MNDSS_DIV_CLK_SRC_MASK); - } else { - clk_src = get_clk_src(i2s_config_ext->mclk, - SKL_MNDSS_DIV_CLK_SRC_MASK); - } - - parent = skl_get_parent_clk(clk_src); - - /* Move to the next nhlt_fmt_cfg */ - fmt_cfg = (struct nhlt_fmt_cfg *)(fmt_cfg->config.caps + - fmt_cfg->config.size); - /* - * Do not copy the config data if there is no parent - * clock available for this clock source select - */ - if (!parent) - continue; - - sclk[id].rate_cfg[rate_index].rate = rate; - sclk[id].rate_cfg[rate_index].config = saved_fmt_cfg; - sclkfs[id].rate_cfg[rate_index].rate = rate; - sclkfs[id].rate_cfg[rate_index].config = saved_fmt_cfg; - sclk[id].parent_name = parent->name; - sclkfs[id].parent_name = parent->name; - - rate_index++; - } - } -} - -static void skl_get_mclk(struct skl_dev *skl, struct skl_ssp_clk *mclk, - struct nhlt_fmt *fmt, u8 id) -{ - struct skl_i2s_config_blob_ext *i2s_config_ext; - struct skl_i2s_config_blob_legacy *i2s_config; - struct nhlt_fmt_cfg *fmt_cfg; - struct skl_clk_parent_src *parent; - u32 clkdiv, div_ratio; - u8 clk_src; - - fmt_cfg = (struct nhlt_fmt_cfg *)fmt->fmt_config; - i2s_config_ext = (struct skl_i2s_config_blob_ext *)fmt_cfg->config.caps; - - /* MCLK Divider Source Select and divider */ - if (is_legacy_blob(i2s_config_ext->hdr.sig)) { - i2s_config = ext_to_legacy_blob(i2s_config_ext); - clk_src = get_clk_src(i2s_config->mclk, - SKL_MCLK_DIV_CLK_SRC_MASK); - clkdiv = i2s_config->mclk.mdivr & - SKL_MCLK_DIV_RATIO_MASK; - } else { - clk_src = get_clk_src(i2s_config_ext->mclk, - SKL_MCLK_DIV_CLK_SRC_MASK); - clkdiv = i2s_config_ext->mclk.mdivr[0] & - SKL_MCLK_DIV_RATIO_MASK; - } - - /* bypass divider */ - div_ratio = 1; - - if (clkdiv != SKL_MCLK_DIV_RATIO_MASK) - /* Divider is 2 + clkdiv */ - div_ratio = clkdiv + 2; - - /* Calculate MCLK rate from source using div value */ - parent = skl_get_parent_clk(clk_src); - if (!parent) - return; - - mclk[id].rate_cfg[0].rate = parent->rate/div_ratio; - mclk[id].rate_cfg[0].config = fmt_cfg; - mclk[id].parent_name = parent->name; -} - -void skl_get_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks) -{ - struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; - struct nhlt_endpoint *epnt; - struct nhlt_fmt *fmt; - int i; - u8 id; - - epnt = (struct nhlt_endpoint *)nhlt->desc; - for (i = 0; i < nhlt->endpoint_count; i++) { - if (epnt->linktype == NHLT_LINK_SSP) { - id = epnt->virtual_bus_id; - - fmt = (struct nhlt_fmt *)(epnt->config.caps - + epnt->config.size); - - skl_get_ssp_clks(skl, ssp_clks, fmt, id); - skl_get_mclk(skl, ssp_clks, fmt, id); - } - epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); - } -} diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c deleted file mode 100644 index 613b27b8da13..000000000000 --- a/sound/soc/intel/skylake/skl-pcm.c +++ /dev/null @@ -1,1507 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-pcm.c -ASoC HDA Platform driver file implementing PCM functionality - * - * Copyright (C) 2014-2015 Intel Corp - * Author: Jeeja KP - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include -#include "skl.h" -#include "skl-topology.h" -#include "skl-sst-dsp.h" -#include "skl-sst-ipc.h" - -#define HDA_MONO 1 -#define HDA_STEREO 2 -#define HDA_QUAD 4 -#define HDA_MAX 8 - -static const struct snd_pcm_hardware azx_pcm_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME | - SNDRV_PCM_INFO_SYNC_START | - SNDRV_PCM_INFO_HAS_WALL_CLOCK | /* legacy */ - SNDRV_PCM_INFO_HAS_LINK_ATIME | - SNDRV_PCM_INFO_NO_PERIOD_WAKEUP), - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S32_LE | - SNDRV_PCM_FMTBIT_S24_LE, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 | - SNDRV_PCM_RATE_8000, - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 8, - .buffer_bytes_max = AZX_MAX_BUF_SIZE, - .period_bytes_min = 128, - .period_bytes_max = AZX_MAX_BUF_SIZE / 2, - .periods_min = 2, - .periods_max = AZX_MAX_FRAG, - .fifo_size = 0, -}; - -static inline -struct hdac_ext_stream *get_hdac_ext_stream(struct snd_pcm_substream *substream) -{ - return substream->runtime->private_data; -} - -static struct hdac_bus *get_bus_ctx(struct snd_pcm_substream *substream) -{ - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_stream *hstream = hdac_stream(stream); - struct hdac_bus *bus = hstream->bus; - return bus; -} - -static int skl_substream_alloc_pages(struct hdac_bus *bus, - struct snd_pcm_substream *substream, - size_t size) -{ - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - - hdac_stream(stream)->bufsize = 0; - hdac_stream(stream)->period_bytes = 0; - hdac_stream(stream)->format_val = 0; - - return 0; -} - -static void skl_set_pcm_constrains(struct hdac_bus *bus, - struct snd_pcm_runtime *runtime) -{ - snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - - /* avoid wrap-around with wall-clock */ - snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_TIME, - 20, 178000000); -} - -static enum hdac_ext_stream_type skl_get_host_stream_type(struct hdac_bus *bus) -{ - if (bus->ppcap) - return HDAC_EXT_STREAM_TYPE_HOST; - else - return HDAC_EXT_STREAM_TYPE_COUPLED; -} - -/* - * check if the stream opened is marked as ignore_suspend by machine, if so - * then enable suspend_active refcount - * - * The count supend_active does not need lock as it is used in open/close - * and suspend context - */ -static void skl_set_suspend_active(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai, bool enable) -{ - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct snd_soc_dapm_widget *w; - struct skl_dev *skl = bus_to_skl(bus); - - w = snd_soc_dai_get_widget(dai, substream->stream); - - if (w->ignore_suspend && enable) - skl->supend_active++; - else if (w->ignore_suspend && !enable) - skl->supend_active--; -} - -int skl_pcm_host_dma_prepare(struct device *dev, struct skl_pipe_params *params) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - unsigned int format_val; - struct hdac_stream *hstream; - struct hdac_ext_stream *stream; - unsigned int bits; - int err; - - hstream = snd_hdac_get_stream(bus, params->stream, - params->host_dma_id + 1); - if (!hstream) - return -EINVAL; - - stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(bus, stream, true); - - bits = snd_hdac_stream_format_bits(params->format, SNDRV_PCM_SUBFORMAT_STD, - params->host_bps); - format_val = snd_hdac_stream_format(params->ch, bits, params->s_freq); - - dev_dbg(dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", - format_val, params->s_freq, params->ch, params->format); - - snd_hdac_stream_reset(hdac_stream(stream)); - err = snd_hdac_stream_set_params(hdac_stream(stream), format_val); - if (err < 0) - return err; - - err = snd_hdac_ext_host_stream_setup(stream, false); - if (err < 0) - return err; - - hdac_stream(stream)->prepared = 1; - - return 0; -} - -int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - unsigned int format_val; - struct hdac_stream *hstream; - struct hdac_ext_stream *stream; - struct hdac_ext_link *link; - unsigned char stream_tag; - unsigned int bits; - - hstream = snd_hdac_get_stream(bus, params->stream, - params->link_dma_id + 1); - if (!hstream) - return -EINVAL; - - stream = stream_to_hdac_ext_stream(hstream); - snd_hdac_ext_stream_decouple(bus, stream, true); - - bits = snd_hdac_stream_format_bits(params->format, SNDRV_PCM_SUBFORMAT_STD, - params->link_bps); - format_val = snd_hdac_stream_format(params->ch, bits, params->s_freq); - - dev_dbg(dev, "format_val=%d, rate=%d, ch=%d, format=%d\n", - format_val, params->s_freq, params->ch, params->format); - - snd_hdac_ext_stream_reset(stream); - - snd_hdac_ext_stream_setup(stream, format_val); - - stream_tag = hstream->stream_tag; - if (stream->hstream.direction == SNDRV_PCM_STREAM_PLAYBACK) { - list_for_each_entry(link, &bus->hlink_list, list) { - if (link->index == params->link_index) - snd_hdac_ext_bus_link_set_stream_id(link, - stream_tag); - } - } - - stream->link_prepared = 1; - - return 0; -} - -static int skl_pcm_open(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct hdac_ext_stream *stream; - struct snd_pcm_runtime *runtime = substream->runtime; - struct skl_dma_params *dma_params; - struct skl_dev *skl = get_skl_ctx(dai->dev); - struct skl_module_cfg *mconfig; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - - stream = snd_hdac_ext_stream_assign(bus, substream, - skl_get_host_stream_type(bus)); - if (stream == NULL) - return -EBUSY; - - skl_set_pcm_constrains(bus, runtime); - - /* - * disable WALLCLOCK timestamps for capture streams - * until we figure out how to handle digital inputs - */ - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_WALL_CLOCK; /* legacy */ - runtime->hw.info &= ~SNDRV_PCM_INFO_HAS_LINK_ATIME; - } - - runtime->private_data = stream; - - dma_params = kzalloc(sizeof(*dma_params), GFP_KERNEL); - if (!dma_params) - return -ENOMEM; - - dma_params->stream_tag = hdac_stream(stream)->stream_tag; - snd_soc_dai_set_dma_data(dai, substream, dma_params); - - dev_dbg(dai->dev, "stream tag set in dma params=%d\n", - dma_params->stream_tag); - skl_set_suspend_active(substream, dai, true); - snd_pcm_set_sync(substream); - - mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (!mconfig) { - kfree(dma_params); - return -EINVAL; - } - - skl_tplg_d0i3_get(skl, mconfig->d0i3_caps); - - return 0; -} - -static int skl_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct skl_dev *skl = get_skl_ctx(dai->dev); - struct skl_module_cfg *mconfig; - int ret; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - - mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - - /* - * In case of XRUN recovery or in the case when the application - * calls prepare another time, reset the FW pipe to clean state - */ - if (mconfig && - (substream->runtime->state == SNDRV_PCM_STATE_XRUN || - mconfig->pipe->state == SKL_PIPE_CREATED || - mconfig->pipe->state == SKL_PIPE_PAUSED)) { - - ret = skl_reset_pipe(skl, mconfig->pipe); - - if (ret < 0) - return ret; - - ret = skl_pcm_host_dma_prepare(dai->dev, - mconfig->pipe->p_params); - if (ret < 0) - return ret; - } - - return 0; -} - -static int skl_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct skl_pipe_params p_params = {0}; - struct skl_module_cfg *m_cfg; - int ret, dma_id; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - ret = skl_substream_alloc_pages(bus, substream, - params_buffer_bytes(params)); - if (ret < 0) - return ret; - - dev_dbg(dai->dev, "format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - - dma_id = hdac_stream(stream)->stream_tag - 1; - dev_dbg(dai->dev, "dma_id=%d\n", dma_id); - - p_params.s_fmt = snd_pcm_format_width(params_format(params)); - p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); - p_params.ch = params_channels(params); - p_params.s_freq = params_rate(params); - p_params.host_dma_id = dma_id; - p_params.stream = substream->stream; - p_params.format = params_format(params); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - p_params.host_bps = dai->driver->playback.sig_bits; - else - p_params.host_bps = dai->driver->capture.sig_bits; - - - m_cfg = skl_tplg_fe_get_cpr_module(dai, p_params.stream); - if (m_cfg) - skl_tplg_update_pipe_params(dai->dev, m_cfg, &p_params); - - return 0; -} - -static void skl_pcm_close(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct skl_dma_params *dma_params = NULL; - struct skl_dev *skl = bus_to_skl(bus); - struct skl_module_cfg *mconfig; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - - snd_hdac_ext_stream_release(stream, skl_get_host_stream_type(bus)); - - dma_params = snd_soc_dai_get_dma_data(dai, substream); - /* - * now we should set this to NULL as we are freeing by the - * dma_params - */ - snd_soc_dai_set_dma_data(dai, substream, NULL); - skl_set_suspend_active(substream, dai, false); - - /* - * check if close is for "Reference Pin" and set back the - * CGCTL.MISCBDCGE if disabled by driver - */ - if (!strncmp(dai->name, "Reference Pin", 13) && - skl->miscbdcg_disabled) { - skl->enable_miscbdcge(dai->dev, true); - skl->miscbdcg_disabled = false; - } - - mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (mconfig) - skl_tplg_d0i3_put(skl, mconfig->d0i3_caps); - - kfree(dma_params); -} - -static int skl_pcm_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct skl_dev *skl = get_skl_ctx(dai->dev); - struct skl_module_cfg *mconfig; - int ret; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - - mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - - if (mconfig) { - ret = skl_reset_pipe(skl, mconfig->pipe); - if (ret < 0) - dev_err(dai->dev, "%s:Reset failed ret =%d", - __func__, ret); - } - - snd_hdac_stream_cleanup(hdac_stream(stream)); - hdac_stream(stream)->prepared = 0; - - return 0; -} - -static int skl_be_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct skl_pipe_params p_params = {0}; - - p_params.s_fmt = snd_pcm_format_width(params_format(params)); - p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); - p_params.ch = params_channels(params); - p_params.s_freq = params_rate(params); - p_params.stream = substream->stream; - - return skl_tplg_be_update_params(dai, &p_params); -} - -static int skl_decoupled_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct hdac_bus *bus = get_bus_ctx(substream); - struct hdac_ext_stream *stream; - int start; - unsigned long cookie; - struct hdac_stream *hstr; - - stream = get_hdac_ext_stream(substream); - hstr = hdac_stream(stream); - - if (!hstr->prepared) - return -EPIPE; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - start = 1; - break; - - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - start = 0; - break; - - default: - return -EINVAL; - } - - spin_lock_irqsave(&bus->reg_lock, cookie); - - if (start) { - snd_hdac_stream_start(hdac_stream(stream)); - snd_hdac_stream_timecounter_init(hstr, 0); - } else { - snd_hdac_stream_stop(hdac_stream(stream)); - } - - spin_unlock_irqrestore(&bus->reg_lock, cookie); - - return 0; -} - -static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct skl_dev *skl = get_skl_ctx(dai->dev); - struct skl_module_cfg *mconfig; - struct hdac_bus *bus = get_bus_ctx(substream); - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - struct hdac_stream *hstream = hdac_stream(stream); - struct snd_soc_dapm_widget *w; - int ret; - - mconfig = skl_tplg_fe_get_cpr_module(dai, substream->stream); - if (!mconfig) - return -EIO; - - w = snd_soc_dai_get_widget(dai, substream->stream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_RESUME: - if (!w->ignore_suspend) { - /* - * enable DMA Resume enable bit for the stream, set the - * dpib & lpib position to resume before starting the - * DMA - */ - snd_hdac_stream_drsm_enable(bus, true, hstream->index); - snd_hdac_stream_set_dpibr(bus, hstream, hstream->lpib); - snd_hdac_stream_set_lpib(hstream, hstream->lpib); - } - fallthrough; - - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - /* - * Start HOST DMA and Start FE Pipe.This is to make sure that - * there are no underrun/overrun in the case when the FE - * pipeline is started but there is a delay in starting the - * DMA channel on the host. - */ - ret = skl_decoupled_trigger(substream, cmd); - if (ret < 0) - return ret; - return skl_run_pipe(skl, mconfig->pipe); - - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - /* - * Stop FE Pipe first and stop DMA. This is to make sure that - * there are no underrun/overrun in the case if there is a delay - * between the two operations. - */ - ret = skl_stop_pipe(skl, mconfig->pipe); - if (ret < 0) - return ret; - - ret = skl_decoupled_trigger(substream, cmd); - if (ret < 0) - return ret; - - if ((cmd == SNDRV_PCM_TRIGGER_SUSPEND) && !w->ignore_suspend) { - /* save the dpib and lpib positions */ - hstream->dpib = readl(bus->remap_addr + - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hstream->index)); - - hstream->lpib = snd_hdac_stream_get_pos_lpib(hstream); - - snd_hdac_ext_stream_decouple(bus, stream, false); - } - break; - - default: - return -EINVAL; - } - - return 0; -} - - -static int skl_link_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct hdac_ext_stream *link_dev; - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - struct skl_pipe_params p_params = {0}; - struct hdac_ext_link *link; - int stream_tag; - - link_dev = snd_hdac_ext_stream_assign(bus, substream, - HDAC_EXT_STREAM_TYPE_LINK); - if (!link_dev) - return -EBUSY; - - snd_soc_dai_set_dma_data(dai, substream, (void *)link_dev); - - link = snd_hdac_ext_bus_get_hlink_by_name(bus, codec_dai->component->name); - if (!link) - return -EINVAL; - - stream_tag = hdac_stream(link_dev)->stream_tag; - - /* set the hdac_stream in the codec dai */ - snd_soc_dai_set_stream(codec_dai, hdac_stream(link_dev), substream->stream); - - p_params.s_fmt = snd_pcm_format_width(params_format(params)); - p_params.s_cont = snd_pcm_format_physical_width(params_format(params)); - p_params.ch = params_channels(params); - p_params.s_freq = params_rate(params); - p_params.stream = substream->stream; - p_params.link_dma_id = stream_tag - 1; - p_params.link_index = link->index; - p_params.format = params_format(params); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - p_params.link_bps = codec_dai->driver->playback.sig_bits; - else - p_params.link_bps = codec_dai->driver->capture.sig_bits; - - return skl_tplg_be_update_params(dai, &p_params); -} - -static int skl_link_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct skl_dev *skl = get_skl_ctx(dai->dev); - struct skl_module_cfg *mconfig = NULL; - - /* In case of XRUN recovery, reset the FW pipe to clean state */ - mconfig = skl_tplg_be_get_cpr_module(dai, substream->stream); - if (mconfig && !mconfig->pipe->passthru && - (substream->runtime->state == SNDRV_PCM_STATE_XRUN)) - skl_reset_pipe(skl, mconfig->pipe); - - return 0; -} - -static int skl_link_pcm_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct hdac_ext_stream *link_dev = - snd_soc_dai_get_dma_data(dai, substream); - struct hdac_bus *bus = get_bus_ctx(substream); - struct hdac_ext_stream *stream = get_hdac_ext_stream(substream); - - dev_dbg(dai->dev, "In %s cmd=%d\n", __func__, cmd); - switch (cmd) { - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - snd_hdac_ext_stream_start(link_dev); - break; - - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - snd_hdac_ext_stream_clear(link_dev); - if (cmd == SNDRV_PCM_TRIGGER_SUSPEND) - snd_hdac_ext_stream_decouple(bus, stream, false); - break; - - default: - return -EINVAL; - } - return 0; -} - -static int skl_link_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct hdac_ext_stream *link_dev = - snd_soc_dai_get_dma_data(dai, substream); - struct hdac_ext_link *link; - unsigned char stream_tag; - - dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name); - - link_dev->link_prepared = 0; - - link = snd_hdac_ext_bus_get_hlink_by_name(bus, snd_soc_rtd_to_codec(rtd, 0)->component->name); - if (!link) - return -EINVAL; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - stream_tag = hdac_stream(link_dev)->stream_tag; - snd_hdac_ext_bus_link_clear_stream_id(link, stream_tag); - } - - snd_hdac_ext_stream_release(link_dev, HDAC_EXT_STREAM_TYPE_LINK); - return 0; -} - -static const struct snd_soc_dai_ops skl_pcm_dai_ops = { - .startup = skl_pcm_open, - .shutdown = skl_pcm_close, - .prepare = skl_pcm_prepare, - .hw_params = skl_pcm_hw_params, - .hw_free = skl_pcm_hw_free, - .trigger = skl_pcm_trigger, -}; - -static const struct snd_soc_dai_ops skl_dmic_dai_ops = { - .hw_params = skl_be_hw_params, -}; - -static const struct snd_soc_dai_ops skl_be_ssp_dai_ops = { - .hw_params = skl_be_hw_params, -}; - -static const struct snd_soc_dai_ops skl_link_dai_ops = { - .prepare = skl_link_pcm_prepare, - .hw_params = skl_link_hw_params, - .hw_free = skl_link_hw_free, - .trigger = skl_link_pcm_trigger, -}; - -static struct snd_soc_dai_driver skl_fe_dai[] = { -{ - .name = "System Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "System Playback", - .channels_min = HDA_MONO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, - .sig_bits = 32, - }, - .capture = { - .stream_name = "System Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - .sig_bits = 32, - }, -}, -{ - .name = "System Pin2", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "Headset Playback", - .channels_min = HDA_MONO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 | - SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, -}, -{ - .name = "Echoref Pin", - .ops = &skl_pcm_dai_ops, - .capture = { - .stream_name = "Echoreference Capture", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000 | - SNDRV_PCM_RATE_8000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, -}, -{ - .name = "Reference Pin", - .ops = &skl_pcm_dai_ops, - .capture = { - .stream_name = "Reference Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_QUAD, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - .sig_bits = 32, - }, -}, -{ - .name = "Deepbuffer Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "Deepbuffer Playback", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - .sig_bits = 32, - }, -}, -{ - .name = "LowLatency Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "Low Latency Playback", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - .sig_bits = 32, - }, -}, -{ - .name = "DMIC Pin", - .ops = &skl_pcm_dai_ops, - .capture = { - .stream_name = "DMIC Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_QUAD, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - .sig_bits = 32, - }, -}, -{ - .name = "HDMI1 Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "HDMI1 Playback", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .sig_bits = 32, - }, -}, -{ - .name = "HDMI2 Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "HDMI2 Playback", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .sig_bits = 32, - }, -}, -{ - .name = "HDMI3 Pin", - .ops = &skl_pcm_dai_ops, - .playback = { - .stream_name = "HDMI3 Playback", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | - SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | - SNDRV_PCM_RATE_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - .sig_bits = 32, - }, -}, -}; - -/* BE CPU Dais */ -static struct snd_soc_dai_driver skl_platform_dai[] = { -{ - .name = "SSP0 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp0 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp0 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "SSP1 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp1 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp1 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "SSP2 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp2 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp2 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "SSP3 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp3 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp3 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "SSP4 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp4 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp4 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "SSP5 Pin", - .ops = &skl_be_ssp_dai_ops, - .playback = { - .stream_name = "ssp5 Tx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, - .capture = { - .stream_name = "ssp5 Rx", - .channels_min = HDA_STEREO, - .channels_max = HDA_STEREO, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "iDisp1 Pin", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "iDisp1 Tx", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | - SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "iDisp2 Pin", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "iDisp2 Tx", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| - SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | - SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "iDisp3 Pin", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "iDisp3 Tx", - .channels_min = HDA_STEREO, - .channels_max = 8, - .rates = SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_16000| - SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE | - SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "DMIC01 Pin", - .ops = &skl_dmic_dai_ops, - .capture = { - .stream_name = "DMIC01 Rx", - .channels_min = HDA_MONO, - .channels_max = HDA_QUAD, - .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE, - }, -}, -{ - .name = "DMIC16k Pin", - .ops = &skl_dmic_dai_ops, - .capture = { - .stream_name = "DMIC16k Rx", - .channels_min = HDA_MONO, - .channels_max = HDA_QUAD, - .rates = SNDRV_PCM_RATE_16000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, - }, -}, -{ - .name = "Analog CPU DAI", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "Analog CPU Playback", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .stream_name = "Analog CPU Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, -}, -{ - .name = "Alt Analog CPU DAI", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "Alt Analog CPU Playback", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .stream_name = "Alt Analog CPU Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, -}, -{ - .name = "Digital CPU DAI", - .ops = &skl_link_dai_ops, - .playback = { - .stream_name = "Digital CPU Playback", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .stream_name = "Digital CPU Capture", - .channels_min = HDA_MONO, - .channels_max = HDA_MAX, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_S32_LE, - }, -}, -}; - -int skl_dai_load(struct snd_soc_component *cmp, int index, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) -{ - dai_drv->ops = &skl_pcm_dai_ops; - - return 0; -} - -static int skl_platform_soc_open(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai_link *dai_link = rtd->dai_link; - - dev_dbg(snd_soc_rtd_to_cpu(rtd, 0)->dev, "In %s:%s\n", __func__, - dai_link->cpus->dai_name); - - snd_soc_set_runtime_hwparams(substream, &azx_pcm_hw); - - return 0; -} - -static int skl_coupled_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct hdac_bus *bus = get_bus_ctx(substream); - struct hdac_ext_stream *stream; - struct snd_pcm_substream *s; - bool start; - int sbits = 0; - unsigned long cookie; - struct hdac_stream *hstr; - - stream = get_hdac_ext_stream(substream); - hstr = hdac_stream(stream); - - dev_dbg(bus->dev, "In %s cmd=%d\n", __func__, cmd); - - if (!hstr->prepared) - return -EPIPE; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - case SNDRV_PCM_TRIGGER_RESUME: - start = true; - break; - - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - start = false; - break; - - default: - return -EINVAL; - } - - snd_pcm_group_for_each_entry(s, substream) { - if (s->pcm->card != substream->pcm->card) - continue; - stream = get_hdac_ext_stream(s); - sbits |= 1 << hdac_stream(stream)->index; - snd_pcm_trigger_done(s, substream); - } - - spin_lock_irqsave(&bus->reg_lock, cookie); - - /* first, set SYNC bits of corresponding streams */ - snd_hdac_stream_sync_trigger(hstr, true, sbits, AZX_REG_SSYNC); - - snd_pcm_group_for_each_entry(s, substream) { - if (s->pcm->card != substream->pcm->card) - continue; - stream = get_hdac_ext_stream(s); - if (start) - snd_hdac_stream_start(hdac_stream(stream)); - else - snd_hdac_stream_stop(hdac_stream(stream)); - } - spin_unlock_irqrestore(&bus->reg_lock, cookie); - - snd_hdac_stream_sync(hstr, start, sbits); - - spin_lock_irqsave(&bus->reg_lock, cookie); - - /* reset SYNC bits */ - snd_hdac_stream_sync_trigger(hstr, false, sbits, AZX_REG_SSYNC); - if (start) - snd_hdac_stream_timecounter_init(hstr, sbits); - spin_unlock_irqrestore(&bus->reg_lock, cookie); - - return 0; -} - -static int skl_platform_soc_trigger(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - int cmd) -{ - struct hdac_bus *bus = get_bus_ctx(substream); - - if (!bus->ppcap) - return skl_coupled_trigger(substream, cmd); - - return 0; -} - -static snd_pcm_uframes_t skl_platform_soc_pointer( - struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - struct hdac_ext_stream *hstream = get_hdac_ext_stream(substream); - struct hdac_bus *bus = get_bus_ctx(substream); - unsigned int pos; - - /* - * Use DPIB for Playback stream as the periodic DMA Position-in- - * Buffer Writes may be scheduled at the same time or later than - * the MSI and does not guarantee to reflect the Position of the - * last buffer that was transferred. Whereas DPIB register in - * HAD space reflects the actual data that is transferred. - * Use the position buffer for capture, as DPIB write gets - * completed earlier than the actual data written to the DDR. - * - * For capture stream following workaround is required to fix the - * incorrect position reporting. - * - * 1. Wait for 20us before reading the DMA position in buffer once - * the interrupt is generated for stream completion as update happens - * on the HDA frame boundary i.e. 20.833uSec. - * 2. Read DPIB register to flush the DMA position value. This dummy - * read is required to flush DMA position value. - * 3. Read the DMA Position-in-Buffer. This value now will be equal to - * or greater than period boundary. - */ - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - pos = readl(bus->remap_addr + AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hdac_stream(hstream)->index)); - } else { - udelay(20); - readl(bus->remap_addr + - AZX_REG_VS_SDXDPIB_XBASE + - (AZX_REG_VS_SDXDPIB_XINTERVAL * - hdac_stream(hstream)->index)); - pos = snd_hdac_stream_get_pos_posbuf(hdac_stream(hstream)); - } - - if (pos >= hdac_stream(hstream)->bufsize) - pos = 0; - - return bytes_to_frames(substream->runtime, pos); -} - -static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, - u64 nsec) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); - u64 codec_frames, codec_nsecs; - - if (!codec_dai->driver->ops->delay) - return nsec; - - codec_frames = codec_dai->driver->ops->delay(substream, codec_dai); - codec_nsecs = div_u64(codec_frames * 1000000000LL, - substream->runtime->rate); - - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) - return nsec + codec_nsecs; - - return (nsec > codec_nsecs) ? nsec - codec_nsecs : 0; -} - -static int skl_platform_soc_get_time_info( - struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct timespec64 *system_ts, struct timespec64 *audio_ts, - struct snd_pcm_audio_tstamp_config *audio_tstamp_config, - struct snd_pcm_audio_tstamp_report *audio_tstamp_report) -{ - struct hdac_ext_stream *sstream = get_hdac_ext_stream(substream); - struct hdac_stream *hstr = hdac_stream(sstream); - u64 nsec; - - if ((substream->runtime->hw.info & SNDRV_PCM_INFO_HAS_LINK_ATIME) && - (audio_tstamp_config->type_requested == SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK)) { - - snd_pcm_gettime(substream->runtime, system_ts); - - nsec = timecounter_read(&hstr->tc); - if (audio_tstamp_config->report_delay) - nsec = skl_adjust_codec_delay(substream, nsec); - - *audio_ts = ns_to_timespec64(nsec); - - audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_LINK; - audio_tstamp_report->accuracy_report = 1; /* rest of struct is valid */ - audio_tstamp_report->accuracy = 42; /* 24MHzWallClk == 42ns resolution */ - - } else { - audio_tstamp_report->actual_type = SNDRV_PCM_AUDIO_TSTAMP_TYPE_DEFAULT; - } - - return 0; -} - -#define MAX_PREALLOC_SIZE (32 * 1024 * 1024) - -static int skl_platform_soc_new(struct snd_soc_component *component, - struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *dai = snd_soc_rtd_to_cpu(rtd, 0); - struct hdac_bus *bus = dev_get_drvdata(dai->dev); - struct snd_pcm *pcm = rtd->pcm; - unsigned int size; - struct skl_dev *skl = bus_to_skl(bus); - - if (dai->driver->playback.channels_min || - dai->driver->capture.channels_min) { - /* buffer pre-allocation */ - size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; - if (size > MAX_PREALLOC_SIZE) - size = MAX_PREALLOC_SIZE; - snd_pcm_set_managed_buffer_all(pcm, - SNDRV_DMA_TYPE_DEV_SG, - &skl->pci->dev, - size, MAX_PREALLOC_SIZE); - } - - return 0; -} - -static int skl_get_module_info(struct skl_dev *skl, - struct skl_module_cfg *mconfig) -{ - struct skl_module_inst_id *pin_id; - guid_t *uuid_mod, *uuid_tplg; - struct skl_module *skl_module; - struct uuid_module *module; - int i, ret = -EIO; - - uuid_mod = (guid_t *)mconfig->guid; - - if (list_empty(&skl->uuid_list)) { - dev_err(skl->dev, "Module list is empty\n"); - return -EIO; - } - - for (i = 0; i < skl->nr_modules; i++) { - skl_module = skl->modules[i]; - uuid_tplg = &skl_module->uuid; - if (guid_equal(uuid_mod, uuid_tplg)) { - mconfig->module = skl_module; - ret = 0; - break; - } - } - - if (skl->nr_modules && ret) - return ret; - - ret = -EIO; - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - mconfig->id.module_id = module->id; - mconfig->module->loadable = module->is_loadable; - ret = 0; - } - - for (i = 0; i < MAX_IN_QUEUE; i++) { - pin_id = &mconfig->m_in_pin[i].id; - if (guid_equal(&pin_id->mod_uuid, &module->uuid)) - pin_id->module_id = module->id; - } - - for (i = 0; i < MAX_OUT_QUEUE; i++) { - pin_id = &mconfig->m_out_pin[i].id; - if (guid_equal(&pin_id->mod_uuid, &module->uuid)) - pin_id->module_id = module->id; - } - } - - return ret; -} - -static int skl_populate_modules(struct skl_dev *skl) -{ - struct skl_pipeline *p; - struct skl_pipe_module *m; - struct snd_soc_dapm_widget *w; - struct skl_module_cfg *mconfig; - int ret = 0; - - list_for_each_entry(p, &skl->ppl_list, node) { - list_for_each_entry(m, &p->pipe->w_list, node) { - w = m->w; - mconfig = w->priv; - - ret = skl_get_module_info(skl, mconfig); - if (ret < 0) { - dev_err(skl->dev, - "query module info failed\n"); - return ret; - } - - skl_tplg_add_moduleid_in_bind_params(skl, w); - } - } - - return ret; -} - -static int skl_platform_soc_probe(struct snd_soc_component *component) -{ - struct hdac_bus *bus = dev_get_drvdata(component->dev); - struct skl_dev *skl = bus_to_skl(bus); - const struct skl_dsp_ops *ops; - int ret; - - ret = pm_runtime_resume_and_get(component->dev); - if (ret < 0 && ret != -EACCES) - return ret; - - if (bus->ppcap) { - skl->component = component; - - /* init debugfs */ - skl->debugfs = skl_debugfs_init(skl); - - ret = skl_tplg_init(component, bus); - if (ret < 0) { - dev_err(component->dev, "Failed to init topology!\n"); - return ret; - } - - /* load the firmwares, since all is set */ - ops = skl_get_dsp_ops(skl->pci->device); - if (!ops) - return -EIO; - - /* - * Disable dynamic clock and power gating during firmware - * and library download - */ - skl->enable_miscbdcge(component->dev, false); - skl->clock_power_gating(component->dev, false); - - ret = ops->init_fw(component->dev, skl); - skl->enable_miscbdcge(component->dev, true); - skl->clock_power_gating(component->dev, true); - if (ret < 0) { - dev_err(component->dev, "Failed to boot first fw: %d\n", ret); - return ret; - } - skl_populate_modules(skl); - skl->update_d0i3c = skl_update_d0i3c; - - if (skl->cfg.astate_cfg != NULL) { - skl_dsp_set_astate_cfg(skl, - skl->cfg.astate_cfg->count, - skl->cfg.astate_cfg); - } - } - pm_runtime_mark_last_busy(component->dev); - pm_runtime_put_autosuspend(component->dev); - - return 0; -} - -static void skl_platform_soc_remove(struct snd_soc_component *component) -{ - struct hdac_bus *bus = dev_get_drvdata(component->dev); - struct skl_dev *skl = bus_to_skl(bus); - - skl_tplg_exit(component, bus); - - skl_debugfs_exit(skl); -} - -static const struct snd_soc_component_driver skl_component = { - .name = "pcm", - .probe = skl_platform_soc_probe, - .remove = skl_platform_soc_remove, - .open = skl_platform_soc_open, - .trigger = skl_platform_soc_trigger, - .pointer = skl_platform_soc_pointer, - .get_time_info = skl_platform_soc_get_time_info, - .pcm_construct = skl_platform_soc_new, - .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ -}; - -int skl_platform_register(struct device *dev) -{ - int ret; - struct snd_soc_dai_driver *dais; - int num_dais = ARRAY_SIZE(skl_platform_dai); - struct hdac_bus *bus = dev_get_drvdata(dev); - struct skl_dev *skl = bus_to_skl(bus); - - skl->dais = kmemdup(skl_platform_dai, sizeof(skl_platform_dai), - GFP_KERNEL); - if (!skl->dais) { - ret = -ENOMEM; - goto err; - } - - if (!skl->use_tplg_pcm) { - dais = krealloc(skl->dais, sizeof(skl_fe_dai) + - sizeof(skl_platform_dai), GFP_KERNEL); - if (!dais) { - kfree(skl->dais); - ret = -ENOMEM; - goto err; - } - - skl->dais = dais; - memcpy(&skl->dais[ARRAY_SIZE(skl_platform_dai)], skl_fe_dai, - sizeof(skl_fe_dai)); - num_dais += ARRAY_SIZE(skl_fe_dai); - } - - ret = devm_snd_soc_register_component(dev, &skl_component, - skl->dais, num_dais); - if (ret) { - kfree(skl->dais); - dev_err(dev, "soc component registration failed %d\n", ret); - } -err: - return ret; -} - -int skl_platform_unregister(struct device *dev) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - struct skl_dev *skl = bus_to_skl(bus); - struct skl_module_deferred_bind *modules, *tmp; - - list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { - list_del(&modules->node); - kfree(modules); - } - - kfree(skl->dais); - - return 0; -} diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c deleted file mode 100644 index 50e93c3707e8..000000000000 --- a/sound/soc/intel/skylake/skl-ssp-clk.c +++ /dev/null @@ -1,428 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2015-17 Intel Corporation - -/* - * skl-ssp-clk.c - ASoC skylake ssp clock driver - */ - -#include -#include -#include -#include -#include -#include -#include -#include "skl.h" -#include "skl-ssp-clk.h" -#include "skl-topology.h" - -#define to_skl_clk(_hw) container_of(_hw, struct skl_clk, hw) - -struct skl_clk_parent { - struct clk_hw *hw; - struct clk_lookup *lookup; -}; - -struct skl_clk { - struct clk_hw hw; - struct clk_lookup *lookup; - unsigned long rate; - struct skl_clk_pdata *pdata; - u32 id; -}; - -struct skl_clk_data { - struct skl_clk_parent parent[SKL_MAX_CLK_SRC]; - struct skl_clk *clk[SKL_MAX_CLK_CNT]; - u8 avail_clk_cnt; -}; - -static int skl_get_clk_type(u32 index) -{ - switch (index) { - case 0 ... (SKL_SCLK_OFS - 1): - return SKL_MCLK; - - case SKL_SCLK_OFS ... (SKL_SCLKFS_OFS - 1): - return SKL_SCLK; - - case SKL_SCLKFS_OFS ... (SKL_MAX_CLK_CNT - 1): - return SKL_SCLK_FS; - - default: - return -EINVAL; - } -} - -static int skl_get_vbus_id(u32 index, u8 clk_type) -{ - switch (clk_type) { - case SKL_MCLK: - return index; - - case SKL_SCLK: - return index - SKL_SCLK_OFS; - - case SKL_SCLK_FS: - return index - SKL_SCLKFS_OFS; - - default: - return -EINVAL; - } -} - -static void skl_fill_clk_ipc(struct skl_clk_rate_cfg_table *rcfg, u8 clk_type) -{ - struct nhlt_fmt_cfg *fmt_cfg; - union skl_clk_ctrl_ipc *ipc; - struct wav_fmt *wfmt; - - if (!rcfg) - return; - - ipc = &rcfg->dma_ctl_ipc; - if (clk_type == SKL_SCLK_FS) { - fmt_cfg = (struct nhlt_fmt_cfg *)rcfg->config; - wfmt = &fmt_cfg->fmt_ext.fmt; - - /* Remove TLV Header size */ - ipc->sclk_fs.hdr.size = sizeof(struct skl_dmactrl_sclkfs_cfg) - - sizeof(struct skl_tlv_hdr); - ipc->sclk_fs.sampling_frequency = wfmt->samples_per_sec; - ipc->sclk_fs.bit_depth = wfmt->bits_per_sample; - ipc->sclk_fs.valid_bit_depth = - fmt_cfg->fmt_ext.sample.valid_bits_per_sample; - ipc->sclk_fs.number_of_channels = wfmt->channels; - } else { - ipc->mclk.hdr.type = DMA_CLK_CONTROLS; - /* Remove TLV Header size */ - ipc->mclk.hdr.size = sizeof(struct skl_dmactrl_mclk_cfg) - - sizeof(struct skl_tlv_hdr); - } -} - -/* Sends dma control IPC to turn the clock ON/OFF */ -static int skl_send_clk_dma_control(struct skl_dev *skl, - struct skl_clk_rate_cfg_table *rcfg, - u32 vbus_id, u8 clk_type, - bool enable) -{ - struct nhlt_specific_cfg *sp_cfg; - u32 i2s_config_size, node_id = 0; - struct nhlt_fmt_cfg *fmt_cfg; - union skl_clk_ctrl_ipc *ipc; - void *i2s_config = NULL; - u8 *data, size; - int ret; - - if (!rcfg) - return -EIO; - - ipc = &rcfg->dma_ctl_ipc; - fmt_cfg = (struct nhlt_fmt_cfg *)rcfg->config; - sp_cfg = &fmt_cfg->config; - - if (clk_type == SKL_SCLK_FS) { - ipc->sclk_fs.hdr.type = - enable ? DMA_TRANSMITION_START : DMA_TRANSMITION_STOP; - data = (u8 *)&ipc->sclk_fs; - size = sizeof(struct skl_dmactrl_sclkfs_cfg); - } else { - /* 1 to enable mclk, 0 to enable sclk */ - if (clk_type == SKL_SCLK) - ipc->mclk.mclk = 0; - else - ipc->mclk.mclk = 1; - - ipc->mclk.keep_running = enable; - ipc->mclk.warm_up_over = enable; - ipc->mclk.clk_stop_over = !enable; - data = (u8 *)&ipc->mclk; - size = sizeof(struct skl_dmactrl_mclk_cfg); - } - - i2s_config_size = sp_cfg->size + size; - i2s_config = kzalloc(i2s_config_size, GFP_KERNEL); - if (!i2s_config) - return -ENOMEM; - - /* copy blob */ - memcpy(i2s_config, sp_cfg->caps, sp_cfg->size); - - /* copy additional dma controls information */ - memcpy(i2s_config + sp_cfg->size, data, size); - - node_id = ((SKL_DMA_I2S_LINK_INPUT_CLASS << 8) | (vbus_id << 4)); - ret = skl_dsp_set_dma_control(skl, (u32 *)i2s_config, - i2s_config_size, node_id); - kfree(i2s_config); - - return ret; -} - -static struct skl_clk_rate_cfg_table *skl_get_rate_cfg( - struct skl_clk_rate_cfg_table *rcfg, - unsigned long rate) -{ - int i; - - for (i = 0; (i < SKL_MAX_CLK_RATES) && rcfg[i].rate; i++) { - if (rcfg[i].rate == rate) - return &rcfg[i]; - } - - return NULL; -} - -static int skl_clk_change_status(struct skl_clk *clkdev, - bool enable) -{ - struct skl_clk_rate_cfg_table *rcfg; - int vbus_id, clk_type; - - clk_type = skl_get_clk_type(clkdev->id); - if (clk_type < 0) - return clk_type; - - vbus_id = skl_get_vbus_id(clkdev->id, clk_type); - if (vbus_id < 0) - return vbus_id; - - rcfg = skl_get_rate_cfg(clkdev->pdata->ssp_clks[clkdev->id].rate_cfg, - clkdev->rate); - if (!rcfg) - return -EINVAL; - - return skl_send_clk_dma_control(clkdev->pdata->pvt_data, rcfg, - vbus_id, clk_type, enable); -} - -static int skl_clk_prepare(struct clk_hw *hw) -{ - struct skl_clk *clkdev = to_skl_clk(hw); - - return skl_clk_change_status(clkdev, true); -} - -static void skl_clk_unprepare(struct clk_hw *hw) -{ - struct skl_clk *clkdev = to_skl_clk(hw); - - skl_clk_change_status(clkdev, false); -} - -static int skl_clk_set_rate(struct clk_hw *hw, unsigned long rate, - unsigned long parent_rate) -{ - struct skl_clk *clkdev = to_skl_clk(hw); - struct skl_clk_rate_cfg_table *rcfg; - int clk_type; - - if (!rate) - return -EINVAL; - - rcfg = skl_get_rate_cfg(clkdev->pdata->ssp_clks[clkdev->id].rate_cfg, - rate); - if (!rcfg) - return -EINVAL; - - clk_type = skl_get_clk_type(clkdev->id); - if (clk_type < 0) - return clk_type; - - skl_fill_clk_ipc(rcfg, clk_type); - clkdev->rate = rate; - - return 0; -} - -static unsigned long skl_clk_recalc_rate(struct clk_hw *hw, - unsigned long parent_rate) -{ - struct skl_clk *clkdev = to_skl_clk(hw); - - if (clkdev->rate) - return clkdev->rate; - - return 0; -} - -/* Not supported by clk driver. Implemented to satisfy clk fw */ -static long skl_clk_round_rate(struct clk_hw *hw, unsigned long rate, - unsigned long *parent_rate) -{ - return rate; -} - -/* - * prepare/unprepare are used instead of enable/disable as IPC will be sent - * in non-atomic context. - */ -static const struct clk_ops skl_clk_ops = { - .prepare = skl_clk_prepare, - .unprepare = skl_clk_unprepare, - .set_rate = skl_clk_set_rate, - .round_rate = skl_clk_round_rate, - .recalc_rate = skl_clk_recalc_rate, -}; - -static void unregister_parent_src_clk(struct skl_clk_parent *pclk, - unsigned int id) -{ - while (id--) { - clkdev_drop(pclk[id].lookup); - clk_hw_unregister_fixed_rate(pclk[id].hw); - } -} - -static void unregister_src_clk(struct skl_clk_data *dclk) -{ - while (dclk->avail_clk_cnt--) - clkdev_drop(dclk->clk[dclk->avail_clk_cnt]->lookup); -} - -static int skl_register_parent_clks(struct device *dev, - struct skl_clk_parent *parent, - struct skl_clk_parent_src *pclk) -{ - int i, ret; - - for (i = 0; i < SKL_MAX_CLK_SRC; i++) { - - /* Register Parent clock */ - parent[i].hw = clk_hw_register_fixed_rate(dev, pclk[i].name, - pclk[i].parent_name, 0, pclk[i].rate); - if (IS_ERR(parent[i].hw)) { - ret = PTR_ERR(parent[i].hw); - goto err; - } - - parent[i].lookup = clkdev_hw_create(parent[i].hw, pclk[i].name, - NULL); - if (!parent[i].lookup) { - clk_hw_unregister_fixed_rate(parent[i].hw); - ret = -ENOMEM; - goto err; - } - } - - return 0; -err: - unregister_parent_src_clk(parent, i); - return ret; -} - -/* Assign fmt_config to clk_data */ -static struct skl_clk *register_skl_clk(struct device *dev, - struct skl_ssp_clk *clk, - struct skl_clk_pdata *clk_pdata, int id) -{ - struct clk_init_data init; - struct skl_clk *clkdev; - int ret; - - clkdev = devm_kzalloc(dev, sizeof(*clkdev), GFP_KERNEL); - if (!clkdev) - return ERR_PTR(-ENOMEM); - - init.name = clk->name; - init.ops = &skl_clk_ops; - init.flags = CLK_SET_RATE_GATE; - init.parent_names = &clk->parent_name; - init.num_parents = 1; - clkdev->hw.init = &init; - clkdev->pdata = clk_pdata; - - clkdev->id = id; - ret = devm_clk_hw_register(dev, &clkdev->hw); - if (ret) { - clkdev = ERR_PTR(ret); - return clkdev; - } - - clkdev->lookup = clkdev_hw_create(&clkdev->hw, init.name, NULL); - if (!clkdev->lookup) - clkdev = ERR_PTR(-ENOMEM); - - return clkdev; -} - -static int skl_clk_dev_probe(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct device *parent_dev = dev->parent; - struct skl_clk_parent_src *parent_clks; - struct skl_clk_pdata *clk_pdata; - struct skl_clk_data *data; - struct skl_ssp_clk *clks; - int ret, i; - - clk_pdata = dev_get_platdata(&pdev->dev); - parent_clks = clk_pdata->parent_clks; - clks = clk_pdata->ssp_clks; - if (!parent_clks || !clks) - return -EIO; - - data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); - if (!data) - return -ENOMEM; - - /* Register Parent clock */ - ret = skl_register_parent_clks(parent_dev, data->parent, parent_clks); - if (ret < 0) - return ret; - - for (i = 0; i < clk_pdata->num_clks; i++) { - /* - * Only register valid clocks - * i.e. for which nhlt entry is present. - */ - if (clks[i].rate_cfg[0].rate == 0) - continue; - - data->clk[data->avail_clk_cnt] = register_skl_clk(dev, - &clks[i], clk_pdata, i); - - if (IS_ERR(data->clk[data->avail_clk_cnt])) { - ret = PTR_ERR(data->clk[data->avail_clk_cnt]); - goto err_unreg_skl_clk; - } - - data->avail_clk_cnt++; - } - - platform_set_drvdata(pdev, data); - - return 0; - -err_unreg_skl_clk: - unregister_src_clk(data); - unregister_parent_src_clk(data->parent, SKL_MAX_CLK_SRC); - - return ret; -} - -static void skl_clk_dev_remove(struct platform_device *pdev) -{ - struct skl_clk_data *data; - - data = platform_get_drvdata(pdev); - unregister_src_clk(data); - unregister_parent_src_clk(data->parent, SKL_MAX_CLK_SRC); -} - -static struct platform_driver skl_clk_driver = { - .driver = { - .name = "skl-ssp-clk", - }, - .probe = skl_clk_dev_probe, - .remove_new = skl_clk_dev_remove, -}; - -module_platform_driver(skl_clk_driver); - -MODULE_DESCRIPTION("Skylake clock driver"); -MODULE_AUTHOR("Jaikrishna Nemallapudi "); -MODULE_AUTHOR("Subhransu S. Prusty "); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:skl-ssp-clk"); diff --git a/sound/soc/intel/skylake/skl-ssp-clk.h b/sound/soc/intel/skylake/skl-ssp-clk.h deleted file mode 100644 index b7852c7f277b..000000000000 --- a/sound/soc/intel/skylake/skl-ssp-clk.h +++ /dev/null @@ -1,108 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * skl-ssp-clk.h - Skylake ssp clock information and ipc structure - * - * Copyright (C) 2017 Intel Corp - * Author: Jaikrishna Nemallapudi - * Author: Subhransu S. Prusty - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#ifndef SOUND_SOC_SKL_SSP_CLK_H -#define SOUND_SOC_SKL_SSP_CLK_H - -#define SKL_MAX_SSP 6 -/* xtal/cardinal/pll, parent of ssp clocks and mclk */ -#define SKL_MAX_CLK_SRC 3 -#define SKL_MAX_SSP_CLK_TYPES 3 /* mclk, sclk, sclkfs */ - -#define SKL_MAX_CLK_CNT (SKL_MAX_SSP * SKL_MAX_SSP_CLK_TYPES) - -/* Max number of configurations supported for each clock */ -#define SKL_MAX_CLK_RATES 10 - -#define SKL_SCLK_OFS SKL_MAX_SSP -#define SKL_SCLKFS_OFS (SKL_SCLK_OFS + SKL_MAX_SSP) - -enum skl_clk_type { - SKL_MCLK, - SKL_SCLK, - SKL_SCLK_FS, -}; - -enum skl_clk_src_type { - SKL_XTAL, - SKL_CARDINAL, - SKL_PLL, -}; - -struct skl_clk_parent_src { - u8 clk_id; - const char *name; - unsigned long rate; - const char *parent_name; -}; - -struct skl_tlv_hdr { - u32 type; - u32 size; -}; - -struct skl_dmactrl_mclk_cfg { - struct skl_tlv_hdr hdr; - /* DMA Clk TLV params */ - u32 clk_warm_up:16; - u32 mclk:1; - u32 warm_up_over:1; - u32 rsvd0:14; - u32 clk_stop_delay:16; - u32 keep_running:1; - u32 clk_stop_over:1; - u32 rsvd1:14; -}; - -struct skl_dmactrl_sclkfs_cfg { - struct skl_tlv_hdr hdr; - /* DMA SClk&FS TLV params */ - u32 sampling_frequency; - u32 bit_depth; - u32 channel_map; - u32 channel_config; - u32 interleaving_style; - u32 number_of_channels : 8; - u32 valid_bit_depth : 8; - u32 sample_type : 8; - u32 reserved : 8; -}; - -union skl_clk_ctrl_ipc { - struct skl_dmactrl_mclk_cfg mclk; - struct skl_dmactrl_sclkfs_cfg sclk_fs; -}; - -struct skl_clk_rate_cfg_table { - unsigned long rate; - union skl_clk_ctrl_ipc dma_ctl_ipc; - void *config; -}; - -/* - * rate for mclk will be in rates[0]. For sclk and sclkfs, rates[] store - * all possible clocks ssp can generate for that platform. - */ -struct skl_ssp_clk { - const char *name; - const char *parent_name; - struct skl_clk_rate_cfg_table rate_cfg[SKL_MAX_CLK_RATES]; -}; - -struct skl_clk_pdata { - struct skl_clk_parent_src *parent_clks; - int num_clks; - struct skl_ssp_clk *ssp_clks; - void *pvt_data; -}; - -#endif /* SOUND_SOC_SKL_SSP_CLK_H */ diff --git a/sound/soc/intel/skylake/skl-sst-cldma.c b/sound/soc/intel/skylake/skl-sst-cldma.c deleted file mode 100644 index b0204ea00f07..000000000000 --- a/sound/soc/intel/skylake/skl-sst-cldma.c +++ /dev/null @@ -1,373 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-sst-cldma.c - Code Loader DMA handler - * - * Copyright (C) 2015, Intel Corporation. - * Author: Subhransu S. Prusty - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" - -static void skl_cldma_int_enable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPIC, - SKL_ADSPIC_CL_DMA, SKL_ADSPIC_CL_DMA); -} - -void skl_cldma_int_disable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits_unlocked(ctx, - SKL_ADSP_REG_ADSPIC, SKL_ADSPIC_CL_DMA, 0); -} - -static void skl_cldma_stream_run(struct sst_dsp *ctx, bool enable) -{ - unsigned char val; - int timeout; - - sst_dsp_shim_update_bits_unlocked(ctx, - SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_RUN_MASK, CL_SD_CTL_RUN(enable)); - - udelay(3); - timeout = 300; - do { - /* waiting for hardware to report that the stream Run bit set */ - val = sst_dsp_shim_read(ctx, SKL_ADSP_REG_CL_SD_CTL) & - CL_SD_CTL_RUN_MASK; - if (enable && val) - break; - else if (!enable && !val) - break; - udelay(3); - } while (--timeout); - - if (timeout == 0) - dev_err(ctx->dev, "Failed to set Run bit=%d enable=%d\n", val, enable); -} - -static void skl_cldma_stream_clear(struct sst_dsp *ctx) -{ - /* make sure Run bit is cleared before setting stream register */ - skl_cldma_stream_run(ctx, 0); - - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_IOCE_MASK, CL_SD_CTL_IOCE(0)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_FEIE_MASK, CL_SD_CTL_FEIE(0)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_DEIE_MASK, CL_SD_CTL_DEIE(0)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_STRM_MASK, CL_SD_CTL_STRM(0)); - - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPL, CL_SD_BDLPLBA(0)); - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPU, 0); - - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_CBL, 0); - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_LVI, 0); -} - -/* Code loader helper APIs */ -static void skl_cldma_setup_bdle(struct sst_dsp *ctx, - struct snd_dma_buffer *dmab_data, - __le32 **bdlp, int size, int with_ioc) -{ - __le32 *bdl = *bdlp; - int remaining = ctx->cl_dev.bufsize; - int offset = 0; - - ctx->cl_dev.frags = 0; - while (remaining > 0) { - phys_addr_t addr; - int chunk; - - addr = snd_sgbuf_get_addr(dmab_data, offset); - bdl[0] = cpu_to_le32(lower_32_bits(addr)); - bdl[1] = cpu_to_le32(upper_32_bits(addr)); - chunk = snd_sgbuf_get_chunk_size(dmab_data, offset, size); - bdl[2] = cpu_to_le32(chunk); - - remaining -= chunk; - bdl[3] = (remaining > 0) ? 0 : cpu_to_le32(0x01); - - bdl += 4; - offset += chunk; - ctx->cl_dev.frags++; - } -} - -/* - * Setup controller - * Configure the registers to update the dma buffer address and - * enable interrupts. - * Note: Using the channel 1 for transfer - */ -static void skl_cldma_setup_controller(struct sst_dsp *ctx, - struct snd_dma_buffer *dmab_bdl, unsigned int max_size, - u32 count) -{ - skl_cldma_stream_clear(ctx); - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPL, - CL_SD_BDLPLBA(dmab_bdl->addr)); - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_BDLPU, - CL_SD_BDLPUBA(dmab_bdl->addr)); - - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_CBL, max_size); - sst_dsp_shim_write(ctx, SKL_ADSP_REG_CL_SD_LVI, count - 1); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_IOCE_MASK, CL_SD_CTL_IOCE(1)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_FEIE_MASK, CL_SD_CTL_FEIE(1)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_DEIE_MASK, CL_SD_CTL_DEIE(1)); - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_CL_SD_CTL, - CL_SD_CTL_STRM_MASK, CL_SD_CTL_STRM(FW_CL_STREAM_NUMBER)); -} - -static void skl_cldma_setup_spb(struct sst_dsp *ctx, - unsigned int size, bool enable) -{ - if (enable) - sst_dsp_shim_update_bits_unlocked(ctx, - SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL, - CL_SPBFIFO_SPBFCCTL_SPIBE_MASK, - CL_SPBFIFO_SPBFCCTL_SPIBE(1)); - - sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_CL_SPBFIFO_SPIB, size); -} - -static void skl_cldma_cleanup_spb(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits_unlocked(ctx, - SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL, - CL_SPBFIFO_SPBFCCTL_SPIBE_MASK, - CL_SPBFIFO_SPBFCCTL_SPIBE(0)); - - sst_dsp_shim_write_unlocked(ctx, SKL_ADSP_REG_CL_SPBFIFO_SPIB, 0); -} - -static void skl_cldma_cleanup(struct sst_dsp *ctx) -{ - skl_cldma_cleanup_spb(ctx); - skl_cldma_stream_clear(ctx); - - ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); - ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_bdl); -} - -int skl_cldma_wait_interruptible(struct sst_dsp *ctx) -{ - int ret = 0; - - if (!wait_event_timeout(ctx->cl_dev.wait_queue, - ctx->cl_dev.wait_condition, - msecs_to_jiffies(SKL_WAIT_TIMEOUT))) { - dev_err(ctx->dev, "%s: Wait timeout\n", __func__); - ret = -EIO; - goto cleanup; - } - - dev_dbg(ctx->dev, "%s: Event wake\n", __func__); - if (ctx->cl_dev.wake_status != SKL_CL_DMA_BUF_COMPLETE) { - dev_err(ctx->dev, "%s: DMA Error\n", __func__); - ret = -EIO; - } - -cleanup: - ctx->cl_dev.wake_status = SKL_CL_DMA_STATUS_NONE; - return ret; -} - -static void skl_cldma_stop(struct sst_dsp *ctx) -{ - skl_cldma_stream_run(ctx, false); -} - -static void skl_cldma_fill_buffer(struct sst_dsp *ctx, unsigned int size, - const void *curr_pos, bool intr_enable, bool trigger) -{ - dev_dbg(ctx->dev, "Size: %x, intr_enable: %d\n", size, intr_enable); - dev_dbg(ctx->dev, "buf_pos_index:%d, trigger:%d\n", - ctx->cl_dev.dma_buffer_offset, trigger); - dev_dbg(ctx->dev, "spib position: %d\n", ctx->cl_dev.curr_spib_pos); - - /* - * Check if the size exceeds buffer boundary. If it exceeds - * max_buffer size, then copy till buffer size and then copy - * remaining buffer from the start of ring buffer. - */ - if (ctx->cl_dev.dma_buffer_offset + size > ctx->cl_dev.bufsize) { - unsigned int size_b = ctx->cl_dev.bufsize - - ctx->cl_dev.dma_buffer_offset; - memcpy(ctx->cl_dev.dmab_data.area + ctx->cl_dev.dma_buffer_offset, - curr_pos, size_b); - size -= size_b; - curr_pos += size_b; - ctx->cl_dev.dma_buffer_offset = 0; - } - - memcpy(ctx->cl_dev.dmab_data.area + ctx->cl_dev.dma_buffer_offset, - curr_pos, size); - - if (ctx->cl_dev.curr_spib_pos == ctx->cl_dev.bufsize) - ctx->cl_dev.dma_buffer_offset = 0; - else - ctx->cl_dev.dma_buffer_offset = ctx->cl_dev.curr_spib_pos; - - ctx->cl_dev.wait_condition = false; - - if (intr_enable) - skl_cldma_int_enable(ctx); - - ctx->cl_dev.ops.cl_setup_spb(ctx, ctx->cl_dev.curr_spib_pos, trigger); - if (trigger) - ctx->cl_dev.ops.cl_trigger(ctx, true); -} - -/* - * The CL dma doesn't have any way to update the transfer status until a BDL - * buffer is fully transferred - * - * So Copying is divided in two parts. - * 1. Interrupt on buffer done where the size to be transferred is more than - * ring buffer size. - * 2. Polling on fw register to identify if data left to transferred doesn't - * fill the ring buffer. Caller takes care of polling the required status - * register to identify the transfer status. - * 3. if wait flag is set, waits for DBL interrupt to copy the next chunk till - * bytes_left is 0. - * if wait flag is not set, doesn't wait for BDL interrupt. after ccopying - * the first chunk return the no of bytes_left to be copied. - */ -static int -skl_cldma_copy_to_buf(struct sst_dsp *ctx, const void *bin, - u32 total_size, bool wait) -{ - int ret; - bool start = true; - unsigned int excess_bytes; - u32 size; - unsigned int bytes_left = total_size; - const void *curr_pos = bin; - - if (total_size <= 0) - return -EINVAL; - - dev_dbg(ctx->dev, "%s: Total binary size: %u\n", __func__, bytes_left); - - while (bytes_left) { - if (bytes_left > ctx->cl_dev.bufsize) { - - /* - * dma transfers only till the write pointer as - * updated in spib - */ - if (ctx->cl_dev.curr_spib_pos == 0) - ctx->cl_dev.curr_spib_pos = ctx->cl_dev.bufsize; - - size = ctx->cl_dev.bufsize; - skl_cldma_fill_buffer(ctx, size, curr_pos, true, start); - - if (wait) { - start = false; - ret = skl_cldma_wait_interruptible(ctx); - if (ret < 0) { - skl_cldma_stop(ctx); - return ret; - } - } - } else { - skl_cldma_int_disable(ctx); - - if ((ctx->cl_dev.curr_spib_pos + bytes_left) - <= ctx->cl_dev.bufsize) { - ctx->cl_dev.curr_spib_pos += bytes_left; - } else { - excess_bytes = bytes_left - - (ctx->cl_dev.bufsize - - ctx->cl_dev.curr_spib_pos); - ctx->cl_dev.curr_spib_pos = excess_bytes; - } - - size = bytes_left; - skl_cldma_fill_buffer(ctx, size, - curr_pos, false, start); - } - bytes_left -= size; - curr_pos = curr_pos + size; - if (!wait) - return bytes_left; - } - - return bytes_left; -} - -void skl_cldma_process_intr(struct sst_dsp *ctx) -{ - u8 cl_dma_intr_status; - - cl_dma_intr_status = - sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_CL_SD_STS); - - if (!(cl_dma_intr_status & SKL_CL_DMA_SD_INT_COMPLETE)) - ctx->cl_dev.wake_status = SKL_CL_DMA_ERR; - else - ctx->cl_dev.wake_status = SKL_CL_DMA_BUF_COMPLETE; - - ctx->cl_dev.wait_condition = true; - wake_up(&ctx->cl_dev.wait_queue); -} - -int skl_cldma_prepare(struct sst_dsp *ctx) -{ - int ret; - __le32 *bdl; - - ctx->cl_dev.bufsize = SKL_MAX_BUFFER_SIZE; - - /* Allocate cl ops */ - ctx->cl_dev.ops.cl_setup_bdle = skl_cldma_setup_bdle; - ctx->cl_dev.ops.cl_setup_controller = skl_cldma_setup_controller; - ctx->cl_dev.ops.cl_setup_spb = skl_cldma_setup_spb; - ctx->cl_dev.ops.cl_cleanup_spb = skl_cldma_cleanup_spb; - ctx->cl_dev.ops.cl_trigger = skl_cldma_stream_run; - ctx->cl_dev.ops.cl_cleanup_controller = skl_cldma_cleanup; - ctx->cl_dev.ops.cl_copy_to_dmabuf = skl_cldma_copy_to_buf; - ctx->cl_dev.ops.cl_stop_dma = skl_cldma_stop; - - /* Allocate buffer*/ - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, ctx->dev, ctx->cl_dev.bufsize, - &ctx->cl_dev.dmab_data); - if (ret < 0) { - dev_err(ctx->dev, "Alloc buffer for base fw failed: %x\n", ret); - return ret; - } - - /* Setup Code loader BDL */ - ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, ctx->dev, BDL_SIZE, &ctx->cl_dev.dmab_bdl); - if (ret < 0) { - dev_err(ctx->dev, "Alloc buffer for blde failed: %x\n", ret); - ctx->dsp_ops.free_dma_buf(ctx->dev, &ctx->cl_dev.dmab_data); - return ret; - } - bdl = (__le32 *)ctx->cl_dev.dmab_bdl.area; - - /* Allocate BDLs */ - ctx->cl_dev.ops.cl_setup_bdle(ctx, &ctx->cl_dev.dmab_data, - &bdl, ctx->cl_dev.bufsize, 1); - ctx->cl_dev.ops.cl_setup_controller(ctx, &ctx->cl_dev.dmab_bdl, - ctx->cl_dev.bufsize, ctx->cl_dev.frags); - - ctx->cl_dev.curr_spib_pos = 0; - ctx->cl_dev.dma_buffer_offset = 0; - init_waitqueue_head(&ctx->cl_dev.wait_queue); - - return ret; -} diff --git a/sound/soc/intel/skylake/skl-sst-cldma.h b/sound/soc/intel/skylake/skl-sst-cldma.h deleted file mode 100644 index d5e285a69baa..000000000000 --- a/sound/soc/intel/skylake/skl-sst-cldma.h +++ /dev/null @@ -1,243 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Intel Code Loader DMA support - * - * Copyright (C) 2015, Intel Corporation. - */ - -#ifndef SKL_SST_CLDMA_H_ -#define SKL_SST_CLDMA_H_ - -#define FW_CL_STREAM_NUMBER 0x1 - -#define DMA_ADDRESS_128_BITS_ALIGNMENT 7 -#define BDL_ALIGN(x) (x >> DMA_ADDRESS_128_BITS_ALIGNMENT) - -#define SKL_ADSPIC_CL_DMA 0x2 -#define SKL_ADSPIS_CL_DMA 0x2 -#define SKL_CL_DMA_SD_INT_DESC_ERR 0x10 /* Descriptor error interrupt */ -#define SKL_CL_DMA_SD_INT_FIFO_ERR 0x08 /* FIFO error interrupt */ -#define SKL_CL_DMA_SD_INT_COMPLETE 0x04 /* Buffer completion interrupt */ - -/* Intel HD Audio Code Loader DMA Registers */ - -#define HDA_ADSP_LOADER_BASE 0x80 - -/* Stream Registers */ -#define SKL_ADSP_REG_CL_SD_CTL (HDA_ADSP_LOADER_BASE + 0x00) -#define SKL_ADSP_REG_CL_SD_STS (HDA_ADSP_LOADER_BASE + 0x03) -#define SKL_ADSP_REG_CL_SD_LPIB (HDA_ADSP_LOADER_BASE + 0x04) -#define SKL_ADSP_REG_CL_SD_CBL (HDA_ADSP_LOADER_BASE + 0x08) -#define SKL_ADSP_REG_CL_SD_LVI (HDA_ADSP_LOADER_BASE + 0x0c) -#define SKL_ADSP_REG_CL_SD_FIFOW (HDA_ADSP_LOADER_BASE + 0x0e) -#define SKL_ADSP_REG_CL_SD_FIFOSIZE (HDA_ADSP_LOADER_BASE + 0x10) -#define SKL_ADSP_REG_CL_SD_FORMAT (HDA_ADSP_LOADER_BASE + 0x12) -#define SKL_ADSP_REG_CL_SD_FIFOL (HDA_ADSP_LOADER_BASE + 0x14) -#define SKL_ADSP_REG_CL_SD_BDLPL (HDA_ADSP_LOADER_BASE + 0x18) -#define SKL_ADSP_REG_CL_SD_BDLPU (HDA_ADSP_LOADER_BASE + 0x1c) - -/* CL: Software Position Based FIFO Capability Registers */ -#define SKL_ADSP_REG_CL_SPBFIFO (HDA_ADSP_LOADER_BASE + 0x20) -#define SKL_ADSP_REG_CL_SPBFIFO_SPBFCH (SKL_ADSP_REG_CL_SPBFIFO + 0x0) -#define SKL_ADSP_REG_CL_SPBFIFO_SPBFCCTL (SKL_ADSP_REG_CL_SPBFIFO + 0x4) -#define SKL_ADSP_REG_CL_SPBFIFO_SPIB (SKL_ADSP_REG_CL_SPBFIFO + 0x8) -#define SKL_ADSP_REG_CL_SPBFIFO_MAXFIFOS (SKL_ADSP_REG_CL_SPBFIFO + 0xc) - -/* CL: Stream Descriptor x Control */ - -/* Stream Reset */ -#define CL_SD_CTL_SRST_SHIFT 0 -#define CL_SD_CTL_SRST_MASK (1 << CL_SD_CTL_SRST_SHIFT) -#define CL_SD_CTL_SRST(x) \ - ((x << CL_SD_CTL_SRST_SHIFT) & CL_SD_CTL_SRST_MASK) - -/* Stream Run */ -#define CL_SD_CTL_RUN_SHIFT 1 -#define CL_SD_CTL_RUN_MASK (1 << CL_SD_CTL_RUN_SHIFT) -#define CL_SD_CTL_RUN(x) \ - ((x << CL_SD_CTL_RUN_SHIFT) & CL_SD_CTL_RUN_MASK) - -/* Interrupt On Completion Enable */ -#define CL_SD_CTL_IOCE_SHIFT 2 -#define CL_SD_CTL_IOCE_MASK (1 << CL_SD_CTL_IOCE_SHIFT) -#define CL_SD_CTL_IOCE(x) \ - ((x << CL_SD_CTL_IOCE_SHIFT) & CL_SD_CTL_IOCE_MASK) - -/* FIFO Error Interrupt Enable */ -#define CL_SD_CTL_FEIE_SHIFT 3 -#define CL_SD_CTL_FEIE_MASK (1 << CL_SD_CTL_FEIE_SHIFT) -#define CL_SD_CTL_FEIE(x) \ - ((x << CL_SD_CTL_FEIE_SHIFT) & CL_SD_CTL_FEIE_MASK) - -/* Descriptor Error Interrupt Enable */ -#define CL_SD_CTL_DEIE_SHIFT 4 -#define CL_SD_CTL_DEIE_MASK (1 << CL_SD_CTL_DEIE_SHIFT) -#define CL_SD_CTL_DEIE(x) \ - ((x << CL_SD_CTL_DEIE_SHIFT) & CL_SD_CTL_DEIE_MASK) - -/* FIFO Limit Change */ -#define CL_SD_CTL_FIFOLC_SHIFT 5 -#define CL_SD_CTL_FIFOLC_MASK (1 << CL_SD_CTL_FIFOLC_SHIFT) -#define CL_SD_CTL_FIFOLC(x) \ - ((x << CL_SD_CTL_FIFOLC_SHIFT) & CL_SD_CTL_FIFOLC_MASK) - -/* Stripe Control */ -#define CL_SD_CTL_STRIPE_SHIFT 16 -#define CL_SD_CTL_STRIPE_MASK (0x3 << CL_SD_CTL_STRIPE_SHIFT) -#define CL_SD_CTL_STRIPE(x) \ - ((x << CL_SD_CTL_STRIPE_SHIFT) & CL_SD_CTL_STRIPE_MASK) - -/* Traffic Priority */ -#define CL_SD_CTL_TP_SHIFT 18 -#define CL_SD_CTL_TP_MASK (1 << CL_SD_CTL_TP_SHIFT) -#define CL_SD_CTL_TP(x) \ - ((x << CL_SD_CTL_TP_SHIFT) & CL_SD_CTL_TP_MASK) - -/* Bidirectional Direction Control */ -#define CL_SD_CTL_DIR_SHIFT 19 -#define CL_SD_CTL_DIR_MASK (1 << CL_SD_CTL_DIR_SHIFT) -#define CL_SD_CTL_DIR(x) \ - ((x << CL_SD_CTL_DIR_SHIFT) & CL_SD_CTL_DIR_MASK) - -/* Stream Number */ -#define CL_SD_CTL_STRM_SHIFT 20 -#define CL_SD_CTL_STRM_MASK (0xf << CL_SD_CTL_STRM_SHIFT) -#define CL_SD_CTL_STRM(x) \ - ((x << CL_SD_CTL_STRM_SHIFT) & CL_SD_CTL_STRM_MASK) - -/* CL: Stream Descriptor x Status */ - -/* Buffer Completion Interrupt Status */ -#define CL_SD_STS_BCIS(x) CL_SD_CTL_IOCE(x) - -/* FIFO Error */ -#define CL_SD_STS_FIFOE(x) CL_SD_CTL_FEIE(x) - -/* Descriptor Error */ -#define CL_SD_STS_DESE(x) CL_SD_CTL_DEIE(x) - -/* FIFO Ready */ -#define CL_SD_STS_FIFORDY(x) CL_SD_CTL_FIFOLC(x) - - -/* CL: Stream Descriptor x Last Valid Index */ -#define CL_SD_LVI_SHIFT 0 -#define CL_SD_LVI_MASK (0xff << CL_SD_LVI_SHIFT) -#define CL_SD_LVI(x) ((x << CL_SD_LVI_SHIFT) & CL_SD_LVI_MASK) - -/* CL: Stream Descriptor x FIFO Eviction Watermark */ -#define CL_SD_FIFOW_SHIFT 0 -#define CL_SD_FIFOW_MASK (0x7 << CL_SD_FIFOW_SHIFT) -#define CL_SD_FIFOW(x) \ - ((x << CL_SD_FIFOW_SHIFT) & CL_SD_FIFOW_MASK) - -/* CL: Stream Descriptor x Buffer Descriptor List Pointer Lower Base Address */ - -/* Protect Bits */ -#define CL_SD_BDLPLBA_PROT_SHIFT 0 -#define CL_SD_BDLPLBA_PROT_MASK (1 << CL_SD_BDLPLBA_PROT_SHIFT) -#define CL_SD_BDLPLBA_PROT(x) \ - ((x << CL_SD_BDLPLBA_PROT_SHIFT) & CL_SD_BDLPLBA_PROT_MASK) - -/* Buffer Descriptor List Lower Base Address */ -#define CL_SD_BDLPLBA_SHIFT 7 -#define CL_SD_BDLPLBA_MASK (0x1ffffff << CL_SD_BDLPLBA_SHIFT) -#define CL_SD_BDLPLBA(x) \ - ((BDL_ALIGN(lower_32_bits(x)) << CL_SD_BDLPLBA_SHIFT) & CL_SD_BDLPLBA_MASK) - -/* Buffer Descriptor List Upper Base Address */ -#define CL_SD_BDLPUBA_SHIFT 0 -#define CL_SD_BDLPUBA_MASK (0xffffffff << CL_SD_BDLPUBA_SHIFT) -#define CL_SD_BDLPUBA(x) \ - ((upper_32_bits(x) << CL_SD_BDLPUBA_SHIFT) & CL_SD_BDLPUBA_MASK) - -/* - * Code Loader - Software Position Based FIFO - * Capability Registers x Software Position Based FIFO Header - */ - -/* Next Capability Pointer */ -#define CL_SPBFIFO_SPBFCH_PTR_SHIFT 0 -#define CL_SPBFIFO_SPBFCH_PTR_MASK (0xff << CL_SPBFIFO_SPBFCH_PTR_SHIFT) -#define CL_SPBFIFO_SPBFCH_PTR(x) \ - ((x << CL_SPBFIFO_SPBFCH_PTR_SHIFT) & CL_SPBFIFO_SPBFCH_PTR_MASK) - -/* Capability Identifier */ -#define CL_SPBFIFO_SPBFCH_ID_SHIFT 16 -#define CL_SPBFIFO_SPBFCH_ID_MASK (0xfff << CL_SPBFIFO_SPBFCH_ID_SHIFT) -#define CL_SPBFIFO_SPBFCH_ID(x) \ - ((x << CL_SPBFIFO_SPBFCH_ID_SHIFT) & CL_SPBFIFO_SPBFCH_ID_MASK) - -/* Capability Version */ -#define CL_SPBFIFO_SPBFCH_VER_SHIFT 28 -#define CL_SPBFIFO_SPBFCH_VER_MASK (0xf << CL_SPBFIFO_SPBFCH_VER_SHIFT) -#define CL_SPBFIFO_SPBFCH_VER(x) \ - ((x << CL_SPBFIFO_SPBFCH_VER_SHIFT) & CL_SPBFIFO_SPBFCH_VER_MASK) - -/* Software Position in Buffer Enable */ -#define CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT 0 -#define CL_SPBFIFO_SPBFCCTL_SPIBE_MASK (1 << CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT) -#define CL_SPBFIFO_SPBFCCTL_SPIBE(x) \ - ((x << CL_SPBFIFO_SPBFCCTL_SPIBE_SHIFT) & CL_SPBFIFO_SPBFCCTL_SPIBE_MASK) - -/* SST IPC SKL defines */ -#define SKL_WAIT_TIMEOUT 500 /* 500 msec */ -#define SKL_MAX_BUFFER_SIZE (32 * PAGE_SIZE) - -enum skl_cl_dma_wake_states { - SKL_CL_DMA_STATUS_NONE = 0, - SKL_CL_DMA_BUF_COMPLETE, - SKL_CL_DMA_ERR, /* TODO: Expand the error states */ -}; - -struct sst_dsp; - -struct skl_cl_dev_ops { - void (*cl_setup_bdle)(struct sst_dsp *ctx, - struct snd_dma_buffer *dmab_data, - __le32 **bdlp, int size, int with_ioc); - void (*cl_setup_controller)(struct sst_dsp *ctx, - struct snd_dma_buffer *dmab_bdl, - unsigned int max_size, u32 page_count); - void (*cl_setup_spb)(struct sst_dsp *ctx, - unsigned int size, bool enable); - void (*cl_cleanup_spb)(struct sst_dsp *ctx); - void (*cl_trigger)(struct sst_dsp *ctx, bool enable); - void (*cl_cleanup_controller)(struct sst_dsp *ctx); - int (*cl_copy_to_dmabuf)(struct sst_dsp *ctx, - const void *bin, u32 size, bool wait); - void (*cl_stop_dma)(struct sst_dsp *ctx); -}; - -/** - * skl_cl_dev - holds information for code loader dma transfer - * - * @dmab_data: buffer pointer - * @dmab_bdl: buffer descriptor list - * @bufsize: ring buffer size - * @frags: Last valid buffer descriptor index in the BDL - * @curr_spib_pos: Current position in ring buffer - * @dma_buffer_offset: dma buffer offset - * @ops: operations supported on CL dma - * @wait_queue: wait queue to wake for wake event - * @wake_status: DMA wake status - * @wait_condition: condition to wait on wait queue - * @cl_dma_lock: for synchronized access to cldma - */ -struct skl_cl_dev { - struct snd_dma_buffer dmab_data; - struct snd_dma_buffer dmab_bdl; - - unsigned int bufsize; - unsigned int frags; - - unsigned int curr_spib_pos; - unsigned int dma_buffer_offset; - struct skl_cl_dev_ops ops; - - wait_queue_head_t wait_queue; - int wake_status; - bool wait_condition; -}; - -#endif /* SKL_SST_CLDMA_H_ */ diff --git a/sound/soc/intel/skylake/skl-sst-dsp.c b/sound/soc/intel/skylake/skl-sst-dsp.c deleted file mode 100644 index 4ae3eae0d1fd..000000000000 --- a/sound/soc/intel/skylake/skl-sst-dsp.c +++ /dev/null @@ -1,462 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-sst-dsp.c - SKL SST library generic function - * - * Copyright (C) 2014-15, Intel Corporation. - * Author:Rafal Redzimski - * Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ -#include - -#include "../common/sst-dsp.h" -#include "../common/sst-ipc.h" -#include "../common/sst-dsp-priv.h" -#include "skl.h" - -/* various timeout values */ -#define SKL_DSP_PU_TO 50 -#define SKL_DSP_PD_TO 50 -#define SKL_DSP_RESET_TO 50 - -void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state) -{ - mutex_lock(&ctx->mutex); - ctx->sst_state = state; - mutex_unlock(&ctx->mutex); -} - -/* - * Initialize core power state and usage count. To be called after - * successful first boot. Hence core 0 will be running and other cores - * will be reset - */ -void skl_dsp_init_core_state(struct sst_dsp *ctx) -{ - struct skl_dev *skl = ctx->thread_context; - int i; - - skl->cores.state[SKL_DSP_CORE0_ID] = SKL_DSP_RUNNING; - skl->cores.usage_count[SKL_DSP_CORE0_ID] = 1; - - for (i = SKL_DSP_CORE0_ID + 1; i < skl->cores.count; i++) { - skl->cores.state[i] = SKL_DSP_RESET; - skl->cores.usage_count[i] = 0; - } -} - -/* Get the mask for all enabled cores */ -unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx) -{ - struct skl_dev *skl = ctx->thread_context; - unsigned int core_mask, en_cores_mask; - u32 val; - - core_mask = SKL_DSP_CORES_MASK(skl->cores.count); - - val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS); - - /* Cores having CPA bit set */ - en_cores_mask = (val & SKL_ADSPCS_CPA_MASK(core_mask)) >> - SKL_ADSPCS_CPA_SHIFT; - - /* And cores having CRST bit cleared */ - en_cores_mask &= (~val & SKL_ADSPCS_CRST_MASK(core_mask)) >> - SKL_ADSPCS_CRST_SHIFT; - - /* And cores having CSTALL bit cleared */ - en_cores_mask &= (~val & SKL_ADSPCS_CSTALL_MASK(core_mask)) >> - SKL_ADSPCS_CSTALL_SHIFT; - en_cores_mask &= core_mask; - - dev_dbg(ctx->dev, "DSP enabled cores mask = %x\n", en_cores_mask); - - return en_cores_mask; -} - -static int -skl_dsp_core_set_reset_state(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, - SKL_ADSP_REG_ADSPCS, SKL_ADSPCS_CRST_MASK(core_mask), - SKL_ADSPCS_CRST_MASK(core_mask)); - - /* poll with timeout to check if operation successful */ - ret = sst_dsp_register_poll(ctx, - SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CRST_MASK(core_mask), - SKL_ADSPCS_CRST_MASK(core_mask), - SKL_DSP_RESET_TO, - "Set reset"); - if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & - SKL_ADSPCS_CRST_MASK(core_mask)) != - SKL_ADSPCS_CRST_MASK(core_mask)) { - dev_err(ctx->dev, "Set reset state failed: core_mask %x\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -int skl_dsp_core_unset_reset_state( - struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - dev_dbg(ctx->dev, "In %s\n", __func__); - - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CRST_MASK(core_mask), 0); - - /* poll with timeout to check if operation successful */ - ret = sst_dsp_register_poll(ctx, - SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CRST_MASK(core_mask), - 0, - SKL_DSP_RESET_TO, - "Unset reset"); - - if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & - SKL_ADSPCS_CRST_MASK(core_mask)) != 0) { - dev_err(ctx->dev, "Unset reset state failed: core_mask %x\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -static bool -is_skl_dsp_core_enable(struct sst_dsp *ctx, unsigned int core_mask) -{ - int val; - bool is_enable; - - val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS); - - is_enable = ((val & SKL_ADSPCS_CPA_MASK(core_mask)) && - (val & SKL_ADSPCS_SPA_MASK(core_mask)) && - !(val & SKL_ADSPCS_CRST_MASK(core_mask)) && - !(val & SKL_ADSPCS_CSTALL_MASK(core_mask))); - - dev_dbg(ctx->dev, "DSP core(s) enabled? %d : core_mask %x\n", - is_enable, core_mask); - - return is_enable; -} - -static int skl_dsp_reset_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* stall core */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CSTALL_MASK(core_mask), - SKL_ADSPCS_CSTALL_MASK(core_mask)); - - /* set reset state */ - return skl_dsp_core_set_reset_state(ctx, core_mask); -} - -int skl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* unset reset state */ - ret = skl_dsp_core_unset_reset_state(ctx, core_mask); - if (ret < 0) - return ret; - - /* run core */ - dev_dbg(ctx->dev, "unstall/run core: core_mask = %x\n", core_mask); - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CSTALL_MASK(core_mask), 0); - - if (!is_skl_dsp_core_enable(ctx, core_mask)) { - skl_dsp_reset_core(ctx, core_mask); - dev_err(ctx->dev, "DSP start core failed: core_mask %x\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -int skl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_SPA_MASK(core_mask), - SKL_ADSPCS_SPA_MASK(core_mask)); - - /* poll with timeout to check if operation successful */ - ret = sst_dsp_register_poll(ctx, - SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CPA_MASK(core_mask), - SKL_ADSPCS_CPA_MASK(core_mask), - SKL_DSP_PU_TO, - "Power up"); - - if ((sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPCS) & - SKL_ADSPCS_CPA_MASK(core_mask)) != - SKL_ADSPCS_CPA_MASK(core_mask)) { - dev_err(ctx->dev, "DSP core power up failed: core_mask %x\n", - core_mask); - ret = -EIO; - } - - return ret; -} - -int skl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask) -{ - /* update bits */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_SPA_MASK(core_mask), 0); - - /* poll with timeout to check if operation successful */ - return sst_dsp_register_poll(ctx, - SKL_ADSP_REG_ADSPCS, - SKL_ADSPCS_CPA_MASK(core_mask), - 0, - SKL_DSP_PD_TO, - "Power down"); -} - -int skl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - /* power up */ - ret = skl_dsp_core_power_up(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "dsp core power up failed: core_mask %x\n", - core_mask); - return ret; - } - - return skl_dsp_start_core(ctx, core_mask); -} - -int skl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask) -{ - int ret; - - ret = skl_dsp_reset_core(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "dsp core reset failed: core_mask %x\n", - core_mask); - return ret; - } - - /* power down core*/ - ret = skl_dsp_core_power_down(ctx, core_mask); - if (ret < 0) { - dev_err(ctx->dev, "dsp core power down fail mask %x: %d\n", - core_mask, ret); - return ret; - } - - if (is_skl_dsp_core_enable(ctx, core_mask)) { - dev_err(ctx->dev, "dsp core disable fail mask %x: %d\n", - core_mask, ret); - ret = -EIO; - } - - return ret; -} - -int skl_dsp_boot(struct sst_dsp *ctx) -{ - int ret; - - if (is_skl_dsp_core_enable(ctx, SKL_DSP_CORE0_MASK)) { - ret = skl_dsp_reset_core(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "dsp core0 reset fail: %d\n", ret); - return ret; - } - - ret = skl_dsp_start_core(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "dsp core0 start fail: %d\n", ret); - return ret; - } - } else { - ret = skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - if (ret < 0) { - dev_err(ctx->dev, "dsp core0 disable fail: %d\n", ret); - return ret; - } - ret = skl_dsp_enable_core(ctx, SKL_DSP_CORE0_MASK); - } - - return ret; -} - -irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id) -{ - struct sst_dsp *ctx = dev_id; - u32 val; - irqreturn_t result = IRQ_NONE; - - spin_lock(&ctx->spinlock); - - val = sst_dsp_shim_read_unlocked(ctx, SKL_ADSP_REG_ADSPIS); - ctx->intr_status = val; - - if (val == 0xffffffff) { - spin_unlock(&ctx->spinlock); - return IRQ_NONE; - } - - if (val & SKL_ADSPIS_IPC) { - skl_ipc_int_disable(ctx); - result = IRQ_WAKE_THREAD; - } - - if (val & SKL_ADSPIS_CL_DMA) { - skl_cldma_int_disable(ctx); - result = IRQ_WAKE_THREAD; - } - - spin_unlock(&ctx->spinlock); - - return result; -} -/* - * skl_dsp_get_core/skl_dsp_put_core will be called inside DAPM context - * within the dapm mutex. Hence no separate lock is used. - */ -int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id) -{ - struct skl_dev *skl = ctx->thread_context; - int ret = 0; - - if (core_id >= skl->cores.count) { - dev_err(ctx->dev, "invalid core id: %d\n", core_id); - return -EINVAL; - } - - skl->cores.usage_count[core_id]++; - - if (skl->cores.state[core_id] == SKL_DSP_RESET) { - ret = ctx->fw_ops.set_state_D0(ctx, core_id); - if (ret < 0) { - dev_err(ctx->dev, "unable to get core%d\n", core_id); - goto out; - } - } - -out: - dev_dbg(ctx->dev, "core id %d state %d usage_count %d\n", - core_id, skl->cores.state[core_id], - skl->cores.usage_count[core_id]); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_dsp_get_core); - -int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id) -{ - struct skl_dev *skl = ctx->thread_context; - int ret = 0; - - if (core_id >= skl->cores.count) { - dev_err(ctx->dev, "invalid core id: %d\n", core_id); - return -EINVAL; - } - - if ((--skl->cores.usage_count[core_id] == 0) && - (skl->cores.state[core_id] != SKL_DSP_RESET)) { - ret = ctx->fw_ops.set_state_D3(ctx, core_id); - if (ret < 0) { - dev_err(ctx->dev, "unable to put core %d: %d\n", - core_id, ret); - skl->cores.usage_count[core_id]++; - } - } - - dev_dbg(ctx->dev, "core id %d state %d usage_count %d\n", - core_id, skl->cores.state[core_id], - skl->cores.usage_count[core_id]); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_dsp_put_core); - -int skl_dsp_wake(struct sst_dsp *ctx) -{ - return skl_dsp_get_core(ctx, SKL_DSP_CORE0_ID); -} -EXPORT_SYMBOL_GPL(skl_dsp_wake); - -int skl_dsp_sleep(struct sst_dsp *ctx) -{ - return skl_dsp_put_core(ctx, SKL_DSP_CORE0_ID); -} -EXPORT_SYMBOL_GPL(skl_dsp_sleep); - -struct sst_dsp *skl_dsp_ctx_init(struct device *dev, - struct sst_dsp_device *sst_dev, int irq) -{ - int ret; - struct sst_dsp *sst; - - sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); - if (sst == NULL) - return NULL; - - spin_lock_init(&sst->spinlock); - mutex_init(&sst->mutex); - sst->dev = dev; - sst->sst_dev = sst_dev; - sst->irq = irq; - sst->ops = sst_dev->ops; - sst->thread_context = sst_dev->thread_context; - - /* Initialise SST Audio DSP */ - if (sst->ops->init) { - ret = sst->ops->init(sst); - if (ret < 0) - return NULL; - } - - return sst; -} - -int skl_dsp_acquire_irq(struct sst_dsp *sst) -{ - struct sst_dsp_device *sst_dev = sst->sst_dev; - int ret; - - /* Register the ISR */ - ret = request_threaded_irq(sst->irq, sst->ops->irq_handler, - sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); - if (ret) - dev_err(sst->dev, "unable to grab threaded IRQ %d, disabling device\n", - sst->irq); - - return ret; -} - -void skl_dsp_free(struct sst_dsp *dsp) -{ - skl_ipc_int_disable(dsp); - - free_irq(dsp->irq, dsp); - skl_ipc_op_int_disable(dsp); - skl_dsp_disable_core(dsp, SKL_DSP_CORE0_MASK); -} -EXPORT_SYMBOL_GPL(skl_dsp_free); - -bool is_skl_dsp_running(struct sst_dsp *ctx) -{ - return (ctx->sst_state == SKL_DSP_RUNNING); -} -EXPORT_SYMBOL_GPL(is_skl_dsp_running); diff --git a/sound/soc/intel/skylake/skl-sst-dsp.h b/sound/soc/intel/skylake/skl-sst-dsp.h deleted file mode 100644 index 1df9ef422f61..000000000000 --- a/sound/soc/intel/skylake/skl-sst-dsp.h +++ /dev/null @@ -1,256 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Skylake SST DSP Support - * - * Copyright (C) 2014-15, Intel Corporation. - */ - -#ifndef __SKL_SST_DSP_H__ -#define __SKL_SST_DSP_H__ - -#include -#include -#include -#include -#include "skl-sst-cldma.h" - -struct sst_dsp; -struct sst_dsp_device; -struct skl_lib_info; -struct skl_dev; - -/* Intel HD Audio General DSP Registers */ -#define SKL_ADSP_GEN_BASE 0x0 -#define SKL_ADSP_REG_ADSPCS (SKL_ADSP_GEN_BASE + 0x04) -#define SKL_ADSP_REG_ADSPIC (SKL_ADSP_GEN_BASE + 0x08) -#define SKL_ADSP_REG_ADSPIS (SKL_ADSP_GEN_BASE + 0x0C) -#define SKL_ADSP_REG_ADSPIC2 (SKL_ADSP_GEN_BASE + 0x10) -#define SKL_ADSP_REG_ADSPIS2 (SKL_ADSP_GEN_BASE + 0x14) - -/* Intel HD Audio Inter-Processor Communication Registers */ -#define SKL_ADSP_IPC_BASE 0x40 -#define SKL_ADSP_REG_HIPCT (SKL_ADSP_IPC_BASE + 0x00) -#define SKL_ADSP_REG_HIPCTE (SKL_ADSP_IPC_BASE + 0x04) -#define SKL_ADSP_REG_HIPCI (SKL_ADSP_IPC_BASE + 0x08) -#define SKL_ADSP_REG_HIPCIE (SKL_ADSP_IPC_BASE + 0x0C) -#define SKL_ADSP_REG_HIPCCTL (SKL_ADSP_IPC_BASE + 0x10) - -/* HIPCI */ -#define SKL_ADSP_REG_HIPCI_BUSY BIT(31) - -/* HIPCIE */ -#define SKL_ADSP_REG_HIPCIE_DONE BIT(30) - -/* HIPCCTL */ -#define SKL_ADSP_REG_HIPCCTL_DONE BIT(1) -#define SKL_ADSP_REG_HIPCCTL_BUSY BIT(0) - -/* HIPCT */ -#define SKL_ADSP_REG_HIPCT_BUSY BIT(31) - -/* FW base IDs */ -#define SKL_INSTANCE_ID 0 -#define SKL_BASE_FW_MODULE_ID 0 - -/* Intel HD Audio SRAM Window 1 */ -#define SKL_ADSP_SRAM1_BASE 0xA000 - -#define SKL_ADSP_MMIO_LEN 0x10000 - -#define SKL_ADSP_W0_STAT_SZ 0x1000 - -#define SKL_ADSP_W0_UP_SZ 0x1000 - -#define SKL_ADSP_W1_SZ 0x1000 - -#define SKL_FW_STS_MASK 0xf - -#define SKL_FW_INIT 0x1 -#define SKL_FW_RFW_START 0xf -#define BXT_FW_ROM_INIT_RETRY 3 -#define BXT_INIT_TIMEOUT 300 - -#define SKL_ADSPIC_IPC 1 -#define SKL_ADSPIS_IPC 1 - -/* Core ID of core0 */ -#define SKL_DSP_CORE0_ID 0 - -/* Mask for a given core index, c = 0.. number of supported cores - 1 */ -#define SKL_DSP_CORE_MASK(c) BIT(c) - -/* - * Core 0 mask = SKL_DSP_CORE_MASK(0); Defined separately - * since Core0 is primary core and it is used often - */ -#define SKL_DSP_CORE0_MASK BIT(0) - -/* - * Mask for a given number of cores - * nc = number of supported cores - */ -#define SKL_DSP_CORES_MASK(nc) GENMASK((nc - 1), 0) - -/* ADSPCS - Audio DSP Control & Status */ - -/* - * Core Reset - asserted high - * CRST Mask for a given core mask pattern, cm - */ -#define SKL_ADSPCS_CRST_SHIFT 0 -#define SKL_ADSPCS_CRST_MASK(cm) ((cm) << SKL_ADSPCS_CRST_SHIFT) - -/* - * Core run/stall - when set to '1' core is stalled - * CSTALL Mask for a given core mask pattern, cm - */ -#define SKL_ADSPCS_CSTALL_SHIFT 8 -#define SKL_ADSPCS_CSTALL_MASK(cm) ((cm) << SKL_ADSPCS_CSTALL_SHIFT) - -/* - * Set Power Active - when set to '1' turn cores on - * SPA Mask for a given core mask pattern, cm - */ -#define SKL_ADSPCS_SPA_SHIFT 16 -#define SKL_ADSPCS_SPA_MASK(cm) ((cm) << SKL_ADSPCS_SPA_SHIFT) - -/* - * Current Power Active - power status of cores, set by hardware - * CPA Mask for a given core mask pattern, cm - */ -#define SKL_ADSPCS_CPA_SHIFT 24 -#define SKL_ADSPCS_CPA_MASK(cm) ((cm) << SKL_ADSPCS_CPA_SHIFT) - -/* DSP Core state */ -enum skl_dsp_states { - SKL_DSP_RUNNING = 1, - /* Running in D0i3 state; can be in streaming or non-streaming D0i3 */ - SKL_DSP_RUNNING_D0I3, /* Running in D0i3 state*/ - SKL_DSP_RESET, -}; - -/* D0i3 substates */ -enum skl_dsp_d0i3_states { - SKL_DSP_D0I3_NONE = -1, /* No D0i3 */ - SKL_DSP_D0I3_NON_STREAMING = 0, - SKL_DSP_D0I3_STREAMING = 1, -}; - -struct skl_dsp_fw_ops { - int (*load_fw)(struct sst_dsp *ctx); - /* FW module parser/loader */ - int (*load_library)(struct sst_dsp *ctx, - struct skl_lib_info *linfo, int lib_count); - int (*parse_fw)(struct sst_dsp *ctx); - int (*set_state_D0)(struct sst_dsp *ctx, unsigned int core_id); - int (*set_state_D3)(struct sst_dsp *ctx, unsigned int core_id); - int (*set_state_D0i3)(struct sst_dsp *ctx); - int (*set_state_D0i0)(struct sst_dsp *ctx); - unsigned int (*get_fw_errcode)(struct sst_dsp *ctx); - int (*load_mod)(struct sst_dsp *ctx, u16 mod_id, u8 *mod_name); - int (*unload_mod)(struct sst_dsp *ctx, u16 mod_id); - -}; - -struct skl_dsp_loader_ops { - int stream_tag; - - int (*alloc_dma_buf)(struct device *dev, - struct snd_dma_buffer *dmab, size_t size); - int (*free_dma_buf)(struct device *dev, - struct snd_dma_buffer *dmab); - int (*prepare)(struct device *dev, unsigned int format, - unsigned int byte_size, - struct snd_dma_buffer *bufp); - int (*trigger)(struct device *dev, bool start, int stream_tag); - - int (*cleanup)(struct device *dev, struct snd_dma_buffer *dmab, - int stream_tag); -}; - -#define MAX_INSTANCE_BUFF 2 - -struct uuid_module { - guid_t uuid; - int id; - int is_loadable; - int max_instance; - u64 pvt_id[MAX_INSTANCE_BUFF]; - int *instance_id; - - struct list_head list; -}; - -struct skl_load_module_info { - u16 mod_id; - const struct firmware *fw; -}; - -struct skl_module_table { - struct skl_load_module_info *mod_info; - unsigned int usage_cnt; - struct list_head list; -}; - -void skl_cldma_process_intr(struct sst_dsp *ctx); -void skl_cldma_int_disable(struct sst_dsp *ctx); -int skl_cldma_prepare(struct sst_dsp *ctx); -int skl_cldma_wait_interruptible(struct sst_dsp *ctx); - -void skl_dsp_set_state_locked(struct sst_dsp *ctx, int state); -struct sst_dsp *skl_dsp_ctx_init(struct device *dev, - struct sst_dsp_device *sst_dev, int irq); -int skl_dsp_acquire_irq(struct sst_dsp *sst); -bool is_skl_dsp_running(struct sst_dsp *ctx); - -unsigned int skl_dsp_get_enabled_cores(struct sst_dsp *ctx); -void skl_dsp_init_core_state(struct sst_dsp *ctx); -int skl_dsp_enable_core(struct sst_dsp *ctx, unsigned int core_mask); -int skl_dsp_disable_core(struct sst_dsp *ctx, unsigned int core_mask); -int skl_dsp_core_power_up(struct sst_dsp *ctx, unsigned int core_mask); -int skl_dsp_core_power_down(struct sst_dsp *ctx, unsigned int core_mask); -int skl_dsp_core_unset_reset_state(struct sst_dsp *ctx, - unsigned int core_mask); -int skl_dsp_start_core(struct sst_dsp *ctx, unsigned int core_mask); - -irqreturn_t skl_dsp_sst_interrupt(int irq, void *dev_id); -int skl_dsp_wake(struct sst_dsp *ctx); -int skl_dsp_sleep(struct sst_dsp *ctx); -void skl_dsp_free(struct sst_dsp *dsp); - -int skl_dsp_get_core(struct sst_dsp *ctx, unsigned int core_id); -int skl_dsp_put_core(struct sst_dsp *ctx, unsigned int core_id); - -int skl_dsp_boot(struct sst_dsp *ctx); -int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp); -int bxt_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp); -int skl_sst_init_fw(struct device *dev, struct skl_dev *skl); -int bxt_sst_init_fw(struct device *dev, struct skl_dev *skl); -void skl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl); -void bxt_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl); - -int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, - unsigned int offset, int index); -int skl_get_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int instance_id); -int skl_put_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int *pvt_id); -int skl_get_pvt_instance_id_map(struct skl_dev *skl, - int module_id, int instance_id); -void skl_freeup_uuid_list(struct skl_dev *skl); - -int skl_dsp_strip_extended_manifest(struct firmware *fw); - -void skl_dsp_set_astate_cfg(struct skl_dev *skl, u32 cnt, void *data); - -int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name, - struct skl_dsp_loader_ops dsp_ops, struct skl_dev **dsp, - struct sst_dsp_device *skl_dev); -int skl_prepare_lib_load(struct skl_dev *skl, struct skl_lib_info *linfo, - struct firmware *stripped_fw, - unsigned int hdr_offset, int index); -void skl_release_library(struct skl_lib_info *linfo, int lib_count); - -#endif /*__SKL_SST_DSP_H__*/ diff --git a/sound/soc/intel/skylake/skl-sst-ipc.c b/sound/soc/intel/skylake/skl-sst-ipc.c deleted file mode 100644 index fd9624ad5f72..000000000000 --- a/sound/soc/intel/skylake/skl-sst-ipc.c +++ /dev/null @@ -1,1071 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-sst-ipc.c - Intel skl IPC Support - * - * Copyright (C) 2014-15, Intel Corporation. - */ -#include - -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "skl.h" -#include "skl-sst-dsp.h" -#include "skl-sst-ipc.h" -#include "sound/hdaudio_ext.h" - - -#define IPC_IXC_STATUS_BITS 24 - -/* Global Message - Generic */ -#define IPC_GLB_TYPE_SHIFT 24 -#define IPC_GLB_TYPE_MASK (0xf << IPC_GLB_TYPE_SHIFT) -#define IPC_GLB_TYPE(x) ((x) << IPC_GLB_TYPE_SHIFT) - -/* Global Message - Reply */ -#define IPC_GLB_REPLY_STATUS_SHIFT 24 -#define IPC_GLB_REPLY_STATUS_MASK ((0x1 << IPC_GLB_REPLY_STATUS_SHIFT) - 1) -#define IPC_GLB_REPLY_STATUS(x) ((x) << IPC_GLB_REPLY_STATUS_SHIFT) - -#define IPC_GLB_REPLY_TYPE_SHIFT 29 -#define IPC_GLB_REPLY_TYPE_MASK 0x1F -#define IPC_GLB_REPLY_TYPE(x) (((x) >> IPC_GLB_REPLY_TYPE_SHIFT) \ - & IPC_GLB_RPLY_TYPE_MASK) - -#define IPC_TIMEOUT_MSECS 3000 - -#define IPC_EMPTY_LIST_SIZE 8 - -#define IPC_MSG_TARGET_SHIFT 30 -#define IPC_MSG_TARGET_MASK 0x1 -#define IPC_MSG_TARGET(x) (((x) & IPC_MSG_TARGET_MASK) \ - << IPC_MSG_TARGET_SHIFT) - -#define IPC_MSG_DIR_SHIFT 29 -#define IPC_MSG_DIR_MASK 0x1 -#define IPC_MSG_DIR(x) (((x) & IPC_MSG_DIR_MASK) \ - << IPC_MSG_DIR_SHIFT) -/* Global Notification Message */ -#define IPC_GLB_NOTIFY_TYPE_SHIFT 16 -#define IPC_GLB_NOTIFY_TYPE_MASK 0xFF -#define IPC_GLB_NOTIFY_TYPE(x) (((x) >> IPC_GLB_NOTIFY_TYPE_SHIFT) \ - & IPC_GLB_NOTIFY_TYPE_MASK) - -#define IPC_GLB_NOTIFY_MSG_TYPE_SHIFT 24 -#define IPC_GLB_NOTIFY_MSG_TYPE_MASK 0x1F -#define IPC_GLB_NOTIFY_MSG_TYPE(x) (((x) >> IPC_GLB_NOTIFY_MSG_TYPE_SHIFT) \ - & IPC_GLB_NOTIFY_MSG_TYPE_MASK) - -#define IPC_GLB_NOTIFY_RSP_SHIFT 29 -#define IPC_GLB_NOTIFY_RSP_MASK 0x1 -#define IPC_GLB_NOTIFY_RSP_TYPE(x) (((x) >> IPC_GLB_NOTIFY_RSP_SHIFT) \ - & IPC_GLB_NOTIFY_RSP_MASK) - -/* Pipeline operations */ - -/* Create pipeline message */ -#define IPC_PPL_MEM_SIZE_SHIFT 0 -#define IPC_PPL_MEM_SIZE_MASK 0x7FF -#define IPC_PPL_MEM_SIZE(x) (((x) & IPC_PPL_MEM_SIZE_MASK) \ - << IPC_PPL_MEM_SIZE_SHIFT) - -#define IPC_PPL_TYPE_SHIFT 11 -#define IPC_PPL_TYPE_MASK 0x1F -#define IPC_PPL_TYPE(x) (((x) & IPC_PPL_TYPE_MASK) \ - << IPC_PPL_TYPE_SHIFT) - -#define IPC_INSTANCE_ID_SHIFT 16 -#define IPC_INSTANCE_ID_MASK 0xFF -#define IPC_INSTANCE_ID(x) (((x) & IPC_INSTANCE_ID_MASK) \ - << IPC_INSTANCE_ID_SHIFT) - -#define IPC_PPL_LP_MODE_SHIFT 0 -#define IPC_PPL_LP_MODE_MASK 0x1 -#define IPC_PPL_LP_MODE(x) (((x) & IPC_PPL_LP_MODE_MASK) \ - << IPC_PPL_LP_MODE_SHIFT) - -/* Set pipeline state message */ -#define IPC_PPL_STATE_SHIFT 0 -#define IPC_PPL_STATE_MASK 0x1F -#define IPC_PPL_STATE(x) (((x) & IPC_PPL_STATE_MASK) \ - << IPC_PPL_STATE_SHIFT) - -/* Module operations primary register */ -#define IPC_MOD_ID_SHIFT 0 -#define IPC_MOD_ID_MASK 0xFFFF -#define IPC_MOD_ID(x) (((x) & IPC_MOD_ID_MASK) \ - << IPC_MOD_ID_SHIFT) - -#define IPC_MOD_INSTANCE_ID_SHIFT 16 -#define IPC_MOD_INSTANCE_ID_MASK 0xFF -#define IPC_MOD_INSTANCE_ID(x) (((x) & IPC_MOD_INSTANCE_ID_MASK) \ - << IPC_MOD_INSTANCE_ID_SHIFT) - -/* Init instance message extension register */ -#define IPC_PARAM_BLOCK_SIZE_SHIFT 0 -#define IPC_PARAM_BLOCK_SIZE_MASK 0xFFFF -#define IPC_PARAM_BLOCK_SIZE(x) (((x) & IPC_PARAM_BLOCK_SIZE_MASK) \ - << IPC_PARAM_BLOCK_SIZE_SHIFT) - -#define IPC_PPL_INSTANCE_ID_SHIFT 16 -#define IPC_PPL_INSTANCE_ID_MASK 0xFF -#define IPC_PPL_INSTANCE_ID(x) (((x) & IPC_PPL_INSTANCE_ID_MASK) \ - << IPC_PPL_INSTANCE_ID_SHIFT) - -#define IPC_CORE_ID_SHIFT 24 -#define IPC_CORE_ID_MASK 0x1F -#define IPC_CORE_ID(x) (((x) & IPC_CORE_ID_MASK) \ - << IPC_CORE_ID_SHIFT) - -#define IPC_DOMAIN_SHIFT 28 -#define IPC_DOMAIN_MASK 0x1 -#define IPC_DOMAIN(x) (((x) & IPC_DOMAIN_MASK) \ - << IPC_DOMAIN_SHIFT) - -/* Bind/Unbind message extension register */ -#define IPC_DST_MOD_ID_SHIFT 0 -#define IPC_DST_MOD_ID(x) (((x) & IPC_MOD_ID_MASK) \ - << IPC_DST_MOD_ID_SHIFT) - -#define IPC_DST_MOD_INSTANCE_ID_SHIFT 16 -#define IPC_DST_MOD_INSTANCE_ID(x) (((x) & IPC_MOD_INSTANCE_ID_MASK) \ - << IPC_DST_MOD_INSTANCE_ID_SHIFT) - -#define IPC_DST_QUEUE_SHIFT 24 -#define IPC_DST_QUEUE_MASK 0x7 -#define IPC_DST_QUEUE(x) (((x) & IPC_DST_QUEUE_MASK) \ - << IPC_DST_QUEUE_SHIFT) - -#define IPC_SRC_QUEUE_SHIFT 27 -#define IPC_SRC_QUEUE_MASK 0x7 -#define IPC_SRC_QUEUE(x) (((x) & IPC_SRC_QUEUE_MASK) \ - << IPC_SRC_QUEUE_SHIFT) -/* Load Module count */ -#define IPC_LOAD_MODULE_SHIFT 0 -#define IPC_LOAD_MODULE_MASK 0xFF -#define IPC_LOAD_MODULE_CNT(x) (((x) & IPC_LOAD_MODULE_MASK) \ - << IPC_LOAD_MODULE_SHIFT) - -/* Save pipeline messgae extension register */ -#define IPC_DMA_ID_SHIFT 0 -#define IPC_DMA_ID_MASK 0x1F -#define IPC_DMA_ID(x) (((x) & IPC_DMA_ID_MASK) \ - << IPC_DMA_ID_SHIFT) -/* Large Config message extension register */ -#define IPC_DATA_OFFSET_SZ_SHIFT 0 -#define IPC_DATA_OFFSET_SZ_MASK 0xFFFFF -#define IPC_DATA_OFFSET_SZ(x) (((x) & IPC_DATA_OFFSET_SZ_MASK) \ - << IPC_DATA_OFFSET_SZ_SHIFT) -#define IPC_DATA_OFFSET_SZ_CLEAR ~(IPC_DATA_OFFSET_SZ_MASK \ - << IPC_DATA_OFFSET_SZ_SHIFT) - -#define IPC_LARGE_PARAM_ID_SHIFT 20 -#define IPC_LARGE_PARAM_ID_MASK 0xFF -#define IPC_LARGE_PARAM_ID(x) (((x) & IPC_LARGE_PARAM_ID_MASK) \ - << IPC_LARGE_PARAM_ID_SHIFT) - -#define IPC_FINAL_BLOCK_SHIFT 28 -#define IPC_FINAL_BLOCK_MASK 0x1 -#define IPC_FINAL_BLOCK(x) (((x) & IPC_FINAL_BLOCK_MASK) \ - << IPC_FINAL_BLOCK_SHIFT) - -#define IPC_INITIAL_BLOCK_SHIFT 29 -#define IPC_INITIAL_BLOCK_MASK 0x1 -#define IPC_INITIAL_BLOCK(x) (((x) & IPC_INITIAL_BLOCK_MASK) \ - << IPC_INITIAL_BLOCK_SHIFT) -#define IPC_INITIAL_BLOCK_CLEAR ~(IPC_INITIAL_BLOCK_MASK \ - << IPC_INITIAL_BLOCK_SHIFT) -/* Set D0ix IPC extension register */ -#define IPC_D0IX_WAKE_SHIFT 0 -#define IPC_D0IX_WAKE_MASK 0x1 -#define IPC_D0IX_WAKE(x) (((x) & IPC_D0IX_WAKE_MASK) \ - << IPC_D0IX_WAKE_SHIFT) - -#define IPC_D0IX_STREAMING_SHIFT 1 -#define IPC_D0IX_STREAMING_MASK 0x1 -#define IPC_D0IX_STREAMING(x) (((x) & IPC_D0IX_STREAMING_MASK) \ - << IPC_D0IX_STREAMING_SHIFT) - - -enum skl_ipc_msg_target { - IPC_FW_GEN_MSG = 0, - IPC_MOD_MSG = 1 -}; - -enum skl_ipc_msg_direction { - IPC_MSG_REQUEST = 0, - IPC_MSG_REPLY = 1 -}; - -/* Global Message Types */ -enum skl_ipc_glb_type { - IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ - IPC_GLB_LOAD_MULTIPLE_MODS = 15, - IPC_GLB_UNLOAD_MULTIPLE_MODS = 16, - IPC_GLB_CREATE_PPL = 17, - IPC_GLB_DELETE_PPL = 18, - IPC_GLB_SET_PPL_STATE = 19, - IPC_GLB_GET_PPL_STATE = 20, - IPC_GLB_GET_PPL_CONTEXT_SIZE = 21, - IPC_GLB_SAVE_PPL = 22, - IPC_GLB_RESTORE_PPL = 23, - IPC_GLB_LOAD_LIBRARY = 24, - IPC_GLB_NOTIFY = 26, - IPC_GLB_MAX_IPC_MSG_NUMBER = 31 /* Maximum message number */ -}; - -enum skl_ipc_glb_reply { - IPC_GLB_REPLY_SUCCESS = 0, - - IPC_GLB_REPLY_UNKNOWN_MSG_TYPE = 1, - IPC_GLB_REPLY_ERROR_INVALID_PARAM = 2, - - IPC_GLB_REPLY_BUSY = 3, - IPC_GLB_REPLY_PENDING = 4, - IPC_GLB_REPLY_FAILURE = 5, - IPC_GLB_REPLY_INVALID_REQUEST = 6, - - IPC_GLB_REPLY_OUT_OF_MEMORY = 7, - IPC_GLB_REPLY_OUT_OF_MIPS = 8, - - IPC_GLB_REPLY_INVALID_RESOURCE_ID = 9, - IPC_GLB_REPLY_INVALID_RESOURCE_STATE = 10, - - IPC_GLB_REPLY_MOD_MGMT_ERROR = 100, - IPC_GLB_REPLY_MOD_LOAD_CL_FAILED = 101, - IPC_GLB_REPLY_MOD_LOAD_INVALID_HASH = 102, - - IPC_GLB_REPLY_MOD_UNLOAD_INST_EXIST = 103, - IPC_GLB_REPLY_MOD_NOT_INITIALIZED = 104, - - IPC_GLB_REPLY_INVALID_CONFIG_PARAM_ID = 120, - IPC_GLB_REPLY_INVALID_CONFIG_DATA_LEN = 121, - IPC_GLB_REPLY_GATEWAY_NOT_INITIALIZED = 140, - IPC_GLB_REPLY_GATEWAY_NOT_EXIST = 141, - IPC_GLB_REPLY_SCLK_ALREADY_RUNNING = 150, - IPC_GLB_REPLY_MCLK_ALREADY_RUNNING = 151, - - IPC_GLB_REPLY_PPL_NOT_INITIALIZED = 160, - IPC_GLB_REPLY_PPL_NOT_EXIST = 161, - IPC_GLB_REPLY_PPL_SAVE_FAILED = 162, - IPC_GLB_REPLY_PPL_RESTORE_FAILED = 163, - - IPC_MAX_STATUS = ((1<tx.data, tx_data, tx_size); -} - -static bool skl_ipc_is_dsp_busy(struct sst_dsp *dsp) -{ - u32 hipci; - - hipci = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCI); - return (hipci & SKL_ADSP_REG_HIPCI_BUSY); -} - -/* Lock to be held by caller */ -static void skl_ipc_tx_msg(struct sst_generic_ipc *ipc, struct ipc_message *msg) -{ - struct skl_ipc_header *header = (struct skl_ipc_header *)(&msg->tx.header); - - if (msg->tx.size) - sst_dsp_outbox_write(ipc->dsp, msg->tx.data, msg->tx.size); - sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCIE, - header->extension); - sst_dsp_shim_write_unlocked(ipc->dsp, SKL_ADSP_REG_HIPCI, - header->primary | SKL_ADSP_REG_HIPCI_BUSY); -} - -int skl_ipc_check_D0i0(struct sst_dsp *dsp, bool state) -{ - int ret; - - /* check D0i3 support */ - if (!dsp->fw_ops.set_state_D0i0) - return 0; - - /* Attempt D0i0 or D0i3 based on state */ - if (state) - ret = dsp->fw_ops.set_state_D0i0(dsp); - else - ret = dsp->fw_ops.set_state_D0i3(dsp); - - return ret; -} - -static struct ipc_message *skl_ipc_reply_get_msg(struct sst_generic_ipc *ipc, - u64 ipc_header) -{ - struct ipc_message *msg = NULL; - struct skl_ipc_header *header = (struct skl_ipc_header *)(&ipc_header); - - if (list_empty(&ipc->rx_list)) { - dev_err(ipc->dev, "ipc: rx list is empty but received 0x%x\n", - header->primary); - goto out; - } - - msg = list_first_entry(&ipc->rx_list, struct ipc_message, list); - - list_del(&msg->list); -out: - return msg; - -} - -int skl_ipc_process_notification(struct sst_generic_ipc *ipc, - struct skl_ipc_header header) -{ - struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc); - - if (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { - switch (IPC_GLB_NOTIFY_TYPE(header.primary)) { - - case IPC_GLB_NOTIFY_UNDERRUN: - dev_err(ipc->dev, "FW Underrun %x\n", header.primary); - break; - - case IPC_GLB_NOTIFY_RESOURCE_EVENT: - dev_err(ipc->dev, "MCPS Budget Violation: %x\n", - header.primary); - break; - - case IPC_GLB_NOTIFY_FW_READY: - skl->boot_complete = true; - wake_up(&skl->boot_wait); - break; - - case IPC_GLB_NOTIFY_PHRASE_DETECTED: - dev_dbg(ipc->dev, "***** Phrase Detected **********\n"); - - /* - * Per HW recomendation, After phrase detection, - * clear the CGCTL.MISCBDCGE. - * - * This will be set back on stream closure - */ - skl->enable_miscbdcge(ipc->dev, false); - skl->miscbdcg_disabled = true; - break; - - default: - dev_err(ipc->dev, "ipc: Unhandled error msg=%x\n", - header.primary); - break; - } - } - - return 0; -} - -struct skl_ipc_err_map { - const char *msg; - enum skl_ipc_glb_reply reply; - int err; -}; - -static struct skl_ipc_err_map skl_err_map[] = { - {"DSP out of memory", IPC_GLB_REPLY_OUT_OF_MEMORY, -ENOMEM}, - {"DSP busy", IPC_GLB_REPLY_BUSY, -EBUSY}, - {"SCLK already running", IPC_GLB_REPLY_SCLK_ALREADY_RUNNING, - IPC_GLB_REPLY_SCLK_ALREADY_RUNNING}, - {"MCLK already running", IPC_GLB_REPLY_MCLK_ALREADY_RUNNING, - IPC_GLB_REPLY_MCLK_ALREADY_RUNNING}, -}; - -static int skl_ipc_set_reply_error_code(struct sst_generic_ipc *ipc, u32 reply) -{ - int i; - - for (i = 0; i < ARRAY_SIZE(skl_err_map); i++) { - if (skl_err_map[i].reply == reply) - break; - } - - if (i == ARRAY_SIZE(skl_err_map)) { - dev_err(ipc->dev, "ipc FW reply: %d FW Error Code: %u\n", - reply, - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); - return -EINVAL; - } - - if (skl_err_map[i].err < 0) - dev_err(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", - skl_err_map[i].msg, - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); - else - dev_info(ipc->dev, "ipc FW reply: %s FW Error Code: %u\n", - skl_err_map[i].msg, - ipc->dsp->fw_ops.get_fw_errcode(ipc->dsp)); - - return skl_err_map[i].err; -} - -void skl_ipc_process_reply(struct sst_generic_ipc *ipc, - struct skl_ipc_header header) -{ - struct ipc_message *msg; - u32 reply = header.primary & IPC_GLB_REPLY_STATUS_MASK; - u64 *ipc_header = (u64 *)(&header); - struct skl_dev *skl = container_of(ipc, struct skl_dev, ipc); - unsigned long flags; - - spin_lock_irqsave(&ipc->dsp->spinlock, flags); - msg = skl_ipc_reply_get_msg(ipc, *ipc_header); - spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); - if (msg == NULL) { - dev_dbg(ipc->dev, "ipc: rx list is empty\n"); - return; - } - - msg->rx.header = *ipc_header; - /* first process the header */ - if (reply == IPC_GLB_REPLY_SUCCESS) { - dev_dbg(ipc->dev, "ipc FW reply %x: success\n", header.primary); - /* copy the rx data from the mailbox */ - sst_dsp_inbox_read(ipc->dsp, msg->rx.data, msg->rx.size); - switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { - case IPC_GLB_LOAD_MULTIPLE_MODS: - case IPC_GLB_LOAD_LIBRARY: - skl->mod_load_complete = true; - skl->mod_load_status = true; - wake_up(&skl->mod_load_wait); - break; - - default: - break; - - } - } else { - msg->errno = skl_ipc_set_reply_error_code(ipc, reply); - switch (IPC_GLB_NOTIFY_MSG_TYPE(header.primary)) { - case IPC_GLB_LOAD_MULTIPLE_MODS: - case IPC_GLB_LOAD_LIBRARY: - skl->mod_load_complete = true; - skl->mod_load_status = false; - wake_up(&skl->mod_load_wait); - break; - - default: - break; - - } - } - - spin_lock_irqsave(&ipc->dsp->spinlock, flags); - sst_ipc_tx_msg_reply_complete(ipc, msg); - spin_unlock_irqrestore(&ipc->dsp->spinlock, flags); -} - -irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context) -{ - struct sst_dsp *dsp = context; - struct skl_dev *skl = dsp->thread_context; - struct sst_generic_ipc *ipc = &skl->ipc; - struct skl_ipc_header header = {0}; - u32 hipcie, hipct, hipcte; - int ipc_irq = 0; - - if (dsp->intr_status & SKL_ADSPIS_CL_DMA) - skl_cldma_process_intr(dsp); - - /* Here we handle IPC interrupts only */ - if (!(dsp->intr_status & SKL_ADSPIS_IPC)) - return IRQ_NONE; - - hipcie = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCIE); - hipct = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCT); - hipcte = sst_dsp_shim_read_unlocked(dsp, SKL_ADSP_REG_HIPCTE); - - /* reply message from DSP */ - if (hipcie & SKL_ADSP_REG_HIPCIE_DONE) { - sst_dsp_shim_update_bits(dsp, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_DONE, 0); - - /* clear DONE bit - tell DSP we have completed the operation */ - sst_dsp_shim_update_bits_forced(dsp, SKL_ADSP_REG_HIPCIE, - SKL_ADSP_REG_HIPCIE_DONE, SKL_ADSP_REG_HIPCIE_DONE); - - ipc_irq = 1; - - /* unmask Done interrupt */ - sst_dsp_shim_update_bits(dsp, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_DONE, SKL_ADSP_REG_HIPCCTL_DONE); - } - - /* New message from DSP */ - if (hipct & SKL_ADSP_REG_HIPCT_BUSY) { - header.primary = hipct; - header.extension = hipcte; - dev_dbg(dsp->dev, "IPC irq: Firmware respond primary:%x\n", - header.primary); - dev_dbg(dsp->dev, "IPC irq: Firmware respond extension:%x\n", - header.extension); - - if (IPC_GLB_NOTIFY_RSP_TYPE(header.primary)) { - /* Handle Immediate reply from DSP Core */ - skl_ipc_process_reply(ipc, header); - } else { - dev_dbg(dsp->dev, "IPC irq: Notification from firmware\n"); - skl_ipc_process_notification(ipc, header); - } - /* clear busy interrupt */ - sst_dsp_shim_update_bits_forced(dsp, SKL_ADSP_REG_HIPCT, - SKL_ADSP_REG_HIPCT_BUSY, SKL_ADSP_REG_HIPCT_BUSY); - ipc_irq = 1; - } - - if (ipc_irq == 0) - return IRQ_NONE; - - skl_ipc_int_enable(dsp); - - /* continue to send any remaining messages... */ - schedule_work(&ipc->kwork); - - return IRQ_HANDLED; -} - -void skl_ipc_int_enable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_ADSPIC, - SKL_ADSPIC_IPC, SKL_ADSPIC_IPC); -} - -void skl_ipc_int_disable(struct sst_dsp *ctx) -{ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_ADSPIC, - SKL_ADSPIC_IPC, 0); -} - -void skl_ipc_op_int_enable(struct sst_dsp *ctx) -{ - /* enable IPC DONE interrupt */ - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_DONE, SKL_ADSP_REG_HIPCCTL_DONE); - - /* Enable IPC BUSY interrupt */ - sst_dsp_shim_update_bits(ctx, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_BUSY, SKL_ADSP_REG_HIPCCTL_BUSY); -} - -void skl_ipc_op_int_disable(struct sst_dsp *ctx) -{ - /* disable IPC DONE interrupt */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_DONE, 0); - - /* Disable IPC BUSY interrupt */ - sst_dsp_shim_update_bits_unlocked(ctx, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_BUSY, 0); - -} - -bool skl_ipc_int_status(struct sst_dsp *ctx) -{ - return sst_dsp_shim_read_unlocked(ctx, - SKL_ADSP_REG_ADSPIS) & SKL_ADSPIS_IPC; -} - -int skl_ipc_init(struct device *dev, struct skl_dev *skl) -{ - struct sst_generic_ipc *ipc; - int err; - - ipc = &skl->ipc; - ipc->dsp = skl->dsp; - ipc->dev = dev; - - ipc->tx_data_max_size = SKL_ADSP_W1_SZ; - ipc->rx_data_max_size = SKL_ADSP_W0_UP_SZ; - - err = sst_ipc_init(ipc); - if (err) - return err; - - ipc->ops.tx_msg = skl_ipc_tx_msg; - ipc->ops.tx_data_copy = skl_ipc_tx_data_copy; - ipc->ops.is_dsp_busy = skl_ipc_is_dsp_busy; - - return 0; -} - -void skl_ipc_free(struct sst_generic_ipc *ipc) -{ - /* Disable IPC DONE interrupt */ - sst_dsp_shim_update_bits(ipc->dsp, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_DONE, 0); - - /* Disable IPC BUSY interrupt */ - sst_dsp_shim_update_bits(ipc->dsp, SKL_ADSP_REG_HIPCCTL, - SKL_ADSP_REG_HIPCCTL_BUSY, 0); - - sst_ipc_fini(ipc); -} - -int skl_ipc_create_pipeline(struct sst_generic_ipc *ipc, - u16 ppl_mem_size, u8 ppl_type, u8 instance_id, u8 lp_mode) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_CREATE_PPL); - header.primary |= IPC_INSTANCE_ID(instance_id); - header.primary |= IPC_PPL_TYPE(ppl_type); - header.primary |= IPC_PPL_MEM_SIZE(ppl_mem_size); - - header.extension = IPC_PPL_LP_MODE(lp_mode); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: create pipeline fail, err: %d\n", ret); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_create_pipeline); - -int skl_ipc_delete_pipeline(struct sst_generic_ipc *ipc, u8 instance_id) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_DELETE_PPL); - header.primary |= IPC_INSTANCE_ID(instance_id); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: delete pipeline failed, err %d\n", ret); - return ret; - } - - return 0; -} -EXPORT_SYMBOL_GPL(skl_ipc_delete_pipeline); - -int skl_ipc_set_pipeline_state(struct sst_generic_ipc *ipc, - u8 instance_id, enum skl_ipc_pipeline_state state) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_SET_PPL_STATE); - header.primary |= IPC_INSTANCE_ID(instance_id); - header.primary |= IPC_PPL_STATE(state); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: set pipeline state failed, err: %d\n", ret); - return ret; - } - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_set_pipeline_state); - -int -skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, u8 instance_id, int dma_id) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_SAVE_PPL); - header.primary |= IPC_INSTANCE_ID(instance_id); - - header.extension = IPC_DMA_ID(dma_id); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: save pipeline failed, err: %d\n", ret); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_save_pipeline); - -int skl_ipc_restore_pipeline(struct sst_generic_ipc *ipc, u8 instance_id) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_RESTORE_PPL); - header.primary |= IPC_INSTANCE_ID(instance_id); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s header=%d\n", __func__, header.primary); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: restore pipeline failed, err: %d\n", ret); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_restore_pipeline); - -int skl_ipc_set_dx(struct sst_generic_ipc *ipc, u8 instance_id, - u16 module_id, struct skl_ipc_dxstate_info *dx) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_MOD_SET_DX); - header.primary |= IPC_MOD_INSTANCE_ID(instance_id); - header.primary |= IPC_MOD_ID(module_id); - - request.header = *(u64 *)(&header); - request.data = dx; - request.size = sizeof(*dx); - - dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, - header.primary, header.extension); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: set dx failed, err %d\n", ret); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_set_dx); - -int skl_ipc_init_instance(struct sst_generic_ipc *ipc, - struct skl_ipc_init_instance_msg *msg, void *param_data) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request; - int ret; - u32 *buffer = (u32 *)param_data; - /* param_block_size must be in dwords */ - u16 param_block_size = msg->param_data_size / sizeof(u32); - - print_hex_dump_debug("Param data:", DUMP_PREFIX_NONE, - 16, 4, buffer, param_block_size, false); - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_MOD_INIT_INSTANCE); - header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); - header.primary |= IPC_MOD_ID(msg->module_id); - - header.extension = IPC_CORE_ID(msg->core_id); - header.extension |= IPC_PPL_INSTANCE_ID(msg->ppl_instance_id); - header.extension |= IPC_PARAM_BLOCK_SIZE(param_block_size); - header.extension |= IPC_DOMAIN(msg->domain); - - request.header = *(u64 *)(&header); - request.data = param_data; - request.size = msg->param_data_size; - - dev_dbg(ipc->dev, "In %s primary =%x ext=%x\n", __func__, - header.primary, header.extension); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - - if (ret < 0) { - dev_err(ipc->dev, "ipc: init instance failed\n"); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_init_instance); - -int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc, - struct skl_ipc_bind_unbind_msg *msg) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - u8 bind_unbind = msg->bind ? IPC_MOD_BIND : IPC_MOD_UNBIND; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(bind_unbind); - header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); - header.primary |= IPC_MOD_ID(msg->module_id); - - header.extension = IPC_DST_MOD_ID(msg->dst_module_id); - header.extension |= IPC_DST_MOD_INSTANCE_ID(msg->dst_instance_id); - header.extension |= IPC_DST_QUEUE(msg->dst_queue); - header.extension |= IPC_SRC_QUEUE(msg->src_queue); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s hdr=%x ext=%x\n", __func__, header.primary, - header.extension); - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, "ipc: bind/unbind failed\n"); - return ret; - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_bind_unbind); - -/* - * In order to load a module we need to send IPC to initiate that. DMA will - * performed to load the module memory. The FW supports multiple module load - * at single shot, so we can send IPC with N modules represented by - * module_cnt - */ -int skl_ipc_load_modules(struct sst_generic_ipc *ipc, - u8 module_cnt, void *data) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_LOAD_MULTIPLE_MODS); - header.primary |= IPC_LOAD_MODULE_CNT(module_cnt); - - request.header = *(u64 *)(&header); - request.data = data; - request.size = sizeof(u16) * module_cnt; - - ret = sst_ipc_tx_message_nowait(ipc, request); - if (ret < 0) - dev_err(ipc->dev, "ipc: load modules failed :%d\n", ret); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_load_modules); - -int skl_ipc_unload_modules(struct sst_generic_ipc *ipc, u8 module_cnt, - void *data) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_UNLOAD_MULTIPLE_MODS); - header.primary |= IPC_LOAD_MODULE_CNT(module_cnt); - - request.header = *(u64 *)(&header); - request.data = data; - request.size = sizeof(u16) * module_cnt; - - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) - dev_err(ipc->dev, "ipc: unload modules failed :%d\n", ret); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_unload_modules); - -int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request; - int ret = 0; - size_t sz_remaining, tx_size, data_offset; - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_MOD_LARGE_CONFIG_SET); - header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); - header.primary |= IPC_MOD_ID(msg->module_id); - - header.extension = IPC_DATA_OFFSET_SZ(msg->param_data_size); - header.extension |= IPC_LARGE_PARAM_ID(msg->large_param_id); - header.extension |= IPC_FINAL_BLOCK(0); - header.extension |= IPC_INITIAL_BLOCK(1); - - sz_remaining = msg->param_data_size; - data_offset = 0; - while (sz_remaining != 0) { - tx_size = sz_remaining > SKL_ADSP_W1_SZ - ? SKL_ADSP_W1_SZ : sz_remaining; - if (tx_size == sz_remaining) - header.extension |= IPC_FINAL_BLOCK(1); - - dev_dbg(ipc->dev, "In %s primary=%#x ext=%#x\n", __func__, - header.primary, header.extension); - dev_dbg(ipc->dev, "transmitting offset: %#x, size: %#x\n", - (unsigned)data_offset, (unsigned)tx_size); - - request.header = *(u64 *)(&header); - request.data = ((char *)param) + data_offset; - request.size = tx_size; - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - if (ret < 0) { - dev_err(ipc->dev, - "ipc: set large config fail, err: %d\n", ret); - return ret; - } - sz_remaining -= tx_size; - data_offset = msg->param_data_size - sz_remaining; - - /* clear the fields */ - header.extension &= IPC_INITIAL_BLOCK_CLEAR; - header.extension &= IPC_DATA_OFFSET_SZ_CLEAR; - /* fill the fields */ - header.extension |= IPC_INITIAL_BLOCK(0); - header.extension |= IPC_DATA_OFFSET_SZ(data_offset); - } - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_set_large_config); - -int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, - u32 **payload, size_t *bytes) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request, reply = {0}; - unsigned int *buf; - int ret; - - reply.data = kzalloc(SKL_ADSP_W1_SZ, GFP_KERNEL); - if (!reply.data) - return -ENOMEM; - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_MOD_LARGE_CONFIG_GET); - header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); - header.primary |= IPC_MOD_ID(msg->module_id); - - header.extension = IPC_DATA_OFFSET_SZ(msg->param_data_size); - header.extension |= IPC_LARGE_PARAM_ID(msg->large_param_id); - header.extension |= IPC_FINAL_BLOCK(1); - header.extension |= IPC_INITIAL_BLOCK(1); - - request.header = *(u64 *)&header; - request.data = *payload; - request.size = *bytes; - reply.size = SKL_ADSP_W1_SZ; - - ret = sst_ipc_tx_message_wait(ipc, request, &reply); - if (ret < 0) - dev_err(ipc->dev, "ipc: get large config fail, err: %d\n", ret); - - reply.size = (reply.header >> 32) & IPC_DATA_OFFSET_SZ_MASK; - buf = krealloc(reply.data, reply.size, GFP_KERNEL); - if (!buf) { - kfree(reply.data); - return -ENOMEM; - } - *payload = buf; - *bytes = reply.size; - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_get_large_config); - -int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc, - u8 dma_id, u8 table_id, bool wait) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret = 0; - - header.primary = IPC_MSG_TARGET(IPC_FW_GEN_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_GLB_LOAD_LIBRARY); - header.primary |= IPC_MOD_INSTANCE_ID(table_id); - header.primary |= IPC_MOD_ID(dma_id); - request.header = *(u64 *)(&header); - - if (wait) - ret = sst_ipc_tx_message_wait(ipc, request, NULL); - else - ret = sst_ipc_tx_message_nowait(ipc, request); - - if (ret < 0) - dev_err(ipc->dev, "ipc: load lib failed\n"); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_sst_ipc_load_library); - -int skl_ipc_set_d0ix(struct sst_generic_ipc *ipc, struct skl_ipc_d0ix_msg *msg) -{ - struct skl_ipc_header header = {0}; - struct sst_ipc_message request = {0}; - int ret; - - header.primary = IPC_MSG_TARGET(IPC_MOD_MSG); - header.primary |= IPC_MSG_DIR(IPC_MSG_REQUEST); - header.primary |= IPC_GLB_TYPE(IPC_MOD_SET_D0IX); - header.primary |= IPC_MOD_INSTANCE_ID(msg->instance_id); - header.primary |= IPC_MOD_ID(msg->module_id); - - header.extension = IPC_D0IX_WAKE(msg->wake); - header.extension |= IPC_D0IX_STREAMING(msg->streaming); - request.header = *(u64 *)(&header); - - dev_dbg(ipc->dev, "In %s primary=%x ext=%x\n", __func__, - header.primary, header.extension); - - /* - * Use the nopm IPC here as we dont want it checking for D0iX - */ - ret = sst_ipc_tx_message_nopm(ipc, request, NULL); - if (ret < 0) - dev_err(ipc->dev, "ipc: set d0ix failed, err %d\n", ret); - - return ret; -} -EXPORT_SYMBOL_GPL(skl_ipc_set_d0ix); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h deleted file mode 100644 index aaaab3b3ec42..000000000000 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ /dev/null @@ -1,169 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Intel SKL IPC Support - * - * Copyright (C) 2014-15, Intel Corporation. - */ - -#ifndef __SKL_IPC_H -#define __SKL_IPC_H - -#include -#include "../common/sst-ipc.h" -#include "skl-sst-dsp.h" - -struct sst_dsp; -struct sst_generic_ipc; - -enum skl_ipc_pipeline_state { - PPL_INVALID_STATE = 0, - PPL_UNINITIALIZED = 1, - PPL_RESET = 2, - PPL_PAUSED = 3, - PPL_RUNNING = 4, - PPL_ERROR_STOP = 5, - PPL_SAVED = 6, - PPL_RESTORED = 7 -}; - -struct skl_ipc_dxstate_info { - u32 core_mask; - u32 dx_mask; -}; - -struct skl_ipc_header { - u32 primary; - u32 extension; -}; - -struct skl_dsp_cores { - unsigned int count; - enum skl_dsp_states *state; - int *usage_count; -}; - -/** - * skl_d0i3_data: skl D0i3 counters data struct - * - * @streaming: Count of usecases that can attempt streaming D0i3 - * @non_streaming: Count of usecases that can attempt non-streaming D0i3 - * @non_d0i3: Count of usecases that cannot attempt D0i3 - * @state: current state - * @work: D0i3 worker thread - */ -struct skl_d0i3_data { - int streaming; - int non_streaming; - int non_d0i3; - enum skl_dsp_d0i3_states state; - struct delayed_work work; -}; - -#define SKL_LIB_NAME_LENGTH 128 -#define SKL_MAX_LIB 16 - -struct skl_lib_info { - char name[SKL_LIB_NAME_LENGTH]; - const struct firmware *fw; -}; - -struct skl_ipc_init_instance_msg { - u32 module_id; - u32 instance_id; - u16 param_data_size; - u8 ppl_instance_id; - u8 core_id; - u8 domain; -}; - -struct skl_ipc_bind_unbind_msg { - u32 module_id; - u32 instance_id; - u32 dst_module_id; - u32 dst_instance_id; - u8 src_queue; - u8 dst_queue; - bool bind; -}; - -struct skl_ipc_large_config_msg { - u32 module_id; - u32 instance_id; - u32 large_param_id; - u32 param_data_size; -}; - -struct skl_ipc_d0ix_msg { - u32 module_id; - u32 instance_id; - u8 streaming; - u8 wake; -}; - -#define SKL_IPC_BOOT_MSECS 3000 - -#define SKL_IPC_D3_MASK 0 -#define SKL_IPC_D0_MASK 3 - -irqreturn_t skl_dsp_irq_thread_handler(int irq, void *context); - -int skl_ipc_create_pipeline(struct sst_generic_ipc *ipc, - u16 ppl_mem_size, u8 ppl_type, u8 instance_id, u8 lp_mode); - -int skl_ipc_delete_pipeline(struct sst_generic_ipc *ipc, u8 instance_id); - -int skl_ipc_set_pipeline_state(struct sst_generic_ipc *ipc, - u8 instance_id, enum skl_ipc_pipeline_state state); - -int skl_ipc_save_pipeline(struct sst_generic_ipc *ipc, - u8 instance_id, int dma_id); - -int skl_ipc_restore_pipeline(struct sst_generic_ipc *ipc, u8 instance_id); - -int skl_ipc_init_instance(struct sst_generic_ipc *ipc, - struct skl_ipc_init_instance_msg *msg, void *param_data); - -int skl_ipc_bind_unbind(struct sst_generic_ipc *ipc, - struct skl_ipc_bind_unbind_msg *msg); - -int skl_ipc_load_modules(struct sst_generic_ipc *ipc, - u8 module_cnt, void *data); - -int skl_ipc_unload_modules(struct sst_generic_ipc *ipc, - u8 module_cnt, void *data); - -int skl_ipc_set_dx(struct sst_generic_ipc *ipc, - u8 instance_id, u16 module_id, struct skl_ipc_dxstate_info *dx); - -int skl_ipc_set_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, u32 *param); - -int skl_ipc_get_large_config(struct sst_generic_ipc *ipc, - struct skl_ipc_large_config_msg *msg, - u32 **payload, size_t *bytes); - -int skl_sst_ipc_load_library(struct sst_generic_ipc *ipc, - u8 dma_id, u8 table_id, bool wait); - -int skl_ipc_set_d0ix(struct sst_generic_ipc *ipc, - struct skl_ipc_d0ix_msg *msg); - -int skl_ipc_check_D0i0(struct sst_dsp *dsp, bool state); - -void skl_ipc_int_enable(struct sst_dsp *ctx); -void skl_ipc_op_int_enable(struct sst_dsp *ctx); -void skl_ipc_op_int_disable(struct sst_dsp *ctx); -void skl_ipc_int_disable(struct sst_dsp *ctx); - -bool skl_ipc_int_status(struct sst_dsp *ctx); -void skl_ipc_free(struct sst_generic_ipc *ipc); -int skl_ipc_init(struct device *dev, struct skl_dev *skl); -void skl_clear_module_cnt(struct sst_dsp *ctx); - -void skl_ipc_process_reply(struct sst_generic_ipc *ipc, - struct skl_ipc_header header); -int skl_ipc_process_notification(struct sst_generic_ipc *ipc, - struct skl_ipc_header header); -void skl_ipc_tx_data_copy(struct ipc_message *msg, char *tx_data, - size_t tx_size); -#endif /* __SKL_IPC_H */ diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c deleted file mode 100644 index b776c58dcf47..000000000000 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ /dev/null @@ -1,425 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-sst-utils.c - SKL sst utils functions - * - * Copyright (C) 2016 Intel Corp - */ - -#include -#include -#include -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "skl.h" - -#define DEFAULT_HASH_SHA256_LEN 32 - -/* FW Extended Manifest Header id = $AE1 */ -#define SKL_EXT_MANIFEST_HEADER_MAGIC 0x31454124 - -union seg_flags { - u32 ul; - struct { - u32 contents : 1; - u32 alloc : 1; - u32 load : 1; - u32 read_only : 1; - u32 code : 1; - u32 data : 1; - u32 _rsvd0 : 2; - u32 type : 4; - u32 _rsvd1 : 4; - u32 length : 16; - } r; -} __packed; - -struct segment_desc { - union seg_flags flags; - u32 v_base_addr; - u32 file_offset; -}; - -struct module_type { - u32 load_type : 4; - u32 auto_start : 1; - u32 domain_ll : 1; - u32 domain_dp : 1; - u32 rsvd : 25; -} __packed; - -struct adsp_module_entry { - u32 struct_id; - u8 name[8]; - u8 uuid[16]; - struct module_type type; - u8 hash1[DEFAULT_HASH_SHA256_LEN]; - u32 entry_point; - u16 cfg_offset; - u16 cfg_count; - u32 affinity_mask; - u16 instance_max_count; - u16 instance_bss_size; - struct segment_desc segments[3]; -} __packed; - -struct adsp_fw_hdr { - u32 id; - u32 len; - u8 name[8]; - u32 preload_page_count; - u32 fw_image_flags; - u32 feature_mask; - u16 major; - u16 minor; - u16 hotfix; - u16 build; - u32 num_modules; - u32 hw_buf_base; - u32 hw_buf_length; - u32 load_offset; -} __packed; - -struct skl_ext_manifest_hdr { - u32 id; - u32 len; - u16 version_major; - u16 version_minor; - u32 entries; -}; - -static int skl_get_pvtid_map(struct uuid_module *module, int instance_id) -{ - int pvt_id; - - for (pvt_id = 0; pvt_id < module->max_instance; pvt_id++) { - if (module->instance_id[pvt_id] == instance_id) - return pvt_id; - } - return -EINVAL; -} - -int skl_get_pvt_instance_id_map(struct skl_dev *skl, - int module_id, int instance_id) -{ - struct uuid_module *module; - - list_for_each_entry(module, &skl->uuid_list, list) { - if (module->id == module_id) - return skl_get_pvtid_map(module, instance_id); - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(skl_get_pvt_instance_id_map); - -static inline int skl_getid_32(struct uuid_module *module, u64 *val, - int word1_mask, int word2_mask) -{ - int index, max_inst, pvt_id; - u32 mask_val; - - max_inst = module->max_instance; - mask_val = (u32)(*val >> word1_mask); - - if (mask_val != 0xffffffff) { - index = ffz(mask_val); - pvt_id = index + word1_mask + word2_mask; - if (pvt_id <= (max_inst - 1)) { - *val |= 1ULL << (index + word1_mask); - return pvt_id; - } - } - - return -EINVAL; -} - -static inline int skl_pvtid_128(struct uuid_module *module) -{ - int j, i, word1_mask, word2_mask = 0, pvt_id; - - for (j = 0; j < MAX_INSTANCE_BUFF; j++) { - word1_mask = 0; - - for (i = 0; i < 2; i++) { - pvt_id = skl_getid_32(module, &module->pvt_id[j], - word1_mask, word2_mask); - if (pvt_id >= 0) - return pvt_id; - - word1_mask += 32; - if ((word1_mask + word2_mask) >= module->max_instance) - return -EINVAL; - } - - word2_mask += 64; - if (word2_mask >= module->max_instance) - return -EINVAL; - } - - return -EINVAL; -} - -/** - * skl_get_pvt_id: generate a private id for use as module id - * - * @skl: driver context - * @uuid_mod: module's uuid - * @instance_id: module's instance id - * - * This generates a 128 bit private unique id for a module TYPE so that - * module instance is unique - */ -int skl_get_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int instance_id) -{ - struct uuid_module *module; - int pvt_id; - - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - - pvt_id = skl_pvtid_128(module); - if (pvt_id >= 0) { - module->instance_id[pvt_id] = instance_id; - - return pvt_id; - } - } - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(skl_get_pvt_id); - -/** - * skl_put_pvt_id: free up the private id allocated - * - * @skl: driver context - * @uuid_mod: module's uuid - * @pvt_id: module pvt id - * - * This frees a 128 bit private unique id previously generated - */ -int skl_put_pvt_id(struct skl_dev *skl, guid_t *uuid_mod, int *pvt_id) -{ - int i; - struct uuid_module *module; - - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid_mod, &module->uuid)) { - - if (*pvt_id != 0) - i = (*pvt_id) / 64; - else - i = 0; - - module->pvt_id[i] &= ~(1 << (*pvt_id)); - *pvt_id = -1; - return 0; - } - } - - return -EINVAL; -} -EXPORT_SYMBOL_GPL(skl_put_pvt_id); - -/* - * Parse the firmware binary to get the UUID, module id - * and loadable flags - */ -int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, - unsigned int offset, int index) -{ - struct adsp_fw_hdr *adsp_hdr; - struct adsp_module_entry *mod_entry; - int i, num_entry, size; - const char *buf; - struct skl_dev *skl = ctx->thread_context; - struct uuid_module *module; - struct firmware stripped_fw; - unsigned int safe_file; - int ret; - - /* Get the FW pointer to derive ADSP header */ - stripped_fw.data = fw->data; - stripped_fw.size = fw->size; - - skl_dsp_strip_extended_manifest(&stripped_fw); - - buf = stripped_fw.data; - - /* check if we have enough space in file to move to header */ - safe_file = sizeof(*adsp_hdr) + offset; - if (stripped_fw.size <= safe_file) { - dev_err(ctx->dev, "Small fw file size, No space for hdr\n"); - return -EINVAL; - } - - adsp_hdr = (struct adsp_fw_hdr *)(buf + offset); - - /* check 1st module entry is in file */ - safe_file += adsp_hdr->len + sizeof(*mod_entry); - if (stripped_fw.size <= safe_file) { - dev_err(ctx->dev, "Small fw file size, No module entry\n"); - return -EINVAL; - } - - mod_entry = (struct adsp_module_entry *)(buf + offset + adsp_hdr->len); - - num_entry = adsp_hdr->num_modules; - - /* check all entries are in file */ - safe_file += num_entry * sizeof(*mod_entry); - if (stripped_fw.size <= safe_file) { - dev_err(ctx->dev, "Small fw file size, No modules\n"); - return -EINVAL; - } - - - /* - * Read the UUID(GUID) from FW Manifest. - * - * The 16 byte UUID format is: XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX - * Populate the UUID table to store module_id and loadable flags - * for the module. - */ - - for (i = 0; i < num_entry; i++, mod_entry++) { - module = kzalloc(sizeof(*module), GFP_KERNEL); - if (!module) { - ret = -ENOMEM; - goto free_uuid_list; - } - - import_guid(&module->uuid, mod_entry->uuid); - - module->id = (i | (index << 12)); - module->is_loadable = mod_entry->type.load_type; - module->max_instance = mod_entry->instance_max_count; - size = sizeof(int) * mod_entry->instance_max_count; - module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); - if (!module->instance_id) { - ret = -ENOMEM; - kfree(module); - goto free_uuid_list; - } - - list_add_tail(&module->list, &skl->uuid_list); - - dev_dbg(ctx->dev, - "Adding uuid :%pUL mod id: %d Loadable: %d\n", - &module->uuid, module->id, module->is_loadable); - } - - return 0; - -free_uuid_list: - skl_freeup_uuid_list(skl); - return ret; -} - -void skl_freeup_uuid_list(struct skl_dev *skl) -{ - struct uuid_module *uuid, *_uuid; - - list_for_each_entry_safe(uuid, _uuid, &skl->uuid_list, list) { - list_del(&uuid->list); - kfree(uuid); - } -} - -/* - * some firmware binary contains some extended manifest. This needs - * to be stripped in that case before we load and use that image. - * - * Get the module id for the module by checking - * the table for the UUID for the module - */ -int skl_dsp_strip_extended_manifest(struct firmware *fw) -{ - struct skl_ext_manifest_hdr *hdr; - - /* check if fw file is greater than header we are looking */ - if (fw->size < sizeof(hdr)) { - pr_err("%s: Firmware file small, no hdr\n", __func__); - return -EINVAL; - } - - hdr = (struct skl_ext_manifest_hdr *)fw->data; - - if (hdr->id == SKL_EXT_MANIFEST_HEADER_MAGIC) { - fw->size -= hdr->len; - fw->data += hdr->len; - } - - return 0; -} - -int skl_sst_ctx_init(struct device *dev, int irq, const char *fw_name, - struct skl_dsp_loader_ops dsp_ops, struct skl_dev **dsp, - struct sst_dsp_device *skl_dev) -{ - struct skl_dev *skl = *dsp; - struct sst_dsp *sst; - - skl->dev = dev; - skl_dev->thread_context = skl; - INIT_LIST_HEAD(&skl->uuid_list); - skl->dsp = skl_dsp_ctx_init(dev, skl_dev, irq); - if (!skl->dsp) { - dev_err(skl->dev, "%s: no device\n", __func__); - return -ENODEV; - } - - sst = skl->dsp; - sst->fw_name = fw_name; - sst->dsp_ops = dsp_ops; - init_waitqueue_head(&skl->mod_load_wait); - INIT_LIST_HEAD(&sst->module_list); - - skl->is_first_boot = true; - - return 0; -} - -int skl_prepare_lib_load(struct skl_dev *skl, struct skl_lib_info *linfo, - struct firmware *stripped_fw, - unsigned int hdr_offset, int index) -{ - int ret; - struct sst_dsp *dsp = skl->dsp; - - if (linfo->fw == NULL) { - ret = request_firmware(&linfo->fw, linfo->name, - skl->dev); - if (ret < 0) { - dev_err(skl->dev, "Request lib %s failed:%d\n", - linfo->name, ret); - return ret; - } - } - - if (skl->is_first_boot) { - ret = snd_skl_parse_uuids(dsp, linfo->fw, hdr_offset, index); - if (ret < 0) - return ret; - } - - stripped_fw->data = linfo->fw->data; - stripped_fw->size = linfo->fw->size; - skl_dsp_strip_extended_manifest(stripped_fw); - - return 0; -} - -void skl_release_library(struct skl_lib_info *linfo, int lib_count) -{ - int i; - - /* library indices start from 1 to N. 0 represents base FW */ - for (i = 1; i < lib_count; i++) { - if (linfo[i].fw) { - release_firmware(linfo[i].fw); - linfo[i].fw = NULL; - } - } -} diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c deleted file mode 100644 index 39d027ac16ec..000000000000 --- a/sound/soc/intel/skylake/skl-sst.c +++ /dev/null @@ -1,599 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-sst.c - HDA DSP library functions for SKL platform - * - * Copyright (C) 2014-15, Intel Corporation. - * Author:Rafal Redzimski - * Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" -#include "../common/sst-ipc.h" -#include "skl.h" - -#define SKL_BASEFW_TIMEOUT 300 -#define SKL_INIT_TIMEOUT 1000 - -/* Intel HD Audio SRAM Window 0*/ -#define SKL_ADSP_SRAM0_BASE 0x8000 - -/* Firmware status window */ -#define SKL_ADSP_FW_STATUS SKL_ADSP_SRAM0_BASE -#define SKL_ADSP_ERROR_CODE (SKL_ADSP_FW_STATUS + 0x4) - -#define SKL_NUM_MODULES 1 - -static bool skl_check_fw_status(struct sst_dsp *ctx, u32 status) -{ - u32 cur_sts; - - cur_sts = sst_dsp_shim_read(ctx, SKL_ADSP_FW_STATUS) & SKL_FW_STS_MASK; - - return (cur_sts == status); -} - -static int skl_transfer_firmware(struct sst_dsp *ctx, - const void *basefw, u32 base_fw_size) -{ - int ret = 0; - - ret = ctx->cl_dev.ops.cl_copy_to_dmabuf(ctx, basefw, base_fw_size, - true); - if (ret < 0) - return ret; - - ret = sst_dsp_register_poll(ctx, - SKL_ADSP_FW_STATUS, - SKL_FW_STS_MASK, - SKL_FW_RFW_START, - SKL_BASEFW_TIMEOUT, - "Firmware boot"); - - ctx->cl_dev.ops.cl_stop_dma(ctx); - - return ret; -} - -#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 - -static int skl_load_base_firmware(struct sst_dsp *ctx) -{ - int ret = 0, i; - struct skl_dev *skl = ctx->thread_context; - struct firmware stripped_fw; - u32 reg; - - skl->boot_complete = false; - init_waitqueue_head(&skl->boot_wait); - - if (ctx->fw == NULL) { - ret = request_firmware(&ctx->fw, ctx->fw_name, ctx->dev); - if (ret < 0) { - dev_err(ctx->dev, "Request firmware failed %d\n", ret); - return -EIO; - } - } - - /* prase uuids on first boot */ - if (skl->is_first_boot) { - ret = snd_skl_parse_uuids(ctx, ctx->fw, SKL_ADSP_FW_BIN_HDR_OFFSET, 0); - if (ret < 0) { - dev_err(ctx->dev, "UUID parsing err: %d\n", ret); - release_firmware(ctx->fw); - skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - return ret; - } - } - - /* check for extended manifest */ - stripped_fw.data = ctx->fw->data; - stripped_fw.size = ctx->fw->size; - - skl_dsp_strip_extended_manifest(&stripped_fw); - - ret = skl_dsp_boot(ctx); - if (ret < 0) { - dev_err(ctx->dev, "Boot dsp core failed ret: %d\n", ret); - goto skl_load_base_firmware_failed; - } - - ret = skl_cldma_prepare(ctx); - if (ret < 0) { - dev_err(ctx->dev, "CL dma prepare failed : %d\n", ret); - goto skl_load_base_firmware_failed; - } - - /* enable Interrupt */ - skl_ipc_int_enable(ctx); - skl_ipc_op_int_enable(ctx); - - /* check ROM Status */ - for (i = SKL_INIT_TIMEOUT; i > 0; --i) { - if (skl_check_fw_status(ctx, SKL_FW_INIT)) { - dev_dbg(ctx->dev, - "ROM loaded, we can continue with FW loading\n"); - break; - } - mdelay(1); - } - if (!i) { - reg = sst_dsp_shim_read(ctx, SKL_ADSP_FW_STATUS); - dev_err(ctx->dev, - "Timeout waiting for ROM init done, reg:0x%x\n", reg); - ret = -EIO; - goto transfer_firmware_failed; - } - - ret = skl_transfer_firmware(ctx, stripped_fw.data, stripped_fw.size); - if (ret < 0) { - dev_err(ctx->dev, "Transfer firmware failed%d\n", ret); - goto transfer_firmware_failed; - } else { - ret = wait_event_timeout(skl->boot_wait, skl->boot_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - if (ret == 0) { - dev_err(ctx->dev, "DSP boot failed, FW Ready timed-out\n"); - ret = -EIO; - goto transfer_firmware_failed; - } - - dev_dbg(ctx->dev, "Download firmware successful%d\n", ret); - skl->fw_loaded = true; - } - return 0; -transfer_firmware_failed: - ctx->cl_dev.ops.cl_cleanup_controller(ctx); -skl_load_base_firmware_failed: - skl_dsp_disable_core(ctx, SKL_DSP_CORE0_MASK); - release_firmware(ctx->fw); - ctx->fw = NULL; - return ret; -} - -static int skl_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) -{ - int ret; - struct skl_ipc_dxstate_info dx; - struct skl_dev *skl = ctx->thread_context; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - - /* If core0 is being turned on, we need to load the FW */ - if (core_id == SKL_DSP_CORE0_ID) { - ret = skl_load_base_firmware(ctx); - if (ret < 0) { - dev_err(ctx->dev, "unable to load firmware\n"); - return ret; - } - - /* load libs as they are also lost on D3 */ - if (skl->lib_count > 1) { - ret = ctx->fw_ops.load_library(ctx, skl->lib_info, - skl->lib_count); - if (ret < 0) { - dev_err(ctx->dev, "reload libs failed: %d\n", - ret); - return ret; - } - - } - } - - /* - * If any core other than core 0 is being moved to D0, enable the - * core and send the set dx IPC for the core. - */ - if (core_id != SKL_DSP_CORE0_ID) { - ret = skl_dsp_enable_core(ctx, core_mask); - if (ret < 0) - return ret; - - dx.core_mask = core_mask; - dx.dx_mask = core_mask; - - ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, - SKL_BASE_FW_MODULE_ID, &dx); - if (ret < 0) { - dev_err(ctx->dev, "Failed to set dsp to D0:core id= %d\n", - core_id); - skl_dsp_disable_core(ctx, core_mask); - } - } - - skl->cores.state[core_id] = SKL_DSP_RUNNING; - - return 0; -} - -static int skl_set_dsp_D3(struct sst_dsp *ctx, unsigned int core_id) -{ - int ret; - struct skl_ipc_dxstate_info dx; - struct skl_dev *skl = ctx->thread_context; - unsigned int core_mask = SKL_DSP_CORE_MASK(core_id); - - dx.core_mask = core_mask; - dx.dx_mask = SKL_IPC_D3_MASK; - - ret = skl_ipc_set_dx(&skl->ipc, SKL_INSTANCE_ID, SKL_BASE_FW_MODULE_ID, &dx); - if (ret < 0) - dev_err(ctx->dev, "set Dx core %d fail: %d\n", core_id, ret); - - if (core_id == SKL_DSP_CORE0_ID) { - /* disable Interrupt */ - ctx->cl_dev.ops.cl_cleanup_controller(ctx); - skl_cldma_int_disable(ctx); - skl_ipc_op_int_disable(ctx); - skl_ipc_int_disable(ctx); - } - - ret = skl_dsp_disable_core(ctx, core_mask); - if (ret < 0) - return ret; - - skl->cores.state[core_id] = SKL_DSP_RESET; - return ret; -} - -static unsigned int skl_get_errorcode(struct sst_dsp *ctx) -{ - return sst_dsp_shim_read(ctx, SKL_ADSP_ERROR_CODE); -} - -/* - * since get/set_module are called from DAPM context, - * we don't need lock for usage count - */ -static int skl_get_module(struct sst_dsp *ctx, u16 mod_id) -{ - struct skl_module_table *module; - - list_for_each_entry(module, &ctx->module_list, list) { - if (module->mod_info->mod_id == mod_id) - return ++module->usage_cnt; - } - - return -EINVAL; -} - -static int skl_put_module(struct sst_dsp *ctx, u16 mod_id) -{ - struct skl_module_table *module; - - list_for_each_entry(module, &ctx->module_list, list) { - if (module->mod_info->mod_id == mod_id) - return --module->usage_cnt; - } - - return -EINVAL; -} - -static struct skl_module_table *skl_fill_module_table(struct sst_dsp *ctx, - char *mod_name, int mod_id) -{ - const struct firmware *fw; - struct skl_module_table *skl_module; - unsigned int size; - int ret; - - ret = request_firmware(&fw, mod_name, ctx->dev); - if (ret < 0) { - dev_err(ctx->dev, "Request Module %s failed :%d\n", - mod_name, ret); - return NULL; - } - - skl_module = devm_kzalloc(ctx->dev, sizeof(*skl_module), GFP_KERNEL); - if (skl_module == NULL) { - release_firmware(fw); - return NULL; - } - - size = sizeof(*skl_module->mod_info); - skl_module->mod_info = devm_kzalloc(ctx->dev, size, GFP_KERNEL); - if (skl_module->mod_info == NULL) { - release_firmware(fw); - return NULL; - } - - skl_module->mod_info->mod_id = mod_id; - skl_module->mod_info->fw = fw; - list_add(&skl_module->list, &ctx->module_list); - - return skl_module; -} - -/* get a module from it's unique ID */ -static struct skl_module_table *skl_module_get_from_id( - struct sst_dsp *ctx, u16 mod_id) -{ - struct skl_module_table *module; - - if (list_empty(&ctx->module_list)) { - dev_err(ctx->dev, "Module list is empty\n"); - return NULL; - } - - list_for_each_entry(module, &ctx->module_list, list) { - if (module->mod_info->mod_id == mod_id) - return module; - } - - return NULL; -} - -static int skl_transfer_module(struct sst_dsp *ctx, const void *data, - u32 size, u16 mod_id, u8 table_id, bool is_module) -{ - int ret, bytes_left, curr_pos; - struct skl_dev *skl = ctx->thread_context; - skl->mod_load_complete = false; - - bytes_left = ctx->cl_dev.ops.cl_copy_to_dmabuf(ctx, data, size, false); - if (bytes_left < 0) - return bytes_left; - - /* check is_module flag to load module or library */ - if (is_module) - ret = skl_ipc_load_modules(&skl->ipc, SKL_NUM_MODULES, &mod_id); - else - ret = skl_sst_ipc_load_library(&skl->ipc, 0, table_id, false); - - if (ret < 0) { - dev_err(ctx->dev, "Failed to Load %s with err %d\n", - is_module ? "module" : "lib", ret); - goto out; - } - - /* - * if bytes_left > 0 then wait for BDL complete interrupt and - * copy the next chunk till bytes_left is 0. if bytes_left is - * zero, then wait for load module IPC reply - */ - while (bytes_left > 0) { - curr_pos = size - bytes_left; - - ret = skl_cldma_wait_interruptible(ctx); - if (ret < 0) - goto out; - - bytes_left = ctx->cl_dev.ops.cl_copy_to_dmabuf(ctx, - data + curr_pos, - bytes_left, false); - } - - ret = wait_event_timeout(skl->mod_load_wait, skl->mod_load_complete, - msecs_to_jiffies(SKL_IPC_BOOT_MSECS)); - if (ret == 0 || !skl->mod_load_status) { - dev_err(ctx->dev, "Module Load failed\n"); - ret = -EIO; - } - -out: - ctx->cl_dev.ops.cl_stop_dma(ctx); - - return ret; -} - -static int -skl_load_library(struct sst_dsp *ctx, struct skl_lib_info *linfo, int lib_count) -{ - struct skl_dev *skl = ctx->thread_context; - struct firmware stripped_fw; - int ret, i; - - /* library indices start from 1 to N. 0 represents base FW */ - for (i = 1; i < lib_count; i++) { - ret = skl_prepare_lib_load(skl, &skl->lib_info[i], &stripped_fw, - SKL_ADSP_FW_BIN_HDR_OFFSET, i); - if (ret < 0) - goto load_library_failed; - ret = skl_transfer_module(ctx, stripped_fw.data, - stripped_fw.size, 0, i, false); - if (ret < 0) - goto load_library_failed; - } - - return 0; - -load_library_failed: - skl_release_library(linfo, lib_count); - return ret; -} - -static int skl_load_module(struct sst_dsp *ctx, u16 mod_id, u8 *guid) -{ - struct skl_module_table *module_entry = NULL; - int ret = 0; - char mod_name[64]; /* guid str = 32 chars + 4 hyphens */ - - snprintf(mod_name, sizeof(mod_name), "intel/dsp_fw_%pUL.bin", guid); - - module_entry = skl_module_get_from_id(ctx, mod_id); - if (module_entry == NULL) { - module_entry = skl_fill_module_table(ctx, mod_name, mod_id); - if (module_entry == NULL) { - dev_err(ctx->dev, "Failed to Load module\n"); - return -EINVAL; - } - } - - if (!module_entry->usage_cnt) { - ret = skl_transfer_module(ctx, module_entry->mod_info->fw->data, - module_entry->mod_info->fw->size, - mod_id, 0, true); - if (ret < 0) { - dev_err(ctx->dev, "Failed to Load module\n"); - return ret; - } - } - - ret = skl_get_module(ctx, mod_id); - - return ret; -} - -static int skl_unload_module(struct sst_dsp *ctx, u16 mod_id) -{ - int usage_cnt; - struct skl_dev *skl = ctx->thread_context; - int ret = 0; - - usage_cnt = skl_put_module(ctx, mod_id); - if (usage_cnt < 0) { - dev_err(ctx->dev, "Module bad usage cnt!:%d\n", usage_cnt); - return -EIO; - } - - /* if module is used by others return, no need to unload */ - if (usage_cnt > 0) - return 0; - - ret = skl_ipc_unload_modules(&skl->ipc, - SKL_NUM_MODULES, &mod_id); - if (ret < 0) { - dev_err(ctx->dev, "Failed to UnLoad module\n"); - skl_get_module(ctx, mod_id); - return ret; - } - - return ret; -} - -void skl_clear_module_cnt(struct sst_dsp *ctx) -{ - struct skl_module_table *module; - - if (list_empty(&ctx->module_list)) - return; - - list_for_each_entry(module, &ctx->module_list, list) { - module->usage_cnt = 0; - } -} -EXPORT_SYMBOL_GPL(skl_clear_module_cnt); - -static void skl_clear_module_table(struct sst_dsp *ctx) -{ - struct skl_module_table *module, *tmp; - - if (list_empty(&ctx->module_list)) - return; - - list_for_each_entry_safe(module, tmp, &ctx->module_list, list) { - list_del(&module->list); - release_firmware(module->mod_info->fw); - } -} - -static const struct skl_dsp_fw_ops skl_fw_ops = { - .set_state_D0 = skl_set_dsp_D0, - .set_state_D3 = skl_set_dsp_D3, - .load_fw = skl_load_base_firmware, - .get_fw_errcode = skl_get_errorcode, - .load_library = skl_load_library, - .load_mod = skl_load_module, - .unload_mod = skl_unload_module, -}; - -static struct sst_ops skl_ops = { - .irq_handler = skl_dsp_sst_interrupt, - .write = sst_shim32_write, - .read = sst_shim32_read, - .free = skl_dsp_free, -}; - -static struct sst_dsp_device skl_dev = { - .thread = skl_dsp_irq_thread_handler, - .ops = &skl_ops, -}; - -int skl_sst_dsp_init(struct device *dev, void __iomem *mmio_base, int irq, - const char *fw_name, struct skl_dsp_loader_ops dsp_ops, - struct skl_dev **dsp) -{ - struct skl_dev *skl; - struct sst_dsp *sst; - int ret; - - ret = skl_sst_ctx_init(dev, irq, fw_name, dsp_ops, dsp, &skl_dev); - if (ret < 0) { - dev_err(dev, "%s: no device\n", __func__); - return ret; - } - - skl = *dsp; - sst = skl->dsp; - sst->addr.lpe = mmio_base; - sst->addr.shim = mmio_base; - sst->addr.sram0_base = SKL_ADSP_SRAM0_BASE; - sst->addr.sram1_base = SKL_ADSP_SRAM1_BASE; - sst->addr.w0_stat_sz = SKL_ADSP_W0_STAT_SZ; - sst->addr.w0_up_sz = SKL_ADSP_W0_UP_SZ; - - sst_dsp_mailbox_init(sst, (SKL_ADSP_SRAM0_BASE + SKL_ADSP_W0_STAT_SZ), - SKL_ADSP_W0_UP_SZ, SKL_ADSP_SRAM1_BASE, SKL_ADSP_W1_SZ); - - ret = skl_ipc_init(dev, skl); - if (ret) { - skl_dsp_free(sst); - return ret; - } - - sst->fw_ops = skl_fw_ops; - - return skl_dsp_acquire_irq(sst); -} -EXPORT_SYMBOL_GPL(skl_sst_dsp_init); - -int skl_sst_init_fw(struct device *dev, struct skl_dev *skl) -{ - int ret; - struct sst_dsp *sst = skl->dsp; - - ret = sst->fw_ops.load_fw(sst); - if (ret < 0) { - dev_err(dev, "Load base fw failed : %d\n", ret); - return ret; - } - - skl_dsp_init_core_state(sst); - - if (skl->lib_count > 1) { - ret = sst->fw_ops.load_library(sst, skl->lib_info, - skl->lib_count); - if (ret < 0) { - dev_err(dev, "Load Library failed : %x\n", ret); - return ret; - } - } - skl->is_first_boot = false; - - return 0; -} -EXPORT_SYMBOL_GPL(skl_sst_init_fw); - -void skl_sst_dsp_cleanup(struct device *dev, struct skl_dev *skl) -{ - - if (skl->dsp->fw) - release_firmware(skl->dsp->fw); - skl_clear_module_table(skl->dsp); - skl_freeup_uuid_list(skl); - skl_ipc_free(&skl->ipc); - skl->dsp->ops->free(skl->dsp); - if (skl->boot_complete) { - skl->dsp->cl_dev.ops.cl_cleanup_controller(skl->dsp); - skl_cldma_int_disable(skl->dsp); - } -} -EXPORT_SYMBOL_GPL(skl_sst_dsp_cleanup); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Skylake IPC driver"); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c deleted file mode 100644 index 602ef4321122..000000000000 --- a/sound/soc/intel/skylake/skl-topology.c +++ /dev/null @@ -1,3605 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl-topology.c - Implements Platform component ALSA controls/widget - * handlers. - * - * Copyright (C) 2014-2015 Intel Corp - * Author: Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "skl-sst-dsp.h" -#include "skl-sst-ipc.h" -#include "skl-topology.h" -#include "skl.h" -#include "../common/sst-dsp.h" -#include "../common/sst-dsp-priv.h" - -#define SKL_CH_FIXUP_MASK (1 << 0) -#define SKL_RATE_FIXUP_MASK (1 << 1) -#define SKL_FMT_FIXUP_MASK (1 << 2) -#define SKL_IN_DIR_BIT_MASK BIT(0) -#define SKL_PIN_COUNT_MASK GENMASK(7, 4) - -static const int mic_mono_list[] = { -0, 1, 2, 3, -}; -static const int mic_stereo_list[][SKL_CH_STEREO] = { -{0, 1}, {0, 2}, {0, 3}, {1, 2}, {1, 3}, {2, 3}, -}; -static const int mic_trio_list[][SKL_CH_TRIO] = { -{0, 1, 2}, {0, 1, 3}, {0, 2, 3}, {1, 2, 3}, -}; -static const int mic_quatro_list[][SKL_CH_QUATRO] = { -{0, 1, 2, 3}, -}; - -#define CHECK_HW_PARAMS(ch, freq, bps, prm_ch, prm_freq, prm_bps) \ - ((ch == prm_ch) && (bps == prm_bps) && (freq == prm_freq)) - -void skl_tplg_d0i3_get(struct skl_dev *skl, enum d0i3_capability caps) -{ - struct skl_d0i3_data *d0i3 = &skl->d0i3; - - switch (caps) { - case SKL_D0I3_NONE: - d0i3->non_d0i3++; - break; - - case SKL_D0I3_STREAMING: - d0i3->streaming++; - break; - - case SKL_D0I3_NON_STREAMING: - d0i3->non_streaming++; - break; - } -} - -void skl_tplg_d0i3_put(struct skl_dev *skl, enum d0i3_capability caps) -{ - struct skl_d0i3_data *d0i3 = &skl->d0i3; - - switch (caps) { - case SKL_D0I3_NONE: - d0i3->non_d0i3--; - break; - - case SKL_D0I3_STREAMING: - d0i3->streaming--; - break; - - case SKL_D0I3_NON_STREAMING: - d0i3->non_streaming--; - break; - } -} - -/* - * SKL DSP driver modelling uses only few DAPM widgets so for rest we will - * ignore. This helpers checks if the SKL driver handles this widget type - */ -static int is_skl_dsp_widget_type(struct snd_soc_dapm_widget *w, - struct device *dev) -{ - if (w->dapm->dev != dev) - return false; - - switch (w->id) { - case snd_soc_dapm_dai_link: - case snd_soc_dapm_dai_in: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai_out: - case snd_soc_dapm_switch: - case snd_soc_dapm_output: - case snd_soc_dapm_mux: - - return false; - default: - return true; - } -} - -static void skl_dump_mconfig(struct skl_dev *skl, struct skl_module_cfg *mcfg) -{ - struct skl_module_iface *iface = &mcfg->module->formats[mcfg->fmt_idx]; - - dev_dbg(skl->dev, "Dumping config\n"); - dev_dbg(skl->dev, "Input Format:\n"); - dev_dbg(skl->dev, "channels = %d\n", iface->inputs[0].fmt.channels); - dev_dbg(skl->dev, "s_freq = %d\n", iface->inputs[0].fmt.s_freq); - dev_dbg(skl->dev, "ch_cfg = %d\n", iface->inputs[0].fmt.ch_cfg); - dev_dbg(skl->dev, "valid bit depth = %d\n", - iface->inputs[0].fmt.valid_bit_depth); - dev_dbg(skl->dev, "Output Format:\n"); - dev_dbg(skl->dev, "channels = %d\n", iface->outputs[0].fmt.channels); - dev_dbg(skl->dev, "s_freq = %d\n", iface->outputs[0].fmt.s_freq); - dev_dbg(skl->dev, "valid bit depth = %d\n", - iface->outputs[0].fmt.valid_bit_depth); - dev_dbg(skl->dev, "ch_cfg = %d\n", iface->outputs[0].fmt.ch_cfg); -} - -static void skl_tplg_update_chmap(struct skl_module_fmt *fmt, int chs) -{ - int slot_map = 0xFFFFFFFF; - int start_slot = 0; - int i; - - for (i = 0; i < chs; i++) { - /* - * For 2 channels with starting slot as 0, slot map will - * look like 0xFFFFFF10. - */ - slot_map &= (~(0xF << (4 * i)) | (start_slot << (4 * i))); - start_slot++; - } - fmt->ch_map = slot_map; -} - -static void skl_tplg_update_params(struct skl_module_fmt *fmt, - struct skl_pipe_params *params, int fixup) -{ - if (fixup & SKL_RATE_FIXUP_MASK) - fmt->s_freq = params->s_freq; - if (fixup & SKL_CH_FIXUP_MASK) { - fmt->channels = params->ch; - skl_tplg_update_chmap(fmt, fmt->channels); - } - if (fixup & SKL_FMT_FIXUP_MASK) { - fmt->valid_bit_depth = skl_get_bit_depth(params->s_fmt); - - /* - * 16 bit is 16 bit container whereas 24 bit is in 32 bit - * container so update bit depth accordingly - */ - switch (fmt->valid_bit_depth) { - case SKL_DEPTH_16BIT: - fmt->bit_depth = fmt->valid_bit_depth; - break; - - default: - fmt->bit_depth = SKL_DEPTH_32BIT; - break; - } - } - -} - -/* - * A pipeline may have modules which impact the pcm parameters, like SRC, - * channel converter, format converter. - * We need to calculate the output params by applying the 'fixup' - * Topology will tell driver which type of fixup is to be applied by - * supplying the fixup mask, so based on that we calculate the output - * - * Now In FE the pcm hw_params is source/target format. Same is applicable - * for BE with its hw_params invoked. - * here based on FE, BE pipeline and direction we calculate the input and - * outfix and then apply that for a module - */ -static void skl_tplg_update_params_fixup(struct skl_module_cfg *m_cfg, - struct skl_pipe_params *params, bool is_fe) -{ - int in_fixup, out_fixup; - struct skl_module_fmt *in_fmt, *out_fmt; - - /* Fixups will be applied to pin 0 only */ - in_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].inputs[0].fmt; - out_fmt = &m_cfg->module->formats[m_cfg->fmt_idx].outputs[0].fmt; - - if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (is_fe) { - in_fixup = m_cfg->params_fixup; - out_fixup = (~m_cfg->converter) & - m_cfg->params_fixup; - } else { - out_fixup = m_cfg->params_fixup; - in_fixup = (~m_cfg->converter) & - m_cfg->params_fixup; - } - } else { - if (is_fe) { - out_fixup = m_cfg->params_fixup; - in_fixup = (~m_cfg->converter) & - m_cfg->params_fixup; - } else { - in_fixup = m_cfg->params_fixup; - out_fixup = (~m_cfg->converter) & - m_cfg->params_fixup; - } - } - - skl_tplg_update_params(in_fmt, params, in_fixup); - skl_tplg_update_params(out_fmt, params, out_fixup); -} - -/* - * A module needs input and output buffers, which are dependent upon pcm - * params, so once we have calculate params, we need buffer calculation as - * well. - */ -static void skl_tplg_update_buffer_size(struct skl_dev *skl, - struct skl_module_cfg *mcfg) -{ - int multiplier = 1; - struct skl_module_fmt *in_fmt, *out_fmt; - struct skl_module_res *res; - - /* Since fixups is applied to pin 0 only, ibs, obs needs - * change for pin 0 only - */ - res = &mcfg->module->resources[mcfg->res_idx]; - in_fmt = &mcfg->module->formats[mcfg->fmt_idx].inputs[0].fmt; - out_fmt = &mcfg->module->formats[mcfg->fmt_idx].outputs[0].fmt; - - if (mcfg->m_type == SKL_MODULE_TYPE_SRCINT) - multiplier = 5; - - res->ibs = DIV_ROUND_UP(in_fmt->s_freq, 1000) * - in_fmt->channels * (in_fmt->bit_depth >> 3) * - multiplier; - - res->obs = DIV_ROUND_UP(out_fmt->s_freq, 1000) * - out_fmt->channels * (out_fmt->bit_depth >> 3) * - multiplier; -} - -static u8 skl_tplg_be_dev_type(int dev_type) -{ - int ret; - - switch (dev_type) { - case SKL_DEVICE_BT: - ret = NHLT_DEVICE_BT; - break; - - case SKL_DEVICE_DMIC: - ret = NHLT_DEVICE_DMIC; - break; - - case SKL_DEVICE_I2S: - ret = NHLT_DEVICE_I2S; - break; - - default: - ret = NHLT_DEVICE_INVALID; - break; - } - - return ret; -} - -static int skl_tplg_update_be_blob(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *m_cfg = w->priv; - int link_type, dir; - u32 ch, s_freq, s_fmt, s_cont; - struct nhlt_specific_cfg *cfg; - u8 dev_type = skl_tplg_be_dev_type(m_cfg->dev_type); - int fmt_idx = m_cfg->fmt_idx; - struct skl_module_iface *m_iface = &m_cfg->module->formats[fmt_idx]; - - /* check if we already have blob */ - if (m_cfg->formats_config[SKL_PARAM_INIT].caps_size > 0) - return 0; - - dev_dbg(skl->dev, "Applying default cfg blob\n"); - switch (m_cfg->dev_type) { - case SKL_DEVICE_DMIC: - link_type = NHLT_LINK_DMIC; - dir = SNDRV_PCM_STREAM_CAPTURE; - s_freq = m_iface->inputs[0].fmt.s_freq; - s_fmt = m_iface->inputs[0].fmt.valid_bit_depth; - s_cont = m_iface->inputs[0].fmt.bit_depth; - ch = m_iface->inputs[0].fmt.channels; - break; - - case SKL_DEVICE_I2S: - link_type = NHLT_LINK_SSP; - if (m_cfg->hw_conn_type == SKL_CONN_SOURCE) { - dir = SNDRV_PCM_STREAM_PLAYBACK; - s_freq = m_iface->outputs[0].fmt.s_freq; - s_fmt = m_iface->outputs[0].fmt.valid_bit_depth; - s_cont = m_iface->outputs[0].fmt.bit_depth; - ch = m_iface->outputs[0].fmt.channels; - } else { - dir = SNDRV_PCM_STREAM_CAPTURE; - s_freq = m_iface->inputs[0].fmt.s_freq; - s_fmt = m_iface->inputs[0].fmt.valid_bit_depth; - s_cont = m_iface->inputs[0].fmt.bit_depth; - ch = m_iface->inputs[0].fmt.channels; - } - break; - - default: - return -EINVAL; - } - - /* update the blob based on virtual bus_id and default params */ - cfg = intel_nhlt_get_endpoint_blob(skl->dev, skl->nhlt, m_cfg->vbus_id, - link_type, s_fmt, s_cont, ch, - s_freq, dir, dev_type); - if (cfg) { - m_cfg->formats_config[SKL_PARAM_INIT].caps_size = cfg->size; - m_cfg->formats_config[SKL_PARAM_INIT].caps = (u32 *)&cfg->caps; - } else { - dev_err(skl->dev, "Blob NULL for id %x type %d dirn %d\n", - m_cfg->vbus_id, link_type, dir); - dev_err(skl->dev, "PCM: ch %d, freq %d, fmt %d/%d\n", - ch, s_freq, s_fmt, s_cont); - return -EIO; - } - - return 0; -} - -static void skl_tplg_update_module_params(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *m_cfg = w->priv; - struct skl_pipe_params *params = m_cfg->pipe->p_params; - int p_conn_type = m_cfg->pipe->conn_type; - bool is_fe; - - if (!m_cfg->params_fixup) - return; - - dev_dbg(skl->dev, "Mconfig for widget=%s BEFORE updation\n", - w->name); - - skl_dump_mconfig(skl, m_cfg); - - if (p_conn_type == SKL_PIPE_CONN_TYPE_FE) - is_fe = true; - else - is_fe = false; - - skl_tplg_update_params_fixup(m_cfg, params, is_fe); - skl_tplg_update_buffer_size(skl, m_cfg); - - dev_dbg(skl->dev, "Mconfig for widget=%s AFTER updation\n", - w->name); - - skl_dump_mconfig(skl, m_cfg); -} - -/* - * some modules can have multiple params set from user control and - * need to be set after module is initialized. If set_param flag is - * set module params will be done after module is initialised. - */ -static int skl_tplg_set_module_params(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - int i, ret; - struct skl_module_cfg *mconfig = w->priv; - const struct snd_kcontrol_new *k; - struct soc_bytes_ext *sb; - struct skl_algo_data *bc; - struct skl_specific_cfg *sp_cfg; - - if (mconfig->formats_config[SKL_PARAM_SET].caps_size > 0 && - mconfig->formats_config[SKL_PARAM_SET].set_params == SKL_PARAM_SET) { - sp_cfg = &mconfig->formats_config[SKL_PARAM_SET]; - ret = skl_set_module_params(skl, sp_cfg->caps, - sp_cfg->caps_size, - sp_cfg->param_id, mconfig); - if (ret < 0) - return ret; - } - - for (i = 0; i < w->num_kcontrols; i++) { - k = &w->kcontrol_news[i]; - if (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - sb = (void *) k->private_value; - bc = (struct skl_algo_data *)sb->dobj.private; - - if (bc->set_params == SKL_PARAM_SET) { - ret = skl_set_module_params(skl, - (u32 *)bc->params, bc->size, - bc->param_id, mconfig); - if (ret < 0) - return ret; - } - } - } - - return 0; -} - -/* - * some module param can set from user control and this is required as - * when module is initailzed. if module param is required in init it is - * identifed by set_param flag. if set_param flag is not set, then this - * parameter needs to set as part of module init. - */ -static int skl_tplg_set_module_init_data(struct snd_soc_dapm_widget *w) -{ - const struct snd_kcontrol_new *k; - struct soc_bytes_ext *sb; - struct skl_algo_data *bc; - struct skl_module_cfg *mconfig = w->priv; - int i; - - for (i = 0; i < w->num_kcontrols; i++) { - k = &w->kcontrol_news[i]; - if (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - sb = (struct soc_bytes_ext *)k->private_value; - bc = (struct skl_algo_data *)sb->dobj.private; - - if (bc->set_params != SKL_PARAM_INIT) - continue; - - mconfig->formats_config[SKL_PARAM_INIT].caps = - (u32 *)bc->params; - mconfig->formats_config[SKL_PARAM_INIT].caps_size = - bc->size; - - break; - } - } - - return 0; -} - -static int skl_tplg_module_prepare(struct skl_dev *skl, struct skl_pipe *pipe, - struct snd_soc_dapm_widget *w, struct skl_module_cfg *mcfg) -{ - switch (mcfg->dev_type) { - case SKL_DEVICE_HDAHOST: - return skl_pcm_host_dma_prepare(skl->dev, pipe->p_params); - - case SKL_DEVICE_HDALINK: - return skl_pcm_link_dma_prepare(skl->dev, pipe->p_params); - } - - return 0; -} - -/* - * Inside a pipe instance, we can have various modules. These modules need - * to instantiated in DSP by invoking INIT_MODULE IPC, which is achieved by - * skl_init_module() routine, so invoke that for all modules in a pipeline - */ -static int -skl_tplg_init_pipe_modules(struct skl_dev *skl, struct skl_pipe *pipe) -{ - struct skl_pipe_module *w_module; - struct snd_soc_dapm_widget *w; - struct skl_module_cfg *mconfig; - u8 cfg_idx; - int ret = 0; - - list_for_each_entry(w_module, &pipe->w_list, node) { - guid_t *uuid_mod; - w = w_module->w; - mconfig = w->priv; - - /* check if module ids are populated */ - if (mconfig->id.module_id < 0) { - dev_err(skl->dev, - "module %pUL id not populated\n", - (guid_t *)mconfig->guid); - return -EIO; - } - - cfg_idx = mconfig->pipe->cur_config_idx; - mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; - mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; - - if (mconfig->module->loadable && skl->dsp->fw_ops.load_mod) { - ret = skl->dsp->fw_ops.load_mod(skl->dsp, - mconfig->id.module_id, mconfig->guid); - if (ret < 0) - return ret; - } - - /* prepare the DMA if the module is gateway cpr */ - ret = skl_tplg_module_prepare(skl, pipe, w, mconfig); - if (ret < 0) - return ret; - - /* update blob if blob is null for be with default value */ - skl_tplg_update_be_blob(w, skl); - - /* - * apply fix/conversion to module params based on - * FE/BE params - */ - skl_tplg_update_module_params(w, skl); - uuid_mod = (guid_t *)mconfig->guid; - mconfig->id.pvt_id = skl_get_pvt_id(skl, uuid_mod, - mconfig->id.instance_id); - if (mconfig->id.pvt_id < 0) - return ret; - skl_tplg_set_module_init_data(w); - - ret = skl_dsp_get_core(skl->dsp, mconfig->core_id); - if (ret < 0) { - dev_err(skl->dev, "Failed to wake up core %d ret=%d\n", - mconfig->core_id, ret); - return ret; - } - - ret = skl_init_module(skl, mconfig); - if (ret < 0) { - skl_put_pvt_id(skl, uuid_mod, &mconfig->id.pvt_id); - goto err; - } - - ret = skl_tplg_set_module_params(w, skl); - if (ret < 0) - goto err; - } - - return 0; -err: - skl_dsp_put_core(skl->dsp, mconfig->core_id); - return ret; -} - -static int skl_tplg_unload_pipe_modules(struct skl_dev *skl, - struct skl_pipe *pipe) -{ - int ret = 0; - struct skl_pipe_module *w_module; - struct skl_module_cfg *mconfig; - - list_for_each_entry(w_module, &pipe->w_list, node) { - guid_t *uuid_mod; - mconfig = w_module->w->priv; - uuid_mod = (guid_t *)mconfig->guid; - - if (mconfig->module->loadable && skl->dsp->fw_ops.unload_mod) { - ret = skl->dsp->fw_ops.unload_mod(skl->dsp, - mconfig->id.module_id); - if (ret < 0) - return -EIO; - } - skl_put_pvt_id(skl, uuid_mod, &mconfig->id.pvt_id); - - ret = skl_dsp_put_core(skl->dsp, mconfig->core_id); - if (ret < 0) { - /* don't return; continue with other modules */ - dev_err(skl->dev, "Failed to sleep core %d ret=%d\n", - mconfig->core_id, ret); - } - } - - /* no modules to unload in this path, so return */ - return ret; -} - -static void skl_tplg_set_pipe_config_idx(struct skl_pipe *pipe, int idx) -{ - pipe->cur_config_idx = idx; - pipe->memory_pages = pipe->configs[idx].mem_pages; -} - -/* - * Here, we select pipe format based on the pipe type and pipe - * direction to determine the current config index for the pipeline. - * The config index is then used to select proper module resources. - * Intermediate pipes currently have a fixed format hence we select the - * 0th configuratation by default for such pipes. - */ -static int -skl_tplg_get_pipe_config(struct skl_dev *skl, struct skl_module_cfg *mconfig) -{ - struct skl_pipe *pipe = mconfig->pipe; - struct skl_pipe_params *params = pipe->p_params; - struct skl_path_config *pconfig = &pipe->configs[0]; - struct skl_pipe_fmt *fmt = NULL; - bool in_fmt = false; - int i; - - if (pipe->nr_cfgs == 0) { - skl_tplg_set_pipe_config_idx(pipe, 0); - return 0; - } - - if (pipe->conn_type == SKL_PIPE_CONN_TYPE_NONE || pipe->nr_cfgs == 1) { - dev_dbg(skl->dev, "No conn_type or just 1 pathcfg, taking 0th for %d\n", - pipe->ppl_id); - skl_tplg_set_pipe_config_idx(pipe, 0); - return 0; - } - - if ((pipe->conn_type == SKL_PIPE_CONN_TYPE_FE && - pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (pipe->conn_type == SKL_PIPE_CONN_TYPE_BE && - pipe->direction == SNDRV_PCM_STREAM_CAPTURE)) - in_fmt = true; - - for (i = 0; i < pipe->nr_cfgs; i++) { - pconfig = &pipe->configs[i]; - if (in_fmt) - fmt = &pconfig->in_fmt; - else - fmt = &pconfig->out_fmt; - - if (CHECK_HW_PARAMS(params->ch, params->s_freq, params->s_fmt, - fmt->channels, fmt->freq, fmt->bps)) { - skl_tplg_set_pipe_config_idx(pipe, i); - dev_dbg(skl->dev, "Using pipe config: %d\n", i); - return 0; - } - } - - dev_err(skl->dev, "Invalid pipe config: %d %d %d for pipe: %d\n", - params->ch, params->s_freq, params->s_fmt, pipe->ppl_id); - return -EINVAL; -} - -/* - * Mixer module represents a pipeline. So in the Pre-PMU event of mixer we - * need create the pipeline. So we do following: - * - Create the pipeline - * - Initialize the modules in pipeline - * - finally bind all modules together - */ -static int skl_tplg_mixer_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - int ret; - struct skl_module_cfg *mconfig = w->priv; - struct skl_pipe_module *w_module; - struct skl_pipe *s_pipe = mconfig->pipe; - struct skl_module_cfg *src_module = NULL, *dst_module, *module; - struct skl_module_deferred_bind *modules; - - ret = skl_tplg_get_pipe_config(skl, mconfig); - if (ret < 0) - return ret; - - /* - * Create a list of modules for pipe. - * This list contains modules from source to sink - */ - ret = skl_create_pipeline(skl, mconfig->pipe); - if (ret < 0) - return ret; - - /* Init all pipe modules from source to sink */ - ret = skl_tplg_init_pipe_modules(skl, s_pipe); - if (ret < 0) - return ret; - - /* Bind modules from source to sink */ - list_for_each_entry(w_module, &s_pipe->w_list, node) { - dst_module = w_module->w->priv; - - if (src_module == NULL) { - src_module = dst_module; - continue; - } - - ret = skl_bind_modules(skl, src_module, dst_module); - if (ret < 0) - return ret; - - src_module = dst_module; - } - - /* - * When the destination module is initialized, check for these modules - * in deferred bind list. If found, bind them. - */ - list_for_each_entry(w_module, &s_pipe->w_list, node) { - if (list_empty(&skl->bind_list)) - break; - - list_for_each_entry(modules, &skl->bind_list, node) { - module = w_module->w->priv; - if (modules->dst == module) - skl_bind_modules(skl, modules->src, - modules->dst); - } - } - - return 0; -} - -static int skl_fill_sink_instance_id(struct skl_dev *skl, u32 *params, - int size, struct skl_module_cfg *mcfg) -{ - int i, pvt_id; - - if (mcfg->m_type == SKL_MODULE_TYPE_KPB) { - struct skl_kpb_params *kpb_params = - (struct skl_kpb_params *)params; - struct skl_mod_inst_map *inst = kpb_params->u.map; - - for (i = 0; i < kpb_params->num_modules; i++) { - pvt_id = skl_get_pvt_instance_id_map(skl, inst->mod_id, - inst->inst_id); - if (pvt_id < 0) - return -EINVAL; - - inst->inst_id = pvt_id; - inst++; - } - } - - return 0; -} -/* - * Some modules require params to be set after the module is bound to - * all pins connected. - * - * The module provider initializes set_param flag for such modules and we - * send params after binding - */ -static int skl_tplg_set_module_bind_params(struct snd_soc_dapm_widget *w, - struct skl_module_cfg *mcfg, struct skl_dev *skl) -{ - int i, ret; - struct skl_module_cfg *mconfig = w->priv; - const struct snd_kcontrol_new *k; - struct soc_bytes_ext *sb; - struct skl_algo_data *bc; - struct skl_specific_cfg *sp_cfg; - u32 *params; - - /* - * check all out/in pins are in bind state. - * if so set the module param - */ - for (i = 0; i < mcfg->module->max_output_pins; i++) { - if (mcfg->m_out_pin[i].pin_state != SKL_PIN_BIND_DONE) - return 0; - } - - for (i = 0; i < mcfg->module->max_input_pins; i++) { - if (mcfg->m_in_pin[i].pin_state != SKL_PIN_BIND_DONE) - return 0; - } - - if (mconfig->formats_config[SKL_PARAM_BIND].caps_size > 0 && - mconfig->formats_config[SKL_PARAM_BIND].set_params == - SKL_PARAM_BIND) { - sp_cfg = &mconfig->formats_config[SKL_PARAM_BIND]; - ret = skl_set_module_params(skl, sp_cfg->caps, - sp_cfg->caps_size, - sp_cfg->param_id, mconfig); - if (ret < 0) - return ret; - } - - for (i = 0; i < w->num_kcontrols; i++) { - k = &w->kcontrol_news[i]; - if (k->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - sb = (void *) k->private_value; - bc = (struct skl_algo_data *)sb->dobj.private; - - if (bc->set_params == SKL_PARAM_BIND) { - params = kmemdup(bc->params, bc->max, GFP_KERNEL); - if (!params) - return -ENOMEM; - - skl_fill_sink_instance_id(skl, params, bc->max, - mconfig); - - ret = skl_set_module_params(skl, params, - bc->max, bc->param_id, mconfig); - kfree(params); - - if (ret < 0) - return ret; - } - } - } - - return 0; -} - -static int skl_get_module_id(struct skl_dev *skl, guid_t *uuid) -{ - struct uuid_module *module; - - list_for_each_entry(module, &skl->uuid_list, list) { - if (guid_equal(uuid, &module->uuid)) - return module->id; - } - - return -EINVAL; -} - -static int skl_tplg_find_moduleid_from_uuid(struct skl_dev *skl, - const struct snd_kcontrol_new *k) -{ - struct soc_bytes_ext *sb = (void *) k->private_value; - struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private; - struct skl_kpb_params *uuid_params, *params; - struct hdac_bus *bus = skl_to_bus(skl); - int i, size, module_id; - - if (bc->set_params == SKL_PARAM_BIND && bc->max) { - uuid_params = (struct skl_kpb_params *)bc->params; - size = struct_size(params, u.map, uuid_params->num_modules); - - params = devm_kzalloc(bus->dev, size, GFP_KERNEL); - if (!params) - return -ENOMEM; - - params->num_modules = uuid_params->num_modules; - - for (i = 0; i < uuid_params->num_modules; i++) { - module_id = skl_get_module_id(skl, - &uuid_params->u.map_uuid[i].mod_uuid); - if (module_id < 0) { - devm_kfree(bus->dev, params); - return -EINVAL; - } - - params->u.map[i].mod_id = module_id; - params->u.map[i].inst_id = - uuid_params->u.map_uuid[i].inst_id; - } - - devm_kfree(bus->dev, bc->params); - bc->params = (char *)params; - bc->max = size; - } - - return 0; -} - -/* - * Retrieve the module id from UUID mentioned in the - * post bind params - */ -void skl_tplg_add_moduleid_in_bind_params(struct skl_dev *skl, - struct snd_soc_dapm_widget *w) -{ - struct skl_module_cfg *mconfig = w->priv; - int i; - - /* - * Post bind params are used for only for KPB - * to set copier instances to drain the data - * in fast mode - */ - if (mconfig->m_type != SKL_MODULE_TYPE_KPB) - return; - - for (i = 0; i < w->num_kcontrols; i++) - if ((w->kcontrol_news[i].access & - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) && - (skl_tplg_find_moduleid_from_uuid(skl, - &w->kcontrol_news[i]) < 0)) - dev_err(skl->dev, - "%s: invalid kpb post bind params\n", - __func__); -} - -static int skl_tplg_module_add_deferred_bind(struct skl_dev *skl, - struct skl_module_cfg *src, struct skl_module_cfg *dst) -{ - struct skl_module_deferred_bind *m_list, *modules; - int i; - - /* only supported for module with static pin connection */ - for (i = 0; i < dst->module->max_input_pins; i++) { - struct skl_module_pin *pin = &dst->m_in_pin[i]; - - if (pin->is_dynamic) - continue; - - if ((pin->id.module_id == src->id.module_id) && - (pin->id.instance_id == src->id.instance_id)) { - - if (!list_empty(&skl->bind_list)) { - list_for_each_entry(modules, &skl->bind_list, node) { - if (modules->src == src && modules->dst == dst) - return 0; - } - } - - m_list = kzalloc(sizeof(*m_list), GFP_KERNEL); - if (!m_list) - return -ENOMEM; - - m_list->src = src; - m_list->dst = dst; - - list_add(&m_list->node, &skl->bind_list); - } - } - - return 0; -} - -static int skl_tplg_bind_sinks(struct snd_soc_dapm_widget *w, - struct skl_dev *skl, - struct snd_soc_dapm_widget *src_w, - struct skl_module_cfg *src_mconfig) -{ - struct snd_soc_dapm_path *p; - struct snd_soc_dapm_widget *sink = NULL, *next_sink = NULL; - struct skl_module_cfg *sink_mconfig; - int ret; - - snd_soc_dapm_widget_for_each_sink_path(w, p) { - if (!p->connect) - continue; - - dev_dbg(skl->dev, - "%s: src widget=%s\n", __func__, w->name); - dev_dbg(skl->dev, - "%s: sink widget=%s\n", __func__, p->sink->name); - - next_sink = p->sink; - - if (!is_skl_dsp_widget_type(p->sink, skl->dev)) - return skl_tplg_bind_sinks(p->sink, skl, src_w, src_mconfig); - - /* - * here we will check widgets in sink pipelines, so that - * can be any widgets type and we are only interested if - * they are ones used for SKL so check that first - */ - if ((p->sink->priv != NULL) && - is_skl_dsp_widget_type(p->sink, skl->dev)) { - - sink = p->sink; - sink_mconfig = sink->priv; - - /* - * Modules other than PGA leaf can be connected - * directly or via switch to a module in another - * pipeline. EX: reference path - * when the path is enabled, the dst module that needs - * to be bound may not be initialized. if the module is - * not initialized, add these modules in the deferred - * bind list and when the dst module is initialised, - * bind this module to the dst_module in deferred list. - */ - if (((src_mconfig->m_state == SKL_MODULE_INIT_DONE) - && (sink_mconfig->m_state == SKL_MODULE_UNINIT))) { - - ret = skl_tplg_module_add_deferred_bind(skl, - src_mconfig, sink_mconfig); - - if (ret < 0) - return ret; - - } - - - if (src_mconfig->m_state == SKL_MODULE_UNINIT || - sink_mconfig->m_state == SKL_MODULE_UNINIT) - continue; - - /* Bind source to sink, mixin is always source */ - ret = skl_bind_modules(skl, src_mconfig, sink_mconfig); - if (ret) - return ret; - - /* set module params after bind */ - skl_tplg_set_module_bind_params(src_w, - src_mconfig, skl); - skl_tplg_set_module_bind_params(sink, - sink_mconfig, skl); - - /* Start sinks pipe first */ - if (sink_mconfig->pipe->state != SKL_PIPE_STARTED) { - if (sink_mconfig->pipe->conn_type != - SKL_PIPE_CONN_TYPE_FE) - ret = skl_run_pipe(skl, - sink_mconfig->pipe); - if (ret) - return ret; - } - } - } - - if (!sink && next_sink) - return skl_tplg_bind_sinks(next_sink, skl, src_w, src_mconfig); - - return 0; -} - -/* - * A PGA represents a module in a pipeline. So in the Pre-PMU event of PGA - * we need to do following: - * - Bind to sink pipeline - * Since the sink pipes can be running and we don't get mixer event on - * connect for already running mixer, we need to find the sink pipes - * here and bind to them. This way dynamic connect works. - * - Start sink pipeline, if not running - * - Then run current pipe - */ -static int skl_tplg_pga_dapm_pre_pmu_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *src_mconfig; - int ret = 0; - - src_mconfig = w->priv; - - /* - * find which sink it is connected to, bind with the sink, - * if sink is not started, start sink pipe first, then start - * this pipe - */ - ret = skl_tplg_bind_sinks(w, skl, w, src_mconfig); - if (ret) - return ret; - - /* Start source pipe last after starting all sinks */ - if (src_mconfig->pipe->conn_type != SKL_PIPE_CONN_TYPE_FE) - return skl_run_pipe(skl, src_mconfig->pipe); - - return 0; -} - -static struct snd_soc_dapm_widget *skl_get_src_dsp_widget( - struct snd_soc_dapm_widget *w, struct skl_dev *skl) -{ - struct snd_soc_dapm_path *p; - struct snd_soc_dapm_widget *src_w = NULL; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - src_w = p->source; - if (!p->connect) - continue; - - dev_dbg(skl->dev, "sink widget=%s\n", w->name); - dev_dbg(skl->dev, "src widget=%s\n", p->source->name); - - /* - * here we will check widgets in sink pipelines, so that can - * be any widgets type and we are only interested if they are - * ones used for SKL so check that first - */ - if ((p->source->priv != NULL) && - is_skl_dsp_widget_type(p->source, skl->dev)) { - return p->source; - } - } - - if (src_w != NULL) - return skl_get_src_dsp_widget(src_w, skl); - - return NULL; -} - -/* - * in the Post-PMU event of mixer we need to do following: - * - Check if this pipe is running - * - if not, then - * - bind this pipeline to its source pipeline - * if source pipe is already running, this means it is a dynamic - * connection and we need to bind only to that pipe - * - start this pipeline - */ -static int skl_tplg_mixer_dapm_post_pmu_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - int ret = 0; - struct snd_soc_dapm_widget *source, *sink; - struct skl_module_cfg *src_mconfig, *sink_mconfig; - int src_pipe_started = 0; - - sink = w; - sink_mconfig = sink->priv; - - /* - * If source pipe is already started, that means source is driving - * one more sink before this sink got connected, Since source is - * started, bind this sink to source and start this pipe. - */ - source = skl_get_src_dsp_widget(w, skl); - if (source != NULL) { - src_mconfig = source->priv; - sink_mconfig = sink->priv; - src_pipe_started = 1; - - /* - * check pipe state, then no need to bind or start the - * pipe - */ - if (src_mconfig->pipe->state != SKL_PIPE_STARTED) - src_pipe_started = 0; - } - - if (src_pipe_started) { - ret = skl_bind_modules(skl, src_mconfig, sink_mconfig); - if (ret) - return ret; - - /* set module params after bind */ - skl_tplg_set_module_bind_params(source, src_mconfig, skl); - skl_tplg_set_module_bind_params(sink, sink_mconfig, skl); - - if (sink_mconfig->pipe->conn_type != SKL_PIPE_CONN_TYPE_FE) - ret = skl_run_pipe(skl, sink_mconfig->pipe); - } - - return ret; -} - -/* - * in the Pre-PMD event of mixer we need to do following: - * - Stop the pipe - * - find the source connections and remove that from dapm_path_list - * - unbind with source pipelines if still connected - */ -static int skl_tplg_mixer_dapm_pre_pmd_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *src_mconfig, *sink_mconfig; - int ret = 0, i; - - sink_mconfig = w->priv; - - /* Stop the pipe */ - ret = skl_stop_pipe(skl, sink_mconfig->pipe); - if (ret) - return ret; - - for (i = 0; i < sink_mconfig->module->max_input_pins; i++) { - if (sink_mconfig->m_in_pin[i].pin_state == SKL_PIN_BIND_DONE) { - src_mconfig = sink_mconfig->m_in_pin[i].tgt_mcfg; - if (!src_mconfig) - continue; - - ret = skl_unbind_modules(skl, - src_mconfig, sink_mconfig); - } - } - - return ret; -} - -/* - * in the Post-PMD event of mixer we need to do following: - * - Unbind the modules within the pipeline - * - Delete the pipeline (modules are not required to be explicitly - * deleted, pipeline delete is enough here - */ -static int skl_tplg_mixer_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *mconfig = w->priv; - struct skl_pipe_module *w_module; - struct skl_module_cfg *src_module = NULL, *dst_module; - struct skl_pipe *s_pipe = mconfig->pipe; - struct skl_module_deferred_bind *modules, *tmp; - - if (s_pipe->state == SKL_PIPE_INVALID) - return -EINVAL; - - list_for_each_entry(w_module, &s_pipe->w_list, node) { - if (list_empty(&skl->bind_list)) - break; - - src_module = w_module->w->priv; - - list_for_each_entry_safe(modules, tmp, &skl->bind_list, node) { - /* - * When the destination module is deleted, Unbind the - * modules from deferred bind list. - */ - if (modules->dst == src_module) { - skl_unbind_modules(skl, modules->src, - modules->dst); - } - - /* - * When the source module is deleted, remove this entry - * from the deferred bind list. - */ - if (modules->src == src_module) { - list_del(&modules->node); - modules->src = NULL; - modules->dst = NULL; - kfree(modules); - } - } - } - - list_for_each_entry(w_module, &s_pipe->w_list, node) { - dst_module = w_module->w->priv; - - if (src_module == NULL) { - src_module = dst_module; - continue; - } - - skl_unbind_modules(skl, src_module, dst_module); - src_module = dst_module; - } - - skl_delete_pipe(skl, mconfig->pipe); - - list_for_each_entry(w_module, &s_pipe->w_list, node) { - src_module = w_module->w->priv; - src_module->m_state = SKL_MODULE_UNINIT; - } - - return skl_tplg_unload_pipe_modules(skl, s_pipe); -} - -/* - * in the Post-PMD event of PGA we need to do following: - * - Stop the pipeline - * - In source pipe is connected, unbind with source pipelines - */ -static int skl_tplg_pga_dapm_post_pmd_event(struct snd_soc_dapm_widget *w, - struct skl_dev *skl) -{ - struct skl_module_cfg *src_mconfig, *sink_mconfig; - int ret = 0, i; - - src_mconfig = w->priv; - - /* Stop the pipe since this is a mixin module */ - ret = skl_stop_pipe(skl, src_mconfig->pipe); - if (ret) - return ret; - - for (i = 0; i < src_mconfig->module->max_output_pins; i++) { - if (src_mconfig->m_out_pin[i].pin_state == SKL_PIN_BIND_DONE) { - sink_mconfig = src_mconfig->m_out_pin[i].tgt_mcfg; - if (!sink_mconfig) - continue; - /* - * This is a connecter and if path is found that means - * unbind between source and sink has not happened yet - */ - ret = skl_unbind_modules(skl, src_mconfig, - sink_mconfig); - } - } - - return ret; -} - -/* - * In modelling, we assume there will be ONLY one mixer in a pipeline. If a - * second one is required that is created as another pipe entity. - * The mixer is responsible for pipe management and represent a pipeline - * instance - */ -static int skl_tplg_mixer_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct skl_dev *skl = get_skl_ctx(dapm->dev); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - return skl_tplg_mixer_dapm_pre_pmu_event(w, skl); - - case SND_SOC_DAPM_POST_PMU: - return skl_tplg_mixer_dapm_post_pmu_event(w, skl); - - case SND_SOC_DAPM_PRE_PMD: - return skl_tplg_mixer_dapm_pre_pmd_event(w, skl); - - case SND_SOC_DAPM_POST_PMD: - return skl_tplg_mixer_dapm_post_pmd_event(w, skl); - } - - return 0; -} - -/* - * In modelling, we assumed rest of the modules in pipeline are PGA. But we - * are interested in last PGA (leaf PGA) in a pipeline to disconnect with - * the sink when it is running (two FE to one BE or one FE to two BE) - * scenarios - */ -static int skl_tplg_pga_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) - -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct skl_dev *skl = get_skl_ctx(dapm->dev); - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - return skl_tplg_pga_dapm_pre_pmu_event(w, skl); - - case SND_SOC_DAPM_POST_PMD: - return skl_tplg_pga_dapm_post_pmd_event(w, skl); - } - - return 0; -} - -static int skl_tplg_multi_config_set_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol, - bool is_set) -{ - struct snd_soc_component *component = - snd_soc_kcontrol_component(kcontrol); - struct hdac_bus *bus = snd_soc_component_get_drvdata(component); - struct skl_dev *skl = bus_to_skl(bus); - struct skl_pipeline *ppl; - struct skl_pipe *pipe = NULL; - struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; - u32 *pipe_id; - - if (!ec) - return -EINVAL; - - if (is_set && ucontrol->value.enumerated.item[0] > ec->items) - return -EINVAL; - - pipe_id = ec->dobj.private; - - list_for_each_entry(ppl, &skl->ppl_list, node) { - if (ppl->pipe->ppl_id == *pipe_id) { - pipe = ppl->pipe; - break; - } - } - if (!pipe) - return -EIO; - - if (is_set) - skl_tplg_set_pipe_config_idx(pipe, ucontrol->value.enumerated.item[0]); - else - ucontrol->value.enumerated.item[0] = pipe->cur_config_idx; - - return 0; -} - -static int skl_tplg_multi_config_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false); -} - -static int skl_tplg_multi_config_set(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true); -} - -static int skl_tplg_multi_config_get_dmic(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - return skl_tplg_multi_config_set_get(kcontrol, ucontrol, false); -} - -static int skl_tplg_multi_config_set_dmic(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - return skl_tplg_multi_config_set_get(kcontrol, ucontrol, true); -} - -static int skl_tplg_tlv_control_get(struct snd_kcontrol *kcontrol, - unsigned int __user *data, unsigned int size) -{ - struct soc_bytes_ext *sb = - (struct soc_bytes_ext *)kcontrol->private_value; - struct skl_algo_data *bc = (struct skl_algo_data *)sb->dobj.private; - struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); - struct skl_module_cfg *mconfig = w->priv; - struct skl_dev *skl = get_skl_ctx(w->dapm->dev); - - if (w->power) - skl_get_module_params(skl, (u32 *)bc->params, - bc->size, bc->param_id, mconfig); - - /* decrement size for TLV header */ - size -= 2 * sizeof(u32); - - /* check size as we don't want to send kernel data */ - if (size > bc->max) - size = bc->max; - - if (bc->params) { - if (copy_to_user(data, &bc->param_id, sizeof(u32))) - return -EFAULT; - if (copy_to_user(data + 1, &size, sizeof(u32))) - return -EFAULT; - if (copy_to_user(data + 2, bc->params, size)) - return -EFAULT; - } - - return 0; -} - -#define SKL_PARAM_VENDOR_ID 0xff - -static int skl_tplg_tlv_control_set(struct snd_kcontrol *kcontrol, - const unsigned int __user *data, unsigned int size) -{ - struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); - struct skl_module_cfg *mconfig = w->priv; - struct soc_bytes_ext *sb = - (struct soc_bytes_ext *)kcontrol->private_value; - struct skl_algo_data *ac = (struct skl_algo_data *)sb->dobj.private; - struct skl_dev *skl = get_skl_ctx(w->dapm->dev); - - if (ac->params) { - if (size > ac->max) - return -EINVAL; - ac->size = size; - - if (copy_from_user(ac->params, data, size)) - return -EFAULT; - - if (w->power) - return skl_set_module_params(skl, - (u32 *)ac->params, ac->size, - ac->param_id, mconfig); - } - - return 0; -} - -static int skl_tplg_mic_control_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); - struct skl_module_cfg *mconfig = w->priv; - struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; - u32 ch_type = *((u32 *)ec->dobj.private); - - if (mconfig->dmic_ch_type == ch_type) - ucontrol->value.enumerated.item[0] = - mconfig->dmic_ch_combo_index; - else - ucontrol->value.enumerated.item[0] = 0; - - return 0; -} - -static int skl_fill_mic_sel_params(struct skl_module_cfg *mconfig, - struct skl_mic_sel_config *mic_cfg, struct device *dev) -{ - struct skl_specific_cfg *sp_cfg = - &mconfig->formats_config[SKL_PARAM_INIT]; - - sp_cfg->caps_size = sizeof(struct skl_mic_sel_config); - sp_cfg->set_params = SKL_PARAM_SET; - sp_cfg->param_id = 0x00; - if (!sp_cfg->caps) { - sp_cfg->caps = devm_kzalloc(dev, sp_cfg->caps_size, GFP_KERNEL); - if (!sp_cfg->caps) - return -ENOMEM; - } - - mic_cfg->mic_switch = SKL_MIC_SEL_SWITCH; - mic_cfg->flags = 0; - memcpy(sp_cfg->caps, mic_cfg, sp_cfg->caps_size); - - return 0; -} - -static int skl_tplg_mic_control_set(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); - struct skl_module_cfg *mconfig = w->priv; - struct skl_mic_sel_config mic_cfg = {0}; - struct soc_enum *ec = (struct soc_enum *)kcontrol->private_value; - u32 ch_type = *((u32 *)ec->dobj.private); - const int *list; - u8 in_ch, out_ch, index; - - mconfig->dmic_ch_type = ch_type; - mconfig->dmic_ch_combo_index = ucontrol->value.enumerated.item[0]; - - /* enum control index 0 is INVALID, so no channels to be set */ - if (mconfig->dmic_ch_combo_index == 0) - return 0; - - /* No valid channel selection map for index 0, so offset by 1 */ - index = mconfig->dmic_ch_combo_index - 1; - - switch (ch_type) { - case SKL_CH_MONO: - if (mconfig->dmic_ch_combo_index > ARRAY_SIZE(mic_mono_list)) - return -EINVAL; - - list = &mic_mono_list[index]; - break; - - case SKL_CH_STEREO: - if (mconfig->dmic_ch_combo_index > ARRAY_SIZE(mic_stereo_list)) - return -EINVAL; - - list = mic_stereo_list[index]; - break; - - case SKL_CH_TRIO: - if (mconfig->dmic_ch_combo_index > ARRAY_SIZE(mic_trio_list)) - return -EINVAL; - - list = mic_trio_list[index]; - break; - - case SKL_CH_QUATRO: - if (mconfig->dmic_ch_combo_index > ARRAY_SIZE(mic_quatro_list)) - return -EINVAL; - - list = mic_quatro_list[index]; - break; - - default: - dev_err(w->dapm->dev, - "Invalid channel %d for mic_select module\n", - ch_type); - return -EINVAL; - - } - - /* channel type enum map to number of chanels for that type */ - for (out_ch = 0; out_ch < ch_type; out_ch++) { - in_ch = list[out_ch]; - mic_cfg.blob[out_ch][in_ch] = SKL_DEFAULT_MIC_SEL_GAIN; - } - - return skl_fill_mic_sel_params(mconfig, &mic_cfg, w->dapm->dev); -} - -/* - * Fill the dma id for host and link. In case of passthrough - * pipeline, this will both host and link in the same - * pipeline, so need to copy the link and host based on dev_type - */ -static void skl_tplg_fill_dma_id(struct skl_module_cfg *mcfg, - struct skl_pipe_params *params) -{ - struct skl_pipe *pipe = mcfg->pipe; - - if (pipe->passthru) { - switch (mcfg->dev_type) { - case SKL_DEVICE_HDALINK: - pipe->p_params->link_dma_id = params->link_dma_id; - pipe->p_params->link_index = params->link_index; - pipe->p_params->link_bps = params->link_bps; - break; - - case SKL_DEVICE_HDAHOST: - pipe->p_params->host_dma_id = params->host_dma_id; - pipe->p_params->host_bps = params->host_bps; - break; - - default: - break; - } - pipe->p_params->s_fmt = params->s_fmt; - pipe->p_params->ch = params->ch; - pipe->p_params->s_freq = params->s_freq; - pipe->p_params->stream = params->stream; - pipe->p_params->format = params->format; - - } else { - memcpy(pipe->p_params, params, sizeof(*params)); - } -} - -/* - * The FE params are passed by hw_params of the DAI. - * On hw_params, the params are stored in Gateway module of the FE and we - * need to calculate the format in DSP module configuration, that - * conversion is done here - */ -int skl_tplg_update_pipe_params(struct device *dev, - struct skl_module_cfg *mconfig, - struct skl_pipe_params *params) -{ - struct skl_module_res *res; - struct skl_dev *skl = get_skl_ctx(dev); - struct skl_module_fmt *format = NULL; - u8 cfg_idx = mconfig->pipe->cur_config_idx; - - res = &mconfig->module->resources[mconfig->res_idx]; - skl_tplg_fill_dma_id(mconfig, params); - mconfig->fmt_idx = mconfig->mod_cfg[cfg_idx].fmt_idx; - mconfig->res_idx = mconfig->mod_cfg[cfg_idx].res_idx; - - if (skl->nr_modules) - return 0; - - if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) - format = &mconfig->module->formats[mconfig->fmt_idx].inputs[0].fmt; - else - format = &mconfig->module->formats[mconfig->fmt_idx].outputs[0].fmt; - - /* set the hw_params */ - format->s_freq = params->s_freq; - format->channels = params->ch; - format->valid_bit_depth = skl_get_bit_depth(params->s_fmt); - - /* - * 16 bit is 16 bit container whereas 24 bit is in 32 bit - * container so update bit depth accordingly - */ - switch (format->valid_bit_depth) { - case SKL_DEPTH_16BIT: - format->bit_depth = format->valid_bit_depth; - break; - - case SKL_DEPTH_24BIT: - case SKL_DEPTH_32BIT: - format->bit_depth = SKL_DEPTH_32BIT; - break; - - default: - dev_err(dev, "Invalid bit depth %x for pipe\n", - format->valid_bit_depth); - return -EINVAL; - } - - if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { - res->ibs = (format->s_freq / 1000) * - (format->channels) * - (format->bit_depth >> 3); - } else { - res->obs = (format->s_freq / 1000) * - (format->channels) * - (format->bit_depth >> 3); - } - - return 0; -} - -/* - * Query the module config for the FE DAI - * This is used to find the hw_params set for that DAI and apply to FE - * pipeline - */ -struct skl_module_cfg * -skl_tplg_fe_get_cpr_module(struct snd_soc_dai *dai, int stream) -{ - struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); - struct snd_soc_dapm_path *p = NULL; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - snd_soc_dapm_widget_for_each_sink_path(w, p) { - if (p->connect && p->sink->power && - !is_skl_dsp_widget_type(p->sink, dai->dev)) - continue; - - if (p->sink->priv) { - dev_dbg(dai->dev, "set params for %s\n", - p->sink->name); - return p->sink->priv; - } - } - } else { - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connect && p->source->power && - !is_skl_dsp_widget_type(p->source, dai->dev)) - continue; - - if (p->source->priv) { - dev_dbg(dai->dev, "set params for %s\n", - p->source->name); - return p->source->priv; - } - } - } - - return NULL; -} - -static struct skl_module_cfg *skl_get_mconfig_pb_cpr( - struct snd_soc_dai *dai, struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p; - struct skl_module_cfg *mconfig = NULL; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (w->endpoints[SND_SOC_DAPM_DIR_OUT] > 0) { - if (p->connect && - (p->sink->id == snd_soc_dapm_aif_out) && - p->source->priv) { - mconfig = p->source->priv; - return mconfig; - } - mconfig = skl_get_mconfig_pb_cpr(dai, p->source); - if (mconfig) - return mconfig; - } - } - return mconfig; -} - -static struct skl_module_cfg *skl_get_mconfig_cap_cpr( - struct snd_soc_dai *dai, struct snd_soc_dapm_widget *w) -{ - struct snd_soc_dapm_path *p; - struct skl_module_cfg *mconfig = NULL; - - snd_soc_dapm_widget_for_each_sink_path(w, p) { - if (w->endpoints[SND_SOC_DAPM_DIR_IN] > 0) { - if (p->connect && - (p->source->id == snd_soc_dapm_aif_in) && - p->sink->priv) { - mconfig = p->sink->priv; - return mconfig; - } - mconfig = skl_get_mconfig_cap_cpr(dai, p->sink); - if (mconfig) - return mconfig; - } - } - return mconfig; -} - -struct skl_module_cfg * -skl_tplg_be_get_cpr_module(struct snd_soc_dai *dai, int stream) -{ - struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, stream); - struct skl_module_cfg *mconfig; - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - mconfig = skl_get_mconfig_pb_cpr(dai, w); - } else { - mconfig = skl_get_mconfig_cap_cpr(dai, w); - } - return mconfig; -} - -static u8 skl_tplg_be_link_type(int dev_type) -{ - int ret; - - switch (dev_type) { - case SKL_DEVICE_BT: - ret = NHLT_LINK_SSP; - break; - - case SKL_DEVICE_DMIC: - ret = NHLT_LINK_DMIC; - break; - - case SKL_DEVICE_I2S: - ret = NHLT_LINK_SSP; - break; - - case SKL_DEVICE_HDALINK: - ret = NHLT_LINK_HDA; - break; - - default: - ret = NHLT_LINK_INVALID; - break; - } - - return ret; -} - -/* - * Fill the BE gateway parameters - * The BE gateway expects a blob of parameters which are kept in the ACPI - * NHLT blob, so query the blob for interface type (i2s/pdm) and instance. - * The port can have multiple settings so pick based on the pipeline - * parameters - */ -static int skl_tplg_be_fill_pipe_params(struct snd_soc_dai *dai, - struct skl_module_cfg *mconfig, - struct skl_pipe_params *params) -{ - struct nhlt_specific_cfg *cfg; - struct skl_pipe *pipe = mconfig->pipe; - struct skl_pipe_params save = *pipe->p_params; - struct skl_pipe_fmt *pipe_fmt; - struct skl_dev *skl = get_skl_ctx(dai->dev); - int link_type = skl_tplg_be_link_type(mconfig->dev_type); - u8 dev_type = skl_tplg_be_dev_type(mconfig->dev_type); - int ret; - - skl_tplg_fill_dma_id(mconfig, params); - - if (link_type == NHLT_LINK_HDA) - return 0; - - *pipe->p_params = *params; - ret = skl_tplg_get_pipe_config(skl, mconfig); - if (ret) - goto err; - - dev_dbg(skl->dev, "%s using pipe config: %d\n", __func__, pipe->cur_config_idx); - if (pipe->direction == SNDRV_PCM_STREAM_PLAYBACK) - pipe_fmt = &pipe->configs[pipe->cur_config_idx].out_fmt; - else - pipe_fmt = &pipe->configs[pipe->cur_config_idx].in_fmt; - - /* update the blob based on virtual bus_id*/ - cfg = intel_nhlt_get_endpoint_blob(dai->dev, skl->nhlt, - mconfig->vbus_id, link_type, - pipe_fmt->bps, params->s_cont, - pipe_fmt->channels, pipe_fmt->freq, - pipe->direction, dev_type); - if (cfg) { - mconfig->formats_config[SKL_PARAM_INIT].caps_size = cfg->size; - mconfig->formats_config[SKL_PARAM_INIT].caps = (u32 *)&cfg->caps; - } else { - dev_err(dai->dev, "Blob NULL for id:%d type:%d dirn:%d ch:%d, freq:%d, fmt:%d\n", - mconfig->vbus_id, link_type, params->stream, - params->ch, params->s_freq, params->s_fmt); - ret = -EINVAL; - goto err; - } - - return 0; - -err: - *pipe->p_params = save; - return ret; -} - -static int skl_tplg_be_set_src_pipe_params(struct snd_soc_dai *dai, - struct snd_soc_dapm_widget *w, - struct skl_pipe_params *params) -{ - struct snd_soc_dapm_path *p; - int ret = -EIO; - - snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connect && is_skl_dsp_widget_type(p->source, dai->dev) && - p->source->priv) { - - ret = skl_tplg_be_fill_pipe_params(dai, - p->source->priv, params); - if (ret < 0) - return ret; - } else { - ret = skl_tplg_be_set_src_pipe_params(dai, - p->source, params); - if (ret < 0) - return ret; - } - } - - return ret; -} - -static int skl_tplg_be_set_sink_pipe_params(struct snd_soc_dai *dai, - struct snd_soc_dapm_widget *w, struct skl_pipe_params *params) -{ - struct snd_soc_dapm_path *p; - int ret = -EIO; - - snd_soc_dapm_widget_for_each_sink_path(w, p) { - if (p->connect && is_skl_dsp_widget_type(p->sink, dai->dev) && - p->sink->priv) { - - ret = skl_tplg_be_fill_pipe_params(dai, - p->sink->priv, params); - if (ret < 0) - return ret; - } else { - ret = skl_tplg_be_set_sink_pipe_params( - dai, p->sink, params); - if (ret < 0) - return ret; - } - } - - return ret; -} - -/* - * BE hw_params can be a source parameters (capture) or sink parameters - * (playback). Based on sink and source we need to either find the source - * list or the sink list and set the pipeline parameters - */ -int skl_tplg_be_update_params(struct snd_soc_dai *dai, - struct skl_pipe_params *params) -{ - struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(dai, params->stream); - - if (params->stream == SNDRV_PCM_STREAM_PLAYBACK) { - return skl_tplg_be_set_src_pipe_params(dai, w, params); - } else { - return skl_tplg_be_set_sink_pipe_params(dai, w, params); - } -} - -static const struct snd_soc_tplg_widget_events skl_tplg_widget_ops[] = { - {SKL_MIXER_EVENT, skl_tplg_mixer_event}, - {SKL_VMIXER_EVENT, skl_tplg_mixer_event}, - {SKL_PGA_EVENT, skl_tplg_pga_event}, -}; - -static const struct snd_soc_tplg_bytes_ext_ops skl_tlv_ops[] = { - {SKL_CONTROL_TYPE_BYTE_TLV, skl_tplg_tlv_control_get, - skl_tplg_tlv_control_set}, -}; - -static const struct snd_soc_tplg_kcontrol_ops skl_tplg_kcontrol_ops[] = { - { - .id = SKL_CONTROL_TYPE_MIC_SELECT, - .get = skl_tplg_mic_control_get, - .put = skl_tplg_mic_control_set, - }, - { - .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT, - .get = skl_tplg_multi_config_get, - .put = skl_tplg_multi_config_set, - }, - { - .id = SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC, - .get = skl_tplg_multi_config_get_dmic, - .put = skl_tplg_multi_config_set_dmic, - } -}; - -static int skl_tplg_fill_pipe_cfg(struct device *dev, - struct skl_pipe *pipe, u32 tkn, - u32 tkn_val, int conf_idx, int dir) -{ - struct skl_pipe_fmt *fmt; - struct skl_path_config *config; - - switch (dir) { - case SKL_DIR_IN: - fmt = &pipe->configs[conf_idx].in_fmt; - break; - - case SKL_DIR_OUT: - fmt = &pipe->configs[conf_idx].out_fmt; - break; - - default: - dev_err(dev, "Invalid direction: %d\n", dir); - return -EINVAL; - } - - config = &pipe->configs[conf_idx]; - - switch (tkn) { - case SKL_TKN_U32_CFG_FREQ: - fmt->freq = tkn_val; - break; - - case SKL_TKN_U8_CFG_CHAN: - fmt->channels = tkn_val; - break; - - case SKL_TKN_U8_CFG_BPS: - fmt->bps = tkn_val; - break; - - case SKL_TKN_U32_PATH_MEM_PGS: - config->mem_pages = tkn_val; - break; - - default: - dev_err(dev, "Invalid token config: %d\n", tkn); - return -EINVAL; - } - - return 0; -} - -static int skl_tplg_fill_pipe_tkn(struct device *dev, - struct skl_pipe *pipe, u32 tkn, - u32 tkn_val) -{ - - switch (tkn) { - case SKL_TKN_U32_PIPE_CONN_TYPE: - pipe->conn_type = tkn_val; - break; - - case SKL_TKN_U32_PIPE_PRIORITY: - pipe->pipe_priority = tkn_val; - break; - - case SKL_TKN_U32_PIPE_MEM_PGS: - pipe->memory_pages = tkn_val; - break; - - case SKL_TKN_U32_PMODE: - pipe->lp_mode = tkn_val; - break; - - case SKL_TKN_U32_PIPE_DIRECTION: - pipe->direction = tkn_val; - break; - - case SKL_TKN_U32_NUM_CONFIGS: - pipe->nr_cfgs = tkn_val; - break; - - default: - dev_err(dev, "Token not handled %d\n", tkn); - return -EINVAL; - } - - return 0; -} - -/* - * Add pipeline by parsing the relevant tokens - * Return an existing pipe if the pipe already exists. - */ -static int skl_tplg_add_pipe(struct device *dev, - struct skl_module_cfg *mconfig, struct skl_dev *skl, - struct snd_soc_tplg_vendor_value_elem *tkn_elem) -{ - struct skl_pipeline *ppl; - struct skl_pipe *pipe; - struct skl_pipe_params *params; - - list_for_each_entry(ppl, &skl->ppl_list, node) { - if (ppl->pipe->ppl_id == tkn_elem->value) { - mconfig->pipe = ppl->pipe; - return -EEXIST; - } - } - - ppl = devm_kzalloc(dev, sizeof(*ppl), GFP_KERNEL); - if (!ppl) - return -ENOMEM; - - pipe = devm_kzalloc(dev, sizeof(*pipe), GFP_KERNEL); - if (!pipe) - return -ENOMEM; - - params = devm_kzalloc(dev, sizeof(*params), GFP_KERNEL); - if (!params) - return -ENOMEM; - - pipe->p_params = params; - pipe->ppl_id = tkn_elem->value; - INIT_LIST_HEAD(&pipe->w_list); - - ppl->pipe = pipe; - list_add(&ppl->node, &skl->ppl_list); - - mconfig->pipe = pipe; - mconfig->pipe->state = SKL_PIPE_INVALID; - - return 0; -} - -static int skl_tplg_get_uuid(struct device *dev, guid_t *guid, - struct snd_soc_tplg_vendor_uuid_elem *uuid_tkn) -{ - if (uuid_tkn->token == SKL_TKN_UUID) { - import_guid(guid, uuid_tkn->uuid); - return 0; - } - - dev_err(dev, "Not an UUID token %d\n", uuid_tkn->token); - - return -EINVAL; -} - -static int skl_tplg_fill_pin(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_module_pin *m_pin, - int pin_index) -{ - int ret; - - switch (tkn_elem->token) { - case SKL_TKN_U32_PIN_MOD_ID: - m_pin[pin_index].id.module_id = tkn_elem->value; - break; - - case SKL_TKN_U32_PIN_INST_ID: - m_pin[pin_index].id.instance_id = tkn_elem->value; - break; - - case SKL_TKN_UUID: - ret = skl_tplg_get_uuid(dev, &m_pin[pin_index].id.mod_uuid, - (struct snd_soc_tplg_vendor_uuid_elem *)tkn_elem); - if (ret < 0) - return ret; - - break; - - default: - dev_err(dev, "%d Not a pin token\n", tkn_elem->token); - return -EINVAL; - } - - return 0; -} - -/* - * Parse for pin config specific tokens to fill up the - * module private data - */ -static int skl_tplg_fill_pins_info(struct device *dev, - struct skl_module_cfg *mconfig, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - int dir, int pin_count) -{ - int ret; - struct skl_module_pin *m_pin; - - switch (dir) { - case SKL_DIR_IN: - m_pin = mconfig->m_in_pin; - break; - - case SKL_DIR_OUT: - m_pin = mconfig->m_out_pin; - break; - - default: - dev_err(dev, "Invalid direction value\n"); - return -EINVAL; - } - - ret = skl_tplg_fill_pin(dev, tkn_elem, m_pin, pin_count); - if (ret < 0) - return ret; - - m_pin[pin_count].in_use = false; - m_pin[pin_count].pin_state = SKL_PIN_UNBIND; - - return 0; -} - -/* - * Fill up input/output module config format based - * on the direction - */ -static int skl_tplg_fill_fmt(struct device *dev, - struct skl_module_fmt *dst_fmt, - u32 tkn, u32 value) -{ - switch (tkn) { - case SKL_TKN_U32_FMT_CH: - dst_fmt->channels = value; - break; - - case SKL_TKN_U32_FMT_FREQ: - dst_fmt->s_freq = value; - break; - - case SKL_TKN_U32_FMT_BIT_DEPTH: - dst_fmt->bit_depth = value; - break; - - case SKL_TKN_U32_FMT_SAMPLE_SIZE: - dst_fmt->valid_bit_depth = value; - break; - - case SKL_TKN_U32_FMT_CH_CONFIG: - dst_fmt->ch_cfg = value; - break; - - case SKL_TKN_U32_FMT_INTERLEAVE: - dst_fmt->interleaving_style = value; - break; - - case SKL_TKN_U32_FMT_SAMPLE_TYPE: - dst_fmt->sample_type = value; - break; - - case SKL_TKN_U32_FMT_CH_MAP: - dst_fmt->ch_map = value; - break; - - default: - dev_err(dev, "Invalid token %d\n", tkn); - return -EINVAL; - } - - return 0; -} - -static int skl_tplg_widget_fill_fmt(struct device *dev, - struct skl_module_iface *fmt, - u32 tkn, u32 val, u32 dir, int fmt_idx) -{ - struct skl_module_fmt *dst_fmt; - - if (!fmt) - return -EINVAL; - - switch (dir) { - case SKL_DIR_IN: - dst_fmt = &fmt->inputs[fmt_idx].fmt; - break; - - case SKL_DIR_OUT: - dst_fmt = &fmt->outputs[fmt_idx].fmt; - break; - - default: - dev_err(dev, "Invalid direction: %d\n", dir); - return -EINVAL; - } - - return skl_tplg_fill_fmt(dev, dst_fmt, tkn, val); -} - -static void skl_tplg_fill_pin_dynamic_val( - struct skl_module_pin *mpin, u32 pin_count, u32 value) -{ - int i; - - for (i = 0; i < pin_count; i++) - mpin[i].is_dynamic = value; -} - -/* - * Resource table in the manifest has pin specific resources - * like pin and pin buffer size - */ -static int skl_tplg_manifest_pin_res_tkn(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_module_res *res, int pin_idx, int dir) -{ - struct skl_module_pin_resources *m_pin; - - switch (dir) { - case SKL_DIR_IN: - m_pin = &res->input[pin_idx]; - break; - - case SKL_DIR_OUT: - m_pin = &res->output[pin_idx]; - break; - - default: - dev_err(dev, "Invalid pin direction: %d\n", dir); - return -EINVAL; - } - - switch (tkn_elem->token) { - case SKL_TKN_MM_U32_RES_PIN_ID: - m_pin->pin_index = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_PIN_BUF: - m_pin->buf_size = tkn_elem->value; - break; - - default: - dev_err(dev, "Invalid token: %d\n", tkn_elem->token); - return -EINVAL; - } - - return 0; -} - -/* - * Fill module specific resources from the manifest's resource - * table like CPS, DMA size, mem_pages. - */ -static int skl_tplg_fill_res_tkn(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_module_res *res, - int pin_idx, int dir) -{ - int ret, tkn_count = 0; - - if (!res) - return -EINVAL; - - switch (tkn_elem->token) { - case SKL_TKN_MM_U32_DMA_SIZE: - res->dma_buffer_size = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_CPC: - res->cpc = tkn_elem->value; - break; - - case SKL_TKN_U32_MEM_PAGES: - res->is_pages = tkn_elem->value; - break; - - case SKL_TKN_U32_OBS: - res->obs = tkn_elem->value; - break; - - case SKL_TKN_U32_IBS: - res->ibs = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_RES_PIN_ID: - case SKL_TKN_MM_U32_PIN_BUF: - ret = skl_tplg_manifest_pin_res_tkn(dev, tkn_elem, res, - pin_idx, dir); - if (ret < 0) - return ret; - break; - - case SKL_TKN_MM_U32_CPS: - case SKL_TKN_U32_MAX_MCPS: - /* ignore unused tokens */ - break; - - default: - dev_err(dev, "Not a res type token: %d", tkn_elem->token); - return -EINVAL; - - } - tkn_count++; - - return tkn_count; -} - -/* - * Parse tokens to fill up the module private data - */ -static int skl_tplg_get_token(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_dev *skl, struct skl_module_cfg *mconfig) -{ - int tkn_count = 0; - int ret; - static int is_pipe_exists; - static int pin_index, dir, conf_idx; - struct skl_module_iface *iface = NULL; - struct skl_module_res *res = NULL; - int res_idx = mconfig->res_idx; - int fmt_idx = mconfig->fmt_idx; - - /* - * If the manifest structure contains no modules, fill all - * the module data to 0th index. - * res_idx and fmt_idx are default set to 0. - */ - if (skl->nr_modules == 0) { - res = &mconfig->module->resources[res_idx]; - iface = &mconfig->module->formats[fmt_idx]; - } - - if (tkn_elem->token > SKL_TKN_MAX) - return -EINVAL; - - switch (tkn_elem->token) { - case SKL_TKN_U8_IN_QUEUE_COUNT: - mconfig->module->max_input_pins = tkn_elem->value; - break; - - case SKL_TKN_U8_OUT_QUEUE_COUNT: - mconfig->module->max_output_pins = tkn_elem->value; - break; - - case SKL_TKN_U8_DYN_IN_PIN: - if (!mconfig->m_in_pin) - mconfig->m_in_pin = - devm_kcalloc(dev, MAX_IN_QUEUE, - sizeof(*mconfig->m_in_pin), - GFP_KERNEL); - if (!mconfig->m_in_pin) - return -ENOMEM; - - skl_tplg_fill_pin_dynamic_val(mconfig->m_in_pin, MAX_IN_QUEUE, - tkn_elem->value); - break; - - case SKL_TKN_U8_DYN_OUT_PIN: - if (!mconfig->m_out_pin) - mconfig->m_out_pin = - devm_kcalloc(dev, MAX_IN_QUEUE, - sizeof(*mconfig->m_in_pin), - GFP_KERNEL); - if (!mconfig->m_out_pin) - return -ENOMEM; - - skl_tplg_fill_pin_dynamic_val(mconfig->m_out_pin, MAX_OUT_QUEUE, - tkn_elem->value); - break; - - case SKL_TKN_U8_TIME_SLOT: - mconfig->time_slot = tkn_elem->value; - break; - - case SKL_TKN_U8_CORE_ID: - mconfig->core_id = tkn_elem->value; - break; - - case SKL_TKN_U8_MOD_TYPE: - mconfig->m_type = tkn_elem->value; - break; - - case SKL_TKN_U8_DEV_TYPE: - mconfig->dev_type = tkn_elem->value; - break; - - case SKL_TKN_U8_HW_CONN_TYPE: - mconfig->hw_conn_type = tkn_elem->value; - break; - - case SKL_TKN_U16_MOD_INST_ID: - mconfig->id.instance_id = - tkn_elem->value; - break; - - case SKL_TKN_U32_MEM_PAGES: - case SKL_TKN_U32_MAX_MCPS: - case SKL_TKN_U32_OBS: - case SKL_TKN_U32_IBS: - ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_index, dir); - if (ret < 0) - return ret; - - break; - - case SKL_TKN_U32_VBUS_ID: - mconfig->vbus_id = tkn_elem->value; - break; - - case SKL_TKN_U32_PARAMS_FIXUP: - mconfig->params_fixup = tkn_elem->value; - break; - - case SKL_TKN_U32_CONVERTER: - mconfig->converter = tkn_elem->value; - break; - - case SKL_TKN_U32_D0I3_CAPS: - mconfig->d0i3_caps = tkn_elem->value; - break; - - case SKL_TKN_U32_PIPE_ID: - ret = skl_tplg_add_pipe(dev, - mconfig, skl, tkn_elem); - - if (ret < 0) { - if (ret == -EEXIST) { - is_pipe_exists = 1; - break; - } - return is_pipe_exists; - } - - break; - - case SKL_TKN_U32_PIPE_CONFIG_ID: - conf_idx = tkn_elem->value; - break; - - case SKL_TKN_U32_PIPE_CONN_TYPE: - case SKL_TKN_U32_PIPE_PRIORITY: - case SKL_TKN_U32_PIPE_MEM_PGS: - case SKL_TKN_U32_PMODE: - case SKL_TKN_U32_PIPE_DIRECTION: - case SKL_TKN_U32_NUM_CONFIGS: - if (is_pipe_exists) { - ret = skl_tplg_fill_pipe_tkn(dev, mconfig->pipe, - tkn_elem->token, tkn_elem->value); - if (ret < 0) - return ret; - } - - break; - - case SKL_TKN_U32_PATH_MEM_PGS: - case SKL_TKN_U32_CFG_FREQ: - case SKL_TKN_U8_CFG_CHAN: - case SKL_TKN_U8_CFG_BPS: - if (mconfig->pipe->nr_cfgs) { - ret = skl_tplg_fill_pipe_cfg(dev, mconfig->pipe, - tkn_elem->token, tkn_elem->value, - conf_idx, dir); - if (ret < 0) - return ret; - } - break; - - case SKL_TKN_CFG_MOD_RES_ID: - mconfig->mod_cfg[conf_idx].res_idx = tkn_elem->value; - break; - - case SKL_TKN_CFG_MOD_FMT_ID: - mconfig->mod_cfg[conf_idx].fmt_idx = tkn_elem->value; - break; - - /* - * SKL_TKN_U32_DIR_PIN_COUNT token has the value for both - * direction and the pin count. The first four bits represent - * direction and next four the pin count. - */ - case SKL_TKN_U32_DIR_PIN_COUNT: - dir = tkn_elem->value & SKL_IN_DIR_BIT_MASK; - pin_index = (tkn_elem->value & - SKL_PIN_COUNT_MASK) >> 4; - - break; - - case SKL_TKN_U32_FMT_CH: - case SKL_TKN_U32_FMT_FREQ: - case SKL_TKN_U32_FMT_BIT_DEPTH: - case SKL_TKN_U32_FMT_SAMPLE_SIZE: - case SKL_TKN_U32_FMT_CH_CONFIG: - case SKL_TKN_U32_FMT_INTERLEAVE: - case SKL_TKN_U32_FMT_SAMPLE_TYPE: - case SKL_TKN_U32_FMT_CH_MAP: - ret = skl_tplg_widget_fill_fmt(dev, iface, tkn_elem->token, - tkn_elem->value, dir, pin_index); - - if (ret < 0) - return ret; - - break; - - case SKL_TKN_U32_PIN_MOD_ID: - case SKL_TKN_U32_PIN_INST_ID: - case SKL_TKN_UUID: - ret = skl_tplg_fill_pins_info(dev, - mconfig, tkn_elem, dir, - pin_index); - if (ret < 0) - return ret; - - break; - - case SKL_TKN_U32_FMT_CFG_IDX: - if (tkn_elem->value > SKL_MAX_PARAMS_TYPES) - return -EINVAL; - - mconfig->fmt_cfg_idx = tkn_elem->value; - break; - - case SKL_TKN_U32_CAPS_SIZE: - mconfig->formats_config[mconfig->fmt_cfg_idx].caps_size = - tkn_elem->value; - - break; - - case SKL_TKN_U32_CAPS_SET_PARAMS: - mconfig->formats_config[mconfig->fmt_cfg_idx].set_params = - tkn_elem->value; - break; - - case SKL_TKN_U32_CAPS_PARAMS_ID: - mconfig->formats_config[mconfig->fmt_cfg_idx].param_id = - tkn_elem->value; - break; - - case SKL_TKN_U32_PROC_DOMAIN: - mconfig->domain = - tkn_elem->value; - - break; - - case SKL_TKN_U32_DMA_BUF_SIZE: - mconfig->dma_buffer_size = tkn_elem->value; - break; - - case SKL_TKN_U8_IN_PIN_TYPE: - case SKL_TKN_U8_OUT_PIN_TYPE: - case SKL_TKN_U8_CONN_TYPE: - break; - - default: - dev_err(dev, "Token %d not handled\n", - tkn_elem->token); - return -EINVAL; - } - - tkn_count++; - - return tkn_count; -} - -/* - * Parse the vendor array for specific tokens to construct - * module private data - */ -static int skl_tplg_get_tokens(struct device *dev, - char *pvt_data, struct skl_dev *skl, - struct skl_module_cfg *mconfig, int block_size) -{ - struct snd_soc_tplg_vendor_array *array; - struct snd_soc_tplg_vendor_value_elem *tkn_elem; - int tkn_count = 0, ret; - int off = 0, tuple_size = 0; - bool is_module_guid = true; - - if (block_size <= 0) - return -EINVAL; - - while (tuple_size < block_size) { - array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); - - off += array->size; - - switch (array->type) { - case SND_SOC_TPLG_TUPLE_TYPE_STRING: - dev_warn(dev, "no string tokens expected for skl tplg\n"); - continue; - - case SND_SOC_TPLG_TUPLE_TYPE_UUID: - if (is_module_guid) { - ret = skl_tplg_get_uuid(dev, (guid_t *)mconfig->guid, - array->uuid); - is_module_guid = false; - } else { - ret = skl_tplg_get_token(dev, array->value, skl, - mconfig); - } - - if (ret < 0) - return ret; - - tuple_size += sizeof(*array->uuid); - - continue; - - default: - tkn_elem = array->value; - tkn_count = 0; - break; - } - - while (tkn_count <= (array->num_elems - 1)) { - ret = skl_tplg_get_token(dev, tkn_elem, - skl, mconfig); - - if (ret < 0) - return ret; - - tkn_count = tkn_count + ret; - tkn_elem++; - } - - tuple_size += tkn_count * sizeof(*tkn_elem); - } - - return off; -} - -/* - * Every data block is preceded by a descriptor to read the number - * of data blocks, they type of the block and it's size - */ -static int skl_tplg_get_desc_blocks(struct device *dev, - struct snd_soc_tplg_vendor_array *array) -{ - struct snd_soc_tplg_vendor_value_elem *tkn_elem; - - tkn_elem = array->value; - - switch (tkn_elem->token) { - case SKL_TKN_U8_NUM_BLOCKS: - case SKL_TKN_U8_BLOCK_TYPE: - case SKL_TKN_U16_BLOCK_SIZE: - return tkn_elem->value; - - default: - dev_err(dev, "Invalid descriptor token %d\n", tkn_elem->token); - break; - } - - return -EINVAL; -} - -static int skl_tplg_get_caps_data(struct device *dev, char *data, - struct skl_module_cfg *mconfig) -{ - int idx = mconfig->fmt_cfg_idx; - - if (mconfig->formats_config[idx].caps_size > 0) { - mconfig->formats_config[idx].caps = - devm_kzalloc(dev, mconfig->formats_config[idx].caps_size, - GFP_KERNEL); - if (!mconfig->formats_config[idx].caps) - return -ENOMEM; - memcpy(mconfig->formats_config[idx].caps, data, - mconfig->formats_config[idx].caps_size); - } - - return mconfig->formats_config[idx].caps_size; -} - -/* - * Parse the private data for the token and corresponding value. - * The private data can have multiple data blocks. So, a data block - * is preceded by a descriptor for number of blocks and a descriptor - * for the type and size of the suceeding data block. - */ -static int skl_tplg_get_pvt_data(struct snd_soc_tplg_dapm_widget *tplg_w, - struct skl_dev *skl, struct device *dev, - struct skl_module_cfg *mconfig) -{ - struct snd_soc_tplg_vendor_array *array; - int num_blocks, block_size, block_type, off = 0; - char *data; - int ret; - - /* Read the NUM_DATA_BLOCKS descriptor */ - array = (struct snd_soc_tplg_vendor_array *)tplg_w->priv.data; - ret = skl_tplg_get_desc_blocks(dev, array); - if (ret < 0) - return ret; - num_blocks = ret; - - off += array->size; - /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ - while (num_blocks > 0) { - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - - ret = skl_tplg_get_desc_blocks(dev, array); - - if (ret < 0) - return ret; - block_type = ret; - off += array->size; - - array = (struct snd_soc_tplg_vendor_array *) - (tplg_w->priv.data + off); - - ret = skl_tplg_get_desc_blocks(dev, array); - - if (ret < 0) - return ret; - block_size = ret; - off += array->size; - - data = (tplg_w->priv.data + off); - - if (block_type == SKL_TYPE_TUPLE) { - ret = skl_tplg_get_tokens(dev, data, - skl, mconfig, block_size); - } else { - ret = skl_tplg_get_caps_data(dev, data, mconfig); - } - - if (ret < 0) - return ret; - - --num_blocks; - off += ret; - } - - return 0; -} - -static void skl_clear_pin_config(struct snd_soc_component *component, - struct snd_soc_dapm_widget *w) -{ - int i; - struct skl_module_cfg *mconfig; - struct skl_pipe *pipe; - - if (!strncmp(w->dapm->component->name, component->name, - strlen(component->name))) { - mconfig = w->priv; - pipe = mconfig->pipe; - for (i = 0; i < mconfig->module->max_input_pins; i++) { - mconfig->m_in_pin[i].in_use = false; - mconfig->m_in_pin[i].pin_state = SKL_PIN_UNBIND; - } - for (i = 0; i < mconfig->module->max_output_pins; i++) { - mconfig->m_out_pin[i].in_use = false; - mconfig->m_out_pin[i].pin_state = SKL_PIN_UNBIND; - } - pipe->state = SKL_PIPE_INVALID; - mconfig->m_state = SKL_MODULE_UNINIT; - } -} - -void skl_cleanup_resources(struct skl_dev *skl) -{ - struct snd_soc_component *soc_component = skl->component; - struct snd_soc_dapm_widget *w; - struct snd_soc_card *card; - - if (soc_component == NULL) - return; - - card = soc_component->card; - if (!snd_soc_card_is_instantiated(card)) - return; - - list_for_each_entry(w, &card->widgets, list) { - if (is_skl_dsp_widget_type(w, skl->dev) && w->priv != NULL) - skl_clear_pin_config(soc_component, w); - } - - skl_clear_module_cnt(skl->dsp); -} - -/* - * Topology core widget load callback - * - * This is used to save the private data for each widget which gives - * information to the driver about module and pipeline parameters which DSP - * FW expects like ids, resource values, formats etc - */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, - struct snd_soc_dapm_widget *w, - struct snd_soc_tplg_dapm_widget *tplg_w) -{ - int ret; - struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); - struct skl_dev *skl = bus_to_skl(bus); - struct skl_module_cfg *mconfig; - - if (!tplg_w->priv.size) - goto bind_event; - - mconfig = devm_kzalloc(bus->dev, sizeof(*mconfig), GFP_KERNEL); - - if (!mconfig) - return -ENOMEM; - - if (skl->nr_modules == 0) { - mconfig->module = devm_kzalloc(bus->dev, - sizeof(*mconfig->module), GFP_KERNEL); - if (!mconfig->module) - return -ENOMEM; - } - - w->priv = mconfig; - - /* - * module binary can be loaded later, so set it to query when - * module is load for a use case - */ - mconfig->id.module_id = -1; - - /* To provide backward compatibility, set default as SKL_PARAM_INIT */ - mconfig->fmt_cfg_idx = SKL_PARAM_INIT; - - /* Parse private data for tuples */ - ret = skl_tplg_get_pvt_data(tplg_w, skl, bus->dev, mconfig); - if (ret < 0) - return ret; - - skl_debug_init_module(skl->debugfs, w, mconfig); - -bind_event: - if (tplg_w->event_type == 0) { - dev_dbg(bus->dev, "ASoC: No event handler required\n"); - return 0; - } - - ret = snd_soc_tplg_widget_bind_event(w, skl_tplg_widget_ops, - ARRAY_SIZE(skl_tplg_widget_ops), - tplg_w->event_type); - - if (ret) { - dev_err(bus->dev, "%s: No matching event handlers found for %d\n", - __func__, tplg_w->event_type); - return -EINVAL; - } - - return 0; -} - -static int skl_init_algo_data(struct device *dev, struct soc_bytes_ext *be, - struct snd_soc_tplg_bytes_control *bc) -{ - struct skl_algo_data *ac; - struct skl_dfw_algo_data *dfw_ac = - (struct skl_dfw_algo_data *)bc->priv.data; - - ac = devm_kzalloc(dev, sizeof(*ac), GFP_KERNEL); - if (!ac) - return -ENOMEM; - - /* Fill private data */ - ac->max = dfw_ac->max; - ac->param_id = dfw_ac->param_id; - ac->set_params = dfw_ac->set_params; - ac->size = dfw_ac->max; - - if (ac->max) { - ac->params = devm_kzalloc(dev, ac->max, GFP_KERNEL); - if (!ac->params) - return -ENOMEM; - - memcpy(ac->params, dfw_ac->params, ac->max); - } - - be->dobj.private = ac; - return 0; -} - -static int skl_init_enum_data(struct device *dev, struct soc_enum *se, - struct snd_soc_tplg_enum_control *ec) -{ - - void *data; - - if (ec->priv.size) { - data = devm_kzalloc(dev, sizeof(ec->priv.size), GFP_KERNEL); - if (!data) - return -ENOMEM; - memcpy(data, ec->priv.data, ec->priv.size); - se->dobj.private = data; - } - - return 0; - -} - -static int skl_tplg_control_load(struct snd_soc_component *cmpnt, - int index, - struct snd_kcontrol_new *kctl, - struct snd_soc_tplg_ctl_hdr *hdr) -{ - struct soc_bytes_ext *sb; - struct snd_soc_tplg_bytes_control *tplg_bc; - struct snd_soc_tplg_enum_control *tplg_ec; - struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); - struct soc_enum *se; - - switch (hdr->ops.info) { - case SND_SOC_TPLG_CTL_BYTES: - tplg_bc = container_of(hdr, - struct snd_soc_tplg_bytes_control, hdr); - if (kctl->access & SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK) { - sb = (struct soc_bytes_ext *)kctl->private_value; - if (tplg_bc->priv.size) - return skl_init_algo_data( - bus->dev, sb, tplg_bc); - } - break; - - case SND_SOC_TPLG_CTL_ENUM: - tplg_ec = container_of(hdr, - struct snd_soc_tplg_enum_control, hdr); - if (kctl->access & SNDRV_CTL_ELEM_ACCESS_READ) { - se = (struct soc_enum *)kctl->private_value; - if (tplg_ec->priv.size) - skl_init_enum_data(bus->dev, se, tplg_ec); - } - - /* - * now that the control initializations are done, remove - * write permission for the DMIC configuration enums to - * avoid conflicts between NHLT settings and user interaction - */ - - if (hdr->ops.get == SKL_CONTROL_TYPE_MULTI_IO_SELECT_DMIC) - kctl->access = SNDRV_CTL_ELEM_ACCESS_READ; - - break; - - default: - dev_dbg(bus->dev, "Control load not supported %d:%d:%d\n", - hdr->ops.get, hdr->ops.put, hdr->ops.info); - break; - } - - return 0; -} - -static int skl_tplg_fill_str_mfest_tkn(struct device *dev, - struct snd_soc_tplg_vendor_string_elem *str_elem, - struct skl_dev *skl) -{ - int tkn_count = 0; - static int ref_count; - - switch (str_elem->token) { - case SKL_TKN_STR_LIB_NAME: - if (ref_count > skl->lib_count - 1) { - ref_count = 0; - return -EINVAL; - } - - strncpy(skl->lib_info[ref_count].name, - str_elem->string, - ARRAY_SIZE(skl->lib_info[ref_count].name)); - ref_count++; - break; - - default: - dev_err(dev, "Not a string token %d\n", str_elem->token); - break; - } - tkn_count++; - - return tkn_count; -} - -static int skl_tplg_get_str_tkn(struct device *dev, - struct snd_soc_tplg_vendor_array *array, - struct skl_dev *skl) -{ - int tkn_count = 0, ret; - struct snd_soc_tplg_vendor_string_elem *str_elem; - - str_elem = (struct snd_soc_tplg_vendor_string_elem *)array->value; - while (tkn_count < array->num_elems) { - ret = skl_tplg_fill_str_mfest_tkn(dev, str_elem, skl); - str_elem++; - - if (ret < 0) - return ret; - - tkn_count = tkn_count + ret; - } - - return tkn_count; -} - -static int skl_tplg_manifest_fill_fmt(struct device *dev, - struct skl_module_iface *fmt, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - u32 dir, int fmt_idx) -{ - struct skl_module_pin_fmt *dst_fmt; - struct skl_module_fmt *mod_fmt; - int ret; - - if (!fmt) - return -EINVAL; - - switch (dir) { - case SKL_DIR_IN: - dst_fmt = &fmt->inputs[fmt_idx]; - break; - - case SKL_DIR_OUT: - dst_fmt = &fmt->outputs[fmt_idx]; - break; - - default: - dev_err(dev, "Invalid direction: %d\n", dir); - return -EINVAL; - } - - mod_fmt = &dst_fmt->fmt; - - switch (tkn_elem->token) { - case SKL_TKN_MM_U32_INTF_PIN_ID: - dst_fmt->id = tkn_elem->value; - break; - - default: - ret = skl_tplg_fill_fmt(dev, mod_fmt, tkn_elem->token, - tkn_elem->value); - if (ret < 0) - return ret; - break; - } - - return 0; -} - -static int skl_tplg_fill_mod_info(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_module *mod) -{ - - if (!mod) - return -EINVAL; - - switch (tkn_elem->token) { - case SKL_TKN_U8_IN_PIN_TYPE: - mod->input_pin_type = tkn_elem->value; - break; - - case SKL_TKN_U8_OUT_PIN_TYPE: - mod->output_pin_type = tkn_elem->value; - break; - - case SKL_TKN_U8_IN_QUEUE_COUNT: - mod->max_input_pins = tkn_elem->value; - break; - - case SKL_TKN_U8_OUT_QUEUE_COUNT: - mod->max_output_pins = tkn_elem->value; - break; - - case SKL_TKN_MM_U8_NUM_RES: - mod->nr_resources = tkn_elem->value; - break; - - case SKL_TKN_MM_U8_NUM_INTF: - mod->nr_interfaces = tkn_elem->value; - break; - - default: - dev_err(dev, "Invalid mod info token %d", tkn_elem->token); - return -EINVAL; - } - - return 0; -} - - -static int skl_tplg_get_int_tkn(struct device *dev, - struct snd_soc_tplg_vendor_value_elem *tkn_elem, - struct skl_dev *skl) -{ - int tkn_count = 0, ret; - static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx; - struct skl_module_res *res = NULL; - struct skl_module_iface *fmt = NULL; - struct skl_module *mod = NULL; - static struct skl_astate_param *astate_table; - static int astate_cfg_idx, count; - int i; - size_t size; - - if (skl->modules) { - mod = skl->modules[mod_idx]; - res = &mod->resources[res_val_idx]; - fmt = &mod->formats[intf_val_idx]; - } - - switch (tkn_elem->token) { - case SKL_TKN_U32_LIB_COUNT: - skl->lib_count = tkn_elem->value; - break; - - case SKL_TKN_U8_NUM_MOD: - skl->nr_modules = tkn_elem->value; - skl->modules = devm_kcalloc(dev, skl->nr_modules, - sizeof(*skl->modules), GFP_KERNEL); - if (!skl->modules) - return -ENOMEM; - - for (i = 0; i < skl->nr_modules; i++) { - skl->modules[i] = devm_kzalloc(dev, - sizeof(struct skl_module), GFP_KERNEL); - if (!skl->modules[i]) - return -ENOMEM; - } - break; - - case SKL_TKN_MM_U8_MOD_IDX: - mod_idx = tkn_elem->value; - break; - - case SKL_TKN_U32_ASTATE_COUNT: - if (astate_table != NULL) { - dev_err(dev, "More than one entry for A-State count"); - return -EINVAL; - } - - if (tkn_elem->value > SKL_MAX_ASTATE_CFG) { - dev_err(dev, "Invalid A-State count %d\n", - tkn_elem->value); - return -EINVAL; - } - - size = struct_size(skl->cfg.astate_cfg, astate_table, - tkn_elem->value); - skl->cfg.astate_cfg = devm_kzalloc(dev, size, GFP_KERNEL); - if (!skl->cfg.astate_cfg) - return -ENOMEM; - - astate_table = skl->cfg.astate_cfg->astate_table; - count = skl->cfg.astate_cfg->count = tkn_elem->value; - break; - - case SKL_TKN_U32_ASTATE_IDX: - if (tkn_elem->value >= count) { - dev_err(dev, "Invalid A-State index %d\n", - tkn_elem->value); - return -EINVAL; - } - - astate_cfg_idx = tkn_elem->value; - break; - - case SKL_TKN_U32_ASTATE_KCPS: - astate_table[astate_cfg_idx].kcps = tkn_elem->value; - break; - - case SKL_TKN_U32_ASTATE_CLK_SRC: - astate_table[astate_cfg_idx].clk_src = tkn_elem->value; - break; - - case SKL_TKN_U8_IN_PIN_TYPE: - case SKL_TKN_U8_OUT_PIN_TYPE: - case SKL_TKN_U8_IN_QUEUE_COUNT: - case SKL_TKN_U8_OUT_QUEUE_COUNT: - case SKL_TKN_MM_U8_NUM_RES: - case SKL_TKN_MM_U8_NUM_INTF: - ret = skl_tplg_fill_mod_info(dev, tkn_elem, mod); - if (ret < 0) - return ret; - break; - - case SKL_TKN_U32_DIR_PIN_COUNT: - dir = tkn_elem->value & SKL_IN_DIR_BIT_MASK; - pin_idx = (tkn_elem->value & SKL_PIN_COUNT_MASK) >> 4; - break; - - case SKL_TKN_MM_U32_RES_ID: - if (!res) - return -EINVAL; - - res->id = tkn_elem->value; - res_val_idx = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_FMT_ID: - if (!fmt) - return -EINVAL; - - fmt->fmt_idx = tkn_elem->value; - intf_val_idx = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_CPS: - case SKL_TKN_MM_U32_DMA_SIZE: - case SKL_TKN_MM_U32_CPC: - case SKL_TKN_U32_MEM_PAGES: - case SKL_TKN_U32_OBS: - case SKL_TKN_U32_IBS: - case SKL_TKN_MM_U32_RES_PIN_ID: - case SKL_TKN_MM_U32_PIN_BUF: - ret = skl_tplg_fill_res_tkn(dev, tkn_elem, res, pin_idx, dir); - if (ret < 0) - return ret; - - break; - - case SKL_TKN_MM_U32_NUM_IN_FMT: - if (!fmt) - return -EINVAL; - - res->nr_input_pins = tkn_elem->value; - break; - - case SKL_TKN_MM_U32_NUM_OUT_FMT: - if (!fmt) - return -EINVAL; - - res->nr_output_pins = tkn_elem->value; - break; - - case SKL_TKN_U32_FMT_CH: - case SKL_TKN_U32_FMT_FREQ: - case SKL_TKN_U32_FMT_BIT_DEPTH: - case SKL_TKN_U32_FMT_SAMPLE_SIZE: - case SKL_TKN_U32_FMT_CH_CONFIG: - case SKL_TKN_U32_FMT_INTERLEAVE: - case SKL_TKN_U32_FMT_SAMPLE_TYPE: - case SKL_TKN_U32_FMT_CH_MAP: - case SKL_TKN_MM_U32_INTF_PIN_ID: - ret = skl_tplg_manifest_fill_fmt(dev, fmt, tkn_elem, - dir, pin_idx); - if (ret < 0) - return ret; - break; - - default: - dev_err(dev, "Not a manifest token %d\n", tkn_elem->token); - return -EINVAL; - } - tkn_count++; - - return tkn_count; -} - -/* - * Fill the manifest structure by parsing the tokens based on the - * type. - */ -static int skl_tplg_get_manifest_tkn(struct device *dev, - char *pvt_data, struct skl_dev *skl, - int block_size) -{ - int tkn_count = 0, ret; - int off = 0, tuple_size = 0; - u8 uuid_index = 0; - struct snd_soc_tplg_vendor_array *array; - struct snd_soc_tplg_vendor_value_elem *tkn_elem; - - if (block_size <= 0) - return -EINVAL; - - while (tuple_size < block_size) { - array = (struct snd_soc_tplg_vendor_array *)(pvt_data + off); - off += array->size; - switch (array->type) { - case SND_SOC_TPLG_TUPLE_TYPE_STRING: - ret = skl_tplg_get_str_tkn(dev, array, skl); - - if (ret < 0) - return ret; - tkn_count = ret; - - tuple_size += tkn_count * - sizeof(struct snd_soc_tplg_vendor_string_elem); - continue; - - case SND_SOC_TPLG_TUPLE_TYPE_UUID: - if (array->uuid->token != SKL_TKN_UUID) { - dev_err(dev, "Not an UUID token: %d\n", - array->uuid->token); - return -EINVAL; - } - if (uuid_index >= skl->nr_modules) { - dev_err(dev, "Too many UUID tokens\n"); - return -EINVAL; - } - import_guid(&skl->modules[uuid_index++]->uuid, - array->uuid->uuid); - - tuple_size += sizeof(*array->uuid); - continue; - - default: - tkn_elem = array->value; - tkn_count = 0; - break; - } - - while (tkn_count <= array->num_elems - 1) { - ret = skl_tplg_get_int_tkn(dev, - tkn_elem, skl); - if (ret < 0) - return ret; - - tkn_count = tkn_count + ret; - tkn_elem++; - } - tuple_size += (tkn_count * sizeof(*tkn_elem)); - tkn_count = 0; - } - - return off; -} - -/* - * Parse manifest private data for tokens. The private data block is - * preceded by descriptors for type and size of data block. - */ -static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, - struct device *dev, struct skl_dev *skl) -{ - struct snd_soc_tplg_vendor_array *array; - int num_blocks, block_size = 0, block_type, off = 0; - char *data; - int ret; - - /* Read the NUM_DATA_BLOCKS descriptor */ - array = (struct snd_soc_tplg_vendor_array *)manifest->priv.data; - ret = skl_tplg_get_desc_blocks(dev, array); - if (ret < 0) - return ret; - num_blocks = ret; - - off += array->size; - /* Read the BLOCK_TYPE and BLOCK_SIZE descriptor */ - while (num_blocks > 0) { - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - ret = skl_tplg_get_desc_blocks(dev, array); - - if (ret < 0) - return ret; - block_type = ret; - off += array->size; - - array = (struct snd_soc_tplg_vendor_array *) - (manifest->priv.data + off); - - ret = skl_tplg_get_desc_blocks(dev, array); - - if (ret < 0) - return ret; - block_size = ret; - off += array->size; - - data = (manifest->priv.data + off); - - if (block_type == SKL_TYPE_TUPLE) { - ret = skl_tplg_get_manifest_tkn(dev, data, skl, - block_size); - - if (ret < 0) - return ret; - - --num_blocks; - } else { - return -EINVAL; - } - off += ret; - } - - return 0; -} - -static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, - struct snd_soc_tplg_manifest *manifest) -{ - struct hdac_bus *bus = snd_soc_component_get_drvdata(cmpnt); - struct skl_dev *skl = bus_to_skl(bus); - - /* proceed only if we have private data defined */ - if (manifest->priv.size == 0) - return 0; - - skl_tplg_get_manifest_data(manifest, bus->dev, skl); - - if (skl->lib_count > SKL_MAX_LIB) { - dev_err(bus->dev, "Exceeding max Library count. Got:%d\n", - skl->lib_count); - return -EINVAL; - } - - return 0; -} - -static int skl_tplg_complete(struct snd_soc_component *component) -{ - struct snd_soc_dobj *dobj; - struct snd_soc_acpi_mach *mach; - struct snd_ctl_elem_value *val; - int i; - - val = kmalloc(sizeof(*val), GFP_KERNEL); - if (!val) - return -ENOMEM; - - mach = dev_get_platdata(component->card->dev); - list_for_each_entry(dobj, &component->dobj_list, list) { - struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se; - char **texts; - char chan_text[4]; - - if (dobj->type != SND_SOC_DOBJ_ENUM || !kcontrol || - kcontrol->put != skl_tplg_multi_config_set_dmic) - continue; - - se = (struct soc_enum *)kcontrol->private_value; - texts = dobj->control.dtexts; - sprintf(chan_text, "c%d", mach->mach_params.dmic_num); - - for (i = 0; i < se->items; i++) { - if (strstr(texts[i], chan_text)) { - memset(val, 0, sizeof(*val)); - val->value.enumerated.item[0] = i; - kcontrol->put(kcontrol, val); - } - } - } - - kfree(val); - return 0; -} - -static const struct snd_soc_tplg_ops skl_tplg_ops = { - .widget_load = skl_tplg_widget_load, - .control_load = skl_tplg_control_load, - .bytes_ext_ops = skl_tlv_ops, - .bytes_ext_ops_count = ARRAY_SIZE(skl_tlv_ops), - .io_ops = skl_tplg_kcontrol_ops, - .io_ops_count = ARRAY_SIZE(skl_tplg_kcontrol_ops), - .manifest = skl_manifest_load, - .dai_load = skl_dai_load, - .complete = skl_tplg_complete, -}; - -/* - * A pipe can have multiple modules, each of them will be a DAPM widget as - * well. While managing a pipeline we need to get the list of all the - * widgets in a pipelines, so this helper - skl_tplg_create_pipe_widget_list() - * helps to get the SKL type widgets in that pipeline - */ -static int skl_tplg_create_pipe_widget_list(struct snd_soc_component *component) -{ - struct snd_soc_dapm_widget *w; - struct skl_module_cfg *mcfg = NULL; - struct skl_pipe_module *p_module = NULL; - struct skl_pipe *pipe; - - list_for_each_entry(w, &component->card->widgets, list) { - if (is_skl_dsp_widget_type(w, component->dev) && w->priv) { - mcfg = w->priv; - pipe = mcfg->pipe; - - p_module = devm_kzalloc(component->dev, - sizeof(*p_module), GFP_KERNEL); - if (!p_module) - return -ENOMEM; - - p_module->w = w; - list_add_tail(&p_module->node, &pipe->w_list); - } - } - - return 0; -} - -static void skl_tplg_set_pipe_type(struct skl_dev *skl, struct skl_pipe *pipe) -{ - struct skl_pipe_module *w_module; - struct snd_soc_dapm_widget *w; - struct skl_module_cfg *mconfig; - bool host_found = false, link_found = false; - - list_for_each_entry(w_module, &pipe->w_list, node) { - w = w_module->w; - mconfig = w->priv; - - if (mconfig->dev_type == SKL_DEVICE_HDAHOST) - host_found = true; - else if (mconfig->dev_type != SKL_DEVICE_NONE) - link_found = true; - } - - if (host_found && link_found) - pipe->passthru = true; - else - pipe->passthru = false; -} - -/* - * SKL topology init routine - */ -int skl_tplg_init(struct snd_soc_component *component, struct hdac_bus *bus) -{ - int ret; - const struct firmware *fw; - struct skl_dev *skl = bus_to_skl(bus); - struct skl_pipeline *ppl; - - ret = request_firmware(&fw, skl->tplg_name, bus->dev); - if (ret < 0) { - char alt_tplg_name[64]; - - snprintf(alt_tplg_name, sizeof(alt_tplg_name), "%s-tplg.bin", - skl->mach->drv_name); - dev_info(bus->dev, "tplg fw %s load failed with %d, trying alternative tplg name %s", - skl->tplg_name, ret, alt_tplg_name); - - ret = request_firmware(&fw, alt_tplg_name, bus->dev); - if (!ret) - goto component_load; - - dev_info(bus->dev, "tplg %s failed with %d, falling back to dfw_sst.bin", - alt_tplg_name, ret); - - ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); - if (ret < 0) { - dev_err(bus->dev, "Fallback tplg fw %s load failed with %d\n", - "dfw_sst.bin", ret); - return ret; - } - } - -component_load: - ret = snd_soc_tplg_component_load(component, &skl_tplg_ops, fw); - if (ret < 0) { - dev_err(bus->dev, "tplg component load failed%d\n", ret); - goto err; - } - - ret = skl_tplg_create_pipe_widget_list(component); - if (ret < 0) { - dev_err(bus->dev, "tplg create pipe widget list failed%d\n", - ret); - goto err; - } - - list_for_each_entry(ppl, &skl->ppl_list, node) - skl_tplg_set_pipe_type(skl, ppl->pipe); - -err: - release_firmware(fw); - return ret; -} - -void skl_tplg_exit(struct snd_soc_component *component, struct hdac_bus *bus) -{ - struct skl_dev *skl = bus_to_skl(bus); - struct skl_pipeline *ppl, *tmp; - - list_for_each_entry_safe(ppl, tmp, &skl->ppl_list, node) - list_del(&ppl->node); - - /* clean up topology */ - snd_soc_tplg_component_remove(component); -} diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h deleted file mode 100644 index 30a0977af943..000000000000 --- a/sound/soc/intel/skylake/skl-topology.h +++ /dev/null @@ -1,524 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * skl_topology.h - Intel HDA Platform topology header file - * - * Copyright (C) 2014-15 Intel Corp - * Author: Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#ifndef __SKL_TOPOLOGY_H__ -#define __SKL_TOPOLOGY_H__ - -#include - -#include -#include -#include -#include "skl.h" - -#define BITS_PER_BYTE 8 -#define MAX_TS_GROUPS 8 -#define MAX_DMIC_TS_GROUPS 4 -#define MAX_FIXED_DMIC_PARAMS_SIZE 727 - -/* Maximum number of coefficients up down mixer module */ -#define UP_DOWN_MIXER_MAX_COEFF 8 - -#define MODULE_MAX_IN_PINS 8 -#define MODULE_MAX_OUT_PINS 8 - -#define SKL_MIC_CH_SUPPORT 4 -#define SKL_MIC_MAX_CH_SUPPORT 8 -#define SKL_DEFAULT_MIC_SEL_GAIN 0x3FF -#define SKL_MIC_SEL_SWITCH 0x3 - -#define SKL_OUTPUT_PIN 0 -#define SKL_INPUT_PIN 1 -#define SKL_MAX_PATH_CONFIGS 8 -#define SKL_MAX_MODULES_IN_PIPE 8 -#define SKL_MAX_MODULE_FORMATS 32 -#define SKL_MAX_MODULE_RESOURCES 32 - -enum skl_channel_index { - SKL_CHANNEL_LEFT = 0, - SKL_CHANNEL_RIGHT = 1, - SKL_CHANNEL_CENTER = 2, - SKL_CHANNEL_LEFT_SURROUND = 3, - SKL_CHANNEL_CENTER_SURROUND = 3, - SKL_CHANNEL_RIGHT_SURROUND = 4, - SKL_CHANNEL_LFE = 7, - SKL_CHANNEL_INVALID = 0xF, -}; - -enum skl_bitdepth { - SKL_DEPTH_8BIT = 8, - SKL_DEPTH_16BIT = 16, - SKL_DEPTH_24BIT = 24, - SKL_DEPTH_32BIT = 32, - SKL_DEPTH_INVALID -}; - - -enum skl_s_freq { - SKL_FS_8000 = 8000, - SKL_FS_11025 = 11025, - SKL_FS_12000 = 12000, - SKL_FS_16000 = 16000, - SKL_FS_22050 = 22050, - SKL_FS_24000 = 24000, - SKL_FS_32000 = 32000, - SKL_FS_44100 = 44100, - SKL_FS_48000 = 48000, - SKL_FS_64000 = 64000, - SKL_FS_88200 = 88200, - SKL_FS_96000 = 96000, - SKL_FS_128000 = 128000, - SKL_FS_176400 = 176400, - SKL_FS_192000 = 192000, - SKL_FS_INVALID -}; - -#define SKL_MAX_PARAMS_TYPES 4 - -enum skl_widget_type { - SKL_WIDGET_VMIXER = 1, - SKL_WIDGET_MIXER = 2, - SKL_WIDGET_PGA = 3, - SKL_WIDGET_MUX = 4 -}; - -struct skl_audio_data_format { - enum skl_s_freq s_freq; - enum skl_bitdepth bit_depth; - u32 channel_map; - enum skl_ch_cfg ch_cfg; - enum skl_interleaving interleaving; - u8 number_of_channels; - u8 valid_bit_depth; - u8 sample_type; - u8 reserved; -} __packed; - -struct skl_base_cfg { - u32 cpc; - u32 ibs; - u32 obs; - u32 is_pages; - struct skl_audio_data_format audio_fmt; -}; - -struct skl_cpr_gtw_cfg { - u32 node_id; - u32 dma_buffer_size; - u32 config_length; - /* not mandatory; required only for DMIC/I2S */ - struct { - u32 gtw_attrs; - u32 data[]; - } config_data; -} __packed; - -struct skl_dma_control { - u32 node_id; - u32 config_length; - u32 config_data[]; -} __packed; - -struct skl_cpr_cfg { - struct skl_base_cfg base_cfg; - struct skl_audio_data_format out_fmt; - u32 cpr_feature_mask; - struct skl_cpr_gtw_cfg gtw_cfg; -} __packed; - -struct skl_cpr_pin_fmt { - u32 sink_id; - struct skl_audio_data_format src_fmt; - struct skl_audio_data_format dst_fmt; -} __packed; - -struct skl_src_module_cfg { - struct skl_base_cfg base_cfg; - enum skl_s_freq src_cfg; -} __packed; - -struct skl_up_down_mixer_cfg { - struct skl_base_cfg base_cfg; - enum skl_ch_cfg out_ch_cfg; - /* This should be set to 1 if user coefficients are required */ - u32 coeff_sel; - /* Pass the user coeff in this array */ - s32 coeff[UP_DOWN_MIXER_MAX_COEFF]; - u32 ch_map; -} __packed; - -struct skl_pin_format { - u32 pin_idx; - u32 buf_size; - struct skl_audio_data_format audio_fmt; -} __packed; - -struct skl_base_cfg_ext { - u16 nr_input_pins; - u16 nr_output_pins; - u8 reserved[8]; - u32 priv_param_length; - /* Input pin formats followed by output ones. */ - struct skl_pin_format pins_fmt[]; -} __packed; - -struct skl_algo_cfg { - struct skl_base_cfg base_cfg; - char params[]; -} __packed; - -struct skl_base_outfmt_cfg { - struct skl_base_cfg base_cfg; - struct skl_audio_data_format out_fmt; -} __packed; - -enum skl_dma_type { - SKL_DMA_HDA_HOST_OUTPUT_CLASS = 0, - SKL_DMA_HDA_HOST_INPUT_CLASS = 1, - SKL_DMA_HDA_HOST_INOUT_CLASS = 2, - SKL_DMA_HDA_LINK_OUTPUT_CLASS = 8, - SKL_DMA_HDA_LINK_INPUT_CLASS = 9, - SKL_DMA_HDA_LINK_INOUT_CLASS = 0xA, - SKL_DMA_DMIC_LINK_INPUT_CLASS = 0xB, - SKL_DMA_I2S_LINK_OUTPUT_CLASS = 0xC, - SKL_DMA_I2S_LINK_INPUT_CLASS = 0xD, -}; - -union skl_ssp_dma_node { - u8 val; - struct { - u8 time_slot_index:4; - u8 i2s_instance:4; - } dma_node; -}; - -union skl_connector_node_id { - u32 val; - struct { - u32 vindex:8; - u32 dma_type:4; - u32 rsvd:20; - } node; -}; - -struct skl_module_fmt { - u32 channels; - u32 s_freq; - u32 bit_depth; - u32 valid_bit_depth; - u32 ch_cfg; - u32 interleaving_style; - u32 sample_type; - u32 ch_map; -}; - -struct skl_module_cfg; - -struct skl_mod_inst_map { - u16 mod_id; - u16 inst_id; -}; - -struct skl_uuid_inst_map { - u16 inst_id; - u16 reserved; - guid_t mod_uuid; -} __packed; - -struct skl_kpb_params { - u32 num_modules; - union { - DECLARE_FLEX_ARRAY(struct skl_mod_inst_map, map); - DECLARE_FLEX_ARRAY(struct skl_uuid_inst_map, map_uuid); - } u; -}; - -struct skl_module_inst_id { - guid_t mod_uuid; - int module_id; - u32 instance_id; - int pvt_id; -}; - -enum skl_module_pin_state { - SKL_PIN_UNBIND = 0, - SKL_PIN_BIND_DONE = 1, -}; - -struct skl_module_pin { - struct skl_module_inst_id id; - bool is_dynamic; - bool in_use; - enum skl_module_pin_state pin_state; - struct skl_module_cfg *tgt_mcfg; -}; - -struct skl_specific_cfg { - u32 set_params; - u32 param_id; - u32 caps_size; - u32 *caps; -}; - -enum skl_pipe_state { - SKL_PIPE_INVALID = 0, - SKL_PIPE_CREATED = 1, - SKL_PIPE_PAUSED = 2, - SKL_PIPE_STARTED = 3, - SKL_PIPE_RESET = 4 -}; - -struct skl_pipe_module { - struct snd_soc_dapm_widget *w; - struct list_head node; -}; - -struct skl_pipe_params { - u8 host_dma_id; - u8 link_dma_id; - u32 ch; - u32 s_freq; - u32 s_fmt; - u32 s_cont; - u8 linktype; - snd_pcm_format_t format; - int link_index; - int stream; - unsigned int host_bps; - unsigned int link_bps; -}; - -struct skl_pipe_fmt { - u32 freq; - u8 channels; - u8 bps; -}; - -struct skl_pipe_mcfg { - u8 res_idx; - u8 fmt_idx; -}; - -struct skl_path_config { - u8 mem_pages; - struct skl_pipe_fmt in_fmt; - struct skl_pipe_fmt out_fmt; -}; - -struct skl_pipe { - u8 ppl_id; - u8 pipe_priority; - u16 conn_type; - u32 memory_pages; - u8 lp_mode; - struct skl_pipe_params *p_params; - enum skl_pipe_state state; - u8 direction; - u8 cur_config_idx; - u8 nr_cfgs; - struct skl_path_config configs[SKL_MAX_PATH_CONFIGS]; - struct list_head w_list; - bool passthru; -}; - -enum skl_module_state { - SKL_MODULE_UNINIT = 0, - SKL_MODULE_INIT_DONE = 1, - SKL_MODULE_BIND_DONE = 2, -}; - -enum d0i3_capability { - SKL_D0I3_NONE = 0, - SKL_D0I3_STREAMING = 1, - SKL_D0I3_NON_STREAMING = 2, -}; - -struct skl_module_pin_fmt { - u8 id; - struct skl_module_fmt fmt; -}; - -struct skl_module_iface { - u8 fmt_idx; - u8 nr_in_fmt; - u8 nr_out_fmt; - struct skl_module_pin_fmt inputs[MAX_IN_QUEUE]; - struct skl_module_pin_fmt outputs[MAX_OUT_QUEUE]; -}; - -struct skl_module_pin_resources { - u8 pin_index; - u32 buf_size; -}; - -struct skl_module_res { - u8 id; - u32 is_pages; - u32 ibs; - u32 obs; - u32 dma_buffer_size; - u32 cpc; - u8 nr_input_pins; - u8 nr_output_pins; - struct skl_module_pin_resources input[MAX_IN_QUEUE]; - struct skl_module_pin_resources output[MAX_OUT_QUEUE]; -}; - -struct skl_module { - guid_t uuid; - u8 loadable; - u8 input_pin_type; - u8 output_pin_type; - u8 max_input_pins; - u8 max_output_pins; - u8 nr_resources; - u8 nr_interfaces; - struct skl_module_res resources[SKL_MAX_MODULE_RESOURCES]; - struct skl_module_iface formats[SKL_MAX_MODULE_FORMATS]; -}; - -struct skl_module_cfg { - u8 guid[16]; - struct skl_module_inst_id id; - struct skl_module *module; - int res_idx; - int fmt_idx; - int fmt_cfg_idx; - u8 domain; - bool homogenous_inputs; - bool homogenous_outputs; - struct skl_module_fmt in_fmt[MODULE_MAX_IN_PINS]; - struct skl_module_fmt out_fmt[MODULE_MAX_OUT_PINS]; - u8 max_in_queue; - u8 max_out_queue; - u8 in_queue_mask; - u8 out_queue_mask; - u8 in_queue; - u8 out_queue; - u8 is_loadable; - u8 core_id; - u8 dev_type; - u8 dma_id; - u8 time_slot; - u8 dmic_ch_combo_index; - u32 dmic_ch_type; - u32 params_fixup; - u32 converter; - u32 vbus_id; - u32 mem_pages; - enum d0i3_capability d0i3_caps; - u32 dma_buffer_size; /* in milli seconds */ - struct skl_module_pin *m_in_pin; - struct skl_module_pin *m_out_pin; - enum skl_module_type m_type; - enum skl_hw_conn_type hw_conn_type; - enum skl_module_state m_state; - struct skl_pipe *pipe; - struct skl_specific_cfg formats_config[SKL_MAX_PARAMS_TYPES]; - struct skl_pipe_mcfg mod_cfg[SKL_MAX_MODULES_IN_PIPE]; -}; - -struct skl_algo_data { - u32 param_id; - u32 set_params; - u32 max; - u32 size; - char *params; -}; - -struct skl_pipeline { - struct skl_pipe *pipe; - struct list_head node; -}; - -struct skl_module_deferred_bind { - struct skl_module_cfg *src; - struct skl_module_cfg *dst; - struct list_head node; -}; - -struct skl_mic_sel_config { - u16 mic_switch; - u16 flags; - u16 blob[SKL_MIC_MAX_CH_SUPPORT][SKL_MIC_MAX_CH_SUPPORT]; -} __packed; - -enum skl_channel { - SKL_CH_MONO = 1, - SKL_CH_STEREO = 2, - SKL_CH_TRIO = 3, - SKL_CH_QUATRO = 4, -}; - -static inline struct skl_dev *get_skl_ctx(struct device *dev) -{ - struct hdac_bus *bus = dev_get_drvdata(dev); - - return bus_to_skl(bus); -} - -int skl_tplg_be_update_params(struct snd_soc_dai *dai, - struct skl_pipe_params *params); -int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps, - u32 caps_size, u32 node_id); -void skl_tplg_set_be_dmic_config(struct snd_soc_dai *dai, - struct skl_pipe_params *params, int stream); -int skl_tplg_init(struct snd_soc_component *component, - struct hdac_bus *bus); -void skl_tplg_exit(struct snd_soc_component *component, - struct hdac_bus *bus); -struct skl_module_cfg *skl_tplg_fe_get_cpr_module( - struct snd_soc_dai *dai, int stream); -int skl_tplg_update_pipe_params(struct device *dev, - struct skl_module_cfg *mconfig, struct skl_pipe_params *params); - -void skl_tplg_d0i3_get(struct skl_dev *skl, enum d0i3_capability caps); -void skl_tplg_d0i3_put(struct skl_dev *skl, enum d0i3_capability caps); - -int skl_create_pipeline(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_run_pipe(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_pause_pipe(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_delete_pipe(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_stop_pipe(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_reset_pipe(struct skl_dev *skl, struct skl_pipe *pipe); - -int skl_init_module(struct skl_dev *skl, struct skl_module_cfg *mconfig); - -int skl_bind_modules(struct skl_dev *skl, struct skl_module_cfg - *src_mcfg, struct skl_module_cfg *dst_mcfg); - -int skl_unbind_modules(struct skl_dev *skl, struct skl_module_cfg - *src_mcfg, struct skl_module_cfg *dst_mcfg); - -int skl_set_module_params(struct skl_dev *skl, u32 *params, int size, - u32 param_id, struct skl_module_cfg *mcfg); -int skl_get_module_params(struct skl_dev *skl, u32 *params, int size, - u32 param_id, struct skl_module_cfg *mcfg); - -struct skl_module_cfg *skl_tplg_be_get_cpr_module(struct snd_soc_dai *dai, - int stream); -enum skl_bitdepth skl_get_bit_depth(int params); -int skl_pcm_host_dma_prepare(struct device *dev, - struct skl_pipe_params *params); -int skl_pcm_link_dma_prepare(struct device *dev, - struct skl_pipe_params *params); - -int skl_dai_load(struct snd_soc_component *cmp, int index, - struct snd_soc_dai_driver *dai_drv, - struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); -void skl_tplg_add_moduleid_in_bind_params(struct skl_dev *skl, - struct snd_soc_dapm_widget *w); -#endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c deleted file mode 100644 index 117125187793..000000000000 --- a/sound/soc/intel/skylake/skl.c +++ /dev/null @@ -1,1177 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -/* - * skl.c - Implementation of ASoC Intel SKL HD Audio driver - * - * Copyright (C) 2014-2015 Intel Corp - * Author: Jeeja KP - * - * Derived mostly from Intel HDA driver with following copyrights: - * Copyright (c) 2004 Takashi Iwai - * PeiSen Hou - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include -#include "skl.h" -#include "skl-sst-dsp.h" -#include "skl-sst-ipc.h" - -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) -#include "../../../soc/codecs/hdac_hda.h" -#endif -static int skl_pci_binding; -module_param_named(pci_binding, skl_pci_binding, int, 0444); -MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); - -/* - * initialize the PCI registers - */ -static void skl_update_pci_byte(struct pci_dev *pci, unsigned int reg, - unsigned char mask, unsigned char val) -{ - unsigned char data; - - pci_read_config_byte(pci, reg, &data); - data &= ~mask; - data |= (val & mask); - pci_write_config_byte(pci, reg, data); -} - -static void skl_init_pci(struct skl_dev *skl) -{ - struct hdac_bus *bus = skl_to_bus(skl); - - /* - * Clear bits 0-2 of PCI register TCSEL (at offset 0x44) - * TCSEL == Traffic Class Select Register, which sets PCI express QOS - * Ensuring these bits are 0 clears playback static on some HD Audio - * codecs. - * The PCI register TCSEL is defined in the Intel manuals. - */ - dev_dbg(bus->dev, "Clearing TCSEL\n"); - skl_update_pci_byte(skl->pci, AZX_PCIREG_TCSEL, 0x07, 0); -} - -static void update_pci_dword(struct pci_dev *pci, - unsigned int reg, u32 mask, u32 val) -{ - u32 data = 0; - - pci_read_config_dword(pci, reg, &data); - data &= ~mask; - data |= (val & mask); - pci_write_config_dword(pci, reg, data); -} - -/* - * skl_enable_miscbdcge - enable/dsiable CGCTL.MISCBDCGE bits - * - * @dev: device pointer - * @enable: enable/disable flag - */ -static void skl_enable_miscbdcge(struct device *dev, bool enable) -{ - struct pci_dev *pci = to_pci_dev(dev); - u32 val; - - val = enable ? AZX_CGCTL_MISCBDCGE_MASK : 0; - - update_pci_dword(pci, AZX_PCIREG_CGCTL, AZX_CGCTL_MISCBDCGE_MASK, val); -} - -/** - * skl_clock_power_gating: Enable/Disable clock and power gating - * - * @dev: Device pointer - * @enable: Enable/Disable flag - */ -static void skl_clock_power_gating(struct device *dev, bool enable) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - u32 val; - - /* Update PDCGE bit of CGCTL register */ - val = enable ? AZX_CGCTL_ADSPDCGE : 0; - update_pci_dword(pci, AZX_PCIREG_CGCTL, AZX_CGCTL_ADSPDCGE, val); - - /* Update L1SEN bit of EM2 register */ - val = enable ? AZX_REG_VS_EM2_L1SEN : 0; - snd_hdac_chip_updatel(bus, VS_EM2, AZX_REG_VS_EM2_L1SEN, val); - - /* Update ADSPPGD bit of PGCTL register */ - val = enable ? 0 : AZX_PGCTL_ADSPPGD; - update_pci_dword(pci, AZX_PCIREG_PGCTL, AZX_PGCTL_ADSPPGD, val); -} - -/* - * While performing reset, controller may not come back properly causing - * issues, so recommendation is to set CGCTL.MISCBDCGE to 0 then do reset - * (init chip) and then again set CGCTL.MISCBDCGE to 1 - */ -static int skl_init_chip(struct hdac_bus *bus, bool full_reset) -{ - struct hdac_ext_link *hlink; - int ret; - - snd_hdac_set_codec_wakeup(bus, true); - skl_enable_miscbdcge(bus->dev, false); - ret = snd_hdac_bus_init_chip(bus, full_reset); - - /* Reset stream-to-link mapping */ - list_for_each_entry(hlink, &bus->hlink_list, list) - writel(0, hlink->ml_addr + AZX_REG_ML_LOSIDV); - - skl_enable_miscbdcge(bus->dev, true); - snd_hdac_set_codec_wakeup(bus, false); - - return ret; -} - -void skl_update_d0i3c(struct device *dev, bool enable) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - u8 reg; - int timeout = 50; - - reg = snd_hdac_chip_readb(bus, VS_D0I3C); - /* Do not write to D0I3C until command in progress bit is cleared */ - while ((reg & AZX_REG_VS_D0I3C_CIP) && --timeout) { - udelay(10); - reg = snd_hdac_chip_readb(bus, VS_D0I3C); - } - - /* Highly unlikely. But if it happens, flag error explicitly */ - if (!timeout) { - dev_err(bus->dev, "Before D0I3C update: D0I3C CIP timeout\n"); - return; - } - - if (enable) - reg = reg | AZX_REG_VS_D0I3C_I3; - else - reg = reg & (~AZX_REG_VS_D0I3C_I3); - - snd_hdac_chip_writeb(bus, VS_D0I3C, reg); - - timeout = 50; - /* Wait for cmd in progress to be cleared before exiting the function */ - reg = snd_hdac_chip_readb(bus, VS_D0I3C); - while ((reg & AZX_REG_VS_D0I3C_CIP) && --timeout) { - udelay(10); - reg = snd_hdac_chip_readb(bus, VS_D0I3C); - } - - /* Highly unlikely. But if it happens, flag error explicitly */ - if (!timeout) { - dev_err(bus->dev, "After D0I3C update: D0I3C CIP timeout\n"); - return; - } - - dev_dbg(bus->dev, "D0I3C register = 0x%x\n", - snd_hdac_chip_readb(bus, VS_D0I3C)); -} - -/** - * skl_dum_set - set DUM bit in EM2 register - * @bus: HD-audio core bus - * - * Addresses incorrect position reporting for capture streams. - * Used on device power up. - */ -static void skl_dum_set(struct hdac_bus *bus) -{ - /* For the DUM bit to be set, CRST needs to be out of reset state */ - if (!(snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET)) { - skl_enable_miscbdcge(bus->dev, false); - snd_hdac_bus_exit_link_reset(bus); - skl_enable_miscbdcge(bus->dev, true); - } - - snd_hdac_chip_updatel(bus, VS_EM2, AZX_VS_EM2_DUM, AZX_VS_EM2_DUM); -} - -/* called from IRQ */ -static void skl_stream_update(struct hdac_bus *bus, struct hdac_stream *hstr) -{ - snd_pcm_period_elapsed(hstr->substream); -} - -static irqreturn_t skl_interrupt(int irq, void *dev_id) -{ - struct hdac_bus *bus = dev_id; - u32 status; - - if (!pm_runtime_active(bus->dev)) - return IRQ_NONE; - - spin_lock(&bus->reg_lock); - - status = snd_hdac_chip_readl(bus, INTSTS); - if (status == 0 || status == 0xffffffff) { - spin_unlock(&bus->reg_lock); - return IRQ_NONE; - } - - /* clear rirb int */ - status = snd_hdac_chip_readb(bus, RIRBSTS); - if (status & RIRB_INT_MASK) { - if (status & RIRB_INT_RESPONSE) - snd_hdac_bus_update_rirb(bus); - snd_hdac_chip_writeb(bus, RIRBSTS, RIRB_INT_MASK); - } - - spin_unlock(&bus->reg_lock); - - return snd_hdac_chip_readl(bus, INTSTS) ? IRQ_WAKE_THREAD : IRQ_HANDLED; -} - -static irqreturn_t skl_threaded_handler(int irq, void *dev_id) -{ - struct hdac_bus *bus = dev_id; - u32 status; - - status = snd_hdac_chip_readl(bus, INTSTS); - - snd_hdac_bus_handle_stream_irq(bus, status, skl_stream_update); - - return IRQ_HANDLED; -} - -static int skl_acquire_irq(struct hdac_bus *bus, int do_disconnect) -{ - struct skl_dev *skl = bus_to_skl(bus); - int ret; - - ret = request_threaded_irq(skl->pci->irq, skl_interrupt, - skl_threaded_handler, - IRQF_SHARED, - KBUILD_MODNAME, bus); - if (ret) { - dev_err(bus->dev, - "unable to grab IRQ %d, disabling device\n", - skl->pci->irq); - return ret; - } - - bus->irq = skl->pci->irq; - pci_intx(skl->pci, 1); - - return 0; -} - -static int skl_suspend_late(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - struct skl_dev *skl = bus_to_skl(bus); - - return skl_suspend_late_dsp(skl); -} - -#ifdef CONFIG_PM -static int _skl_suspend(struct hdac_bus *bus) -{ - struct skl_dev *skl = bus_to_skl(bus); - struct pci_dev *pci = to_pci_dev(bus->dev); - int ret; - - snd_hdac_ext_bus_link_power_down_all(bus); - - ret = skl_suspend_dsp(skl); - if (ret < 0) - return ret; - - snd_hdac_bus_stop_chip(bus); - update_pci_dword(pci, AZX_PCIREG_PGCTL, - AZX_PGCTL_LSRMD_MASK, AZX_PGCTL_LSRMD_MASK); - skl_enable_miscbdcge(bus->dev, false); - snd_hdac_bus_enter_link_reset(bus); - skl_enable_miscbdcge(bus->dev, true); - skl_cleanup_resources(skl); - - return 0; -} - -static int _skl_resume(struct hdac_bus *bus) -{ - struct skl_dev *skl = bus_to_skl(bus); - - skl_init_pci(skl); - skl_dum_set(bus); - skl_init_chip(bus, true); - - return skl_resume_dsp(skl); -} -#endif - -#ifdef CONFIG_PM_SLEEP -/* - * power management - */ -static int skl_suspend(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - struct skl_dev *skl = bus_to_skl(bus); - int ret; - - /* - * Do not suspend if streams which are marked ignore suspend are - * running, we need to save the state for these and continue - */ - if (skl->supend_active) { - /* turn off the links and stop the CORB/RIRB DMA if it is On */ - snd_hdac_ext_bus_link_power_down_all(bus); - - if (bus->cmd_dma_state) - snd_hdac_bus_stop_cmd_io(bus); - - enable_irq_wake(bus->irq); - pci_save_state(pci); - } else { - ret = _skl_suspend(bus); - if (ret < 0) - return ret; - skl->fw_loaded = false; - } - - return 0; -} - -static int skl_resume(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - struct skl_dev *skl = bus_to_skl(bus); - struct hdac_ext_link *hlink; - int ret; - - /* - * resume only when we are not in suspend active, otherwise need to - * restore the device - */ - if (skl->supend_active) { - pci_restore_state(pci); - snd_hdac_ext_bus_link_power_up_all(bus); - disable_irq_wake(bus->irq); - /* - * turn On the links which are On before active suspend - * and start the CORB/RIRB DMA if On before - * active suspend. - */ - list_for_each_entry(hlink, &bus->hlink_list, list) { - if (hlink->ref_count) - snd_hdac_ext_bus_link_power_up(hlink); - } - - ret = 0; - if (bus->cmd_dma_state) - snd_hdac_bus_init_cmd_io(bus); - } else { - ret = _skl_resume(bus); - } - - return ret; -} -#endif /* CONFIG_PM_SLEEP */ - -#ifdef CONFIG_PM -static int skl_runtime_suspend(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - - dev_dbg(bus->dev, "in %s\n", __func__); - - return _skl_suspend(bus); -} - -static int skl_runtime_resume(struct device *dev) -{ - struct pci_dev *pci = to_pci_dev(dev); - struct hdac_bus *bus = pci_get_drvdata(pci); - - dev_dbg(bus->dev, "in %s\n", __func__); - - return _skl_resume(bus); -} -#endif /* CONFIG_PM */ - -static const struct dev_pm_ops skl_pm = { - SET_SYSTEM_SLEEP_PM_OPS(skl_suspend, skl_resume) - SET_RUNTIME_PM_OPS(skl_runtime_suspend, skl_runtime_resume, NULL) - .suspend_late = skl_suspend_late, -}; - -/* - * destructor - */ -static int skl_free(struct hdac_bus *bus) -{ - struct skl_dev *skl = bus_to_skl(bus); - - skl->init_done = 0; /* to be sure */ - - snd_hdac_stop_streams_and_chip(bus); - - if (bus->irq >= 0) - free_irq(bus->irq, (void *)bus); - snd_hdac_bus_free_stream_pages(bus); - snd_hdac_ext_stream_free_all(bus); - snd_hdac_ext_link_free_all(bus); - - if (bus->remap_addr) - iounmap(bus->remap_addr); - - pci_release_regions(skl->pci); - pci_disable_device(skl->pci); - - snd_hdac_ext_bus_exit(bus); - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - snd_hdac_i915_exit(bus); - } - - return 0; -} - -/* - * For each ssp there are 3 clocks (mclk/sclk/sclkfs). - * e.g. for ssp0, clocks will be named as - * "ssp0_mclk", "ssp0_sclk", "ssp0_sclkfs" - * So for skl+, there are 6 ssps, so 18 clocks will be created. - */ -static struct skl_ssp_clk skl_ssp_clks[] = { - {.name = "ssp0_mclk"}, {.name = "ssp1_mclk"}, {.name = "ssp2_mclk"}, - {.name = "ssp3_mclk"}, {.name = "ssp4_mclk"}, {.name = "ssp5_mclk"}, - {.name = "ssp0_sclk"}, {.name = "ssp1_sclk"}, {.name = "ssp2_sclk"}, - {.name = "ssp3_sclk"}, {.name = "ssp4_sclk"}, {.name = "ssp5_sclk"}, - {.name = "ssp0_sclkfs"}, {.name = "ssp1_sclkfs"}, - {.name = "ssp2_sclkfs"}, - {.name = "ssp3_sclkfs"}, {.name = "ssp4_sclkfs"}, - {.name = "ssp5_sclkfs"}, -}; - -static struct snd_soc_acpi_mach *skl_find_hda_machine(struct skl_dev *skl, - struct snd_soc_acpi_mach *machines) -{ - struct snd_soc_acpi_mach *mach; - - /* point to common table */ - mach = snd_soc_acpi_intel_hda_machines; - - /* all entries in the machine table use the same firmware */ - mach->fw_filename = machines->fw_filename; - - return mach; -} - -static int skl_find_machine(struct skl_dev *skl, void *driver_data) -{ - struct hdac_bus *bus = skl_to_bus(skl); - struct snd_soc_acpi_mach *mach = driver_data; - struct skl_machine_pdata *pdata; - - mach = snd_soc_acpi_find_machine(mach); - if (!mach) { - dev_dbg(bus->dev, "No matching I2S machine driver found\n"); - mach = skl_find_hda_machine(skl, driver_data); - if (!mach) { - dev_err(bus->dev, "No matching machine driver found\n"); - return -ENODEV; - } - } - - skl->mach = mach; - skl->fw_name = mach->fw_filename; - pdata = mach->pdata; - - if (pdata) { - skl->use_tplg_pcm = pdata->use_tplg_pcm; - mach->mach_params.dmic_num = - intel_nhlt_get_dmic_geo(&skl->pci->dev, - skl->nhlt); - } - - return 0; -} - -static int skl_machine_device_register(struct skl_dev *skl) -{ - struct snd_soc_acpi_mach *mach = skl->mach; - struct hdac_bus *bus = skl_to_bus(skl); - struct platform_device *pdev; - int ret; - - pdev = platform_device_alloc(mach->drv_name, -1); - if (pdev == NULL) { - dev_err(bus->dev, "platform device alloc failed\n"); - return -EIO; - } - - mach->mach_params.platform = dev_name(bus->dev); - mach->mach_params.codec_mask = bus->codec_mask; - - ret = platform_device_add_data(pdev, (const void *)mach, sizeof(*mach)); - if (ret) { - dev_err(bus->dev, "failed to add machine device platform data\n"); - platform_device_put(pdev); - return ret; - } - - ret = platform_device_add(pdev); - if (ret) { - dev_err(bus->dev, "failed to add machine device\n"); - platform_device_put(pdev); - return -EIO; - } - - - skl->i2s_dev = pdev; - - return 0; -} - -static void skl_machine_device_unregister(struct skl_dev *skl) -{ - if (skl->i2s_dev) - platform_device_unregister(skl->i2s_dev); -} - -static int skl_dmic_device_register(struct skl_dev *skl) -{ - struct hdac_bus *bus = skl_to_bus(skl); - struct platform_device *pdev; - int ret; - - /* SKL has one dmic port, so allocate dmic device for this */ - pdev = platform_device_alloc("dmic-codec", -1); - if (!pdev) { - dev_err(bus->dev, "failed to allocate dmic device\n"); - return -ENOMEM; - } - - ret = platform_device_add(pdev); - if (ret) { - dev_err(bus->dev, "failed to add dmic device: %d\n", ret); - platform_device_put(pdev); - return ret; - } - skl->dmic_dev = pdev; - - return 0; -} - -static void skl_dmic_device_unregister(struct skl_dev *skl) -{ - if (skl->dmic_dev) - platform_device_unregister(skl->dmic_dev); -} - -static struct skl_clk_parent_src skl_clk_src[] = { - { .clk_id = SKL_XTAL, .name = "xtal" }, - { .clk_id = SKL_CARDINAL, .name = "cardinal", .rate = 24576000 }, - { .clk_id = SKL_PLL, .name = "pll", .rate = 96000000 }, -}; - -struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id) -{ - unsigned int i; - - for (i = 0; i < ARRAY_SIZE(skl_clk_src); i++) { - if (skl_clk_src[i].clk_id == clk_id) - return &skl_clk_src[i]; - } - - return NULL; -} - -static void init_skl_xtal_rate(int pci_id) -{ - switch (pci_id) { - case PCI_DEVICE_ID_INTEL_HDA_SKL_LP: - case PCI_DEVICE_ID_INTEL_HDA_KBL_LP: - skl_clk_src[0].rate = 24000000; - return; - - default: - skl_clk_src[0].rate = 19200000; - return; - } -} - -static int skl_clock_device_register(struct skl_dev *skl) -{ - struct platform_device_info pdevinfo = {NULL}; - struct skl_clk_pdata *clk_pdata; - - if (!skl->nhlt) - return 0; - - clk_pdata = devm_kzalloc(&skl->pci->dev, sizeof(*clk_pdata), - GFP_KERNEL); - if (!clk_pdata) - return -ENOMEM; - - init_skl_xtal_rate(skl->pci->device); - - clk_pdata->parent_clks = skl_clk_src; - clk_pdata->ssp_clks = skl_ssp_clks; - clk_pdata->num_clks = ARRAY_SIZE(skl_ssp_clks); - - /* Query NHLT to fill the rates and parent */ - skl_get_clks(skl, clk_pdata->ssp_clks); - clk_pdata->pvt_data = skl; - - /* Register Platform device */ - pdevinfo.parent = &skl->pci->dev; - pdevinfo.id = -1; - pdevinfo.name = "skl-ssp-clk"; - pdevinfo.data = clk_pdata; - pdevinfo.size_data = sizeof(*clk_pdata); - skl->clk_dev = platform_device_register_full(&pdevinfo); - return PTR_ERR_OR_ZERO(skl->clk_dev); -} - -static void skl_clock_device_unregister(struct skl_dev *skl) -{ - if (skl->clk_dev) - platform_device_unregister(skl->clk_dev); -} - -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) - -#define IDISP_INTEL_VENDOR_ID 0x80860000 - -/* - * load the legacy codec driver - */ -static void load_codec_module(struct hda_codec *codec) -{ -#ifdef MODULE - char modalias[MODULE_NAME_LEN]; - const char *mod = NULL; - - snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias)); - mod = modalias; - dev_dbg(&codec->core.dev, "loading %s codec module\n", mod); - request_module(mod); -#endif -} - -#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ - -static struct hda_codec *skl_codec_device_init(struct hdac_bus *bus, int addr) -{ - struct hda_codec *codec; - int ret; - - codec = snd_hda_codec_device_init(to_hda_bus(bus), addr, "ehdaudio%dD%d", bus->idx, addr); - if (IS_ERR(codec)) { - dev_err(bus->dev, "device init failed for hdac device\n"); - return codec; - } - - codec->core.type = HDA_DEV_ASOC; - - ret = snd_hdac_device_register(&codec->core); - if (ret) { - dev_err(bus->dev, "failed to register hdac device\n"); - put_device(&codec->core.dev); - return ERR_PTR(ret); - } - - return codec; -} - -/* - * Probe the given codec address - */ -static int probe_codec(struct hdac_bus *bus, int addr) -{ - unsigned int cmd = (addr << 28) | (AC_NODE_ROOT << 20) | - (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; - unsigned int res = -1; -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) - struct skl_dev *skl = bus_to_skl(bus); - struct hdac_hda_priv *hda_codec; -#endif - struct hda_codec *codec; - - mutex_lock(&bus->cmd_mutex); - snd_hdac_bus_send_cmd(bus, cmd); - snd_hdac_bus_get_response(bus, addr, &res); - mutex_unlock(&bus->cmd_mutex); - if (res == -1) - return -EIO; - dev_dbg(bus->dev, "codec #%d probed OK: %x\n", addr, res); - -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) - hda_codec = devm_kzalloc(&skl->pci->dev, sizeof(*hda_codec), - GFP_KERNEL); - if (!hda_codec) - return -ENOMEM; - - codec = skl_codec_device_init(bus, addr); - if (IS_ERR(codec)) - return PTR_ERR(codec); - - hda_codec->codec = codec; - hda_codec->dev_index = addr; - dev_set_drvdata(&codec->core.dev, hda_codec); - - /* use legacy bus only for HDA codecs, idisp uses ext bus */ - if ((res & 0xFFFF0000) != IDISP_INTEL_VENDOR_ID) { - codec->core.type = HDA_DEV_LEGACY; - load_codec_module(hda_codec->codec); - } - return 0; -#else - codec = skl_codec_device_init(bus, addr); - return PTR_ERR_OR_ZERO(codec); -#endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ -} - -/* Codec initialization */ -static void skl_codec_create(struct hdac_bus *bus) -{ - int c, max_slots; - - max_slots = HDA_MAX_CODECS; - - /* First try to probe all given codec slots */ - for (c = 0; c < max_slots; c++) { - if ((bus->codec_mask & (1 << c))) { - if (probe_codec(bus, c) < 0) { - /* - * Some BIOSen give you wrong codec addresses - * that don't exist - */ - dev_warn(bus->dev, - "Codec #%d probe error; disabling it...\n", c); - bus->codec_mask &= ~(1 << c); - /* - * More badly, accessing to a non-existing - * codec often screws up the controller bus, - * and disturbs the further communications. - * Thus if an error occurs during probing, - * better to reset the controller bus to get - * back to the sanity state. - */ - snd_hdac_bus_stop_chip(bus); - skl_init_chip(bus, true); - } - } - } -} - -static void skl_probe_work(struct work_struct *work) -{ - struct skl_dev *skl = container_of(work, struct skl_dev, probe_work); - struct hdac_bus *bus = skl_to_bus(skl); - struct hdac_ext_link *hlink; - int err; - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, true); - - skl_init_pci(skl); - skl_dum_set(bus); - - err = skl_init_chip(bus, true); - if (err < 0) { - dev_err(bus->dev, "Init chip failed with err: %d\n", err); - goto out_err; - } - - /* codec detection */ - if (!bus->codec_mask) - dev_info(bus->dev, "no hda codecs found!\n"); - - /* create codec instances */ - skl_codec_create(bus); - - /* register platform dai and controls */ - err = skl_platform_register(bus->dev); - if (err < 0) { - dev_err(bus->dev, "platform register failed: %d\n", err); - goto out_err; - } - - err = skl_machine_device_register(skl); - if (err < 0) { - dev_err(bus->dev, "machine register failed: %d\n", err); - goto out_err; - } - - /* - * we are done probing so decrement link counts - */ - list_for_each_entry(hlink, &bus->hlink_list, list) - snd_hdac_ext_bus_link_put(bus, hlink); - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); - - /* configure PM */ - pm_runtime_put_noidle(bus->dev); - pm_runtime_allow(bus->dev); - skl->init_done = 1; - - return; - -out_err: - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) - snd_hdac_display_power(bus, HDA_CODEC_IDX_CONTROLLER, false); -} - -/* - * constructor - */ -static int skl_create(struct pci_dev *pci, - struct skl_dev **rskl) -{ - struct hdac_ext_bus_ops *ext_ops = NULL; - struct skl_dev *skl; - struct hdac_bus *bus; - struct hda_bus *hbus; - int err; - - *rskl = NULL; - - err = pci_enable_device(pci); - if (err < 0) - return err; - - skl = devm_kzalloc(&pci->dev, sizeof(*skl), GFP_KERNEL); - if (!skl) { - pci_disable_device(pci); - return -ENOMEM; - } - - hbus = skl_to_hbus(skl); - bus = skl_to_bus(skl); - - INIT_LIST_HEAD(&skl->ppl_list); - INIT_LIST_HEAD(&skl->bind_list); - -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) - ext_ops = snd_soc_hdac_hda_get_ops(); -#endif - snd_hdac_ext_bus_init(bus, &pci->dev, NULL, ext_ops); - bus->use_posbuf = 1; - skl->pci = pci; - INIT_WORK(&skl->probe_work, skl_probe_work); - bus->bdl_pos_adj = 0; - - mutex_init(&hbus->prepare_mutex); - hbus->pci = pci; - hbus->mixer_assigned = -1; - hbus->modelname = "sklbus"; - - *rskl = skl; - - return 0; -} - -static int skl_first_init(struct hdac_bus *bus) -{ - struct skl_dev *skl = bus_to_skl(bus); - struct pci_dev *pci = skl->pci; - int err; - unsigned short gcap; - int cp_streams, pb_streams, start_idx; - - err = pci_request_regions(pci, "Skylake HD audio"); - if (err < 0) - return err; - - bus->addr = pci_resource_start(pci, 0); - bus->remap_addr = pci_ioremap_bar(pci, 0); - if (bus->remap_addr == NULL) { - dev_err(bus->dev, "ioremap error\n"); - return -ENXIO; - } - - snd_hdac_bus_parse_capabilities(bus); - - /* check if PPCAP exists */ - if (!bus->ppcap) { - dev_err(bus->dev, "bus ppcap not set, HDAudio or DSP not present?\n"); - return -ENODEV; - } - - if (skl_acquire_irq(bus, 0) < 0) - return -EBUSY; - - pci_set_master(pci); - synchronize_irq(bus->irq); - - gcap = snd_hdac_chip_readw(bus, GCAP); - dev_dbg(bus->dev, "chipset global capabilities = 0x%x\n", gcap); - - /* read number of streams from GCAP register */ - cp_streams = (gcap >> 8) & 0x0f; - pb_streams = (gcap >> 12) & 0x0f; - - if (!pb_streams && !cp_streams) { - dev_err(bus->dev, "no streams found in GCAP definitions?\n"); - return -EIO; - } - - bus->num_streams = cp_streams + pb_streams; - - /* allow 64bit DMA address if supported by H/W */ - if (dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(64))) - dma_set_mask_and_coherent(bus->dev, DMA_BIT_MASK(32)); - dma_set_max_seg_size(bus->dev, UINT_MAX); - - /* initialize streams */ - snd_hdac_ext_stream_init_all - (bus, 0, cp_streams, SNDRV_PCM_STREAM_CAPTURE); - start_idx = cp_streams; - snd_hdac_ext_stream_init_all - (bus, start_idx, pb_streams, SNDRV_PCM_STREAM_PLAYBACK); - - err = snd_hdac_bus_alloc_stream_pages(bus); - if (err < 0) - return err; - - return 0; -} - -static int skl_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) -{ - struct skl_dev *skl; - struct hdac_bus *bus = NULL; - int err; - - switch (skl_pci_binding) { - case SND_SKL_PCI_BIND_AUTO: - err = snd_intel_dsp_driver_probe(pci); - if (err != SND_INTEL_DSP_DRIVER_ANY && - err != SND_INTEL_DSP_DRIVER_SST) - return -ENODEV; - break; - case SND_SKL_PCI_BIND_LEGACY: - dev_info(&pci->dev, "Module parameter forced binding with HDAudio legacy, aborting probe\n"); - return -ENODEV; - case SND_SKL_PCI_BIND_ASOC: - dev_info(&pci->dev, "Module parameter forced binding with SKL driver, bypassed detection logic\n"); - break; - default: - dev_err(&pci->dev, "invalid value for skl_pci_binding module parameter, ignored\n"); - break; - } - - /* we use ext core ops, so provide NULL for ops here */ - err = skl_create(pci, &skl); - if (err < 0) - return err; - - bus = skl_to_bus(skl); - - err = skl_first_init(bus); - if (err < 0) { - dev_err(bus->dev, "skl_first_init failed with err: %d\n", err); - goto out_free; - } - - skl->pci_id = pci->device; - - device_disable_async_suspend(bus->dev); - - skl->nhlt = intel_nhlt_init(bus->dev); - - if (skl->nhlt == NULL) { -#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC) - dev_err(bus->dev, "no nhlt info found\n"); - err = -ENODEV; - goto out_free; -#else - dev_warn(bus->dev, "no nhlt info found, continuing to try to enable HDAudio codec\n"); -#endif - } else { - - err = skl_nhlt_create_sysfs(skl); - if (err < 0) { - dev_err(bus->dev, "skl_nhlt_create_sysfs failed with err: %d\n", err); - goto out_nhlt_free; - } - - skl_nhlt_update_topology_bin(skl); - - /* create device for dsp clk */ - err = skl_clock_device_register(skl); - if (err < 0) { - dev_err(bus->dev, "skl_clock_device_register failed with err: %d\n", err); - goto out_clk_free; - } - } - - pci_set_drvdata(skl->pci, bus); - - - err = skl_find_machine(skl, (void *)pci_id->driver_data); - if (err < 0) { - dev_err(bus->dev, "skl_find_machine failed with err: %d\n", err); - goto out_nhlt_free; - } - - err = skl_init_dsp(skl); - if (err < 0) { - dev_dbg(bus->dev, "error failed to register dsp\n"); - goto out_nhlt_free; - } - skl->enable_miscbdcge = skl_enable_miscbdcge; - skl->clock_power_gating = skl_clock_power_gating; - - if (bus->mlcap) - snd_hdac_ext_bus_get_ml_capabilities(bus); - - /* create device for soc dmic */ - err = skl_dmic_device_register(skl); - if (err < 0) { - dev_err(bus->dev, "skl_dmic_device_register failed with err: %d\n", err); - goto out_dsp_free; - } - - if (IS_ENABLED(CONFIG_SND_SOC_HDAC_HDMI)) { - err = snd_hdac_i915_init(bus); - if (err < 0) - goto out_dmic_unregister; - } - schedule_work(&skl->probe_work); - - return 0; - -out_dmic_unregister: - skl_dmic_device_unregister(skl); -out_dsp_free: - skl_free_dsp(skl); -out_clk_free: - skl_clock_device_unregister(skl); -out_nhlt_free: - if (skl->nhlt) - intel_nhlt_free(skl->nhlt); -out_free: - skl_free(bus); - - return err; -} - -static void skl_shutdown(struct pci_dev *pci) -{ - struct hdac_bus *bus = pci_get_drvdata(pci); - struct hdac_stream *s; - struct hdac_ext_stream *stream; - struct skl_dev *skl; - - if (!bus) - return; - - skl = bus_to_skl(bus); - - if (!skl->init_done) - return; - - snd_hdac_stop_streams(bus); - snd_hdac_ext_bus_link_power_down_all(bus); - skl_dsp_sleep(skl->dsp); - - list_for_each_entry(s, &bus->stream_list, list) { - stream = stream_to_hdac_ext_stream(s); - snd_hdac_ext_stream_decouple(bus, stream, false); - } - - snd_hdac_bus_stop_chip(bus); -} - -static void skl_remove(struct pci_dev *pci) -{ - struct hdac_bus *bus = pci_get_drvdata(pci); - struct skl_dev *skl = bus_to_skl(bus); - - cancel_work_sync(&skl->probe_work); - - pm_runtime_get_noresume(&pci->dev); - - /* codec removal, invoke bus_device_remove */ - snd_hdac_ext_bus_device_remove(bus); - - skl_platform_unregister(&pci->dev); - skl_free_dsp(skl); - skl_machine_device_unregister(skl); - skl_dmic_device_unregister(skl); - skl_clock_device_unregister(skl); - skl_nhlt_remove_sysfs(skl); - if (skl->nhlt) - intel_nhlt_free(skl->nhlt); - skl_free(bus); -} - -/* PCI IDs */ -static const struct pci_device_id skl_ids[] = { -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_SKL) - { PCI_DEVICE_DATA(INTEL, HDA_SKL_LP, &snd_soc_acpi_intel_skl_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) - { PCI_DEVICE_DATA(INTEL, HDA_APL, &snd_soc_acpi_intel_bxt_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_KBL) - { PCI_DEVICE_DATA(INTEL, HDA_KBL_LP, &snd_soc_acpi_intel_kbl_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) - { PCI_DEVICE_DATA(INTEL, HDA_GML, &snd_soc_acpi_intel_glk_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CNL) - { PCI_DEVICE_DATA(INTEL, HDA_CNL_LP, &snd_soc_acpi_intel_cnl_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CFL) - { PCI_DEVICE_DATA(INTEL, HDA_CNL_H, &snd_soc_acpi_intel_cnl_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) - { PCI_DEVICE_DATA(INTEL, HDA_CML_LP, &snd_soc_acpi_intel_cnl_machines) }, -#endif -#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) - { PCI_DEVICE_DATA(INTEL, HDA_CML_H, &snd_soc_acpi_intel_cnl_machines) }, -#endif - { 0, } -}; -MODULE_DEVICE_TABLE(pci, skl_ids); - -/* pci_driver definition */ -static struct pci_driver skl_driver = { - .name = KBUILD_MODNAME, - .id_table = skl_ids, - .probe = skl_probe, - .remove = skl_remove, - .shutdown = skl_shutdown, - .driver = { - .pm = &skl_pm, - }, -}; -module_pci_driver(skl_driver); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel Skylake ASoC HDA driver"); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h deleted file mode 100644 index f55f8b3dbdc3..000000000000 --- a/sound/soc/intel/skylake/skl.h +++ /dev/null @@ -1,207 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * skl.h - HD Audio skylake definitions. - * - * Copyright (C) 2015 Intel Corp - * Author: Jeeja KP - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - * - * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ - */ - -#ifndef __SOUND_SOC_SKL_H -#define __SOUND_SOC_SKL_H - -#include -#include -#include -#include -#include "skl-ssp-clk.h" -#include "skl-sst-ipc.h" - -#define SKL_SUSPEND_DELAY 2000 - -#define SKL_MAX_ASTATE_CFG 3 - -#define AZX_PCIREG_PGCTL 0x44 -#define AZX_PGCTL_LSRMD_MASK (1 << 4) -#define AZX_PGCTL_ADSPPGD BIT(2) -#define AZX_PCIREG_CGCTL 0x48 -#define AZX_CGCTL_MISCBDCGE_MASK (1 << 6) -#define AZX_CGCTL_ADSPDCGE BIT(1) -/* D0I3C Register fields */ -#define AZX_REG_VS_D0I3C_CIP 0x1 /* Command in progress */ -#define AZX_REG_VS_D0I3C_I3 0x4 /* D0i3 enable */ -#define SKL_MAX_DMACTRL_CFG 18 -#define DMA_CLK_CONTROLS 1 -#define DMA_TRANSMITION_START 2 -#define DMA_TRANSMITION_STOP 3 - -#define AZX_VS_EM2_DUM BIT(23) -#define AZX_REG_VS_EM2_L1SEN BIT(13) - -struct skl_debug; - -struct skl_astate_param { - u32 kcps; - u32 clk_src; -}; - -struct skl_astate_config { - u32 count; - struct skl_astate_param astate_table[]; -}; - -struct skl_fw_config { - struct skl_astate_config *astate_cfg; -}; - -struct skl_dev { - struct hda_bus hbus; - struct pci_dev *pci; - - unsigned int init_done:1; /* delayed init status */ - struct platform_device *dmic_dev; - struct platform_device *i2s_dev; - struct platform_device *clk_dev; - struct snd_soc_component *component; - struct snd_soc_dai_driver *dais; - - struct nhlt_acpi_table *nhlt; /* nhlt ptr */ - - struct list_head ppl_list; - struct list_head bind_list; - - const char *fw_name; - char tplg_name[64]; - unsigned short pci_id; - - int supend_active; - - struct work_struct probe_work; - - struct skl_debug *debugfs; - u8 nr_modules; - struct skl_module **modules; - bool use_tplg_pcm; - struct skl_fw_config cfg; - struct snd_soc_acpi_mach *mach; - - struct device *dev; - struct sst_dsp *dsp; - - /* boot */ - wait_queue_head_t boot_wait; - bool boot_complete; - - /* module load */ - wait_queue_head_t mod_load_wait; - bool mod_load_complete; - bool mod_load_status; - - /* IPC messaging */ - struct sst_generic_ipc ipc; - - /* callback for miscbdge */ - void (*enable_miscbdcge)(struct device *dev, bool enable); - /* Is CGCTL.MISCBDCGE disabled */ - bool miscbdcg_disabled; - - /* Populate module information */ - struct list_head uuid_list; - - /* Is firmware loaded */ - bool fw_loaded; - - /* first boot ? */ - bool is_first_boot; - - /* multi-core */ - struct skl_dsp_cores cores; - - /* library info */ - struct skl_lib_info lib_info[SKL_MAX_LIB]; - int lib_count; - - /* Callback to update D0i3C register */ - void (*update_d0i3c)(struct device *dev, bool enable); - - struct skl_d0i3_data d0i3; - - const struct skl_dsp_ops *dsp_ops; - - /* Callback to update dynamic clock and power gating registers */ - void (*clock_power_gating)(struct device *dev, bool enable); -}; - -#define skl_to_bus(s) (&(s)->hbus.core) -#define bus_to_skl(bus) container_of(bus, struct skl_dev, hbus.core) - -#define skl_to_hbus(s) (&(s)->hbus) -#define hbus_to_skl(hbus) container_of((hbus), struct skl_dev, (hbus)) - -/* to pass dai dma data */ -struct skl_dma_params { - u32 format; - u8 stream_tag; -}; - -struct skl_machine_pdata { - bool use_tplg_pcm; /* use dais and dai links from topology */ -}; - -struct skl_dsp_ops { - int id; - unsigned int num_cores; - struct skl_dsp_loader_ops (*loader_ops)(void); - int (*init)(struct device *dev, void __iomem *mmio_base, - int irq, const char *fw_name, - struct skl_dsp_loader_ops loader_ops, - struct skl_dev **skl_sst); - int (*init_fw)(struct device *dev, struct skl_dev *skl); - void (*cleanup)(struct device *dev, struct skl_dev *skl); -}; - -int skl_platform_unregister(struct device *dev); -int skl_platform_register(struct device *dev); - -int skl_nhlt_update_topology_bin(struct skl_dev *skl); -int skl_init_dsp(struct skl_dev *skl); -int skl_free_dsp(struct skl_dev *skl); -int skl_suspend_late_dsp(struct skl_dev *skl); -int skl_suspend_dsp(struct skl_dev *skl); -int skl_resume_dsp(struct skl_dev *skl); -void skl_cleanup_resources(struct skl_dev *skl); -const struct skl_dsp_ops *skl_get_dsp_ops(int pci_id); -void skl_update_d0i3c(struct device *dev, bool enable); -int skl_nhlt_create_sysfs(struct skl_dev *skl); -void skl_nhlt_remove_sysfs(struct skl_dev *skl); -void skl_get_clks(struct skl_dev *skl, struct skl_ssp_clk *ssp_clks); -struct skl_clk_parent_src *skl_get_parent_clk(u8 clk_id); -int skl_dsp_set_dma_control(struct skl_dev *skl, u32 *caps, - u32 caps_size, u32 node_id); - -struct skl_module_cfg; - -#ifdef CONFIG_DEBUG_FS -struct skl_debug *skl_debugfs_init(struct skl_dev *skl); -void skl_debugfs_exit(struct skl_dev *skl); -void skl_debug_init_module(struct skl_debug *d, - struct snd_soc_dapm_widget *w, - struct skl_module_cfg *mconfig); -#else -static inline struct skl_debug *skl_debugfs_init(struct skl_dev *skl) -{ - return NULL; -} - -static inline void skl_debugfs_exit(struct skl_dev *skl) -{} - -static inline void skl_debug_init_module(struct skl_debug *d, - struct snd_soc_dapm_widget *w, - struct skl_module_cfg *mconfig) -{} -#endif - -#endif /* __SOUND_SOC_SKL_H */ From 526139aff1d14c5a2cc0a769c063f439444c61c2 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski Date: Wed, 14 Aug 2024 10:39:29 +0200 Subject: [PATCH 228/410] ASoC: Intel: avs: Enable by default for all SST configurations The skylake-driver is deprecated in favour of the avs-driver. As the latter supports all configurations of its predecessor and more, update the existing selection mechanism to acknowledge the SST flag. Acked-by: Andy Shevchenko Signed-off-by: Cezary Rojewski Link: https://patch.msgid.link/20240814083929.1217319-15-cezary.rojewski@intel.com Signed-off-by: Mark Brown --- sound/soc/intel/avs/core.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index f2dc82a2abc7..da7bac09acb4 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -422,8 +422,14 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) int ret; ret = snd_intel_dsp_driver_probe(pci); - if (ret != SND_INTEL_DSP_DRIVER_ANY && ret != SND_INTEL_DSP_DRIVER_AVS) + switch (ret) { + case SND_INTEL_DSP_DRIVER_ANY: + case SND_INTEL_DSP_DRIVER_SST: + case SND_INTEL_DSP_DRIVER_AVS: + break; + default: return -ENODEV; + } ret = pcim_enable_device(pci); if (ret < 0) From b29ba8f1f9429b775a8b901364a21291588c2a23 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Mon, 19 Aug 2024 12:37:36 +0000 Subject: [PATCH 229/410] ALSA: hda/realtek: Convert existing CS35L56 products to use autodetect fixup function The existing CS35L56 products can make use of the fixup function that works out the component binding details so we can remove the fixed configuration fixup functions. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240819123736.111946-1-simont@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 47 +++++------------------------------ 1 file changed, 6 insertions(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d569626f78a..c573183c69a9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6970,26 +6970,6 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } -static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) -{ - comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); -} - -static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) -{ - comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); -} - -static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) -{ - comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); -} - -static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) -{ - comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); -} - static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { /* @@ -6997,7 +6977,7 @@ static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fix * different codecs and the newer GA403U has a ALC285. */ if (cdc->core.vendor_id == 0x10ec0285) - cs35l56_fixup_i2c_two(cdc, fix, action); + cs35lxx_autodet_fixup(cdc, fix, action); else alc_fixup_inv_dmic(cdc, fix, action); } @@ -7599,9 +7579,6 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, - ALC285_FIXUP_CS35L56_SPI_2, - ALC285_FIXUP_CS35L56_I2C_2, - ALC285_FIXUP_CS35L56_I2C_4, ALC285_FIXUP_ASUS_GA403U, ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC, ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1, @@ -9867,7 +9844,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, - .v.func = cs35l56_fixup_spi_four, + .v.func = cs35lxx_autodet_fixup, .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, @@ -9883,18 +9860,6 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, - [ALC285_FIXUP_CS35L56_SPI_2] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35l56_fixup_spi_two, - }, - [ALC285_FIXUP_CS35L56_I2C_2] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35l56_fixup_i2c_two, - }, - [ALC285_FIXUP_CS35L56_I2C_4] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35l56_fixup_i2c_four, - }, [ALC285_FIXUP_ASUS_GA403U] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_asus_ga403u, @@ -9923,7 +9888,7 @@ static const struct hda_fixup alc269_fixups[] = { { } }, .chained = true, - .chain_id = ALC285_FIXUP_CS35L56_SPI_2 + .chain_id = ALCXXX_FIXUP_CS35LXX }, [ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, @@ -10456,14 +10421,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALCXXX_FIXUP_CS35LXX), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), - SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), - SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALCXXX_FIXUP_CS35LXX), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALCXXX_FIXUP_CS35LXX), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), From e6c1d9068295796e34d59ef08fa80f6ff8f3530a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 18 Aug 2024 19:30:37 +0200 Subject: [PATCH 230/410] ASoC: dt-bindings: samsung,odroid: drop stale clocks Clocks property was present only to allow usage of assigned-clocks in the sound card node, however in upstream DTS the assigned-clocks were moved in commit 4afb06afd768 ("ARM: dts: exynos: move assigned-clock* properties to i2s0 node in Odroid XU4") to respective I2S nodes. Linux drivers never parsed "clocks" so it can be safely dropped. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240818173037.122152-1-krzysztof.kozlowski@linaro.org Acked-by: Rob Herring (Arm) Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/samsung,odroid.yaml | 5 ----- 1 file changed, 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index b77284e3e26a..c3dea852cc8d 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -27,11 +27,6 @@ properties: - const: samsung,odroid-xu4-audio deprecated: true - assigned-clock-parents: true - assigned-clock-rates: true - assigned-clocks: true - clocks: true - cpu: type: object additionalProperties: false From 5f83ee4b1f0c04a8b4125daab8918606df3dc035 Mon Sep 17 00:00:00 2001 From: Dmitry Torokhov Date: Mon, 19 Aug 2024 18:13:47 -0700 Subject: [PATCH 231/410] ASoC: tas5086: use sleeping variants of gpiod API The driver does not access reset GPIO in atomic contexts so it is usable with GPIOs that may sleep during access. Switch to using gpiod_set_value_cansleep(). Also the reset GPIO is configured as output at the time it is acquired, there is no need to use gpiod_direction_output() when executing reset sequence. Signed-off-by: Dmitry Torokhov Link: https://patch.msgid.link/ZsPty8oNMQk4YTl1@google.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 4bc1fdd232bb..b97c0e885713 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -463,9 +463,9 @@ static void tas5086_reset(struct tas5086_private *priv) { if (priv->reset) { /* Reset codec - minimum assertion time is 400ns */ - gpiod_direction_output(priv->reset, 1); + gpiod_set_value_cansleep(priv->reset, 1); udelay(1); - gpiod_set_value(priv->reset, 0); + gpiod_set_value_cansleep(priv->reset, 0); /* Codec needs ~15ms to wake up */ msleep(15); @@ -866,9 +866,10 @@ static void tas5086_remove(struct snd_soc_component *component) { struct tas5086_private *priv = snd_soc_component_get_drvdata(component); - if (priv->reset) + if (priv->reset) { /* Set codec to the reset state */ - gpiod_set_value(priv->reset, 1); + gpiod_set_value_cansleep(priv->reset, 1); + } regulator_bulk_disable(ARRAY_SIZE(priv->supplies), priv->supplies); }; From 1004f34d4f4a59aa5508c3b96069759efa738544 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 19 Aug 2024 11:43:29 +0530 Subject: [PATCH 232/410] ASoC: amd: acp: replace desc->rev check with acp pci revision id Replace acp descriptor structure member 'rev' check with acp pci revision id. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240819061329.1025189-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index 0f6115c8b005..e4d46fdda88b 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -236,7 +236,6 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, unsigned int image_length) { struct snd_sof_dev *sdev = adata->dev; - const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int tx_count, fw_qualifier, val; int ret; @@ -265,8 +264,9 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_DMA_DESTINATION_ADDR, dest_addr); snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SHA_MSG_LENGTH, image_length); - /* psp_send_cmd only required for vangogh platform (rev - 5) */ - if (desc->rev == 5 && !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { + /* psp_send_cmd only required for vangogh platform */ + if (adata->pci_rev == ACP_VANGOGH_PCI_ID && + !(adata->quirks && adata->quirks->skip_iram_dram_size_mod)) { /* Modify IRAM and DRAM size */ ret = psp_send_cmd(adata, MBOX_ACP_IRAM_DRAM_FENCE_COMMAND | IRAM_DRAM_FENCE_2); if (ret) @@ -285,8 +285,8 @@ int configure_and_run_sha_dma(struct acp_dev_data *adata, void *image_addr, return ret; } - /* psp_send_cmd only required for renoir platform (rev - 3) */ - if (desc->rev == 3) { + /* psp_send_cmd only required for renoir platform*/ + if (adata->pci_rev == ACP_RN_PCI_ID) { ret = psp_send_cmd(adata, MBOX_ACP_SHA_DMA_COMMAND); if (ret) return ret; @@ -405,7 +405,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->ext_intr_stat, ACP_ERROR_IRQ_MASK); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_sw0_i2s_err_reason, 0); /* ACP_SW1_I2S_ERROR_REASON is newly added register from rmb platform onwards */ - if (desc->rev >= 6) + if (adata->pci_rev >= ACP_RMB_PCI_ID) snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_SW1_I2S_ERROR_REASON, 0); snd_sof_dsp_write(sdev, ACP_DSP_BAR, desc->acp_error_stat, 0); irq_flag = 1; @@ -431,6 +431,7 @@ static irqreturn_t acp_irq_handler(int irq, void *dev_id) static int acp_power_on(struct snd_sof_dev *sdev) { const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *adata = sdev->pdata->hw_pdata; unsigned int base = desc->pgfsm_base; unsigned int val; unsigned int acp_pgfsm_status_mask, acp_pgfsm_cntl_mask; @@ -441,13 +442,14 @@ static int acp_power_on(struct snd_sof_dev *sdev) if (val == ACP_POWERED_ON) return 0; - switch (desc->rev) { - case 3: - case 5: + switch (adata->pci_rev) { + case ACP_RN_PCI_ID: + case ACP_VANGOGH_PCI_ID: acp_pgfsm_status_mask = ACP3X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP3X_PGFSM_CNTL_POWER_ON_MASK; break; - case 6: + case ACP_RMB_PCI_ID: + case ACP63_PCI_ID: acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; break; From aaf55d12fb51d7aa64abe51e27acac1d3d1853ec Mon Sep 17 00:00:00 2001 From: Frank Li Date: Tue, 20 Aug 2024 14:46:03 -0400 Subject: [PATCH 233/410] ASoC: dt-bindings: Convert tpa6130a2.txt to yaml Convert binding doc tpa6130a2.txt to yaml format. Additional change: - add ref to dai-common.yaml - add i2c node in example Fix below warning: arch/arm64/boot/dts/freescale/imx8mq-zii-ultra-rmb3.dtb: /soc@0/bus@30800000/i2c@30a20000/amp@60: failed to match any schema with compatible: ['ti,tpa6130a2'] Reviewed-by: Rob Herring (Arm) Signed-off-by: Frank Li Link: https://patch.msgid.link/20240820184604.499017-1-Frank.Li@nxp.com Signed-off-by: Mark Brown --- .../bindings/sound/ti,tpa6130a2.yaml | 55 +++++++++++++++++++ .../devicetree/bindings/sound/tpa6130a2.txt | 27 --------- MAINTAINERS | 2 +- 3 files changed, 56 insertions(+), 28 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml new file mode 100644 index 000000000000..a42bf9bde694 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml @@ -0,0 +1,55 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tpa6130a2.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments - tpa6130a2 Codec module + +maintainers: + - Sebastian Reichel + +description: + Stereo, analog input headphone amplifier + +properties: + compatible: + enum: + - ti,tpa6130a2 + - ti,tpa6140a2 + + reg: + maxItems: 1 + + Vdd-supply: + description: power supply regulator + + power-gpio: + description: gpio pin to power the device + +required: + - compatible + - reg + - Vdd-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + #include + + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@60 { + compatible = "ti,tpa6130a2"; + reg = <0x60>; + Vdd-supply = <&vmmc2>; + power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; + }; + }; + diff --git a/Documentation/devicetree/bindings/sound/tpa6130a2.txt b/Documentation/devicetree/bindings/sound/tpa6130a2.txt deleted file mode 100644 index 6dfa740e4b2d..000000000000 --- a/Documentation/devicetree/bindings/sound/tpa6130a2.txt +++ /dev/null @@ -1,27 +0,0 @@ -Texas Instruments - tpa6130a2 Codec module - -The tpa6130a2 serial control bus communicates through I2C protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tpa6130a2" - TPA6130A2 - "ti,tpa6140a2" - TPA6140A2 - - -- reg - - I2C slave address - -- Vdd-supply - - power supply regulator - -Optional properties: - -- power-gpio - gpio pin to power the device - -Example: - -tpa6130a2: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - reg = <0x60>; - Vdd-supply = <&vmmc2>; - power-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; -}; diff --git a/MAINTAINERS b/MAINTAINERS index 446c375617fb..1d12f3fca229 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22579,12 +22579,12 @@ F: Documentation/devicetree/bindings/sound/tas2552.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h From 6f6d8b2d49299492e704030632ab79257685e5d3 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Wed, 21 Aug 2024 12:49:27 +0100 Subject: [PATCH 234/410] ASoC: codecs: wcd934x: make read-only array minCode_param static const Don't populate the read-only array minCode_param on the stack at run time, instead make it static const. Signed-off-by: Colin Ian King Link: https://patch.msgid.link/20240821114927.520193-1-colin.i.king@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 291d0c80a6fc..910852eb9698 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2643,8 +2643,8 @@ static void wcd934x_mbhc_get_result_params(struct wcd934x_codec *wcd934x, s16 c1; s32 x1, d1; int32_t denom; - int minCode_param[] = { - 3277, 1639, 820, 410, 205, 103, 52, 26 + static const int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 }; regmap_update_bits(wcd934x->regmap, WCD934X_ANA_MBHC_ZDET, 0x20, 0x20); From 8a8dcf702673787543173b8ac5dafae2f7f13e87 Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Wed, 21 Aug 2024 15:25:27 +0800 Subject: [PATCH 235/410] ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove Remove unnecessary line feed for tasdevice_codec_remove. Add comma at the end the last member of the array. Signed-off-by: Baojun Xu Link: https://patch.msgid.link/20240821072527.1294-1-baojun.xu@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..eb8732b1bb99 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -582,13 +582,13 @@ static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -730,8 +730,7 @@ static void tasdevice_deinit(void *context) tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; } -static void tasdevice_codec_remove( - struct snd_soc_component *codec) +static void tasdevice_codec_remove(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); From 2d3b218d383e24623070f4439a0af64d200eb740 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Aug 2024 02:23:07 +0000 Subject: [PATCH 236/410] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested() soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87msl6aa2c.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f79abcf36105..baf8b81e7110 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } From 61c80c77b4f35e229347551d13e265752f067151 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 21 Aug 2024 12:16:50 +0530 Subject: [PATCH 237/410] ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure Remove unused structure member 'rev' from sof_amd_acp_desc structure. Signed-off-by: Vijendar Mukunda Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240821064650.2850310-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp.h | 1 - sound/soc/sof/amd/pci-acp63.c | 1 - sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 1 - sound/soc/sof/amd/pci-vangogh.c | 1 - 5 files changed, 5 deletions(-) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 321c40cc6d50..11def07efc0f 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -190,7 +190,6 @@ struct acp_dsp_stream { }; struct sof_amd_acp_desc { - unsigned int rev; const char *name; unsigned int host_bridge_id; u32 pgfsm_base; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index e90658ba2bd7..b54ed61b79ed 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -28,7 +28,6 @@ #define ACP6x_REG_END 0x125C000 static const struct sof_amd_acp_desc acp63_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_ACP63, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_enb = ACP6X_EXTERNAL_INTR_ENB, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index a366f904e6f3..c45256bf4fda 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -28,7 +28,6 @@ #define ACP6X_FUTURE_REG_ACLK_0 0x1854 static const struct sof_amd_acp_desc rembrandt_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_RMB, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 2b7c53470ce8..386a0f1e7ee0 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -28,7 +28,6 @@ #define ACP3X_FUTURE_REG_ACLK_0 0x1860 static const struct sof_amd_acp_desc renoir_chip_info = { - .rev = 3, .host_bridge_id = HOST_BRIDGE_CZN, .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index eba580840100..cb845f81795e 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -26,7 +26,6 @@ #define ACP5X_FUTURE_REG_ACLK_0 0x1864 static const struct sof_amd_acp_desc vangogh_chip_info = { - .rev = 5, .name = "vangogh", .host_bridge_id = HOST_BRIDGE_VGH, .pgfsm_base = ACP5X_PGFSM_BASE, From 23618f5b630a1dde8c465150ddb2fd308b686b08 Mon Sep 17 00:00:00 2001 From: Wu Bo Date: Thu, 22 Aug 2024 03:52:49 -0600 Subject: [PATCH 238/410] ASoC: dwc: change to use devm_clk_get_enabled() helpers Make the code cleaner and avoid call clk_disable_unprepare() Signed-off-by: Wu Bo Link: https://patch.msgid.link/20240822095249.1642713-1-bo.wu@vivo.com Signed-off-by: Mark Brown --- sound/soc/dwc/dwc-i2s.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index c04466f5492e..e6a5eebe5bc9 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -995,16 +995,12 @@ static int dw_i2s_probe(struct platform_device *pdev) goto err_assert_reset; } } - dev->clk = devm_clk_get(&pdev->dev, clk_id); + dev->clk = devm_clk_get_enabled(&pdev->dev, clk_id); if (IS_ERR(dev->clk)) { ret = PTR_ERR(dev->clk); goto err_assert_reset; } - - ret = clk_prepare_enable(dev->clk); - if (ret < 0) - goto err_assert_reset; } dev_set_drvdata(&pdev->dev, dev); @@ -1012,7 +1008,7 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "not able to register dai\n"); - goto err_clk_disable; + goto err_assert_reset; } if (!pdata || dev->is_jh7110) { @@ -1030,16 +1026,13 @@ static int dw_i2s_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "could not register pcm: %d\n", ret); - goto err_clk_disable; + goto err_assert_reset; } } pm_runtime_enable(&pdev->dev); return 0; -err_clk_disable: - if (dev->capability & DW_I2S_MASTER) - clk_disable_unprepare(dev->clk); err_assert_reset: reset_control_assert(dev->reset); return ret; @@ -1049,9 +1042,6 @@ static void dw_i2s_remove(struct platform_device *pdev) { struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); - if (dev->capability & DW_I2S_MASTER) - clk_disable_unprepare(dev->clk); - reset_control_assert(dev->reset); pm_runtime_disable(&pdev->dev); } From 1a9e3b0af301413210319e6946fb4b0b1ad71ccc Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 22 Aug 2024 14:32:02 +0800 Subject: [PATCH 239/410] ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused In case of tas2781, tas2563_dvc_table will be unused, so mark it as __maybe_unused. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240822063205.662-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ From fd69dfe6789f4ed46d1fdb52e223cff83946d997 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Aug 2024 02:13:56 +0000 Subject: [PATCH 240/410] ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition I have been wondering why DPCM needs special flag (= dpcm_playback/capture) to use it. Below is the history why it was added to ASoC. (A) In beginning, there was no dpcm_xxx flag on ASoC. It checks channels_min for DPCM, same as current non-DPCM. Let's name it as "validation check" here. if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { if (cpu_dai->driver->playback.channels_min) playback = 1; if (cpu_dai->driver->capture.channels_min) capture = 1; (B) commit 1e9de42f4324 ("ASoC: dpcm: Explicitly set BE DAI link supported stream directions") force to use dpcm_xxx flag on DPCM. According to this commit log, this is because "Some BE dummy DAI doesn't set channels_min for playback/capture". But we don't know which DAI is it, and not know why it can't/don't have channels_min. Let's name it as "no_chan_DAI" here. According to the code and git-log, it is used as DCPM-BE and is CPU DAI. I think the correct solution was set channels_min on "no_chan_DAI" side, not update ASoC framework side. But everything is under smoke today. if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { playback = rtd->dai_link->dpcm_playback; capture = rtd->dai_link->dpcm_capture; (C) commit 9b5db059366a ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") checks channels_min (= validation check) again. Because DPCM availability was handled by dpcm_xxx flag at that time, but some Sound Card set it even though it wasn't available. Clearly there's a contradiction here. I think correct solution was update Sound Card side instead of ASoC framework. Sound Card side will be updated to handle this issue later (commit 25612477d20b ("ASoC: soc-dai: set dai_link dpcm_ flags with a helper")) if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) { ... playback = rtd->dai_link->dpcm_playback && snd_soc_dai_stream_valid(cpu_dai, ...); capture = rtd->dai_link->dpcm_capture && snd_soc_dai_stream_valid(cpu_dai, ...); This (C) patch should have broken "no_chan_DAI" which doesn't have channels_min, but there was no such report during this 4 years. Possibilities case are as follows - No one is using "no_chan_DAI" - "no_chan_DAI" is no longer exist : was removed ? - "no_chan_DAI" is no longer exist : has channels_min ? Because of these history, this dpcm_xxx is unneeded flag today. But because we have been used it for 10 years since (B), it may have been used differently. For example some DAI available both playback/capture, but it set dpcm_playback flag only, in this case dpcm_xxx flag is used as availability limitation. We can use playback_only flag instead in this case, but it is very difficult to find such DAI today. Let's add grace time to remove dpcm_playback/capture flag. This patch don't use dpcm_xxx flag anymore, and indicates warning to use xxx_only flag if both playback/capture were available but using only one of dpcm_xxx flag, and not using xxx_only flag. Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Kuninori Morimoto Tested-by: Jerome Brunet Link: https://patch.msgid.link/87seuyaahn.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 1 + sound/soc/soc-pcm.c | 65 ++++++++++++++++++++++++++------------------- 2 files changed, 38 insertions(+), 28 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index e844f6afc5b5..81ef380b2e06 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -815,6 +815,7 @@ struct snd_soc_dai_link { /* This DAI link can route to other DAI links at runtime (Frontend)*/ unsigned int dynamic:1; + /* REMOVE ME */ /* DPCM capture and Playback support */ unsigned int dpcm_capture:1; unsigned int dpcm_playback:1; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index baf8b81e7110..c4834c741fa4 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2739,6 +2739,7 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai; + struct snd_soc_dai_link_ch_map *ch_maps; int has_playback = 0; int has_capture = 0; int i; @@ -2749,43 +2750,51 @@ static int soc_get_playback_capture(struct snd_soc_pcm_runtime *rtd, } if (dai_link->dynamic || dai_link->no_pcm) { - int stream; - if (dai_link->dpcm_playback) { - stream = SNDRV_PCM_STREAM_PLAYBACK; + for_each_rtd_ch_maps(rtd, i, ch_maps) { + cpu_dai = snd_soc_rtd_to_cpu(rtd, ch_maps->cpu); - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (snd_soc_dai_stream_valid(cpu_dai, stream)) { - has_playback = 1; - break; - } - } - if (!has_playback) { - dev_err(rtd->card->dev, - "No CPU DAIs support playback for stream %s\n", - dai_link->stream_name); - return -EINVAL; - } + if (snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + has_playback = 1; + + if (snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + has_capture = 1; } - if (dai_link->dpcm_capture) { - stream = SNDRV_PCM_STREAM_CAPTURE; - for_each_rtd_cpu_dais(rtd, i, cpu_dai) { - if (snd_soc_dai_stream_valid(cpu_dai, stream)) { - has_capture = 1; - break; - } + /* + * REMOVE ME + * + * dpcm_xxx flag will be removed soon, Indicates warning if dpcm_xxx flag was used + * as availability limitation + */ + if (has_playback && has_capture) { + if ( dai_link->dpcm_playback && + !dai_link->dpcm_capture && + !dai_link->playback_only) { + dev_warn(rtd->card->dev, + "both playback/capture are available," + " but not using playback_only flag (%s)\n", + dai_link->stream_name); + dev_warn(rtd->card->dev, + "dpcm_playback/capture are no longer needed," + " please use playback/capture_only instead\n"); + has_capture = 0; } - if (!has_capture) { - dev_err(rtd->card->dev, - "No CPU DAIs support capture for stream %s\n", - dai_link->stream_name); - return -EINVAL; + if (!dai_link->dpcm_playback && + dai_link->dpcm_capture && + !dai_link->capture_only) { + dev_warn(rtd->card->dev, + "both playback/capture are available," + " but not using capture_only flag (%s)\n", + dai_link->stream_name); + dev_warn(rtd->card->dev, + "dpcm_playback/capture are no longer needed," + " please use playback/capture_only instead\n"); + has_playback = 0; } } } else { - struct snd_soc_dai_link_ch_map *ch_maps; struct snd_soc_dai *codec_dai; /* Adapt stream for codec2codec links */ From 12806510481497a01d01edd64d7bb53a4d9ec28d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Aug 2024 02:14:02 +0000 Subject: [PATCH 241/410] ASoC: remove snd_soc_dai_link_set_capabilities() dpcm_xxx flags are no longer needed. We need to use xxx_only flags instead if needed, but snd_soc_dai_link_set_capabilities() user adds dpcm_xxx if playback/capture were available. Thus converting dpcm_xxx to xxx_only is not needed. Just remove it. Signed-off-by: Kuninori Morimoto Tested-by: Jerome Brunet Link: https://patch.msgid.link/87r0aiaahh.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dai.h | 1 - sound/soc/fsl/imx-card.c | 3 --- sound/soc/generic/audio-graph-card.c | 2 -- sound/soc/generic/audio-graph-card2.c | 2 -- sound/soc/generic/simple-card.c | 2 -- sound/soc/meson/axg-card.c | 1 - sound/soc/meson/gx-card.c | 1 - sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 --------------------------- 9 files changed, 51 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index ab4e109fe71d..0d1b215f24f4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -219,7 +219,6 @@ void snd_soc_dai_resume(struct snd_soc_dai *dai); int snd_soc_dai_compress_new(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd, int num); bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int stream); -void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link); void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action); static inline void snd_soc_dai_activate(struct snd_soc_dai *dai, diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 0e18ccabe28c..98b37dd2b901 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -650,9 +650,6 @@ static int imx_card_parse_of(struct imx_card_data *data) link->ops = &imx_aif_ops; } - if (link->no_pcm || link->dynamic) - snd_soc_dai_link_set_capabilities(link); - /* Get dai fmt */ ret = simple_util_parse_daifmt(dev, np, codec, NULL, &link->dai_fmt); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3425fbbcbd7e..1bdcfc4d4222 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -279,8 +279,6 @@ static int graph_dai_link_of_dpcm(struct simple_util_priv *priv, graph_parse_convert(dev, ep, &dai_props->adata); - snd_soc_dai_link_set_capabilities(dai_link); - ret = graph_link_init(priv, cpu_ep, codec_ep, li, dai_name); li->link++; diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 56f7f946882e..051adb567397 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -966,8 +966,6 @@ int audio_graph2_link_dpcm(struct simple_util_priv *priv, graph_parse_convert(ep, dai_props); /* at node of */ graph_parse_convert(rep, dai_props); /* at node of */ - snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, lnk, cpu_port, codec_port, li, is_cpu); err: of_node_put(ep); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index d2588f1ea54e..42c60b92cca5 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -291,8 +291,6 @@ static int simple_dai_link_of_dpcm(struct simple_util_priv *priv, simple_parse_convert(dev, np, &dai_props->adata); - snd_soc_dai_link_set_capabilities(dai_link); - ret = simple_link_init(priv, np, codec, li, prefix, dai_name); out_put_node: diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 8c5605c1e34e..09aa36e94c85 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -339,7 +339,6 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->num_c2c_params = 1; } else { dai_link->no_pcm = 1; - snd_soc_dai_link_set_capabilities(dai_link); if (axg_card_cpu_is_tdm_iface(dai_link->cpus->of_node)) ret = axg_card_parse_tdm(card, np, index); } diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index f1539e542638..7edca3e49c8f 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -107,7 +107,6 @@ static int gx_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->num_c2c_params = 1; } else { dai_link->no_pcm = 1; - snd_soc_dai_link_set_capabilities(dai_link); /* Check if the cpu is the i2s encoder and parse i2s data */ if (gx_card_cpu_identify(dai_link->cpus, "I2S Encoder")) ret = gx_card_parse_i2s(card, np, index); diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 56b4a3654aec..928cf5cb5999 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -155,7 +155,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card) if (platform || !codec) { /* DPCM */ - snd_soc_dai_link_set_capabilities(link); link->ignore_suspend = 1; link->nonatomic = 1; } diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 302ca753a1f3..4e08892d24c6 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -479,44 +479,6 @@ bool snd_soc_dai_stream_valid(const struct snd_soc_dai *dai, int dir) return stream->channels_min; } -/* - * snd_soc_dai_link_set_capabilities() - set dai_link properties based on its DAIs - */ -void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) -{ - bool supported[SNDRV_PCM_STREAM_LAST + 1]; - int direction; - - for_each_pcm_streams(direction) { - struct snd_soc_dai_link_component *cpu; - struct snd_soc_dai_link_component *codec; - struct snd_soc_dai *dai; - bool supported_cpu = false; - bool supported_codec = false; - int i; - - for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai_with_mutex(cpu); - if (dai && snd_soc_dai_stream_valid(dai, direction)) { - supported_cpu = true; - break; - } - } - for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai_with_mutex(codec); - if (dai && snd_soc_dai_stream_valid(dai, direction)) { - supported_codec = true; - break; - } - } - supported[direction] = supported_cpu && supported_codec; - } - - dai_link->dpcm_playback = supported[SNDRV_PCM_STREAM_PLAYBACK]; - dai_link->dpcm_capture = supported[SNDRV_PCM_STREAM_CAPTURE]; -} -EXPORT_SYMBOL_GPL(snd_soc_dai_link_set_capabilities); - void snd_soc_dai_action(struct snd_soc_dai *dai, int stream, int action) { From 46fb727a28d8c7195f915150a669d927d463069b Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Wed, 21 Aug 2024 02:14:09 +0000 Subject: [PATCH 242/410] ASoC: amlogic: do not use dpcm_playback/capture flags dpcm_playback/capture flags are being deprecated in ASoC. Use playback/capture_only flags instead Suggested-by: Kuninori Morimoto Signed-off-by: Jerome Brunet Signed-off-by: Kuninori Morimoto Tested-by: Jerome Brunet Link: https://patch.msgid.link/87plq2aahb.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 10 +++++----- sound/soc/meson/meson-card-utils.c | 4 ++-- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 09aa36e94c85..646ab87afac2 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -132,7 +132,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, lb->stream_name = lb->name; lb->cpus->of_node = pad->cpus->of_node; lb->cpus->dai_name = "TDM Loopback"; - lb->dpcm_capture = 1; + lb->capture_only = 1; lb->no_pcm = 1; lb->ops = &axg_card_tdm_be_ops; lb->init = axg_card_tdm_dai_lb_init; @@ -176,7 +176,7 @@ static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, /* Disable playback is the interface has no tx slots */ if (!tx) - link->dpcm_playback = 0; + link->capture_only = 1; for (i = 0, rx = 0; i < AXG_TDM_NUM_LANES; i++) { snprintf(propname, 32, "dai-tdm-slot-rx-mask-%d", i); @@ -186,9 +186,9 @@ static int axg_card_parse_cpu_tdm_slots(struct snd_soc_card *card, /* Disable capture is the interface has no rx slots */ if (!rx) - link->dpcm_capture = 0; + link->playback_only = 1; - /* ... but the interface should at least have one of them */ + /* ... but the interface should at least have one direction */ if (!tx && !rx) { dev_err(card->dev, "tdm link has no cpu slots\n"); return -EINVAL; @@ -275,7 +275,7 @@ static int axg_card_parse_tdm(struct snd_soc_card *card, return ret; /* Add loopback if the pad dai has playback */ - if (link->dpcm_playback) { + if (!link->capture_only) { ret = axg_card_add_tdm_loopback(card, index); if (ret) return ret; diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index ed6c7e2f609c..1a4ef124e4e2 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -186,9 +186,9 @@ int meson_card_set_fe_link(struct snd_soc_card *card, link->dpcm_merged_rate = 1; if (is_playback) - link->dpcm_playback = 1; + link->playback_only = 1; else - link->dpcm_capture = 1; + link->capture_only = 1; return meson_card_set_link_name(card, link, node, "fe"); } From 61e1f74f739546415570ccc1ac14e1b26afe4705 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 21 Aug 2024 02:14:15 +0000 Subject: [PATCH 243/410] ASoC: Intel: sof_sdw: use playback/capture_only flags Prepare for removal of dpcm_playback and dpcm_capture flags in dailinks. [Kuninori adjusted Pierre-Louis's patch] Signed-off-by: Pierre-Louis Bossart Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87o75maah5.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/sdw_utils/soc_sdw_utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 6183629d1754..e8d0f199155d 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -950,8 +950,8 @@ void asoc_sdw_init_dai_link(struct device *dev, struct snd_soc_dai_link *dai_lin dai_links->num_cpus = cpus_num; dai_links->codecs = codecs; dai_links->num_codecs = codecs_num; - dai_links->dpcm_playback = playback; - dai_links->dpcm_capture = capture; + dai_links->playback_only = playback && !capture; + dai_links->capture_only = !playback && capture; dai_links->init = init; dai_links->ops = ops; } From 90dc34da02aca2fe529ae1ed1822e8fc2a0d9ebc Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Thu, 22 Aug 2024 15:55:35 +0100 Subject: [PATCH 244/410] ASoC: cs35l56: Make struct regmap_config const It's now possible to declare instances of struct regmap_config as const data. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20240822145535.336407-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- include/sound/cs35l56.h | 6 +++--- sound/soc/codecs/cs35l56-shared.c | 6 +++--- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index a51acefa785f..94e8185c4795 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -282,9 +282,9 @@ static inline bool cs35l56_is_otp_register(unsigned int reg) return (reg >> 16) == 3; } -extern struct regmap_config cs35l56_regmap_i2c; -extern struct regmap_config cs35l56_regmap_spi; -extern struct regmap_config cs35l56_regmap_sdw; +extern const struct regmap_config cs35l56_regmap_i2c; +extern const struct regmap_config cs35l56_regmap_spi; +extern const struct regmap_config cs35l56_regmap_sdw; extern const struct cirrus_amp_cal_controls cs35l56_calibration_controls; diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index 3744a7c8cea5..e45e9ae01bc6 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -916,7 +916,7 @@ const unsigned int cs35l56_tx_input_values[] = { }; EXPORT_SYMBOL_NS_GPL(cs35l56_tx_input_values, SND_SOC_CS35L56_SHARED); -struct regmap_config cs35l56_regmap_i2c = { +const struct regmap_config cs35l56_regmap_i2c = { .reg_bits = 32, .val_bits = 32, .reg_stride = 4, @@ -932,7 +932,7 @@ struct regmap_config cs35l56_regmap_i2c = { }; EXPORT_SYMBOL_NS_GPL(cs35l56_regmap_i2c, SND_SOC_CS35L56_SHARED); -struct regmap_config cs35l56_regmap_spi = { +const struct regmap_config cs35l56_regmap_spi = { .reg_bits = 32, .val_bits = 32, .pad_bits = 16, @@ -949,7 +949,7 @@ struct regmap_config cs35l56_regmap_spi = { }; EXPORT_SYMBOL_NS_GPL(cs35l56_regmap_spi, SND_SOC_CS35L56_SHARED); -struct regmap_config cs35l56_regmap_sdw = { +const struct regmap_config cs35l56_regmap_sdw = { .reg_bits = 32, .val_bits = 32, .reg_stride = 4, From 5a98c2e5399b125231ebb4594fee5fddfb7db9fd Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 23 Aug 2024 09:45:59 +0200 Subject: [PATCH 245/410] ASoC: dapm-graph: remove the "ROOT" cluster Widgets not belonging to any component are currently represented inside a cluster labeled "ROOT". This is not a correct representation of the actual structure, as these widgets are not necessarily related to each other as the ones inside actual components are. Improve the graphical representation by not adding a cluster around these widgets. Now a dot cluster represents a card component faithfully. This will be particularly important with the upcoming improvements which will visualize the component bias_level. Signed-off-by: Luca Ceresoli Link: https://patch.msgid.link/20240823-dapm-graph-v1-1-989a47308c4c@bootlin.com Signed-off-by: Mark Brown --- tools/sound/dapm-graph | 21 +++++++++++++-------- 1 file changed, 13 insertions(+), 8 deletions(-) diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph index 57d78f6df041..205783d124d3 100755 --- a/tools/sound/dapm-graph +++ b/tools/sound/dapm-graph @@ -150,29 +150,32 @@ process_dapm_widget() # # $1 = temporary work dir # $2 = component directory -# $3 = forced component name (extracted for path if empty) +# $3 = "ROOT" for the root card directory, empty otherwise process_dapm_component() { local tmp_dir="${1}" local c_dir="${2}" local c_name="${3}" + local is_component=0 local dot_file="${tmp_dir}/main.dot" local links_file="${tmp_dir}/links.dot" if [ -z "${c_name}" ]; then + is_component=1 + # Extract directory name into component name: # "./cs42l51.0-004a/dapm" -> "cs42l51.0-004a" c_name="$(basename $(dirname "${c_dir}"))" + + echo "" >> "${dot_file}" + echo " subgraph \"${c_name}\" {" >> "${dot_file}" + echo " cluster = true" >> "${dot_file}" + echo " label = \"${c_name}\"" >> "${dot_file}" + echo " color=dodgerblue" >> "${dot_file}" fi dbg_echo " * Component: ${c_name}" - echo "" >> "${dot_file}" - echo " subgraph \"${c_name}\" {" >> "${dot_file}" - echo " cluster = true" >> "${dot_file}" - echo " label = \"${c_name}\"" >> "${dot_file}" - echo " color=dodgerblue" >> "${dot_file}" - # Create empty file to ensure it will exist in all cases >"${links_file}" @@ -181,7 +184,9 @@ process_dapm_component() process_dapm_widget "${tmp_dir}" "${c_name}" "${w_file}" done - echo " }" >> "${dot_file}" + if [ ${is_component} = 1 ]; then + echo " }" >> "${dot_file}" + fi cat "${links_file}" >> "${dot_file}" } From 64a1e3ddab1ebaa590101b0d7d7fa5d3144da1e8 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 23 Aug 2024 09:46:00 +0200 Subject: [PATCH 246/410] ASoC: dapm-graph: visualize component On/Off bias level Read the bias_level debugfs files (ignored so far) and visualize the On/Off state of each component using different graphic attributes in the generated graph. Signed-off-by: Luca Ceresoli Link: https://patch.msgid.link/20240823-dapm-graph-v1-2-989a47308c4c@bootlin.com Signed-off-by: Mark Brown --- tools/sound/dapm-graph | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph index 205783d124d3..4e90883912d0 100755 --- a/tools/sound/dapm-graph +++ b/tools/sound/dapm-graph @@ -8,6 +8,8 @@ set -eu +STYLE_COMPONENT_ON="color=dodgerblue;style=bold" +STYLE_COMPONENT_OFF="color=gray40;style=filled;fillcolor=gray90" STYLE_NODE_ON="shape=box,style=bold,color=green4" STYLE_NODE_OFF="shape=box,style=filled,color=gray30,fillcolor=gray95" @@ -159,6 +161,7 @@ process_dapm_component() local is_component=0 local dot_file="${tmp_dir}/main.dot" local links_file="${tmp_dir}/links.dot" + local c_attribs="" if [ -z "${c_name}" ]; then is_component=1 @@ -166,16 +169,28 @@ process_dapm_component() # Extract directory name into component name: # "./cs42l51.0-004a/dapm" -> "cs42l51.0-004a" c_name="$(basename $(dirname "${c_dir}"))" + fi + + dbg_echo " * Component: ${c_name}" + + if [ ${is_component} = 1 ]; then + if [ -f "${c_dir}/bias_level" ]; then + c_onoff=$(sed -n -e 1p "${c_dir}/bias_level" | awk '{print $1}') + dbg_echo " - bias_level: ${c_onoff}" + if [ "$c_onoff" = "On" ]; then + c_attribs="${STYLE_COMPONENT_ON}" + elif [ "$c_onoff" = "Off" ]; then + c_attribs="${STYLE_COMPONENT_OFF}" + fi + fi echo "" >> "${dot_file}" echo " subgraph \"${c_name}\" {" >> "${dot_file}" echo " cluster = true" >> "${dot_file}" echo " label = \"${c_name}\"" >> "${dot_file}" - echo " color=dodgerblue" >> "${dot_file}" + echo " ${c_attribs}" >> "${dot_file}" fi - dbg_echo " * Component: ${c_name}" - # Create empty file to ensure it will exist in all cases >"${links_file}" From a14b278a47dd4b263799214c5ae0da6506ed7692 Mon Sep 17 00:00:00 2001 From: Luca Ceresoli Date: Fri, 23 Aug 2024 09:46:01 +0200 Subject: [PATCH 247/410] ASoC: dapm-graph: show path name for non-static routes Many routes are just static, not modifiable at runtime. Show the route name for all the other routes as an edge label in the generated graph. Signed-off-by: Luca Ceresoli Link: https://patch.msgid.link/20240823-dapm-graph-v1-3-989a47308c4c@bootlin.com Signed-off-by: Mark Brown --- tools/sound/dapm-graph | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/tools/sound/dapm-graph b/tools/sound/dapm-graph index 4e90883912d0..f14bdfedee8f 100755 --- a/tools/sound/dapm-graph +++ b/tools/sound/dapm-graph @@ -134,11 +134,17 @@ process_dapm_widget() # Collect any links. We could use "in" links or "out" links, # let's use "in" links if echo "${line}" | grep -q '^in '; then + local w_route=$(echo "$line" | awk -F\" '{print $2}') local w_src=$(echo "$line" | awk -F\" '{print $6 "_" $4}' | sed 's/^(null)_/ROOT_/') dbg_echo " - Input route from: ${w_src}" - echo " \"${w_src}\" -> \"$w_tag\"" >> "${links_file}" + dbg_echo " - Route: ${w_route}" + local w_edge_attrs="" + if [ "${w_route}" != "static" ]; then + w_edge_attrs=" [label=\"${w_route}\"]" + fi + echo " \"${w_src}\" -> \"$w_tag\"${w_edge_attrs}" >> "${links_file}" fi done @@ -220,7 +226,7 @@ process_dapm_tree() echo "digraph G {" > "${dot_file}" echo " fontname=\"sans-serif\"" >> "${dot_file}" echo " node [fontname=\"sans-serif\"]" >> "${dot_file}" - + echo " edge [fontname=\"sans-serif\"]" >> "${dot_file}" # Process root directory (no component) process_dapm_component "${tmp_dir}" "${dapm_dir}/dapm" "ROOT" From e17de785850e3112b2ea6ba786016a61f195bb23 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 23 Aug 2024 11:07:38 +0530 Subject: [PATCH 248/410] ASoC: amd: Add acpi machine id for acp7.0 version based platform Add acpi machine id for ACP7.0 version based platform and configure driver data to enable SOF sound card support on newer boards. Signed-off-by: Vijendar Mukunda Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240823053739.465187-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp-config.c | 12 ++++++++++++ sound/soc/amd/mach-config.h | 1 + 2 files changed, 13 insertions(+) diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c index 42c2322cd11b..365209ea53f3 100644 --- a/sound/soc/amd/acp-config.c +++ b/sound/soc/amd/acp-config.c @@ -321,5 +321,17 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_machines[] = { }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp63_sof_machines); +struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sof_machines[] = { + { + .id = "AMDI1010", + .drv_name = "acp70-dsp", + .pdata = &acp_quirk_data, + .fw_filename = "sof-acp_7_0.ri", + .sof_tplg_filename = "sof-acp_7_0.tplg", + }, + {}, +}; +EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sof_machines); + MODULE_DESCRIPTION("AMD ACP Machine Configuration Module"); MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/amd/mach-config.h b/sound/soc/amd/mach-config.h index 32aa8a6931f4..1a967da35a0f 100644 --- a/sound/soc/amd/mach-config.h +++ b/sound/soc/amd/mach-config.h @@ -24,6 +24,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_amd_rmb_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_vangogh_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp63_sof_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sof_machines[]; struct config_entry { u32 flags; From 490be7ba2a018093fbfa6c2dd80d7d0c190c4c98 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 23 Aug 2024 11:07:39 +0530 Subject: [PATCH 249/410] ASoC: SOF: amd: add support for acp7.0 based platform Add SOF support for ACP7.0 version based platform. Signed-off-by: Vijendar Mukunda Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240823053739.465187-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/Kconfig | 10 ++ sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 24 ++++- sound/soc/sof/amd/acp.c | 65 ++++++++++--- sound/soc/sof/amd/acp.h | 9 ++ sound/soc/sof/amd/acp70.c | 142 +++++++++++++++++++++++++++++ sound/soc/sof/amd/pci-acp70.c | 112 +++++++++++++++++++++++ 7 files changed, 352 insertions(+), 14 deletions(-) create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 848c031ed5fb..f4cafe801017 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -88,4 +88,14 @@ config SND_SOC_SOF_AMD_ACP63 AMD ACP6.3 version based platforms. Say Y if you want to enable SOF on ACP6.3 based platform. If unsure select "N". + +config SND_SOC_SOF_AMD_ACP70 + tristate "SOF support for ACP7.0 platform" + depends on SND_SOC_SOF_PCI + select SND_SOC_SOF_AMD_COMMON + help + Select this option for SOF support on + AMD ACP7.0 version based platforms. + Say Y if you want to enable SOF on ACP7.0 based platform. + endif diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index 63fe0d55fd0e..6ae39fd5a836 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -2,7 +2,7 @@ # This file is provided under a dual BSD/GPLv2 license. When using or # redistributing this file, you may do so under either license. # -# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. +# Copyright(c) 2021, 2023, 2024 Advanced Micro Devices, Inc. All rights reserved. snd-sof-amd-acp-y := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) += acp-probes.o @@ -10,9 +10,11 @@ snd-sof-amd-renoir-y := pci-rn.o renoir.o snd-sof-amd-rembrandt-y := pci-rmb.o rembrandt.o snd-sof-amd-vangogh-y := pci-vangogh.o vangogh.o snd-sof-amd-acp63-y := pci-acp63.o acp63.o +snd-sof-amd-acp70-y := pci-acp70.o acp70.o obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) += snd-sof-amd-renoir.o obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) += snd-sof-amd-rembrandt.o obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) += snd-sof-amd-vangogh.o obj-$(CONFIG_SND_SOC_SOF_AMD_ACP63) += snd-sof-amd-acp63.o +obj-$(CONFIG_SND_SOC_SOF_AMD_ACP70) += snd-sof-amd-acp70.o diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 072b703f9b3f..ecdcae07ace7 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. + * Copyright(c) 2021, 2023, 2024 Advanced Micro Devices, Inc. All rights reserved. * * Author: Ajit Kumar Pandey */ @@ -23,6 +23,17 @@ #define ACP_DMA_CH_STS 0xE8 #define ACP_DMA_CH_GROUP 0xEC #define ACP_DMA_CH_RST_STS 0xF0 +#define ACP70_DMA_CNTL_0 0x00 +#define ACP70_DMA_DSCR_STRT_IDX_0 0x28 +#define ACP70_DMA_DSCR_CNT_0 0x50 +#define ACP70_DMA_PRIO_0 0x78 +#define ACP70_DMA_CUR_DSCR_0 0xA0 +#define ACP70_DMA_ERR_STS_0 0xF0 +#define ACP70_DMA_DESC_BASE_ADDR 0x118 +#define ACP70_DMA_DESC_MAX_NUM_DSCR 0x11C +#define ACP70_DMA_CH_STS 0x120 +#define ACP70_DMA_CH_GROUP 0x124 +#define ACP70_DMA_CH_RST_STS 0x128 /* Registers from ACP_DSP_0 block */ #define ACP_DSP0_RUNSTALL 0x414 @@ -56,11 +67,13 @@ #define ACP3X_PGFSM_BASE 0x141C #define ACP5X_PGFSM_BASE 0x1424 #define ACP6X_PGFSM_BASE 0x1024 +#define ACP70_PGFSM_BASE ACP6X_PGFSM_BASE #define PGFSM_CONTROL_OFFSET 0x0 #define PGFSM_STATUS_OFFSET 0x4 #define ACP3X_CLKMUX_SEL 0x1424 #define ACP5X_CLKMUX_SEL 0x142C #define ACP6X_CLKMUX_SEL 0x102C +#define ACP70_CLKMUX_SEL ACP6X_CLKMUX_SEL /* Registers from ACP_INTR block */ #define ACP3X_EXT_INTR_STAT 0x1808 @@ -69,22 +82,30 @@ #define ACP6X_EXTERNAL_INTR_CNTL 0x1A04 #define ACP6X_EXT_INTR_STAT 0x1A0C #define ACP6X_EXT_INTR_STAT1 0x1A10 +#define ACP70_EXTERNAL_INTR_ENB ACP6X_EXTERNAL_INTR_ENB +#define ACP70_EXTERNAL_INTR_CNTL ACP6X_EXTERNAL_INTR_CNTL +#define ACP70_EXT_INTR_STAT ACP6X_EXT_INTR_STAT +#define ACP70_EXT_INTR_STAT1 ACP6X_EXT_INTR_STAT1 #define ACP3X_DSP_SW_INTR_BASE 0x1814 #define ACP5X_DSP_SW_INTR_BASE 0x1814 #define ACP6X_DSP_SW_INTR_BASE 0x1808 +#define ACP70_DSP_SW_INTR_BASE ACP6X_DSP_SW_INTR_BASE #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 #define ACP3X_ERROR_STATUS 0x18C4 #define ACP6X_ERROR_STATUS 0x1A4C +#define ACP70_ERROR_STATUS ACP6X_ERROR_STATUS #define ACP3X_AXI2DAGB_SEM_0 0x1880 #define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 +#define ACP70_AXI2DAGB_SEM_0 ACP6X_AXI2DAGB_SEM_0 /* ACP common registers to report errors related to I2S & SoundWire interfaces */ #define ACP3X_SW_I2S_ERROR_REASON 0x18C8 #define ACP6X_SW0_I2S_ERROR_REASON 0x18B4 +#define ACP7X_SW0_I2S_ERROR_REASON ACP6X_SW0_I2S_ERROR_REASON #define ACP_SW1_I2S_ERROR_REASON 0x1A50 /* Registers from ACP_SHA block */ @@ -101,6 +122,7 @@ #define ACP_SCRATCH_REG_0 0x10000 #define ACP6X_DSP_FUSION_RUNSTALL 0x0644 +#define ACP70_DSP_FUSION_RUNSTALL ACP6X_DSP_FUSION_RUNSTALL /* Cache window registers */ #define ACP_DSP0_CACHE_OFFSET0 0x0420 diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index e4d46fdda88b..d579c3849392 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -64,13 +64,24 @@ static void init_dma_descriptor(struct acp_dev_data *adata) { struct snd_sof_dev *sdev = adata->dev; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); + struct acp_dev_data *acp_data = sdev->pdata->hw_pdata; unsigned int addr; + unsigned int acp_dma_desc_base_addr, acp_dma_desc_max_num_dscr; addr = desc->sram_pte_offset + sdev->debug_box.offset + offsetof(struct scratch_reg_conf, dma_desc); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_DESC_BASE_ADDR, addr); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_DESC_MAX_NUM_DSCR, ACP_MAX_DESC_CNT); + switch (acp_data->pci_rev) { + case ACP70_PCI_ID: + acp_dma_desc_base_addr = ACP70_DMA_DESC_BASE_ADDR; + acp_dma_desc_max_num_dscr = ACP70_DMA_DESC_MAX_NUM_DSCR; + break; + default: + acp_dma_desc_base_addr = ACP_DMA_DESC_BASE_ADDR; + acp_dma_desc_max_num_dscr = ACP_DMA_DESC_MAX_NUM_DSCR; + } + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_desc_base_addr, addr); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_desc_max_num_dscr, ACP_MAX_DESC_CNT); } static void configure_dma_descriptor(struct acp_dev_data *adata, unsigned short idx, @@ -92,29 +103,51 @@ static int config_dma_channel(struct acp_dev_data *adata, unsigned int ch, unsigned int idx, unsigned int dscr_count) { struct snd_sof_dev *sdev = adata->dev; + struct acp_dev_data *acp_data = sdev->pdata->hw_pdata; const struct sof_amd_acp_desc *desc = get_chip_info(sdev->pdata); unsigned int val, status; + unsigned int acp_dma_cntl_0, acp_dma_ch_rst_sts, acp_dma_dscr_err_sts_0; + unsigned int acp_dma_dscr_cnt_0, acp_dma_prio_0, acp_dma_dscr_strt_idx_0; int ret; - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32), + switch (acp_data->pci_rev) { + case ACP70_PCI_ID: + acp_dma_cntl_0 = ACP70_DMA_CNTL_0; + acp_dma_ch_rst_sts = ACP70_DMA_CH_RST_STS; + acp_dma_dscr_err_sts_0 = ACP70_DMA_ERR_STS_0; + acp_dma_dscr_cnt_0 = ACP70_DMA_DSCR_CNT_0; + acp_dma_prio_0 = ACP70_DMA_PRIO_0; + acp_dma_dscr_strt_idx_0 = ACP70_DMA_DSCR_STRT_IDX_0; + break; + default: + acp_dma_cntl_0 = ACP_DMA_CNTL_0; + acp_dma_ch_rst_sts = ACP_DMA_CH_RST_STS; + acp_dma_dscr_err_sts_0 = ACP_DMA_ERR_STS_0; + acp_dma_dscr_cnt_0 = ACP_DMA_DSCR_CNT_0; + acp_dma_prio_0 = ACP_DMA_PRIO_0; + acp_dma_dscr_strt_idx_0 = ACP_DMA_DSCR_STRT_IDX_0; + } + + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_cntl_0 + ch * sizeof(u32), ACP_DMA_CH_RST | ACP_DMA_CH_GRACEFUL_RST_EN); - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_RST_STS, val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_rst_sts, val, val & (1 << ch), ACP_REG_POLL_INTERVAL, ACP_REG_POLL_TIMEOUT_US); if (ret < 0) { status = snd_sof_dsp_read(sdev, ACP_DSP_BAR, desc->acp_error_stat); - val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_ERR_STS_0 + ch * sizeof(u32)); + val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, acp_dma_dscr_err_sts_0 + + ch * sizeof(u32)); dev_err(sdev->dev, "ACP_DMA_ERR_STS :0x%x ACP_ERROR_STATUS :0x%x\n", val, status); return ret; } - snd_sof_dsp_write(sdev, ACP_DSP_BAR, (ACP_DMA_CNTL_0 + ch * sizeof(u32)), 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_DSCR_CNT_0 + ch * sizeof(u32), dscr_count); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_DSCR_STRT_IDX_0 + ch * sizeof(u32), idx); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_PRIO_0 + ch * sizeof(u32), 0); - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32), ACP_DMA_CH_RUN); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, (acp_dma_cntl_0 + ch * sizeof(u32)), 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_dscr_cnt_0 + ch * sizeof(u32), dscr_count); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_dscr_strt_idx_0 + ch * sizeof(u32), idx); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_prio_0 + ch * sizeof(u32), 0); + snd_sof_dsp_write(sdev, ACP_DSP_BAR, acp_dma_cntl_0 + ch * sizeof(u32), ACP_DMA_CH_RUN); return ret; } @@ -453,6 +486,10 @@ static int acp_power_on(struct snd_sof_dev *sdev) acp_pgfsm_status_mask = ACP6X_PGFSM_STATUS_MASK; acp_pgfsm_cntl_mask = ACP6X_PGFSM_CNTL_POWER_ON_MASK; break; + case ACP70_PCI_ID: + acp_pgfsm_status_mask = ACP70_PGFSM_STATUS_MASK; + acp_pgfsm_cntl_mask = ACP70_PGFSM_CNTL_POWER_ON_MASK; + break; default: return -EINVAL; } @@ -561,8 +598,11 @@ static bool check_acp_sdw_enable_status(struct snd_sof_dev *sdev) int amd_sof_acp_suspend(struct snd_sof_dev *sdev, u32 target_state) { + struct acp_dev_data *acp_data; int ret; + bool enable = false; + acp_data = sdev->pdata->hw_pdata; /* When acp_reset() function is invoked, it will apply ACP SOFT reset and * DSP reset. ACP Soft reset sequence will cause all ACP IP registers will * be reset to default values which will break the ClockStop Mode functionality. @@ -577,8 +617,9 @@ int amd_sof_acp_suspend(struct snd_sof_dev *sdev, u32 target_state) dev_err(sdev->dev, "ACP Reset failed\n"); return ret; } - - snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_CONTROL, 0x00); + if (acp_data->pci_rev == ACP70_PCI_ID) + enable = true; + snd_sof_dsp_write(sdev, ACP_DSP_BAR, ACP_CONTROL, enable); return 0; } diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 11def07efc0f..800594440f73 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -29,6 +29,8 @@ #define ACP3X_PGFSM_STATUS_MASK 0x03 #define ACP6X_PGFSM_CNTL_POWER_ON_MASK 0x07 #define ACP6X_PGFSM_STATUS_MASK 0x0F +#define ACP70_PGFSM_CNTL_POWER_ON_MASK 0x1F +#define ACP70_PGFSM_STATUS_MASK 0xFF #define ACP_POWERED_ON 0x00 #define ACP_ASSERT_RESET 0x01 @@ -42,6 +44,7 @@ #define ACP3X_SRAM_PTE_OFFSET 0x02050000 #define ACP5X_SRAM_PTE_OFFSET 0x02050000 #define ACP6X_SRAM_PTE_OFFSET 0x03800000 +#define ACP70_SRAM_PTE_OFFSET ACP6X_SRAM_PTE_OFFSET #define PAGE_SIZE_4K_ENABLE 0x2 #define ACP_PAGE_SIZE 0x1000 #define ACP_DMA_CH_RUN 0x02 @@ -63,17 +66,20 @@ #define ACP_DRAM_BASE_ADDRESS 0x01000000 #define ACP_DRAM_PAGE_COUNT 128 #define ACP_SRAM_BASE_ADDRESS 0x3806000 +#define ACP7X_SRAM_BASE_ADDRESS 0x380C000 #define ACP_DSP_TO_HOST_IRQ 0x04 #define ACP_RN_PCI_ID 0x01 #define ACP_VANGOGH_PCI_ID 0x50 #define ACP_RMB_PCI_ID 0x6F #define ACP63_PCI_ID 0x63 +#define ACP70_PCI_ID 0x70 #define HOST_BRIDGE_CZN 0x1630 #define HOST_BRIDGE_VGH 0x1645 #define HOST_BRIDGE_RMB 0x14B5 #define HOST_BRIDGE_ACP63 0x14E8 +#define HOST_BRIDGE_ACP70 0x1507 #define ACP_SHA_STAT 0x8000 #define ACP_PSP_TIMEOUT_US 1000000 #define ACP_EXT_INTR_ERROR_STAT 0x20000000 @@ -326,6 +332,9 @@ int sof_rembrandt_ops_init(struct snd_sof_dev *sdev); extern struct snd_sof_dsp_ops sof_acp63_ops; int sof_acp63_ops_init(struct snd_sof_dev *sdev); +extern struct snd_sof_dsp_ops sof_acp70_ops; +int sof_acp70_ops_init(struct snd_sof_dev *sdev); + struct snd_soc_acpi_mach *amd_sof_machine_select(struct snd_sof_dev *sdev); /* Machine configuration */ int snd_amd_acp_find_config(struct pci_dev *pci); diff --git a/sound/soc/sof/amd/acp70.c b/sound/soc/sof/amd/acp70.c new file mode 100644 index 000000000000..7d1842f42c90 --- /dev/null +++ b/sound/soc/sof/amd/acp70.c @@ -0,0 +1,142 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2024 Advanced Micro Devices, Inc. +// +// Authors: Vijendar Mukunda + +/* + * Hardware interface for Audio DSP on ACP7.0 version based platform + */ + +#include +#include + +#include "../ops.h" +#include "../sof-audio.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define I2S_HS_INSTANCE 0 +#define I2S_BT_INSTANCE 1 +#define I2S_SP_INSTANCE 2 +#define PDM_DMIC_INSTANCE 3 +#define I2S_HS_VIRTUAL_INSTANCE 4 + +static struct snd_soc_dai_driver acp70_sof_dai[] = { + [I2S_HS_INSTANCE] = { + .id = I2S_HS_INSTANCE, + .name = "acp-sof-hs", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S HS controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_BT_INSTANCE] = { + .id = I2S_BT_INSTANCE, + .name = "acp-sof-bt", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S BT controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_SP_INSTANCE] = { + .id = I2S_SP_INSTANCE, + .name = "acp-sof-sp", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S SP controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [PDM_DMIC_INSTANCE] = { + .id = PDM_DMIC_INSTANCE, + .name = "acp-sof-dmic", + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_HS_VIRTUAL_INSTANCE] = { + .id = I2S_HS_VIRTUAL_INSTANCE, + .name = "acp-sof-hs-virtual", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + }, +}; + +/* Phoenix ops */ +struct snd_sof_dsp_ops sof_acp70_ops; +EXPORT_SYMBOL_NS(sof_acp70_ops, SND_SOC_SOF_AMD_COMMON); + +int sof_acp70_ops_init(struct snd_sof_dev *sdev) +{ + /* common defaults */ + memcpy(&sof_acp70_ops, &sof_acp_common_ops, sizeof(struct snd_sof_dsp_ops)); + + sof_acp70_ops.drv = acp70_sof_dai; + sof_acp70_ops.num_drv = ARRAY_SIZE(acp70_sof_dai); + + return 0; +} diff --git a/sound/soc/sof/amd/pci-acp70.c b/sound/soc/sof/amd/pci-acp70.c new file mode 100644 index 000000000000..a5d8b6a95a22 --- /dev/null +++ b/sound/soc/sof/amd/pci-acp70.c @@ -0,0 +1,112 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2024 Advanced Micro Devices, Inc. All rights reserved. +// +// Authors: Vijendar Mukunda + +/*. + * PCI interface for ACP7.0 device + */ + +#include +#include +#include +#include +#include + +#include "../ops.h" +#include "../sof-pci-dev.h" +#include "../../amd/mach-config.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define ACP70_FUTURE_REG_ACLK_0 0x1854 +#define ACP70_REG_START 0x1240000 +#define ACP70_REG_END 0x125C000 + +static const struct sof_amd_acp_desc acp70_chip_info = { + .host_bridge_id = HOST_BRIDGE_ACP70, + .pgfsm_base = ACP70_PGFSM_BASE, + .ext_intr_enb = ACP70_EXTERNAL_INTR_ENB, + .ext_intr_cntl = ACP70_EXTERNAL_INTR_CNTL, + .ext_intr_stat = ACP70_EXT_INTR_STAT, + .ext_intr_stat1 = ACP70_EXT_INTR_STAT1, + .dsp_intr_base = ACP70_DSP_SW_INTR_BASE, + .acp_sw0_i2s_err_reason = ACP7X_SW0_I2S_ERROR_REASON, + .sram_pte_offset = ACP70_SRAM_PTE_OFFSET, + .hw_semaphore_offset = ACP70_AXI2DAGB_SEM_0, + .fusion_dsp_offset = ACP70_DSP_FUSION_RUNSTALL, + .probe_reg_offset = ACP70_FUTURE_REG_ACLK_0, + .reg_start_addr = ACP70_REG_START, + .reg_end_addr = ACP70_REG_END, +}; + +static const struct sof_dev_desc acp70_desc = { + .machines = snd_soc_acpi_amd_acp70_sof_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &acp70_chip_info, + .ipc_supported_mask = BIT(SOF_IPC_TYPE_3), + .ipc_default = SOF_IPC_TYPE_3, + .default_fw_path = { + [SOF_IPC_TYPE_3] = "amd/sof", + }, + .default_tplg_path = { + [SOF_IPC_TYPE_3] = "amd/sof-tplg", + }, + .default_fw_filename = { + [SOF_IPC_TYPE_3] = "sof-acp_7_0.ri", + }, + .nocodec_tplg_filename = "sof-acp.tplg", + .ops = &sof_acp70_ops, + .ops_init = sof_acp70_ops_init, +}; + +static int acp70_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + unsigned int flag; + + if (pci->revision != ACP70_PCI_ID) + return -ENODEV; + + flag = snd_amd_acp_find_config(pci); + if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) + return -ENODEV; + + return sof_pci_probe(pci, pci_id); +}; + +static void acp70_pci_remove(struct pci_dev *pci) +{ + sof_pci_remove(pci); +} + +/* PCI IDs */ +static const struct pci_device_id acp70_pci_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_PCI_DEV_ID), + .driver_data = (unsigned long)&acp70_desc}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, acp70_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_amd_acp70_driver = { + .name = KBUILD_MODNAME, + .id_table = acp70_pci_ids, + .probe = acp70_pci_probe, + .remove = acp70_pci_remove, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_amd_acp70_driver); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("ACP70 SOF Driver"); +MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); From f7b1633d6467a08ff78744cb519a92fb24ad9a0c Mon Sep 17 00:00:00 2001 From: Shen Lichuan Date: Mon, 26 Aug 2024 12:34:54 +0800 Subject: [PATCH 250/410] ALSA: usb-audio: Use kmemdup_array instead of kmemdup for multiple allocation Let the kmemdup_array() take care about multiplication and possible overflows. Using kmemdup_array() is more appropriate and makes the code easier to audit. Signed-off-by: Shen Lichuan Link: https://patch.msgid.link/20240826043454.3198-1-shenlichuan@vivo.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e7b68c67852e..53c69f3069c4 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -167,8 +167,8 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } if (fp->nr_rates > 0) { - rate_table = kmemdup(fp->rate_table, - sizeof(int) * fp->nr_rates, GFP_KERNEL); + rate_table = kmemdup_array(fp->rate_table, fp->nr_rates, sizeof(int), + GFP_KERNEL); if (!rate_table) { kfree(fp); return -ENOMEM; From c8dc1016ba0e249e45863c1da3b951efe7c4214a Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Sat, 24 Aug 2024 14:05:00 +0800 Subject: [PATCH 251/410] ASoC: tas2781: replace devm_kzalloc and scnprintf with devm_kstrdup Replace devm_kzalloc and scnprintf with devm_kstrdup. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240824060503.1259-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index eb8732b1bb99..59fcce7fef7b 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kstrdup(tas_priv->dev, "Speaker Profile Id", GFP_KERNEL); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name || !conf_name) { + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", + GFP_KERNEL); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; From 0225d3b9efe3daca332befaa0c4ce2f119297d5a Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Sun, 25 Aug 2024 10:57:45 +0200 Subject: [PATCH 252/410] ASoC: MAINTAINERS: Drop incorrect tlv320aic31xx.txt path tlv320aic31xx.txt was converted to DT schema (YAML) and new file is already matched by wildcard. This fixes get_maintainers.pl self-test warning: ./MAINTAINERS:22739: warning: no file matches F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt Fixes: e486feb7b8ec ("ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml") Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240825085745.21668-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 1d12f3fca229..f8a645bcf2b1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22584,7 +22584,6 @@ F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h From 69a8d0edb9d78bb5515133b7c08f399d6eaff37a Mon Sep 17 00:00:00 2001 From: Shen Lichuan Date: Mon, 26 Aug 2024 13:44:02 +0800 Subject: [PATCH 253/410] ASoC: SOF: topology: Use kmemdup_array instead of kmemdup for multiple allocation Let the kmemdup_array() take care about multiplication and possible overflows. Using kmemdup_array() is more appropriate and makes the code easier to audit. Signed-off-by: Shen Lichuan Link: https://patch.msgid.link/20240826054402.58396-1-shenlichuan@vivo.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index b54382131991..996c3234eaee 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1349,7 +1349,7 @@ static int sof_parse_pin_binding(struct snd_sof_widget *swidget, /* copy pin binding array to swidget only if it is defined in topology */ if (pin_binding[0]) { - pb = kmemdup(pin_binding, num_pins * sizeof(char *), GFP_KERNEL); + pb = kmemdup_array(pin_binding, num_pins, sizeof(char *), GFP_KERNEL); if (!pb) { ret = -ENOMEM; goto err; @@ -1889,9 +1889,9 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_ return -ENOMEM; slink->num_hw_configs = le32_to_cpu(cfg->num_hw_configs); - slink->hw_configs = kmemdup(cfg->hw_config, - sizeof(*slink->hw_configs) * slink->num_hw_configs, - GFP_KERNEL); + slink->hw_configs = kmemdup_array(cfg->hw_config, + slink->num_hw_configs, sizeof(*slink->hw_configs), + GFP_KERNEL); if (!slink->hw_configs) { kfree(slink); return -ENOMEM; From 002353a537a29b9be5bde3c1d9964628f0d20d45 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 17:27:22 +0200 Subject: [PATCH 254/410] drm/bridge: dw-hdmi: Move vmalloc PCM buffer management into the driver The dw-hdmi drm bridge driver is the only one who still uses the ALSA vmalloc helper API functions. A previous attempt to change the way of buffer management wasn't taken for this legacy stuff, as we had little chance for test and some risk of major breaking. Instead, this patch moves the vmalloc buffer stuff into the dw-hdmi driver code itself, so that we can drop them from ALSA core code afterwards. There should be no functional changes. Link: https://lore.kernel.org/20191210154536.29819-1-tiwai@suse.de Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807152725.18948-2-tiwai@suse.de --- .../drm/bridge/synopsys/dw-hdmi-ahb-audio.c | 30 ++++++++++++++++--- 1 file changed, 26 insertions(+), 4 deletions(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c index 67b8d17a722a..221e9a4edb40 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-ahb-audio.c @@ -8,6 +8,7 @@ #include #include #include +#include #include #include @@ -388,15 +389,36 @@ static int dw_hdmi_close(struct snd_pcm_substream *substream) static int dw_hdmi_hw_free(struct snd_pcm_substream *substream) { - return snd_pcm_lib_free_vmalloc_buffer(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; } static int dw_hdmi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { + struct snd_pcm_runtime *runtime = substream->runtime; + size_t size = params_buffer_bytes(params); + /* Allocate the PCM runtime buffer, which is exposed to userspace. */ - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(params)); + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = vzalloc(size); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 1; +} + +static struct page *dw_hdmi_get_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); } static int dw_hdmi_prepare(struct snd_pcm_substream *substream) @@ -515,7 +537,7 @@ static const struct snd_pcm_ops snd_dw_hdmi_ops = { .prepare = dw_hdmi_prepare, .trigger = dw_hdmi_trigger, .pointer = dw_hdmi_pointer, - .page = snd_pcm_lib_get_vmalloc_page, + .page = dw_hdmi_get_page, }; static int snd_dw_hdmi_probe(struct platform_device *pdev) From 5e1c5c5a687bb3351d9b4a3b0ad457f8497b2a0d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Aug 2024 17:27:23 +0200 Subject: [PATCH 255/410] ALSA: pcm: Drop PCM vmalloc buffer helpers As the last-standing user of PCM vmalloc buffer helper API took its own buffer management, we can finally drop those API functions, which were leftover after reorganization of ALSA memalloc code. Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240807152725.18948-3-tiwai@suse.de --- include/sound/pcm.h | 42 ----------------------------- sound/core/pcm_memory.c | 59 ----------------------------------------- 2 files changed, 101 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 384032b6c59c..732121b934fd 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1358,48 +1358,6 @@ snd_pcm_set_fixed_buffer_all(struct snd_pcm *pcm, int type, return snd_pcm_set_managed_buffer_all(pcm, type, data, size, 0); } -int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, - size_t size, gfp_t gfp_flags); -int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); -struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, - unsigned long offset); -/** - * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer - * @substream: the substream to allocate the buffer to - * @size: the requested buffer size, in bytes - * - * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is - * contiguous in kernel virtual space, but not in physical memory. Use this - * if the buffer is accessed by kernel code but not by device DMA. - * - * Return: 1 if the buffer was changed, 0 if not changed, or a negative error - * code. - */ -static inline int snd_pcm_lib_alloc_vmalloc_buffer - (struct snd_pcm_substream *substream, size_t size) -{ - return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size, - GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO); -} - -/** - * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer - * @substream: the substream to allocate the buffer to - * @size: the requested buffer size, in bytes - * - * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses - * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. - * - * Return: 1 if the buffer was changed, 0 if not changed, or a negative error - * code. - */ -static inline int snd_pcm_lib_alloc_vmalloc_32_buffer - (struct snd_pcm_substream *substream, size_t size) -{ - return _snd_pcm_lib_alloc_vmalloc_buffer(substream, size, - GFP_KERNEL | GFP_DMA32 | __GFP_ZERO); -} - #define snd_pcm_get_dma_buf(substream) ((substream)->runtime->dma_buffer_p) /** diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 506386959f08..8e4c68e3bbd0 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -9,7 +9,6 @@ #include #include #include -#include #include #include #include @@ -497,61 +496,3 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) return 0; } EXPORT_SYMBOL(snd_pcm_lib_free_pages); - -int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, - size_t size, gfp_t gfp_flags) -{ - struct snd_pcm_runtime *runtime; - - if (PCM_RUNTIME_CHECK(substream)) - return -EINVAL; - runtime = substream->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = __vmalloc(size, gfp_flags); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 1; -} -EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); - -/** - * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer - * @substream: the substream with a buffer allocated by - * snd_pcm_lib_alloc_vmalloc_buffer() - * - * Return: Zero if successful, or a negative error code on failure. - */ -int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime; - - if (PCM_RUNTIME_CHECK(substream)) - return -EINVAL; - runtime = substream->runtime; - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} -EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); - -/** - * snd_pcm_lib_get_vmalloc_page - map vmalloc buffer offset to page struct - * @substream: the substream with a buffer allocated by - * snd_pcm_lib_alloc_vmalloc_buffer() - * @offset: offset in the buffer - * - * This function is to be used as the page callback in the PCM ops. - * - * Return: The page struct, or %NULL on failure. - */ -struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, - unsigned long offset) -{ - return vmalloc_to_page(substream->runtime->dma_area + offset); -} -EXPORT_SYMBOL(snd_pcm_lib_get_vmalloc_page); From 6785244f3dfd40fb0671c330de4e953cbc2e2846 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 27 Aug 2024 20:31:59 +0800 Subject: [PATCH 256/410] ASoC: Intel: sof_sdw: make sof_sdw_quirk static There's no need to make this variable visible at a higher level. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 2 -- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b1fb6fabd3b7..9b642c6883b0 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -17,7 +17,7 @@ #include "sof_sdw_common.h" #include "../../codecs/rt711.h" -unsigned long sof_sdw_quirk = RT711_JD1; +static unsigned long sof_sdw_quirk = RT711_JD1; static int quirk_override = -1; module_param_named(quirk, quirk_override, int, 0444); MODULE_PARM_DESC(quirk, "Board-specific quirk override"); diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index c10d2711b730..3aa1dcec5172 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -58,8 +58,6 @@ struct intel_mc_ctx { unsigned int sdw_pin_index[SDW_INTEL_MAX_LINKS]; }; -extern unsigned long sof_sdw_quirk; - /* generic HDMI support */ int sof_sdw_hdmi_init(struct snd_soc_pcm_runtime *rtd); From 1ab959bea29c265d7d219346a72ed15b2490d956 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Aug 2024 20:32:00 +0800 Subject: [PATCH 257/410] ASoC: Intel: sof_sdw: add rt1320 amp support Add Realtek rt1320 amp support. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240827123215.258859-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/sdw_utils/soc_sdw_rt_amp.c | 11 ++++++++++- sound/soc/sdw_utils/soc_sdw_utils.c | 19 +++++++++++++++++++ 3 files changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index cb952183f5ed..1141fe1263ce 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -524,6 +524,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_RT1308 select SND_SOC_RT1316_SDW select SND_SOC_RT1318_SDW + select SND_SOC_RT1320_SDW select SND_SOC_RT5682_SDW select SND_SOC_CS42L42_SDW select SND_SOC_CS42L43 diff --git a/sound/soc/sdw_utils/soc_sdw_rt_amp.c b/sound/soc/sdw_utils/soc_sdw_rt_amp.c index 42be01405ab4..6951dfb56526 100644 --- a/sound/soc/sdw_utils/soc_sdw_rt_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_rt_amp.c @@ -160,6 +160,13 @@ static const struct snd_soc_dapm_route rt1318_map[] = { { "Speaker", NULL, "rt1318-2 SPOR" }, }; +static const struct snd_soc_dapm_route rt1320_map[] = { + { "Speaker", NULL, "rt1320-1 SPOL" }, + { "Speaker", NULL, "rt1320-1 SPOR" }, + { "Speaker", NULL, "rt1320-2 SPOL" }, + { "Speaker", NULL, "rt1320-2 SPOR" }, +}; + static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_dai *dai, char *codec_name) { @@ -171,8 +178,10 @@ static const struct snd_soc_dapm_route *get_codec_name_and_route(struct snd_soc_ return rt1308_map; else if (strcmp(codec_name, "rt1316") == 0) return rt1316_map; - else + else if (strcmp(codec_name, "rt1318") == 0) return rt1318_map; + else + return rt1320_map; } int asoc_sdw_rt_amp_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index e8d0f199155d..d59ccb56642c 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -254,6 +254,25 @@ struct asoc_sdw_codec_info codec_info_list[] = { }, .dai_num = 1, }, + { + .part_id = 0x1320, + .dais = { + { + .direction = {true, false}, + .dai_name = "rt1320-aif1", + .dai_type = SOC_SDW_DAI_TYPE_AMP, + .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_UNUSED_DAI_ID}, + .init = asoc_sdw_rt_amp_init, + .exit = asoc_sdw_rt_amp_exit, + .rtd_init = asoc_sdw_rt_amp_spk_rtd_init, + .controls = generic_spk_controls, + .num_controls = ARRAY_SIZE(generic_spk_controls), + .widgets = generic_spk_widgets, + .num_widgets = ARRAY_SIZE(generic_spk_widgets), + }, + }, + .dai_num = 1, + }, { .part_id = 0x714, .version_id = 3, From 14e91ddd5c02d8c3e5a682ebfa0546352b459911 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Tue, 27 Aug 2024 20:32:01 +0800 Subject: [PATCH 258/410] ASoC: Intel: boards: always check the result of acpi_dev_get_first_match_dev() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The code seems mostly copy-pasted, with some machine drivers forgetting to test if the 'adev' result is NULL. Add this check when missing, and use -ENOENT consistently as an error code. Reported-by: Dan Carpenter Closes: https://lore.kernel.org/alsa-devel/918944d2-3d00-465e-a9d1-5d57fc966113@stanley.mountain/T/#u Signed-off-by: Pierre-Louis Bossart Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/bytcht_cx2072x.c | 4 ++++ sound/soc/intel/boards/bytcht_da7213.c | 4 ++++ sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++++ sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++++ sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 4 ++++ 9 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index df3c2a7b64d2..8c2b4ab764bb 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -255,7 +255,11 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); byt_cht_cx2072x_dais[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 08c598b7e1ee..9178bbe8d995 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -258,7 +258,11 @@ static int bytcht_da7213_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 77b91ea4dc32..3539c9ff0fd2 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -562,7 +562,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index db4a33680d94..4479825c08b5 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1693,7 +1693,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) byt_rt5640_dais[dai_index].codecs->name = byt_rt5640_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 8514b79f389b..1f54da98aacf 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -926,7 +926,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) byt_rt5651_dais[dai_index].codecs->name = byt_rt5651_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 1da9ceee4d59..ac23a8b7cafc 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -582,7 +582,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index d68e5bc755de..c6c469d51243 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -479,7 +479,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = drv->codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* Use SSP0 on Bay Trail CR devices */ diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 2a88efaa6d26..b45d0501f109 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -681,7 +681,7 @@ static int sof_es8336_probe(struct platform_device *pdev) dai_links[0].codecs->dai_name = "ES8326 HiFi"; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index b2d02cc92a6a..0a5ce34d7f7b 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -270,7 +270,11 @@ static int sof_wm8804_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); snd_soc_card_set_drvdata(card, ctx); From 5458411d75947a4212e50a401ec0a98d4c6c931b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Aug 2024 20:32:02 +0800 Subject: [PATCH 259/410] ASoC: SOF: Intel: hda: refactoring topology name fixup for HDA mach Move I2S mach's topology name fixup code to the end of machine driver enumeration flow so HDA mach could also use same code to fixup its topology file name as well. No functional change in this commit. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-hda-match.c | 12 +-- sound/soc/sof/intel/hda.c | 84 ++++++++++--------- 2 files changed, 45 insertions(+), 51 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-hda-match.c b/sound/soc/intel/common/soc-acpi-intel-hda-match.c index 007ccd8a60e5..e93336e27beb 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hda-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hda-match.c @@ -13,16 +13,8 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_hda_machines[] = { { /* .id is not used in this file */ .drv_name = "skl_hda_dsp_generic", - - /* .fw_filename is dynamically set in skylake driver */ - - .sof_tplg_filename = "sof-hda-generic.tplg", - - /* - * .machine_quirk and .quirk_data are not used here but - * can be used if we need a more complicated machine driver - * combining HDA+other device (e.g. DMIC). - */ + .sof_tplg_filename = "sof-hda-generic", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, {}, }; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 5a40b8fbbbd3..128687b24bf5 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -558,7 +558,7 @@ static int check_nhlt_ssp_mclk_mask(struct snd_sof_dev *sdev, int ssp_num) return intel_nhlt_ssp_mclk_mask(nhlt, ssp_num); } -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) || IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) static const char *fixup_tplg_name(struct snd_sof_dev *sdev, const char *sof_tplg_filename, @@ -1045,10 +1045,7 @@ static void hda_generic_machine_select(struct snd_sof_dev *sdev, struct snd_soc_acpi_mach *hda_mach; struct snd_sof_pdata *pdata = sdev->pdata; const char *tplg_filename; - const char *idisp_str; - int dmic_num = 0; int codec_num = 0; - int ret; int i; /* codec detection */ @@ -1071,33 +1068,30 @@ static void hda_generic_machine_select(struct snd_sof_dev *sdev, * - one external HDAudio codec */ if (!*mach && codec_num <= 2) { - bool tplg_fixup; + bool tplg_fixup = false; hda_mach = snd_soc_acpi_intel_hda_machines; dev_info(bus->dev, "using HDA machine driver %s now\n", hda_mach->drv_name); - if (codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask)) - idisp_str = "-idisp"; - else - idisp_str = ""; - - /* topology: use the info from hda_machines */ - if (pdata->tplg_filename) { - tplg_fixup = false; - tplg_filename = pdata->tplg_filename; - } else { + /* + * topology: use the info from hda_machines since tplg file name + * is not overwritten + */ + if (!pdata->tplg_filename) tplg_fixup = true; - tplg_filename = hda_mach->sof_tplg_filename; - } - ret = dmic_detect_topology_fixup(sdev, &tplg_filename, idisp_str, &dmic_num, - tplg_fixup); - if (ret < 0) - return; - hda_mach->mach_params.dmic_num = dmic_num; - pdata->tplg_filename = tplg_filename; + if (tplg_fixup && + codec_num == 1 && HDA_IDISP_CODEC(bus->codec_mask)) { + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, + "%s-idisp", + hda_mach->sof_tplg_filename); + if (!tplg_filename) + return; + + hda_mach->sof_tplg_filename = tplg_filename; + } if (codec_num == 2 || (codec_num == 1 && !HDA_IDISP_CODEC(bus->codec_mask))) { @@ -1311,11 +1305,35 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) const char *tplg_filename; const char *tplg_suffix; bool amp_name_valid; + bool i2s_mach_found = false; /* Try I2S or DMIC if it is supported */ - if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC))) + if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC))) { mach = snd_soc_acpi_find_machine(desc->machines); + if (mach) + i2s_mach_found = true; + } + /* + * If I2S fails and no external HDaudio codec is detected, + * try SoundWire if it is supported + */ + if (!mach && !HDA_EXT_CODEC(bus->codec_mask) && + (interface_mask & BIT(SOF_DAI_INTEL_ALH))) + mach = hda_sdw_machine_select(sdev); + + /* + * Choose HDA generic machine driver if mach is NULL. + * Otherwise, set certain mach params. + */ + hda_generic_machine_select(sdev, &mach); + if (!mach) + dev_warn(sdev->dev, "warning: No matching ASoC machine driver found\n"); + + /* + * Fixup tplg file name by appending dmic num, ssp num, codec/amplifier + * name string if quirk flag is set. + */ if (mach) { bool add_extension = false; bool tplg_fixup = false; @@ -1349,7 +1367,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s%s%d%s", sof_pdata->tplg_filename, - "-dmic", + i2s_mach_found ? "-dmic" : "-", mach->mach_params.dmic_num, "ch"); if (!tplg_filename) @@ -1479,22 +1497,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } } - /* - * If I2S fails and no external HDaudio codec is detected, - * try SoundWire if it is supported - */ - if (!mach && !HDA_EXT_CODEC(bus->codec_mask) && - (interface_mask & BIT(SOF_DAI_INTEL_ALH))) - mach = hda_sdw_machine_select(sdev); - - /* - * Choose HDA generic machine driver if mach is NULL. - * Otherwise, set certain mach params. - */ - hda_generic_machine_select(sdev, &mach); - if (!mach) - dev_warn(sdev->dev, "warning: No matching ASoC machine driver found\n"); - return mach; } From 85b66359c5a76ce613bd75460e4646e8a27b1e2f Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Tue, 27 Aug 2024 20:32:03 +0800 Subject: [PATCH 260/410] ASoC: SOF: Intel: hda: refactoring topology name fixup for SDW mach Remove SDW mach's topology name fixup code and use the code in hda_machine_select() to fixup its topology file name. No functional change in this commit. Compared with I2S/HDA mach, SDW mach always fixup topology file name with dmic num without using DMIC quirk flag and pass topology name with file extension to SOF driver. Therefore, we add extra code to remove file extension if it exists. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-6-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/intel/hda.c | 166 +++++++++++--------------------------- 1 file changed, 46 insertions(+), 120 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 128687b24bf5..2f24b5abc91b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -511,6 +511,8 @@ static int check_dmic_num(struct snd_sof_dev *sdev) if (nhlt) dmic_num = intel_nhlt_get_dmic_geo(sdev->dev, nhlt); + dev_info(sdev->dev, "DMICs detected in NHLT tables: %d\n", dmic_num); + /* allow for module parameter override */ if (dmic_num_override != -1) { dev_dbg(sdev->dev, @@ -558,82 +560,6 @@ static int check_nhlt_ssp_mclk_mask(struct snd_sof_dev *sdev, int ssp_num) return intel_nhlt_ssp_mclk_mask(nhlt, ssp_num); } -#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) - -static const char *fixup_tplg_name(struct snd_sof_dev *sdev, - const char *sof_tplg_filename, - const char *idisp_str, - const char *dmic_str) -{ - const char *tplg_filename = NULL; - char *filename, *tmp; - const char *split_ext; - - filename = kstrdup(sof_tplg_filename, GFP_KERNEL); - if (!filename) - return NULL; - - /* this assumes a .tplg extension */ - tmp = filename; - split_ext = strsep(&tmp, "."); - if (split_ext) - tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, - "%s%s%s.tplg", - split_ext, idisp_str, dmic_str); - kfree(filename); - - return tplg_filename; -} - -static int dmic_detect_topology_fixup(struct snd_sof_dev *sdev, - const char **tplg_filename, - const char *idisp_str, - int *dmic_found, - bool tplg_fixup) -{ - const char *dmic_str; - int dmic_num; - - /* first check for DMICs (using NHLT or module parameter) */ - dmic_num = check_dmic_num(sdev); - - switch (dmic_num) { - case 1: - dmic_str = "-1ch"; - break; - case 2: - dmic_str = "-2ch"; - break; - case 3: - dmic_str = "-3ch"; - break; - case 4: - dmic_str = "-4ch"; - break; - default: - dmic_num = 0; - dmic_str = ""; - break; - } - - if (tplg_fixup) { - const char *default_tplg_filename = *tplg_filename; - const char *fixed_tplg_filename; - - fixed_tplg_filename = fixup_tplg_name(sdev, default_tplg_filename, - idisp_str, dmic_str); - if (!fixed_tplg_filename) - return -ENOMEM; - *tplg_filename = fixed_tplg_filename; - } - - dev_info(sdev->dev, "DMICs detected in NHLT tables: %d\n", dmic_num); - *dmic_found = dmic_num; - - return 0; -} -#endif - static int hda_init_caps(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); @@ -1199,45 +1125,10 @@ static struct snd_soc_acpi_mach *hda_sdw_machine_select(struct snd_sof_dev *sdev break; } if (mach && mach->link_mask) { - int dmic_num = 0; - bool tplg_fixup; - const char *tplg_filename; - mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; mach->mach_params.platform = dev_name(sdev->dev); - if (pdata->tplg_filename) { - tplg_fixup = false; - } else { - tplg_fixup = true; - tplg_filename = mach->sof_tplg_filename; - } - - /* - * DMICs use up to 4 pins and are typically pin-muxed with SoundWire - * link 2 and 3, or link 1 and 2, thus we only try to enable dmics - * if all conditions are true: - * a) 2 or fewer links are used by SoundWire - * b) the NHLT table reports the presence of microphones - */ - if (hweight_long(mach->link_mask) <= 2) { - int ret; - - ret = dmic_detect_topology_fixup(sdev, &tplg_filename, "", - &dmic_num, tplg_fixup); - if (ret < 0) - return NULL; - } - if (tplg_fixup) - pdata->tplg_filename = tplg_filename; - mach->mach_params.dmic_num = dmic_num; - - dev_dbg(sdev->dev, - "SoundWire machine driver %s topology %s\n", - mach->drv_name, - pdata->tplg_filename); - return mach; } @@ -1294,6 +1185,19 @@ static int check_tplg_quirk_mask(struct snd_soc_acpi_mach *mach) return 0; } +static char *remove_file_ext(const char *tplg_filename) +{ + char *filename, *tmp; + + filename = kstrdup(tplg_filename, GFP_KERNEL); + if (!filename) + return NULL; + + /* remove file extension if exist */ + tmp = filename; + return strsep(&tmp, "."); +} + struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) { u32 interface_mask = hda_get_interface_mask(sdev); @@ -1306,6 +1210,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) const char *tplg_suffix; bool amp_name_valid; bool i2s_mach_found = false; + bool sdw_mach_found = false; /* Try I2S or DMIC if it is supported */ if (interface_mask & (BIT(SOF_DAI_INTEL_SSP) | BIT(SOF_DAI_INTEL_DMIC))) { @@ -1319,8 +1224,11 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) * try SoundWire if it is supported */ if (!mach && !HDA_EXT_CODEC(bus->codec_mask) && - (interface_mask & BIT(SOF_DAI_INTEL_ALH))) + (interface_mask & BIT(SOF_DAI_INTEL_ALH))) { mach = hda_sdw_machine_select(sdev); + if (mach) + sdw_mach_found = true; + } /* * Choose HDA generic machine driver if mach is NULL. @@ -1335,15 +1243,20 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) * name string if quirk flag is set. */ if (mach) { - bool add_extension = false; bool tplg_fixup = false; + bool dmic_fixup = false; /* * If tplg file name is overridden, use it instead of * the one set in mach table */ if (!sof_pdata->tplg_filename) { - sof_pdata->tplg_filename = mach->sof_tplg_filename; + /* remove file extension if it exists */ + tplg_filename = remove_file_ext(mach->sof_tplg_filename); + if (!tplg_filename) + return NULL; + + sof_pdata->tplg_filename = tplg_filename; tplg_fixup = true; } @@ -1361,8 +1274,25 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) /* report to machine driver if any DMICs are found */ mach->mach_params.dmic_num = check_dmic_num(sdev); + if (sdw_mach_found) { + /* + * DMICs use up to 4 pins and are typically pin-muxed with SoundWire + * link 2 and 3, or link 1 and 2, thus we only try to enable dmics + * if all conditions are true: + * a) 2 or fewer links are used by SoundWire + * b) the NHLT table reports the presence of microphones + */ + if (hweight_long(mach->link_mask) <= 2) + dmic_fixup = true; + else + mach->mach_params.dmic_num = 0; + } else { + if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER) + dmic_fixup = true; + } + if (tplg_fixup && - mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER && + dmic_fixup && mach->mach_params.dmic_num) { tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s%s%d%s", @@ -1374,7 +1304,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) return NULL; sof_pdata->tplg_filename = tplg_filename; - add_extension = true; } if (mach->link_mask) { @@ -1414,7 +1343,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) return NULL; sof_pdata->tplg_filename = tplg_filename; - add_extension = true; mclk_mask = check_nhlt_ssp_mclk_mask(sdev, ssp_num); @@ -1453,7 +1381,6 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) return NULL; sof_pdata->tplg_filename = tplg_filename; - add_extension = true; } @@ -1475,10 +1402,9 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) return NULL; sof_pdata->tplg_filename = tplg_filename; - add_extension = true; } - if (tplg_fixup && add_extension) { + if (tplg_fixup) { tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s%s", sof_pdata->tplg_filename, From 775c1a4aa640c701eb67d3954a375000d2be3af5 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Aug 2024 20:32:04 +0800 Subject: [PATCH 261/410] ASoC: Intel: sof_sdw: move ignore_internal_dmic check earlier dmic links will not be created if ctx->ignore_internal_dmic is set, and dmic_num should be 0 in this case. Move ignore_internal_dmic check earlier where dmic_num is set to get an accurate dmic_num. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240827123215.258859-7-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9b642c6883b0..3781a27bfbed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1102,8 +1102,12 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) hdmi_num = SOF_PRE_TGL_HDMI_COUNT; /* enable dmic01 & dmic16k */ - if (sof_sdw_quirk & SOC_SDW_PCH_DMIC || mach_params->dmic_num) - dmic_num = 2; + if (sof_sdw_quirk & SOC_SDW_PCH_DMIC || mach_params->dmic_num) { + if (ctx->ignore_internal_dmic) + dev_warn(dev, "Ignoring PCH DMIC\n"); + else + dmic_num = 2; + } if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) bt_num = 1; @@ -1148,14 +1152,10 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) } /* dmic */ - if (dmic_num > 0) { - if (ctx->ignore_internal_dmic) { - dev_warn(dev, "Ignoring PCH DMIC\n"); - } else { - ret = create_dmic_dailinks(card, &dai_links, &be_id); - if (ret) - goto err_end; - } + if (dmic_num) { + ret = create_dmic_dailinks(card, &dai_links, &be_id); + if (ret) + goto err_end; } /* HDMI */ From 7db9f6361170a8caa32cc31d894fac1ba8dfc4f3 Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Tue, 27 Aug 2024 20:32:05 +0800 Subject: [PATCH 262/410] ASoC: Intel: sof_sdw: overwrite mach_params->dmic_num mach_params->dmic_num will be used to set the cfg-mics value of card->components string. Overwrite it to the actual number of PCH DMICs used in the device. Signed-off-by: Bard Liao Reviewed-by: Pierre-Louis Bossart Link: https://patch.msgid.link/20240827123215.258859-8-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 3781a27bfbed..b06948c16b3f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1108,6 +1108,11 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) else dmic_num = 2; } + /* + * mach_params->dmic_num will be used to set the cfg-mics value of card->components + * string. Overwrite it to the actual number of PCH DMICs used in the device. + */ + mach_params->dmic_num = dmic_num; if (sof_sdw_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) bt_num = 1; From 65dc80a78c5f42f5648396b218f50374cf1c2770 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Tue, 27 Aug 2024 20:32:06 +0800 Subject: [PATCH 263/410] ASoC: SOF: Intel: hda: support BT link mask in mach_params Add an new variable bt_link_mask to snd_soc_acpi_mach_params structure. SSP port mask of BT offload found in NHLT table will be sent to machine driver to setup BE dai link with correct SSP port number. This patch only detects and enables the BT dailink. The functionality will only be unlocked with a topology file that makes a reference to that BT dailink. For backwards-compatibility reasons, this topology will not be used by default. Chromebooks and Linux users willing to experiment shall use the tplg_name kernel parameter to force the use of an enhanced topology. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-9-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 2 ++ sound/soc/sof/intel/hda.c | 37 ++++++++++++++++++++++++++++++++----- 2 files changed, 34 insertions(+), 5 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b6d301946244..c1552bc6e31c 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -73,6 +73,7 @@ static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) * @subsystem_rev: optional PCI SSID revision value * @subsystem_id_set: true if a value has been written to * subsystem_vendor and subsystem_device. + * @bt_link_mask: BT offload link enabled on the board */ struct snd_soc_acpi_mach_params { u32 acpi_ipc_irq_index; @@ -89,6 +90,7 @@ struct snd_soc_acpi_mach_params { unsigned short subsystem_device; unsigned short subsystem_rev; bool subsystem_id_set; + u32 bt_link_mask; }; /** diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 2f24b5abc91b..d0a41e8e6334 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -444,6 +444,10 @@ static int mclk_id_override = -1; module_param_named(mclk_id, mclk_id_override, int, 0444); MODULE_PARM_DESC(mclk_id, "SOF SSP mclk_id"); +static int bt_link_mask_override; +module_param_named(bt_link_mask, bt_link_mask_override, int, 0444); +MODULE_PARM_DESC(bt_link_mask, "SOF BT offload link mask"); + static int hda_init(struct snd_sof_dev *sdev) { struct hda_bus *hbus; @@ -529,7 +533,7 @@ static int check_dmic_num(struct snd_sof_dev *sdev) return dmic_num; } -static int check_nhlt_ssp_mask(struct snd_sof_dev *sdev) +static int check_nhlt_ssp_mask(struct snd_sof_dev *sdev, u8 device_type) { struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; struct nhlt_acpi_table *nhlt; @@ -540,9 +544,11 @@ static int check_nhlt_ssp_mask(struct snd_sof_dev *sdev) return ssp_mask; if (intel_nhlt_has_endpoint_type(nhlt, NHLT_LINK_SSP)) { - ssp_mask = intel_nhlt_ssp_endpoint_mask(nhlt, NHLT_DEVICE_I2S); + ssp_mask = intel_nhlt_ssp_endpoint_mask(nhlt, device_type); if (ssp_mask) - dev_info(sdev->dev, "NHLT_DEVICE_I2S detected, ssp_mask %#x\n", ssp_mask); + dev_info(sdev->dev, "NHLT device %s(%d) detected, ssp_mask %#x\n", + device_type == NHLT_DEVICE_BT ? "BT" : "I2S", + device_type, ssp_mask); } return ssp_mask; @@ -1235,8 +1241,29 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) * Otherwise, set certain mach params. */ hda_generic_machine_select(sdev, &mach); - if (!mach) + if (!mach) { dev_warn(sdev->dev, "warning: No matching ASoC machine driver found\n"); + return NULL; + } + + /* report BT offload link mask to machine driver */ + mach->mach_params.bt_link_mask = check_nhlt_ssp_mask(sdev, NHLT_DEVICE_BT); + + dev_info(sdev->dev, "BT link detected in NHLT tables: %#x\n", + mach->mach_params.bt_link_mask); + + /* allow for module parameter override */ + if (bt_link_mask_override) { + dev_dbg(sdev->dev, "overriding BT link detected in NHLT tables %#x by kernel param %#x\n", + mach->mach_params.bt_link_mask, bt_link_mask_override); + mach->mach_params.bt_link_mask = bt_link_mask_override; + } + + if (hweight_long(mach->mach_params.bt_link_mask) > 1) { + dev_warn(sdev->dev, "invalid BT link mask %#x found, reset the mask\n", + mach->mach_params.bt_link_mask); + mach->mach_params.bt_link_mask = 0; + } /* * Fixup tplg file name by appending dmic num, ssp num, codec/amplifier @@ -1312,7 +1339,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) } /* report SSP link mask to machine driver */ - mach->mach_params.i2s_link_mask = check_nhlt_ssp_mask(sdev); + mach->mach_params.i2s_link_mask = check_nhlt_ssp_mask(sdev, NHLT_DEVICE_I2S); if (tplg_fixup && mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER && From 0752ba426a8182acbd083ca53fef32ddbf50e6d2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Tue, 27 Aug 2024 20:32:07 +0800 Subject: [PATCH 264/410] ASoC: Intel: skl_hda_dsp_generic: support BT audio offload Add BT offload BE link to dai link array if the BT offload link mask is valid (only one bit set). Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-10-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_common.c | 13 +++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 4 ++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 37 +++++++++++++++++--- 3 files changed, 49 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index e9cefa4ae56d..5eb63f4df241 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -75,6 +75,11 @@ SND_SOC_DAILINK_DEF(dmic_codec, SND_SOC_DAILINK_DEF(dmic16k, DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); +SND_SOC_DAILINK_DEF(bt_offload_pin, + DAILINK_COMP_ARRAY(COMP_CPU(""))); /* initialized in driver probe function */ +SND_SOC_DAILINK_DEF(dummy, + DAILINK_COMP_ARRAY(COMP_DUMMY())); + SND_SOC_DAILINK_DEF(platform, DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); @@ -132,6 +137,14 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .no_pcm = 1, SND_SOC_DAILINK_REG(dmic16k, dmic_codec, platform), }, + { + .name = NULL, /* initialized in driver probe function */ + .id = 8, + .dpcm_playback = 1, + .dpcm_capture = 1, + .no_pcm = 1, + SND_SOC_DAILINK_REG(bt_offload_pin, dummy, platform), + }, }; int skl_hda_hdmi_jack_init(struct snd_soc_card *card) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 19b814dee4ad..9d714f747dca 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -18,7 +18,7 @@ #include "../../codecs/hdac_hda.h" #include "hda_dsp_common.h" -#define HDA_DSP_MAX_BE_DAI_LINKS 7 +#define HDA_DSP_MAX_BE_DAI_LINKS 8 struct skl_hda_hdmi_pcm { struct list_head head; @@ -35,6 +35,8 @@ struct skl_hda_private { const char *platform_name; bool common_hdmi_codec_drv; bool idisp_codec; + bool bt_offload_present; + int ssp_bt; }; extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 88d91c0280bb..927c73bef065 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -95,6 +95,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) #define IDISP_DAI_COUNT 3 #define HDAC_DAI_COUNT 2 #define DMIC_DAI_COUNT 2 +#define BT_DAI_COUNT 1 /* there are two routes per iDisp output */ #define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) @@ -102,11 +103,12 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) #define HDA_CODEC_AUTOSUSPEND_DELAY_MS 1000 -static int skl_hda_fill_card_info(struct snd_soc_card *card, +static int skl_hda_fill_card_info(struct device *dev, struct snd_soc_card *card, struct snd_soc_acpi_mach_params *mach_params) { struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_dai_link *dai_link; + struct snd_soc_dai_link *bt_link; u32 codec_count, codec_mask; int i, num_links, num_route; @@ -120,7 +122,7 @@ static int skl_hda_fill_card_info(struct snd_soc_card *card, if (codec_mask == IDISP_CODEC_MASK) { /* topology with iDisp as the only HDA codec */ - num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT; + num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT + BT_DAI_COUNT; num_route = IDISP_ROUTE_COUNT; /* @@ -129,7 +131,7 @@ static int skl_hda_fill_card_info(struct snd_soc_card *card, * num_links of dai links need to be registered * to ASoC. */ - for (i = 0; i < DMIC_DAI_COUNT; i++) { + for (i = 0; i < (DMIC_DAI_COUNT + BT_DAI_COUNT); i++) { skl_hda_be_dai_links[IDISP_DAI_COUNT + i] = skl_hda_be_dai_links[IDISP_DAI_COUNT + HDAC_DAI_COUNT + i]; @@ -150,6 +152,28 @@ static int skl_hda_fill_card_info(struct snd_soc_card *card, } } + if (!ctx->bt_offload_present) { + /* remove last link since bt audio offload is not supported */ + num_links -= BT_DAI_COUNT; + } else { + if (codec_mask == IDISP_CODEC_MASK) + bt_link = &skl_hda_be_dai_links[IDISP_DAI_COUNT + DMIC_DAI_COUNT]; + else + bt_link = &skl_hda_be_dai_links[IDISP_DAI_COUNT + HDAC_DAI_COUNT + DMIC_DAI_COUNT]; + + /* complete the link name and dai name with SSP port number */ + bt_link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", + ctx->ssp_bt); + if (!bt_link->name) + return -ENOMEM; + + bt_link->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SSP%d Pin", + ctx->ssp_bt); + if (!bt_link->cpus->dai_name) + return -ENOMEM; + } + card->num_links = num_links; card->num_dapm_routes = num_route; @@ -213,7 +237,12 @@ static int skl_hda_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); - ret = skl_hda_fill_card_info(card, &mach->mach_params); + if (hweight_long(mach->mach_params.bt_link_mask) == 1) { + ctx->bt_offload_present = true; + ctx->ssp_bt = fls(mach->mach_params.bt_link_mask) - 1; + } + + ret = skl_hda_fill_card_info(&pdev->dev, card, &mach->mach_params); if (ret < 0) { dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); return ret; From 26254073e74e9ea507fbd1f628adf428e545be6b Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Tue, 27 Aug 2024 20:32:08 +0800 Subject: [PATCH 265/410] ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ARL match table. Adding ES83x6 codec support for ARL platforms and entry in match table. Signed-off-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-11-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index cc87c34e5a08..304f974dc960 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -95,7 +95,20 @@ static const struct snd_soc_acpi_link_adr arl_sdca_rvp[] = { {} }; +static const struct snd_soc_acpi_codecs arl_essx_83x6 = { + .num_codecs = 3, + .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = { + { + .comp_ids = &arl_essx_83x6, + .drv_name = "sof-essx8336", + .sof_tplg_filename = "sof-arl-es8336", /* the tplg suffix is added at run time */ + .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | + SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | + SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_machines); From e1580f48d4a5b739cfbfbce5de637bc34000dcad Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Tue, 27 Aug 2024 20:32:09 +0800 Subject: [PATCH 266/410] ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in ARL match table Adding HDMI-In capture via I2S feature support in ARL platform. Signed-off-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-12-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 10 ++++++++++ sound/soc/intel/common/soc-acpi-intel-arl-match.c | 12 ++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index b45d0501f109..578271b4230a 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -818,6 +818,16 @@ static const struct platform_device_id board_ids[] = { SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | SOF_ES8336_JD_INVERTED), }, + { + .name = "arl_es83x6_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | + SOF_NO_OF_HDMI_CAPTURE_SSP(2) | + SOF_HDMI_CAPTURE_1_SSP(0) | + SOF_HDMI_CAPTURE_2_SSP(2) | + SOF_SSP_HDMI_CAPTURE_PRESENT | + SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK | + SOF_ES8336_JD_INVERTED), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index 304f974dc960..8f69d61ea39c 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -100,7 +100,19 @@ static const struct snd_soc_acpi_codecs arl_essx_83x6 = { .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; +static const struct snd_soc_acpi_codecs arl_lt6911_hdmi = { + .num_codecs = 1, + .codecs = {"INTC10B0"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = { + { + .comp_ids = &arl_essx_83x6, + .drv_name = "arl_es83x6_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &arl_lt6911_hdmi, + .sof_tplg_filename = "sof-arl-es83x6-ssp1-hdmi-ssp02.tplg", + }, { .comp_ids = &arl_essx_83x6, .drv_name = "sof-essx8336", From a2a0312ac9ee3eb477a02f2706673cab6d375b07 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Tue, 27 Aug 2024 20:32:10 +0800 Subject: [PATCH 267/410] ASoC: Intel: skl_hda_dsp_generic: remove hdac-hdmi support Since this machine driver has no longer been enumerated by SKL platform driver, we could remove hdac-hdmi support code just like what we did to other SOF machine drivers. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-13-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 - sound/soc/intel/boards/skl_hda_dsp_common.c | 32 +++++--------------- sound/soc/intel/boards/skl_hda_dsp_common.h | 26 ---------------- sound/soc/intel/boards/skl_hda_dsp_generic.c | 8 +---- 4 files changed, 9 insertions(+), 58 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 1141fe1263ce..793a9170513a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -305,7 +305,6 @@ if SND_SOC_SOF_HDA_AUDIO_CODEC config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "Skylake+ with HDA Codecs" depends on SND_HDA_CODEC_HDMI - select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 5eb63f4df241..d1de772e9304 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -11,7 +11,6 @@ #include #include #include -#include "../../codecs/hdac_hdmi.h" #include "skl_hda_dsp_common.h" #include @@ -35,7 +34,6 @@ int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) if (!pcm->codec_dai) return -EINVAL; - pcm->device = device; list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); return 0; @@ -150,32 +148,18 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { int skl_hda_hdmi_jack_init(struct snd_soc_card *card) { struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component = NULL; + struct snd_soc_component *component; struct skl_hda_hdmi_pcm *pcm; - char jack_name[NAME_SIZE]; - int err; - if (ctx->common_hdmi_codec_drv) - return skl_hda_hdmi_build_controls(card); - - list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { - component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &pcm->hdmi_jack); - - if (err) - return err; - - err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, - &pcm->hdmi_jack); - if (err < 0) - return err; - } + /* HDMI disabled, do not create controls */ + if (list_empty(&ctx->hdmi_pcm_list)) + return 0; + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm, + head); + component = pcm->codec_dai->component; if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); + return hda_dsp_hdmi_build_controls(card, component); } diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 9d714f747dca..8455f953f4b8 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -23,8 +23,6 @@ struct skl_hda_hdmi_pcm { struct list_head head; struct snd_soc_dai *codec_dai; - struct snd_soc_jack hdmi_jack; - int device; }; struct skl_hda_private { @@ -33,7 +31,6 @@ struct skl_hda_private { int pcm_count; int dai_index; const char *platform_name; - bool common_hdmi_codec_drv; bool idisp_codec; bool bt_offload_present; int ssp_bt; @@ -43,27 +40,4 @@ extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; int skl_hda_hdmi_jack_init(struct snd_soc_card *card); int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device); -/* - * Search card topology and register HDMI PCM related controls - * to codec driver. - */ -static inline int skl_hda_hdmi_build_controls(struct snd_soc_card *card) -{ - struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component; - struct skl_hda_hdmi_pcm *pcm; - - /* HDMI disabled, do not create controls */ - if (list_empty(&ctx->hdmi_pcm_list)) - return 0; - - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm, - head); - component = pcm->codec_dai->component; - if (!component) - return -EINVAL; - - return hda_dsp_hdmi_build_controls(card, component); -} - #endif /* __SOUND_SOC_HDA_DSP_COMMON_H */ diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 927c73bef065..860a21915bce 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -13,7 +13,6 @@ #include #include #include -#include "../../codecs/hdac_hdmi.h" #include "skl_hda_dsp_common.h" static const struct snd_soc_dapm_widget skl_hda_widgets[] = { @@ -208,7 +207,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) static int skl_hda_audio_probe(struct platform_device *pdev) { - struct snd_soc_acpi_mach *mach; + struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; struct skl_hda_private *ctx; struct snd_soc_card *card; int ret; @@ -221,10 +220,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev) INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - mach = pdev->dev.platform_data; - if (!mach) - return -EINVAL; - card = &ctx->card; card->name = "hda-dsp", card->owner = THIS_MODULE, @@ -251,7 +246,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev) ctx->pcm_count = card->num_links; ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ ctx->platform_name = mach->mach_params.platform; - ctx->common_hdmi_codec_drv = mach->mach_params.common_hdmi_codec_drv; card->dev = &pdev->dev; if (!snd_soc_acpi_sof_parent(&pdev->dev)) From 690640ef35a42ed96c89ab71788efef4c47766bf Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Tue, 27 Aug 2024 20:32:11 +0800 Subject: [PATCH 268/410] ASoC: Intel: skl_hda_dsp_generic: use sof_hdmi_private to init HDMI Use sof_hdmi_private structure instead of a link list of skl_hda_hdmi_pcm structure for HDMI dai link initialization since hdac-hdmi support is removed. Signed-off-by: Brent Lu Reviewed-by: Pierre-Louis Bossart Reviewed-by: Kai Vehmanen Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-14-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_common.c | 23 ++++++-------------- sound/soc/intel/boards/skl_hda_dsp_common.h | 9 ++------ sound/soc/intel/boards/skl_hda_dsp_generic.c | 13 +++++------ 3 files changed, 15 insertions(+), 30 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index d1de772e9304..5019bdfa5b45 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -21,20 +21,16 @@ int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) { struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct skl_hda_hdmi_pcm *pcm; + struct snd_soc_dai *dai; char dai_name[NAME_SIZE]; - pcm = devm_kzalloc(card->dev, sizeof(*pcm), GFP_KERNEL); - if (!pcm) - return -ENOMEM; - snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d", ctx->dai_index); - pcm->codec_dai = snd_soc_card_get_codec_dai(card, dai_name); - if (!pcm->codec_dai) + dai = snd_soc_card_get_codec_dai(card, dai_name); + if (!dai) return -EINVAL; - list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + ctx->hdmi.hdmi_comp = dai->component; return 0; } @@ -148,18 +144,13 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { int skl_hda_hdmi_jack_init(struct snd_soc_card *card) { struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_component *component; - struct skl_hda_hdmi_pcm *pcm; /* HDMI disabled, do not create controls */ - if (list_empty(&ctx->hdmi_pcm_list)) + if (!ctx->hdmi.idisp_codec) return 0; - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct skl_hda_hdmi_pcm, - head); - component = pcm->codec_dai->component; - if (!component) + if (!ctx->hdmi.hdmi_comp) return -EINVAL; - return hda_dsp_hdmi_build_controls(card, component); + return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp); } diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h index 8455f953f4b8..40ffbccb2fe0 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ b/sound/soc/intel/boards/skl_hda_dsp_common.h @@ -17,21 +17,16 @@ #include #include "../../codecs/hdac_hda.h" #include "hda_dsp_common.h" +#include "sof_hdmi_common.h" #define HDA_DSP_MAX_BE_DAI_LINKS 8 -struct skl_hda_hdmi_pcm { - struct list_head head; - struct snd_soc_dai *codec_dai; -}; - struct skl_hda_private { struct snd_soc_card card; - struct list_head hdmi_pcm_list; + struct sof_hdmi_private hdmi; int pcm_count; int dai_index; const char *platform_name; - bool idisp_codec; bool bt_offload_present; int ssp_bt; }; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 860a21915bce..225867bb3310 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -75,7 +75,7 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) link->platforms->name = ctx->platform_name; link->nonatomic = 1; - if (!ctx->idisp_codec) + if (!ctx->hdmi.idisp_codec) return 0; if (strstr(link->name, "HDMI")) { @@ -98,7 +98,6 @@ skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) /* there are two routes per iDisp output */ #define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) -#define IDISP_CODEC_MASK 0x4 #define HDA_CODEC_AUTOSUSPEND_DELAY_MS 1000 @@ -113,10 +112,9 @@ static int skl_hda_fill_card_info(struct device *dev, struct snd_soc_card *card, codec_mask = mach_params->codec_mask; codec_count = hweight_long(codec_mask); - ctx->idisp_codec = !!(codec_mask & IDISP_CODEC_MASK); if (!codec_count || codec_count > 2 || - (codec_count == 2 && !ctx->idisp_codec)) + (codec_count == 2 && !ctx->hdmi.idisp_codec)) return -EINVAL; if (codec_mask == IDISP_CODEC_MASK) { @@ -141,7 +139,7 @@ static int skl_hda_fill_card_info(struct device *dev, struct snd_soc_card *card, num_route = ARRAY_SIZE(skl_hda_map); card->dapm_widgets = skl_hda_widgets; card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - if (!ctx->idisp_codec) { + if (!ctx->hdmi.idisp_codec) { card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; num_route -= IDISP_ROUTE_COUNT; for (i = 0; i < IDISP_DAI_COUNT; i++) { @@ -218,8 +216,6 @@ static int skl_hda_audio_probe(struct platform_device *pdev) if (!ctx) return -ENOMEM; - INIT_LIST_HEAD(&ctx->hdmi_pcm_list); - card = &ctx->card; card->name = "hda-dsp", card->owner = THIS_MODULE, @@ -232,6 +228,9 @@ static int skl_hda_audio_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ctx); + if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) + ctx->hdmi.idisp_codec = true; + if (hweight_long(mach->mach_params.bt_link_mask) == 1) { ctx->bt_offload_present = true; ctx->ssp_bt = fls(mach->mach_params.bt_link_mask) - 1; From c0524067653d07e78bc215220bc111b6e5e4a94d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 27 Aug 2024 20:32:12 +0800 Subject: [PATCH 269/410] ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops Add some new match table entries on Arrowlake for some coming cs42l43 laptops. Signed-off-by: Charles Keepax Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-15-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-arl-match.c | 219 ++++++++++++++++++ 1 file changed, 219 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index 8f69d61ea39c..9936a4b4eee7 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -15,6 +15,112 @@ static const struct snd_soc_acpi_endpoint single_endpoint = { .group_id = 0, }; +static const struct snd_soc_acpi_endpoint spk_l_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 0, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_r_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 1, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device cs35l56_2_lr_adr[] = { + { + .adr = 0x00023001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00023101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_lr_adr[] = { + { + .adr = 0x00033001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033401FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_2_r1_adr[] = { + { + .adr = 0x00023101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + }, +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l1_adr[] = { + { + .adr = 0x00033301fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, +}; + static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { { /* Jack Playback Endpoint */ .num = 0, @@ -51,6 +157,15 @@ static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { } }; +static const struct snd_soc_acpi_adr_device cs42l43_2_adr[] = { + { + .adr = 0x00023001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { { .adr = 0x000020025D071100ull, @@ -77,6 +192,80 @@ static const struct snd_soc_acpi_link_adr arl_cs42l43_l0[] = { }, }; +static const struct snd_soc_acpi_link_adr arl_cs42l43_l2[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs42l43_2_adr), + .adr_d = cs42l43_2_adr, + }, +}; + +static const struct snd_soc_acpi_link_adr arl_cs42l43_l2_cs35l56_l3[] = { + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs42l43_2_adr), + .adr_d = cs42l43_2_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_lr_adr), + .adr_d = cs35l56_3_lr_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr arl_cs42l43_l0_cs35l56_l2[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_lr_adr), + .adr_d = cs35l56_2_lr_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr arl_cs42l43_l0_cs35l56_l23[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr arl_cs42l43_l0_cs35l56_2_l23[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r1_adr), + .adr_d = cs35l56_2_r1_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l1_adr), + .adr_d = cs35l56_3_l1_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr arl_rvp[] = { { .mask = BIT(0), @@ -127,12 +316,42 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_machines); /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_sdw_machines[] = { + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = arl_cs42l43_l0_cs35l56_l23, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-cs42l43-l0-cs35l56-l23.tplg", + }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = arl_cs42l43_l0_cs35l56_2_l23, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-cs42l43-l0-cs35l56-l23.tplg", + }, + { + .link_mask = BIT(0) | BIT(2), + .links = arl_cs42l43_l0_cs35l56_l2, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-cs42l43-l0-cs35l56-l2.tplg", + }, { .link_mask = BIT(0), .links = arl_cs42l43_l0, .drv_name = "sof_sdw", .sof_tplg_filename = "sof-arl-cs42l43-l0.tplg", }, + { + .link_mask = BIT(2), + .links = arl_cs42l43_l2, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-cs42l43-l2.tplg", + }, + { + .link_mask = BIT(2) | BIT(3), + .links = arl_cs42l43_l2_cs35l56_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-arl-cs42l43-l2-cs35l56-l3.tplg", + }, { .link_mask = 0x1, /* link0 required */ .links = arl_rvp, From 9ed85cb8c3af96028b2ba6209e9a8d2707e54f5d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 27 Aug 2024 20:32:13 +0800 Subject: [PATCH 270/410] ASoC: Intel: soc-acpi: adl: Add match entries for new cs42l43 laptops Add some new match table entries on Alderlake for some coming cs42l43 laptops. Signed-off-by: Charles Keepax Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-16-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-adl-match.c | 105 ++++++++++++++++++ 1 file changed, 105 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 4167b2e9bc6a..bb1324fb588e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -35,6 +35,86 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1, }; +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1, +}; + +static const struct snd_soc_acpi_adr_device cs35l56_2_r_adr[] = { + { + .adr = 0x00023201FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00023301FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_l_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { + { /* Jack Playback Endpoint */ + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* DMIC Capture Endpoint */ + .num = 1, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Jack Capture Endpoint */ + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + { /* Speaker Playback Endpoint */ + .num = 3, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { + { + .adr = 0x00003001FA424301ull, + .num_endpoints = ARRAY_SIZE(cs42l43_endpoints), + .endpoints = cs42l43_endpoints, + .name_prefix = "cs42l43" + } +}; + static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { { .adr = 0x000020025D071100ull, @@ -416,6 +496,25 @@ static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = { } }; +static const struct snd_soc_acpi_link_adr adl_cs42l43_l0_cs35l56_l23[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_r_adr), + .adr_d = cs35l56_2_r_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_l_adr), + .adr_d = cs35l56_3_l_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr adl_rvp[] = { { .mask = BIT(0), @@ -560,6 +659,12 @@ EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); /* this table is used when there is no I2S codec present */ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = adl_cs42l43_l0_cs35l56_l23, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-adl-cs42l43-l0-cs35l56-l23.tplg", + }, { .link_mask = 0xF, /* 4 active links required */ .links = adl_default, From 1be6b1c689575e80a92d4f3bb85e22b805355059 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 27 Aug 2024 20:32:14 +0800 Subject: [PATCH 271/410] ASoC: Intel: soc-acpi: lnl: Add match entries for new cs42l43 laptops Add some new match table entries on Lunarlake for some coming cs42l43 laptops. Signed-off-by: Charles Keepax Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-17-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- .../intel/common/soc-acpi-intel-lnl-match.c | 104 ++++++++++++++++++ 1 file changed, 104 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c index e6ffcd5be6c5..2522e6f11749 100644 --- a/sound/soc/intel/common/soc-acpi-intel-lnl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-lnl-match.c @@ -36,6 +36,20 @@ static const struct snd_soc_acpi_endpoint spk_r_endpoint = { .group_id = 1, }; +static const struct snd_soc_acpi_endpoint spk_2_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 2, + .group_id = 1, +}; + +static const struct snd_soc_acpi_endpoint spk_3_endpoint = { + .num = 0, + .aggregated = 1, + .group_position = 3, + .group_id = 1, +}; + static const struct snd_soc_acpi_endpoint rt712_endpoints[] = { { .num = 0, @@ -103,6 +117,51 @@ static const struct snd_soc_acpi_endpoint cs42l43_endpoints[] = { }, }; +static const struct snd_soc_acpi_adr_device cs35l56_2_l_adr[] = { + { + .adr = 0x00023001FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00023101FA355601ull, + .num_endpoints = 1, + .endpoints = &spk_2_endpoint, + .name_prefix = "AMP2" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_r_adr[] = { + { + .adr = 0x00033201fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP3" + }, + { + .adr = 0x00033301fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_3_endpoint, + .name_prefix = "AMP4" + } +}; + +static const struct snd_soc_acpi_adr_device cs35l56_3_lr_adr[] = { + { + .adr = 0x00033001fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "AMP1" + }, + { + .adr = 0x00033101fa355601ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "AMP2" + } +}; + static const struct snd_soc_acpi_adr_device cs42l43_0_adr[] = { { .adr = 0x00003001FA424301ull, @@ -210,6 +269,39 @@ static const struct snd_soc_acpi_link_adr lnl_cs42l43_l0[] = { }, }; +static const struct snd_soc_acpi_link_adr lnl_cs42l43_l0_cs35l56_l3[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_lr_adr), + .adr_d = cs35l56_3_lr_adr, + }, + {} +}; + +static const struct snd_soc_acpi_link_adr lnl_cs42l43_l0_cs35l56_l23[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(cs42l43_0_adr), + .adr_d = cs42l43_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(cs35l56_2_l_adr), + .adr_d = cs35l56_2_l_adr, + }, + { + .mask = BIT(3), + .num_adr = ARRAY_SIZE(cs35l56_3_r_adr), + .adr_d = cs35l56_3_r_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr lnl_rvp[] = { { .mask = BIT(0), @@ -312,6 +404,18 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_lnl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-lnl-rt711-l0-rt1316-l23-rt714-l1.tplg", }, + { + .link_mask = BIT(0) | BIT(2) | BIT(3), + .links = lnl_cs42l43_l0_cs35l56_l23, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-cs42l43-l0-cs35l56-l23.tplg", + }, + { + .link_mask = BIT(0) | BIT(3), + .links = lnl_cs42l43_l0_cs35l56_l3, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-lnl-cs42l43-l0-cs35l56-l3.tplg", + }, { .link_mask = BIT(0), .links = lnl_cs42l43_l0, From 9307694f340e518cac0e007f39dd9ff0736e6144 Mon Sep 17 00:00:00 2001 From: Maciej Strozek Date: Tue, 27 Aug 2024 20:32:15 +0800 Subject: [PATCH 272/410] ASoC: Intel: sof_sdw: Add quirks from some new Dell laptops Add quirks for some new Dell laptops using cs42l43's speaker outputs. Signed-off-by: Maciej Strozek Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240827123215.258859-18-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 58 ++++++++++++++++++++++++++++++++ 1 file changed, 58 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index b06948c16b3f..855109d22131 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -480,6 +480,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { .driver_data = (void *)(SOF_SDW_TGL_HDMI | RT711_JD2), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF9") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, /* MeteorLake devices */ { .callback = sof_sdw_quirk_cb, @@ -540,6 +548,56 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(SOC_SDW_SIDECAR_AMPS), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CDB") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CDC") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CDD") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF8") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + + /* ArrowLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE8") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CF7") + }, + .driver_data = (void *)(SOC_SDW_CODEC_SPKR), + }, {} }; From 4dd4baa4408a15d50233e85bae611d576ef77b92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 26 Aug 2024 23:41:52 +0000 Subject: [PATCH 273/410] ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top This patch moves snd_soc_dpcm_can_be_xxx() functions to top of soc-pcm.c This is prepare for cleanup. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87ikvmdf7j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 204 ++++++++++++++++++++++---------------------- 1 file changed, 102 insertions(+), 102 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index c4834c741fa4..f72a492f959e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,6 +49,108 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } +/* is the current PCM operation for this FE ? */ +int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) +{ + if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); + +/* is the current PCM operation for this BE ? */ +int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || + ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) && + be->dpcm[stream].runtime_update)) + return 1; + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); + +static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, + int stream, + const enum snd_soc_dpcm_state *states, + int num_states) +{ + struct snd_soc_dpcm *dpcm; + int state; + int ret = 1; + int i; + + for_each_dpcm_fe(be, stream, dpcm) { + + if (dpcm->fe == fe) + continue; + + state = dpcm->fe->dpcm[stream].state; + for (i = 0; i < num_states; i++) { + if (state == states[i]) { + ret = 0; + break; + } + } + } + + /* it's safe to do this BE DAI */ + return ret; +} + +/* + * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE + * are not running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); + +/* + * We can only change hw params a BE DAI if any of it's FE are not prepared, + * running, paused or suspended for the specified stream direction. + */ +int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_SUSPEND, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); + #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2969,27 +3071,6 @@ out: return ret; } -/* is the current PCM operation for this FE ? */ -int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) -{ - if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) - return 1; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); - -/* is the current PCM operation for this BE ? */ -int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) -{ - if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || - ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_BE) && - be->dpcm[stream].runtime_update)) - return 1; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); - /* get the substream for this BE */ struct snd_pcm_substream * snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream) @@ -2997,84 +3078,3 @@ struct snd_pcm_substream * return be->pcm->streams[stream].substream; } EXPORT_SYMBOL_GPL(snd_soc_dpcm_get_substream); - -static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, - int stream, - const enum snd_soc_dpcm_state *states, - int num_states) -{ - struct snd_soc_dpcm *dpcm; - int state; - int ret = 1; - int i; - - for_each_dpcm_fe(be, stream, dpcm) { - - if (dpcm->fe == fe) - continue; - - state = dpcm->fe->dpcm[stream].state; - for (i = 0; i < num_states; i++) { - if (state == states[i]) { - ret = 0; - break; - } - } - } - - /* it's safe to do this BE DAI */ - return ret; -} - -/* - * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE - * are not running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) -{ - const enum snd_soc_dpcm_state state[] = { - SND_SOC_DPCM_STATE_START, - SND_SOC_DPCM_STATE_PAUSED, - SND_SOC_DPCM_STATE_SUSPEND, - }; - - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); - -/* - * We can only change hw params a BE DAI if any of it's FE are not prepared, - * running, paused or suspended for the specified stream direction. - */ -int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) -{ - const enum snd_soc_dpcm_state state[] = { - SND_SOC_DPCM_STATE_START, - SND_SOC_DPCM_STATE_PAUSED, - SND_SOC_DPCM_STATE_SUSPEND, - SND_SOC_DPCM_STATE_PREPARE, - }; - - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); - -/* - * We can only prepare a BE DAI if any of it's FE are not prepared, - * running or paused for the specified stream direction. - */ -int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) -{ - const enum snd_soc_dpcm_state state[] = { - SND_SOC_DPCM_STATE_START, - SND_SOC_DPCM_STATE_PAUSED, - SND_SOC_DPCM_STATE_PREPARE, - }; - - return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); -} -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); From 290f31e943a29c93532b1684652b04fd60d0f696 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 26 Aug 2024 23:41:57 +0000 Subject: [PATCH 274/410] ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. Let's makes it static function. One note is that snd_soc_dpcm_fe_can_update() is not used in upstream. Use #if-endif and keep it for future support. Signed-off-by: Kuninori Morimoto Link: https://patch.msgid.link/87h6b6df7e.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc-dpcm.h | 18 ------------------ sound/soc/soc-pcm.c | 23 ++++++++++------------- 2 files changed, 10 insertions(+), 31 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 773f2db8c31c..c6fb350b4b06 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -113,24 +113,6 @@ struct snd_soc_dpcm_runtime { #define for_each_dpcm_be_rollback(fe, stream, _dpcm) \ list_for_each_entry_continue_reverse(_dpcm, &(fe)->dpcm[stream].be_clients, list_be) -/* can this BE stop and free */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream); - -/* can this BE perform a hw_params() */ -int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream); - -/* can this BE perform prepare */ -int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream); - -/* is the current PCM operation for this FE ? */ -int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); - -/* is the current PCM operation for this BE ? */ -int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream); /* get the substream for this BE */ struct snd_pcm_substream * diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f72a492f959e..7a59121fc323 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -50,16 +50,17 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, } /* is the current PCM operation for this FE ? */ -int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) +#if 0 +static int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream) { if (fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) return 1; return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dpcm_fe_can_update); +#endif /* is the current PCM operation for this BE ? */ -int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, +static int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream) { if ((fe->dpcm[stream].runtime_update == SND_SOC_DPCM_UPDATE_FE) || @@ -68,7 +69,6 @@ int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe, return 1; return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dpcm_be_can_update); static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, @@ -103,8 +103,8 @@ static int snd_soc_dpcm_check_state(struct snd_soc_pcm_runtime *fe, * We can only hw_free, stop, pause or suspend a BE DAI if any of it's FE * are not running, paused or suspended for the specified stream direction. */ -int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, @@ -114,14 +114,13 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_free_stop); /* * We can only change hw params a BE DAI if any of it's FE are not prepared, * running, paused or suspended for the specified stream direction. */ -int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, @@ -132,14 +131,13 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); /* * We can only prepare a BE DAI if any of it's FE are not prepared, * running or paused for the specified stream direction. */ -int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, - struct snd_soc_pcm_runtime *be, int stream) +static int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) { const enum snd_soc_dpcm_state state[] = { SND_SOC_DPCM_STATE_START, @@ -149,7 +147,6 @@ int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } -EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); #define DPCM_MAX_BE_USERS 8 From 75560718e83bede12d70b2a6b6940089477f58e4 Mon Sep 17 00:00:00 2001 From: Hongbo Li Date: Wed, 28 Aug 2024 20:28:29 +0800 Subject: [PATCH 275/410] ASoC: dapm: Use IS_ERR_OR_NULL() helper function Use the IS_ERR_OR_NULL() helper instead of open-coding a NULL and an error pointer checks to simplify the code and improve readability. Signed-off-by: Hongbo Li Link: https://patch.msgid.link/20240828122829.3697502-1-lihongbo22@huawei.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d7d6dbb9d9ea..ad289f91b13a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2254,7 +2254,7 @@ static const struct file_operations dapm_bias_fops = { void snd_soc_dapm_debugfs_init(struct snd_soc_dapm_context *dapm, struct dentry *parent) { - if (!parent || IS_ERR(parent)) + if (IS_ERR_OR_NULL(parent)) return; dapm->debugfs_dapm = debugfs_create_dir("dapm", parent); From 4b1d9019b26fd654ebc2d0d2e100ed56ef1821f0 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Wed, 28 Aug 2024 15:53:54 +0200 Subject: [PATCH 276/410] ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property The sound card design is based on reference PLL frequencies that are the root of all clock rates calculations. Today, those frequencies are currently specified in DT via assigned-clocks, because they correspond to the basic audio use-case. It makes no sense to setup clock rates for a sound card without referencing the clocks for the sound card, mainly because at some point more complex audio use cases will be supported and those root rates would need to change. To solve this situation, let's legitimize the presence of assigned-clocks in the sound card by documenting those clocks, as it describes a true dependency of the sound card and paths the way of more complex audio uses-cases involving those root frequencies. Signed-off-by: Neil Armstrong Acked-by: Conor Dooley Link: https://patch.msgid.link/20240828-topic-amlogic-upstream-bindings-fixes-audio-snd-card-v2-1-58159abf0779@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/amlogic,axg-sound-card.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml index 5db718e4d0e7..4f13e8ab50b2 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml @@ -26,6 +26,13 @@ properties: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name + clocks: + minItems: 1 + maxItems: 3 + description: + Base PLL clocks of audio susbsytem, used to configure base clock + frequencies for different audio use-cases. + patternProperties: "^dai-link-[0-9]+$": type: object From f189c972f86b00318cf2547b62e461cb98374e34 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Wed, 28 Aug 2024 15:53:55 +0200 Subject: [PATCH 277/410] ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property The sound card design is based on reference PLL frequencies that are the root of all clock rates calculations. Today, those frequencies are currently specified in DT via assigned-clocks, because they correspond to the basic audio use-case. It makes no sense to setup clock rates for a sound card without referencing the clocks for the sound card, mainly because at some point more complex audio use cases will be supported and those root rates would need to change. To solve this situation, let's legitimize the presence of assigned-clocks in the sound card by documenting those clocks, as it describes a true dependency of the sound card and paths the way of more complex audio uses-cases involving those root frequencies. Signed-off-by: Neil Armstrong Acked-by: Conor Dooley Link: https://patch.msgid.link/20240828-topic-amlogic-upstream-bindings-fixes-audio-snd-card-v2-2-58159abf0779@linaro.org Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml index 0ecdaf7190e9..413b47778181 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml @@ -27,6 +27,13 @@ properties: A list off component DAPM widget. Each entry is a pair of strings, the first being the widget type, the second being the widget name + clocks: + minItems: 1 + maxItems: 3 + description: + Base PLL clocks of audio susbsytem, used to configure base clock + frequencies for different audio use-cases. + patternProperties: "^dai-link-[0-9]+$": type: object From 125b749221aa54db069a7805f08d99b55c74c88c Mon Sep 17 00:00:00 2001 From: Lukasz Majewski Date: Wed, 28 Aug 2024 11:27:09 +0200 Subject: [PATCH 278/410] ASoC: dt-bindings: Convert mxs-saif.txt to fsl,saif.yaml (imx28 saif) The 'fsl,imx28-saif' compatible has already the mxs-saif.txt description. This patch converts (and removes it) this file to fsl,saif.yaml (to follow current fsl convention). Changes for the mxs-saif.txt: - Adds 'clocks', '#clock-cells' and '#sound-dai-cells' properties - Provide device description Signed-off-by: Lukasz Majewski Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240828092709.2626359-1-lukma@denx.de Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/fsl,saif.yaml | 83 +++++++++++++++++++ .../devicetree/bindings/sound/mxs-saif.txt | 41 --------- 2 files changed, 83 insertions(+), 41 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,saif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mxs-saif.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,saif.yaml b/Documentation/devicetree/bindings/sound/fsl,saif.yaml new file mode 100644 index 000000000000..0b5db6bb1b7c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,saif.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,saif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale MXS Serial Audio Interface (SAIF) + +maintainers: + - Lukasz Majewski + +allOf: + - $ref: dai-common.yaml# + +description: + The SAIF is based on I2S module that is used to communicate with audio codecs, + but only with half-duplex manner (i.e. it can either transmit or receive PCM + audio). + +properties: + compatible: + const: fsl,imx28-saif + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + interrupts: + maxItems: 1 + + dmas: + maxItems: 1 + + dma-names: + const: rx-tx + + "#clock-cells": + description: Configure the I2S device as MCLK clock provider. + const: 0 + + clocks: + maxItems: 1 + + fsl,saif-master: + description: Indicate that saif is a slave and its phandle points to master + $ref: /schemas/types.yaml#/definitions/phandle + +required: + - compatible + - reg + - "#sound-dai-cells" + - interrupts + - dmas + - dma-names + - clocks + +unevaluatedProperties: false + +examples: + - | + saif0: saif@80042000 { + compatible = "fsl,imx28-saif"; + reg = <0x80042000 2000>; + #sound-dai-cells = <0>; + interrupts = <59>; + dmas = <&dma_apbx 4>; + dma-names = "rx-tx"; + #clock-cells = <0>; + clocks = <&clks 53>; + }; + - | + saif1: saif@80046000 { + compatible = "fsl,imx28-saif"; + reg = <0x80046000 2000>; + #sound-dai-cells = <0>; + interrupts = <58>; + dmas = <&dma_apbx 5>; + dma-names = "rx-tx"; + clocks = <&clks 53>; + fsl,saif-master = <&saif0>; + }; diff --git a/Documentation/devicetree/bindings/sound/mxs-saif.txt b/Documentation/devicetree/bindings/sound/mxs-saif.txt deleted file mode 100644 index 7ba07a118e37..000000000000 --- a/Documentation/devicetree/bindings/sound/mxs-saif.txt +++ /dev/null @@ -1,41 +0,0 @@ -* Freescale MXS Serial Audio Interface (SAIF) - -Required properties: -- compatible: Should be "fsl,-saif" -- reg: Should contain registers location and length -- interrupts: Should contain ERROR interrupt number -- dmas: DMA specifier, consisting of a phandle to DMA controller node - and SAIF DMA channel ID. - Refer to dma.txt and fsl-mxs-dma.txt for details. -- dma-names: Must be "rx-tx". - -Optional properties: -- fsl,saif-master: phandle to the master SAIF. It's only required for - the slave SAIF. - -Note: Each SAIF controller should have an alias correctly numbered -in "aliases" node. - -Example: - -aliases { - saif0 = &saif0; - saif1 = &saif1; -}; - -saif0: saif@80042000 { - compatible = "fsl,imx28-saif"; - reg = <0x80042000 2000>; - interrupts = <59>; - dmas = <&dma_apbx 4>; - dma-names = "rx-tx"; -}; - -saif1: saif@80046000 { - compatible = "fsl,imx28-saif"; - reg = <0x80046000 2000>; - interrupts = <58>; - dmas = <&dma_apbx 5>; - dma-names = "rx-tx"; - fsl,saif-master = <&saif0>; -}; From 7817eb1ad353d732bac546b39316f2422c76fff4 Mon Sep 17 00:00:00 2001 From: Nikita Shubin Date: Thu, 29 Aug 2024 10:52:57 +0300 Subject: [PATCH 279/410] ASoC: dt-bindings: cirrus,cs4271: Convert to dtschema Convert the Cirrus Logic CS4271 audio CODEC bindings to DT schema. Add missing spi-cpha, spi-cpol, '#sound-dai-cells' and port, as they are already being used in the DTS and the driver for this device. Switch to 'reset-gpios' and drop legacy 'reset-gpio' used in original bindings. Based on Animesh Agarwal cs42xx8 conversion patch. Cc: Alexander Sverdlin Link: https://lore.kernel.org/all/20240715-ep93xx-v11-0-4e924efda795@maquefel.me Signed-off-by: Nikita Shubin Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240829-cs4271-yaml-v3-1-f1624cc838f6@maquefel.me Signed-off-by: Mark Brown --- .../bindings/sound/cirrus,cs4271.yaml | 101 ++++++++++++++++++ .../devicetree/bindings/sound/cs4271.txt | 57 ---------- 2 files changed, 101 insertions(+), 57 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml new file mode 100644 index 000000000000..68fbf5cc208f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/cirrus,cs4271.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Cirrus Logic CS4271 audio CODEC + +maintainers: + - Alexander Sverdlin + - Nikita Shubin + +description: + The CS4271 is a stereo audio codec. This device supports both the I2C + and the SPI bus. + +allOf: + - $ref: dai-common.yaml# + - $ref: /schemas/spi/spi-peripheral-props.yaml# + +properties: + compatible: + const: cirrus,cs4271 + + reg: + maxItems: 1 + + spi-cpha: true + + spi-cpol: true + + '#sound-dai-cells': + const: 0 + + reset-gpios: + description: + This pin will be deasserted before communication to the codec starts. + maxItems: 1 + + va-supply: + description: Analog power supply. + + vd-supply: + description: Digital power supply. + + vl-supply: + description: Serial Control Port power supply. + + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + + cirrus,amuteb-eq-bmutec: + description: + When given, the Codec's AMUTEB=BMUTEC flag is enabled. + type: boolean + + cirrus,enable-soft-reset: + description: | + The CS4271 requires its LRCLK and MCLK to be stable before its RESET + line is de-asserted. That also means that clocks cannot be changed + without putting the chip back into hardware reset, which also requires + a complete re-initialization of all registers. + + One (undocumented) workaround is to assert and de-assert the PDN bit + in the MODE2 register. This workaround can be enabled with this DT + property. + + Note that this is not needed in case the clocks are stable + throughout the entire runtime of the codec. + type: boolean + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include + spi { + #address-cells = <1>; + #size-cells = <0>; + codec@0 { + compatible = "cirrus,cs4271"; + reg = <0>; + #sound-dai-cells = <0>; + spi-max-frequency = <6000000>; + spi-cpol; + spi-cpha; + reset-gpios = <&gpio0 1 GPIO_ACTIVE_LOW>; + port { + endpoint { + remote-endpoint = <&i2s_ep>; + }; + }; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/cs4271.txt b/Documentation/devicetree/bindings/sound/cs4271.txt deleted file mode 100644 index 6e699ceabacd..000000000000 --- a/Documentation/devicetree/bindings/sound/cs4271.txt +++ /dev/null @@ -1,57 +0,0 @@ -Cirrus Logic CS4271 DT bindings - -This driver supports both the I2C and the SPI bus. - -Required properties: - - - compatible: "cirrus,cs4271" - -For required properties on SPI, please consult -Documentation/devicetree/bindings/spi/spi-bus.txt - -Required properties on I2C: - - - reg: the i2c address - - -Optional properties: - - - reset-gpio: a GPIO spec to define which pin is connected to the chip's - !RESET pin - - cirrus,amuteb-eq-bmutec: When given, the Codec's AMUTEB=BMUTEC flag - is enabled. - - cirrus,enable-soft-reset: - The CS4271 requires its LRCLK and MCLK to be stable before its RESET - line is de-asserted. That also means that clocks cannot be changed - without putting the chip back into hardware reset, which also requires - a complete re-initialization of all registers. - - One (undocumented) workaround is to assert and de-assert the PDN bit - in the MODE2 register. This workaround can be enabled with this DT - property. - - Note that this is not needed in case the clocks are stable - throughout the entire runtime of the codec. - - - vd-supply: Digital power - - vl-supply: Logic power - - va-supply: Analog Power - -Examples: - - codec_i2c: cs4271@10 { - compatible = "cirrus,cs4271"; - reg = <0x10>; - reset-gpio = <&gpio 23 0>; - vd-supply = <&vdd_3v3_reg>; - vl-supply = <&vdd_3v3_reg>; - va-supply = <&vdd_3v3_reg>; - }; - - codec_spi: cs4271@0 { - compatible = "cirrus,cs4271"; - reg = <0x0>; - reset-gpio = <&gpio 23 0>; - spi-max-frequency = <6000000>; - }; - From 6b99dc62d940758455b0c7e7ffbf3a63fecc87cb Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski Date: Fri, 9 Aug 2024 13:01:22 +0200 Subject: [PATCH 280/410] ASoC: codecs: wsa884x: Implement temperature reading and hwmon Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240809110122.137761-1-krzysztof.kozlowski@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wsa884x.c | 201 +++++++++++++++++++++++++++++++++++++ 1 file changed, 201 insertions(+) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 8db1380d1f10..86df5152c547 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include +#include #include #include +#include #include #include #include +#include #include #include #include @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1660,6 +1701,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1671,6 +1716,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1810,6 +1859,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1866,6 +2053,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1923,6 +2112,18 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + if (IS_REACHABLE(CONFIG_HWMON)) { + struct device *hwmon; + + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", + wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + } + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); From 2186fe21e57aada45f365a79cdb6d21140a539e2 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Thu, 29 Aug 2024 16:11:14 +0000 Subject: [PATCH 281/410] ALSA: hda/realtek: Autodetect Cirrus Logic companion amplifier bindings We do not need model specific HDA quirks to construct the component binding details of Cirrus Logic companion amplifiers as this information is already present in the ACPI. Quirks are then only required for special workarounds not described in the ACPI such as internal configuration of the Realtek codecs. The codec pointer is now initialized in hda_component_manager_init() so that we can detect when companion amplifiers are found in the ACPI but the SSID invokes a quirk that also attempts to create the component binding. Signed-off-by: Simon Trimmer Link: https://patch.msgid.link/20240829161114.140938-1-simont@opensource.cirrus.com Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_component.c | 19 +++- sound/pci/hda/hda_component.h | 2 +- sound/pci/hda/patch_realtek.c | 168 ++++++++++++++-------------------- 3 files changed, 87 insertions(+), 102 deletions(-) diff --git a/sound/pci/hda/hda_component.c b/sound/pci/hda/hda_component.c index b7dfdb10d156..2d6b7b0b355d 100644 --- a/sound/pci/hda/hda_component.c +++ b/sound/pci/hda/hda_component.c @@ -142,7 +142,6 @@ int hda_component_manager_bind(struct hda_codec *cdc, /* Init shared and component specific data */ memset(parent->comps, 0, sizeof(parent->comps)); - parent->codec = cdc; mutex_lock(&parent->mutex); ret = component_bind_all(hda_codec_dev(cdc), parent); @@ -163,6 +162,13 @@ int hda_component_manager_init(struct hda_codec *cdc, struct hda_scodec_match *sm; int ret, i; + if (parent->codec) { + codec_err(cdc, "Component binding already created (SSID: %x)\n", + cdc->core.subsystem_id); + return -EINVAL; + } + parent->codec = cdc; + mutex_init(&parent->mutex); for (i = 0; i < count; i++) { @@ -185,12 +191,19 @@ int hda_component_manager_init(struct hda_codec *cdc, } EXPORT_SYMBOL_NS_GPL(hda_component_manager_init, SND_HDA_SCODEC_COMPONENT); -void hda_component_manager_free(struct hda_codec *cdc, +void hda_component_manager_free(struct hda_component_parent *parent, const struct component_master_ops *ops) { - struct device *dev = hda_codec_dev(cdc); + struct device *dev; + + if (!parent->codec) + return; + + dev = hda_codec_dev(parent->codec); component_master_del(dev, ops); + + parent->codec = NULL; } EXPORT_SYMBOL_NS_GPL(hda_component_manager_free, SND_HDA_SCODEC_COMPONENT); diff --git a/sound/pci/hda/hda_component.h b/sound/pci/hda/hda_component.h index 9f786608144c..7ee37154749f 100644 --- a/sound/pci/hda/hda_component.h +++ b/sound/pci/hda/hda_component.h @@ -75,7 +75,7 @@ int hda_component_manager_init(struct hda_codec *cdc, const char *match_str, const struct component_master_ops *ops); -void hda_component_manager_free(struct hda_codec *cdc, +void hda_component_manager_free(struct hda_component_parent *parent, const struct component_master_ops *ops); int hda_component_manager_bind(struct hda_codec *cdc, struct hda_component_parent *parent); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c55c49db2434..2d4db5ce8624 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -949,7 +949,18 @@ static int alc_init(struct hda_codec *codec) return 0; } -#define alc_free snd_hda_gen_free +/* forward declaration */ +static const struct component_master_ops comp_master_ops; + +static void alc_free(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (spec) + hda_component_manager_free(&spec->comps, &comp_master_ops); + + snd_hda_gen_free(codec); +} static inline void alc_shutup(struct hda_codec *codec) { @@ -6879,14 +6890,12 @@ static void comp_generic_fixup(struct hda_codec *cdc, int action, const char *bu spec->gen.pcm_playback_hook = comp_generic_playback_hook; break; case HDA_FIXUP_ACT_FREE: - hda_component_manager_free(cdc, &comp_master_ops); + hda_component_manager_free(&spec->comps, &comp_master_ops); break; } } -static void cs35lxx_autodet_fixup(struct hda_codec *cdc, - const struct hda_fixup *fix, - int action) +static void find_cirrus_companion_amps(struct hda_codec *cdc) { struct device *dev = hda_codec_dev(cdc); struct acpi_device *adev; @@ -6901,67 +6910,53 @@ static void cs35lxx_autodet_fixup(struct hda_codec *cdc, char *match; int i, count = 0, count_devindex = 0; - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - for (i = 0; i < ARRAY_SIZE(acpi_ids); ++i) { - adev = acpi_dev_get_first_match_dev(acpi_ids[i].hid, NULL, -1); - if (adev) - break; - } - if (!adev) { - dev_err(dev, "Failed to find ACPI entry for a Cirrus Amp\n"); - return; - } - - count = i2c_acpi_client_count(adev); - if (count > 0) { - bus = "i2c"; - } else { - count = acpi_spi_count_resources(adev); - if (count > 0) - bus = "spi"; - } - - fwnode = fwnode_handle_get(acpi_fwnode_handle(adev)); - acpi_dev_put(adev); - - if (!bus) { - dev_err(dev, "Did not find any buses for %s\n", acpi_ids[i].hid); - return; - } - - if (!fwnode) { - dev_err(dev, "Could not get fwnode for %s\n", acpi_ids[i].hid); - return; - } - - /* - * When available the cirrus,dev-index property is an accurate - * count of the amps in a system and is used in preference to - * the count of bus devices that can contain additional address - * alias entries. - */ - count_devindex = fwnode_property_count_u32(fwnode, "cirrus,dev-index"); - if (count_devindex > 0) - count = count_devindex; - - match = devm_kasprintf(dev, GFP_KERNEL, "-%%s:00-%s.%%d", acpi_ids[i].name); - if (!match) - return; - dev_info(dev, "Found %d %s on %s (%s)\n", count, acpi_ids[i].hid, bus, match); - comp_generic_fixup(cdc, action, bus, acpi_ids[i].hid, match, count); - - break; - case HDA_FIXUP_ACT_FREE: - /* - * Pass the action on to comp_generic_fixup() so that - * hda_component_manager functions can be called in just once - * place. In this context the bus, hid, match_str or count - * values do not need to be calculated. - */ - comp_generic_fixup(cdc, action, NULL, NULL, NULL, 0); - break; + for (i = 0; i < ARRAY_SIZE(acpi_ids); ++i) { + adev = acpi_dev_get_first_match_dev(acpi_ids[i].hid, NULL, -1); + if (adev) + break; } + if (!adev) { + codec_dbg(cdc, "Did not find ACPI entry for a Cirrus Amp\n"); + return; + } + + count = i2c_acpi_client_count(adev); + if (count > 0) { + bus = "i2c"; + } else { + count = acpi_spi_count_resources(adev); + if (count > 0) + bus = "spi"; + } + + fwnode = fwnode_handle_get(acpi_fwnode_handle(adev)); + acpi_dev_put(adev); + + if (!bus) { + codec_err(cdc, "Did not find any buses for %s\n", acpi_ids[i].hid); + return; + } + + if (!fwnode) { + codec_err(cdc, "Could not get fwnode for %s\n", acpi_ids[i].hid); + return; + } + + /* + * When available the cirrus,dev-index property is an accurate + * count of the amps in a system and is used in preference to + * the count of bus devices that can contain additional address + * alias entries. + */ + count_devindex = fwnode_property_count_u32(fwnode, "cirrus,dev-index"); + if (count_devindex > 0) + count = count_devindex; + + match = devm_kasprintf(dev, GFP_KERNEL, "-%%s:00-%s.%%d", acpi_ids[i].name); + if (!match) + return; + codec_info(cdc, "Found %d %s on %s (%s)\n", count, acpi_ids[i].hid, bus, match); + comp_generic_fixup(cdc, HDA_FIXUP_ACT_PRE_PROBE, bus, acpi_ids[i].hid, match, count); } static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) @@ -7002,9 +6997,7 @@ static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fix * The same SSID has been re-used in different hardware, they have * different codecs and the newer GA403U has a ALC285. */ - if (cdc->core.vendor_id == 0x10ec0285) - cs35lxx_autodet_fixup(cdc, fix, action); - else + if (cdc->core.vendor_id != 0x10ec0285) alc_fixup_inv_dmic(cdc, fix, action); } @@ -7601,7 +7594,6 @@ enum { ALC2XX_FIXUP_HEADSET_MIC, ALC289_FIXUP_DELL_CS35L41_SPI_2, ALC294_FIXUP_CS35L41_I2C_2, - ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, @@ -7614,7 +7606,6 @@ enum { ALC256_FIXUP_CHROME_BOOK, ALC287_FIXUP_LENOVO_14ARP8_LEGION_IAH7, ALC287_FIXUP_LENOVO_SSID_17AA3820, - ALCXXX_FIXUP_CS35LXX, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9868,12 +9859,6 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, }, - [ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35lxx_autodet_fixup, - .chained = true, - .chain_id = ALC285_FIXUP_HP_GPIO_LED, - }, [ALC256_FIXUP_ACER_SFG16_MICMUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_acer_sfg16_micmute_led, @@ -9913,8 +9898,6 @@ static const struct hda_fixup alc269_fixups[] = { { 0x1b, 0x03a11c30 }, { } }, - .chained = true, - .chain_id = ALCXXX_FIXUP_CS35LXX }, [ALC285_FIXUP_ASUS_GA403U_I2C_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, @@ -9938,10 +9921,6 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc287_fixup_lenovo_ssid_17aa3820, }, - [ALCXXX_FIXUP_CS35LXX] = { - .type = HDA_FIXUP_FUNC, - .v.func = cs35lxx_autodet_fixup, - }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10323,8 +10302,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8c66, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c67, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8c68, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), @@ -10358,17 +10337,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x8d01, "HP ZBook Power 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d08, "HP EliteBook 1045 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d85, "HP EliteBook 1040 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d86, "HP Elite x360 1040 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d8c, "HP EliteBook 830 13 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d8d, "HP Elite x360 830 13 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d8e, "HP EliteBook 840 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d8f, "HP EliteBook 840 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d90, "HP EliteBook 860 16 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d91, "HP ZBook Firefly 14 G12", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x103c, 0x8d92, "HP ZBook Firefly 16 G12", ALCXXX_FIXUP_CS35LXX), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -10448,14 +10416,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALCXXX_FIXUP_CS35LXX), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), - SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALCXXX_FIXUP_CS35LXX), - SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALCXXX_FIXUP_CS35LXX), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), @@ -11648,6 +11613,13 @@ static int patch_alc269(struct hda_codec *codec) snd_hda_pick_pin_fixup(codec, alc269_fallback_pin_fixup_tbl, alc269_fixups, false); snd_hda_pick_fixup(codec, NULL, alc269_fixup_vendor_tbl, alc269_fixups); + + /* + * Check whether ACPI describes companion amplifiers that require + * component binding + */ + find_cirrus_companion_amps(codec); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); From 99c9767c0444098a806529204c7a2556f89d0c04 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Fri, 30 Aug 2024 15:48:19 +0100 Subject: [PATCH 282/410] ASoC: cs-amp-lib: Add KUnit test case for empty calibration entries Add a test case for commit bb4485562f59 ("ASoC: cs-amp-lib: Ignore empty UEFI calibration entries"). Any entries in the calibration blob that have calTime==0 are empty entries. So they must not be returned by a lookup. Signed-off-by: Richard Fitzgerald Link: https://patch.msgid.link/20240830144819.118362-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs-amp-lib-test.c | 44 ++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/cs-amp-lib-test.c b/sound/soc/codecs/cs-amp-lib-test.c index 8169ec88a8ba..a6e8348a1bd5 100644 --- a/sound/soc/codecs/cs-amp-lib-test.c +++ b/sound/soc/codecs/cs-amp-lib-test.c @@ -515,6 +515,49 @@ static void cs_amp_lib_test_get_efi_cal_zero_not_matched_test(struct kunit *test kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); } +/* + * If an entry has a timestamp of 0 it should be ignored even if it has + * a matching target UID. + */ +static void cs_amp_lib_test_get_efi_cal_empty_entry_test(struct kunit *test) +{ + struct cs_amp_lib_test_priv *priv = test->priv; + struct cirrus_amp_cal_data result_data; + u64 uid; + + cs_amp_lib_test_init_dummy_cal_blob(test, 8); + + /* Mark the 3rd entry invalid by zeroing calTime */ + priv->cal_blob->data[2].calTime[0] = 0; + priv->cal_blob->data[2].calTime[1] = 0; + + /* Get the UID value of the 3rd entry */ + uid = priv->cal_blob->data[2].calTarget[1]; + uid <<= 32; + uid |= priv->cal_blob->data[2].calTarget[0]; + + /* Redirect calls to get EFI data */ + kunit_activate_static_stub(test, + cs_amp_test_hooks->get_efi_variable, + cs_amp_lib_test_get_efi_variable); + + /* Lookup by UID should not find it */ + KUNIT_EXPECT_EQ(test, + cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, + uid, -1, + &result_data), + -ENOENT); + + /* Get by index should ignore it */ + KUNIT_EXPECT_EQ(test, + cs_amp_get_efi_calibration_data(&priv->amp_pdev.dev, + 0, 2, + &result_data), + -ENOENT); + + kunit_deactivate_static_stub(test, cs_amp_test_hooks->get_efi_variable); +} + static const struct cirrus_amp_cal_controls cs_amp_lib_test_calibration_controls = { .alg_id = 0x9f210, .mem_region = WMFW_ADSP2_YM, @@ -696,6 +739,7 @@ static struct kunit_case cs_amp_lib_test_cases[] = { cs_amp_lib_test_get_cal_gen_params), KUNIT_CASE_PARAM(cs_amp_lib_test_get_efi_cal_by_index_fallback_test, cs_amp_lib_test_get_cal_gen_params), + KUNIT_CASE(cs_amp_lib_test_get_efi_cal_empty_entry_test), /* Tests for writing calibration data */ KUNIT_CASE(cs_amp_lib_test_write_cal_data_test), From 61bc4deff033181992408f973b48fca08757d3ff Mon Sep 17 00:00:00 2001 From: Hongbo Li Date: Sat, 31 Aug 2024 16:06:39 +0800 Subject: [PATCH 283/410] ALSA: pcm: replace simple_strtoul to kstrtoul As mentioned in [1], "...simple_strtol(), simple_strtoll(), simple_strtoul(), and simple_strtoull() functions explicitly ignore overflows, which may lead to unexpected results in callers." Hence, the use of those functions is discouraged. This patch replace the use of the simple_strtoul with the safer alternatives kstrtoul. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html#simple-strtol-simple-strtoll-simple-strtoul-simple-strtoull Signed-off-by: Hongbo Li Link: https://patch.msgid.link/20240831080639.3985143-1-lihongbo22@huawei.com Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 8e4c68e3bbd0..73d4fc49a0ca 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -193,7 +193,10 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, } if (!snd_info_get_line(buffer, line, sizeof(line))) { snd_info_get_str(str, line, sizeof(str)); - size = simple_strtoul(str, NULL, 10) * 1024; + buffer->error = kstrtoul(str, 10, &size); + if (buffer->error != 0) + return; + size *= 1024; if ((size != 0 && size < 8192) || size > substream->dma_max) { buffer->error = -EINVAL; return; From 43b42ed438bfff6bb5a51cc27a1658c03cd223fd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 1 Sep 2024 15:45:12 +0200 Subject: [PATCH 284/410] ALSA: pcm: Fix the previous conversion to kstrtoul() The previous replacement from simple_strtoul() to kstrtoul() forgot that the passed pointer must be an unsigned long int pointer, while the value used there is a sized_t pointer. Fix it. Fixes: 61bc4deff033 ("ALSA: pcm: replace simple_strtoul to kstrtoul") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202409010425.YPS7cWeJ-lkp@intel.com/ Link: https://patch.msgid.link/20240901134524.27107-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 73d4fc49a0ca..5001181208df 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -183,7 +183,7 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, struct snd_pcm_substream *substream = entry->private_data; struct snd_card *card = substream->pcm->card; char line[64], str[64]; - size_t size; + unsigned long size; struct snd_dma_buffer new_dmab; guard(mutex)(&substream->pcm->open_mutex); From 2657539a27149b22aec6766831fc69ad36548cb1 Mon Sep 17 00:00:00 2001 From: Colin Ian King Date: Sun, 1 Sep 2024 17:21:25 +0100 Subject: [PATCH 285/410] ALSA: ali5451: Remove trailing space after \n newline There is a extraneous space after a newline in a dev_dbg message. Remove it. Signed-off-by: Colin Ian King Link: https://patch.msgid.link/20240901162125.144069-1-colin.i.king@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/ali5451/ali5451.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index 31e51e2df655..793d2f13267e 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -292,7 +292,7 @@ static int snd_ali_codec_ready(struct snd_ali *codec, } snd_ali_5451_poke(codec, port, res & ~0x8000); - dev_dbg(codec->card->dev, "ali_codec_ready: codec is not ready.\n "); + dev_dbg(codec->card->dev, "ali_codec_ready: codec is not ready.\n"); return -EIO; } From 3606f92de365c0189c13e25a94654257751732ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Sep 2024 08:22:16 +0200 Subject: [PATCH 286/410] ALSA: pcm: Fix yet more compile warning at replacement with kstrtoul() The previous fix brought yet another compile warning at pr_debug() call with the changed size. Reported-by: Stephen Rothwell Closes: https://lore.kernel.org/20240902132904.5ee173f3@canb.auug.org.au Fixes: 43b42ed438bf ("ALSA: pcm: Fix the previous conversion to kstrtoul()") Tested-by: Stephen Rothwell Link: https://patch.msgid.link/20240902062217.9777-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 5001181208df..ea3941f8666b 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -212,7 +212,7 @@ static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, substream->stream, size, &new_dmab) < 0) { buffer->error = -ENOMEM; - pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", + pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %lu\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p', substream->number, substream->pcm->name, size); From f48bd50a1c8d7d733b83a8fbabb3041e0d505fc7 Mon Sep 17 00:00:00 2001 From: Jinjie Ruan Date: Mon, 2 Sep 2024 15:16:22 +0800 Subject: [PATCH 287/410] ALSA: core: timer: Use NSEC_PER_SEC macro 1000000000L is number of ns per second, use NSEC_PER_SEC macro to replace it to make it more readable Signed-off-by: Jinjie Ruan Link: https://patch.msgid.link/20240902071622.3519787-1-ruanjinjie@huawei.com Signed-off-by: Takashi Iwai --- sound/core/timer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 7610f102360d..4315d1e98ebf 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1174,7 +1174,7 @@ static int snd_timer_s_close(struct snd_timer *timer) static const struct snd_timer_hardware snd_timer_system = { .flags = SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_WORK, - .resolution = 1000000000L / HZ, + .resolution = NSEC_PER_SEC / HZ, .ticks = 10000000L, .close = snd_timer_s_close, .start = snd_timer_s_start, From 40a024b81d1cbad6bc8cd960481f025b43712b01 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 2 Sep 2024 09:52:27 +0200 Subject: [PATCH 288/410] ALSA: core: Drop superfluous no_free_ptr() for memdup_user() errors We used to wrap with no_free_ptr() for the return value from memdup_user() with errors where the auto cleanup is applied. This was a workaround because the initial implementation of kfree auto-cleanup checked only NULL pointers. Since recently, though, the kfree auto-cleanup checks with IS_ERR_OR_NULL() (by the commit cd7eb8f83fcf ("mm/slab: make __free(kfree) accept error pointers")), hence those workarounds became superfluous. Let's drop them now. Link: https://patch.msgid.link/20240902075246.3743-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 2 +- sound/core/control.c | 4 ++-- sound/core/pcm_native.c | 10 +++++----- sound/core/timer.c | 4 ++-- 4 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index f0008fa2d839..b8c0d6edbdd1 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -581,7 +581,7 @@ snd_compr_set_params(struct snd_compr_stream *stream, unsigned long arg) */ params = memdup_user((void __user *)arg, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(no_free_ptr(params)); + return PTR_ERR(params); retval = snd_compress_check_input(params); if (retval) diff --git a/sound/core/control.c b/sound/core/control.c index 7d7d592a8428..4f55f64c42e1 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1249,7 +1249,7 @@ static int snd_ctl_elem_read_user(struct snd_card *card, control = memdup_user(_control, sizeof(*control)); if (IS_ERR(control)) - return PTR_ERR(no_free_ptr(control)); + return PTR_ERR(control); result = snd_power_ref_and_wait(card); if (result) @@ -1326,7 +1326,7 @@ static int snd_ctl_elem_write_user(struct snd_ctl_file *file, control = memdup_user(_control, sizeof(*control)); if (IS_ERR(control)) - return PTR_ERR(no_free_ptr(control)); + return PTR_ERR(control); card = file->card; result = snd_power_ref_and_wait(card); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4057f9f10aee..44381514f695 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -582,7 +582,7 @@ static int snd_pcm_hw_refine_user(struct snd_pcm_substream *substream, params = memdup_user(_params, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(no_free_ptr(params)); + return PTR_ERR(params); err = snd_pcm_hw_refine(substream, params); if (err < 0) @@ -872,7 +872,7 @@ static int snd_pcm_hw_params_user(struct snd_pcm_substream *substream, params = memdup_user(_params, sizeof(*params)); if (IS_ERR(params)) - return PTR_ERR(no_free_ptr(params)); + return PTR_ERR(params); err = snd_pcm_hw_params(substream, params); if (err < 0) @@ -3243,7 +3243,7 @@ static int snd_pcm_xfern_frames_ioctl(struct snd_pcm_substream *substream, bufs = memdup_user(xfern.bufs, sizeof(void *) * runtime->channels); if (IS_ERR(bufs)) - return PTR_ERR(no_free_ptr(bufs)); + return PTR_ERR(bufs); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) result = snd_pcm_lib_writev(substream, bufs, xfern.frames); else @@ -4032,7 +4032,7 @@ static int snd_pcm_hw_refine_old_user(struct snd_pcm_substream *substream, oparams = memdup_user(_oparams, sizeof(*oparams)); if (IS_ERR(oparams)) - return PTR_ERR(no_free_ptr(oparams)); + return PTR_ERR(oparams); snd_pcm_hw_convert_from_old_params(params, oparams); err = snd_pcm_hw_refine(substream, params); if (err < 0) @@ -4061,7 +4061,7 @@ static int snd_pcm_hw_params_old_user(struct snd_pcm_substream *substream, oparams = memdup_user(_oparams, sizeof(*oparams)); if (IS_ERR(oparams)) - return PTR_ERR(no_free_ptr(oparams)); + return PTR_ERR(oparams); snd_pcm_hw_convert_from_old_params(params, oparams); err = snd_pcm_hw_params(substream, params); diff --git a/sound/core/timer.c b/sound/core/timer.c index 4315d1e98ebf..668c40bac318 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1615,7 +1615,7 @@ static int snd_timer_user_ginfo(struct file *file, ginfo = memdup_user(_ginfo, sizeof(*ginfo)); if (IS_ERR(ginfo)) - return PTR_ERR(no_free_ptr(ginfo)); + return PTR_ERR(ginfo); tid = ginfo->tid; memset(ginfo, 0, sizeof(*ginfo)); @@ -2190,7 +2190,7 @@ static int snd_utimer_ioctl_create(struct file *file, utimer_info = memdup_user(_utimer_info, sizeof(*utimer_info)); if (IS_ERR(utimer_info)) - return PTR_ERR(no_free_ptr(utimer_info)); + return PTR_ERR(utimer_info); err = snd_utimer_create(utimer_info, &utimer); if (err < 0) From f465d10cd73188611ba952fd4df2e2c879998334 Mon Sep 17 00:00:00 2001 From: Shuming Fan Date: Mon, 2 Sep 2024 17:08:45 +0800 Subject: [PATCH 289/410] ASoC: rt1320: Add support for version C This patch added the support for version C. Signed-off-by: Shuming Fan Link: https://patch.msgid.link/20240902090845.1862354-1-shumingf@realtek.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt1320-sdw.c | 2147 ++++++++++++++++++++++++++++++++- sound/soc/codecs/rt1320-sdw.h | 3 + 2 files changed, 2127 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/rt1320-sdw.c b/sound/soc/codecs/rt1320-sdw.c index 2916fa77b791..f4e1ea29c265 100644 --- a/sound/soc/codecs/rt1320-sdw.c +++ b/sound/soc/codecs/rt1320-sdw.c @@ -91,6 +91,2027 @@ static const struct reg_sequence rt1320_blind_write[] = { { 0xd486, 0xc3 }, }; +static const struct reg_sequence rt1320_vc_blind_write[] = { + { 0xc003, 0xe0 }, + { 0xe80a, 0x01 }, + { 0xc5c3, 0xf3 }, + { 0xc057, 0x51 }, + { 0xc054, 0x35 }, + { 0xca05, 0xd6 }, + { 0xca07, 0x07 }, + { 0xca25, 0xd6 }, + { 0xca27, 0x07 }, + { 0xc604, 0x40 }, + { 0xc609, 0x40 }, + { 0xc046, 0xff }, + { 0xc045, 0xff }, + { 0xda81, 0x14 }, + { 0xda8d, 0x14 }, + { 0xc044, 0xff }, + { 0xc043, 0xff }, + { 0xc042, 0xff }, + { 0xc041, 0x7f }, + { 0xc040, 0xff }, + { 0xcc10, 0x01 }, + { 0xc700, 0xf0 }, + { 0xc701, 0x13 }, + { 0xc901, 0x09 }, + { 0xc900, 0xd0 }, + { 0xde03, 0x05 }, + { 0xdd0b, 0x0d }, + { 0xdd0a, 0xff }, + { 0xdd09, 0x0d }, + { 0xdd08, 0xff }, + { 0xc570, 0x08 }, + { 0xc086, 0x02 }, + { 0xc085, 0x7f }, + { 0xc084, 0x00 }, + { 0xc081, 0xfe }, + { 0xf084, 0x0f }, + { 0xf083, 0xff }, + { 0xf082, 0xff }, + { 0xf081, 0xff }, + { 0xf080, 0xff }, + { 0xe802, 0xf8 }, + { 0xe803, 0xbe }, + { 0xc003, 0xc0 }, + { 0xd470, 0xec }, + { 0xd471, 0x3a }, + { 0xd474, 0x11 }, + { 0xd475, 0x32 }, + { 0xd478, 0x64 }, + { 0xd479, 0x20 }, + { 0xd47a, 0x10 }, + { 0xd47c, 0xff }, + { 0xc019, 0x10 }, + { 0xd487, 0x0b }, + { 0xd487, 0x3b }, + { 0xd486, 0xc3 }, + { 0xc598, 0x04 }, + { 0xd500, 0x00 }, + { 0xd500, 0x17 }, + { 0xd600, 0x01 }, + { 0xd601, 0x02 }, + { 0xd602, 0x03 }, + { 0xd603, 0x04 }, + { 0xd64c, 0x03 }, + { 0xd64d, 0x03 }, + { 0xd64e, 0x03 }, + { 0xd64f, 0x03 }, + { 0xd650, 0x03 }, + { 0xd651, 0x03 }, + { 0xd652, 0x03 }, + { 0xd610, 0x01 }, + { 0xd608, 0x03 }, + { 0xd609, 0x00 }, +}; + +static const struct reg_sequence rt1320_vc_patch_code_write[] = { + { 0x10007000, 0x37 }, + { 0x10007001, 0x77 }, + { 0x10007002, 0x00 }, + { 0x10007003, 0x10 }, + { 0x10007004, 0xb7 }, + { 0x10007005, 0xe7 }, + { 0x10007006, 0x00 }, + { 0x10007007, 0x10 }, + { 0x10007008, 0x13 }, + { 0x10007009, 0x07 }, + { 0x1000700a, 0x87 }, + { 0x1000700b, 0x48 }, + { 0x1000700c, 0x23 }, + { 0x1000700d, 0xa6 }, + { 0x1000700e, 0xe7 }, + { 0x1000700f, 0xee }, + { 0x10007010, 0x37 }, + { 0x10007011, 0x77 }, + { 0x10007012, 0x00 }, + { 0x10007013, 0x10 }, + { 0x10007014, 0x13 }, + { 0x10007015, 0x07 }, + { 0x10007016, 0x87 }, + { 0x10007017, 0x56 }, + { 0x10007018, 0x23 }, + { 0x10007019, 0xac }, + { 0x1000701a, 0xe7 }, + { 0x1000701b, 0xde }, + { 0x1000701c, 0x37 }, + { 0x1000701d, 0x77 }, + { 0x1000701e, 0x00 }, + { 0x1000701f, 0x10 }, + { 0x10007020, 0x13 }, + { 0x10007021, 0x07 }, + { 0x10007022, 0xc7 }, + { 0x10007023, 0x5f }, + { 0x10007024, 0x23 }, + { 0x10007025, 0xae }, + { 0x10007026, 0xe7 }, + { 0x10007027, 0xdc }, + { 0x10007028, 0x37 }, + { 0x10007029, 0x87 }, + { 0x1000702a, 0x00 }, + { 0x1000702b, 0x10 }, + { 0x1000702c, 0x13 }, + { 0x1000702d, 0x07 }, + { 0x1000702e, 0xc7 }, + { 0x1000702f, 0x86 }, + { 0x10007030, 0x23 }, + { 0x10007031, 0xae }, + { 0x10007032, 0xe7 }, + { 0x10007033, 0xe6 }, + { 0x10007034, 0x37 }, + { 0x10007035, 0x77 }, + { 0x10007036, 0x00 }, + { 0x10007037, 0x10 }, + { 0x10007038, 0x13 }, + { 0x10007039, 0x07 }, + { 0x1000703a, 0x07 }, + { 0x1000703b, 0x40 }, + { 0x1000703c, 0x23 }, + { 0x1000703d, 0xa6 }, + { 0x1000703e, 0xe7 }, + { 0x1000703f, 0xe8 }, + { 0x10007040, 0x37 }, + { 0x10007041, 0x77 }, + { 0x10007042, 0x00 }, + { 0x10007043, 0x10 }, + { 0x10007044, 0x13 }, + { 0x10007045, 0x07 }, + { 0x10007046, 0xc7 }, + { 0x10007047, 0x63 }, + { 0x10007048, 0x23 }, + { 0x10007049, 0xa2 }, + { 0x1000704a, 0xe7 }, + { 0x1000704b, 0xec }, + { 0x1000704c, 0x37 }, + { 0x1000704d, 0x77 }, + { 0x1000704e, 0x00 }, + { 0x1000704f, 0x10 }, + { 0x10007050, 0x13 }, + { 0x10007051, 0x07 }, + { 0x10007052, 0x47 }, + { 0x10007053, 0x6f }, + { 0x10007054, 0x23 }, + { 0x10007055, 0xa6 }, + { 0x10007056, 0xe7 }, + { 0x10007057, 0xec }, + { 0x10007058, 0x37 }, + { 0x10007059, 0x77 }, + { 0x1000705a, 0x00 }, + { 0x1000705b, 0x10 }, + { 0x1000705c, 0x13 }, + { 0x1000705d, 0x07 }, + { 0x1000705e, 0x07 }, + { 0x1000705f, 0x44 }, + { 0x10007060, 0x23 }, + { 0x10007061, 0xa8 }, + { 0x10007062, 0xe7 }, + { 0x10007063, 0xec }, + { 0x10007064, 0x37 }, + { 0x10007065, 0x87 }, + { 0x10007066, 0x00 }, + { 0x10007067, 0x10 }, + { 0x10007068, 0x13 }, + { 0x10007069, 0x07 }, + { 0x1000706a, 0x87 }, + { 0x1000706b, 0x84 }, + { 0x1000706c, 0x23 }, + { 0x1000706d, 0xa8 }, + { 0x1000706e, 0xe7 }, + { 0x1000706f, 0xee }, + { 0x10007070, 0x37 }, + { 0x10007071, 0x87 }, + { 0x10007072, 0x00 }, + { 0x10007073, 0x10 }, + { 0x10007074, 0x13 }, + { 0x10007075, 0x07 }, + { 0x10007076, 0x47 }, + { 0x10007077, 0x97 }, + { 0x10007078, 0x23 }, + { 0x10007079, 0xaa }, + { 0x1000707a, 0xe7 }, + { 0x1000707b, 0xee }, + { 0x1000707c, 0x67 }, + { 0x1000707d, 0x80 }, + { 0x1000707e, 0x00 }, + { 0x1000707f, 0x00 }, + { 0x10007400, 0xb7 }, + { 0x10007401, 0xd6 }, + { 0x10007402, 0x00 }, + { 0x10007403, 0x00 }, + { 0x10007404, 0x83 }, + { 0x10007405, 0xc7 }, + { 0x10007406, 0x06 }, + { 0x10007407, 0x47 }, + { 0x10007408, 0x93 }, + { 0x10007409, 0xf7 }, + { 0x1000740a, 0x87 }, + { 0x1000740b, 0x00 }, + { 0x1000740c, 0x63 }, + { 0x1000740d, 0x88 }, + { 0x1000740e, 0x07 }, + { 0x1000740f, 0x02 }, + { 0x10007410, 0x03 }, + { 0x10007411, 0xc7 }, + { 0x10007412, 0x31 }, + { 0x10007413, 0x43 }, + { 0x10007414, 0x63 }, + { 0x10007415, 0x14 }, + { 0x10007416, 0x07 }, + { 0x10007417, 0x02 }, + { 0x10007418, 0x13 }, + { 0x10007419, 0x07 }, + { 0x1000741a, 0x10 }, + { 0x1000741b, 0x00 }, + { 0x1000741c, 0xa3 }, + { 0x1000741d, 0x89 }, + { 0x1000741e, 0xe1 }, + { 0x1000741f, 0x42 }, + { 0x10007420, 0x37 }, + { 0x10007421, 0xc7 }, + { 0x10007422, 0x00 }, + { 0x10007423, 0x00 }, + { 0x10007424, 0x03 }, + { 0x10007425, 0x46 }, + { 0x10007426, 0x07 }, + { 0x10007427, 0x06 }, + { 0x10007428, 0x23 }, + { 0x10007429, 0x8a }, + { 0x1000742a, 0xc1 }, + { 0x1000742b, 0x42 }, + { 0x1000742c, 0x83 }, + { 0x1000742d, 0xc7 }, + { 0x1000742e, 0x46 }, + { 0x1000742f, 0x47 }, + { 0x10007430, 0x93 }, + { 0x10007431, 0xf7 }, + { 0x10007432, 0xf7 }, + { 0x10007433, 0x0f }, + { 0x10007434, 0x23 }, + { 0x10007435, 0x00 }, + { 0x10007436, 0xf7 }, + { 0x10007437, 0x06 }, + { 0x10007438, 0x23 }, + { 0x10007439, 0x89 }, + { 0x1000743a, 0x01 }, + { 0x1000743b, 0x42 }, + { 0x1000743c, 0x67 }, + { 0x1000743d, 0x80 }, + { 0x1000743e, 0x00 }, + { 0x1000743f, 0x00 }, + { 0x10007440, 0x37 }, + { 0x10007441, 0xc7 }, + { 0x10007442, 0x00 }, + { 0x10007443, 0x00 }, + { 0x10007444, 0x83 }, + { 0x10007445, 0x27 }, + { 0x10007446, 0xc7 }, + { 0x10007447, 0x5f }, + { 0x10007448, 0x13 }, + { 0x10007449, 0x05 }, + { 0x1000744a, 0x00 }, + { 0x1000744b, 0x00 }, + { 0x1000744c, 0x23 }, + { 0x1000744d, 0xa2 }, + { 0x1000744e, 0xf1 }, + { 0x1000744f, 0x42 }, + { 0x10007450, 0xb7 }, + { 0x10007451, 0x06 }, + { 0x10007452, 0x00 }, + { 0x10007453, 0x10 }, + { 0x10007454, 0xb3 }, + { 0x10007455, 0xf7 }, + { 0x10007456, 0xd7 }, + { 0x10007457, 0x00 }, + { 0x10007458, 0x63 }, + { 0x10007459, 0x86 }, + { 0x1000745a, 0x07 }, + { 0x1000745b, 0x02 }, + { 0x1000745c, 0x83 }, + { 0x1000745d, 0x47 }, + { 0x1000745e, 0x07 }, + { 0x1000745f, 0x56 }, + { 0x10007460, 0x93 }, + { 0x10007461, 0xf7 }, + { 0x10007462, 0x87 }, + { 0x10007463, 0x01 }, + { 0x10007464, 0x63 }, + { 0x10007465, 0x80 }, + { 0x10007466, 0x07 }, + { 0x10007467, 0x02 }, + { 0x10007468, 0x83 }, + { 0x10007469, 0x47 }, + { 0x1000746a, 0x17 }, + { 0x1000746b, 0x08 }, + { 0x1000746c, 0x93 }, + { 0x1000746d, 0xf7 }, + { 0x1000746e, 0x47 }, + { 0x1000746f, 0x00 }, + { 0x10007470, 0x63 }, + { 0x10007471, 0x8a }, + { 0x10007472, 0x07 }, + { 0x10007473, 0x00 }, + { 0x10007474, 0xb7 }, + { 0x10007475, 0xc7 }, + { 0x10007476, 0xc2 }, + { 0x10007477, 0x3f }, + { 0x10007478, 0x03 }, + { 0x10007479, 0xa5 }, + { 0x1000747a, 0x47 }, + { 0x1000747b, 0xfc }, + { 0x1000747c, 0x13 }, + { 0x1000747d, 0x55 }, + { 0x1000747e, 0x25 }, + { 0x1000747f, 0x00 }, + { 0x10007480, 0x13 }, + { 0x10007481, 0x75 }, + { 0x10007482, 0x15 }, + { 0x10007483, 0x00 }, + { 0x10007484, 0x67 }, + { 0x10007485, 0x80 }, + { 0x10007486, 0x00 }, + { 0x10007487, 0x00 }, + { 0x10007488, 0x03 }, + { 0x10007489, 0xa7 }, + { 0x1000748a, 0x81 }, + { 0x1000748b, 0x57 }, + { 0x1000748c, 0x13 }, + { 0x1000748d, 0x01 }, + { 0x1000748e, 0x01 }, + { 0x1000748f, 0xff }, + { 0x10007490, 0x23 }, + { 0x10007491, 0x26 }, + { 0x10007492, 0x11 }, + { 0x10007493, 0x00 }, + { 0x10007494, 0x23 }, + { 0x10007495, 0x24 }, + { 0x10007496, 0x81 }, + { 0x10007497, 0x00 }, + { 0x10007498, 0x23 }, + { 0x10007499, 0x22 }, + { 0x1000749a, 0x91 }, + { 0x1000749b, 0x00 }, + { 0x1000749c, 0x93 }, + { 0x1000749d, 0x07 }, + { 0x1000749e, 0xa0 }, + { 0x1000749f, 0x05 }, + { 0x100074a0, 0x63 }, + { 0x100074a1, 0x14 }, + { 0x100074a2, 0xf7 }, + { 0x100074a3, 0x04 }, + { 0x100074a4, 0x37 }, + { 0x100074a5, 0x07 }, + { 0x100074a6, 0x00 }, + { 0x100074a7, 0x11 }, + { 0x100074a8, 0x83 }, + { 0x100074a9, 0x47 }, + { 0x100074aa, 0x07 }, + { 0x100074ab, 0x01 }, + { 0x100074ac, 0x13 }, + { 0x100074ad, 0x06 }, + { 0x100074ae, 0x30 }, + { 0x100074af, 0x00 }, + { 0x100074b0, 0x93 }, + { 0x100074b1, 0xf7 }, + { 0x100074b2, 0xf7 }, + { 0x100074b3, 0x0f }, + { 0x100074b4, 0x63 }, + { 0x100074b5, 0x9a }, + { 0x100074b6, 0xc7 }, + { 0x100074b7, 0x02 }, + { 0x100074b8, 0x03 }, + { 0x100074b9, 0x47 }, + { 0x100074ba, 0x87 }, + { 0x100074bb, 0x01 }, + { 0x100074bc, 0x13 }, + { 0x100074bd, 0x77 }, + { 0x100074be, 0xf7 }, + { 0x100074bf, 0x0f }, + { 0x100074c0, 0x63 }, + { 0x100074c1, 0x14 }, + { 0x100074c2, 0xf7 }, + { 0x100074c3, 0x02 }, + { 0x100074c4, 0x37 }, + { 0x100074c5, 0xd7 }, + { 0x100074c6, 0x00 }, + { 0x100074c7, 0x00 }, + { 0x100074c8, 0x83 }, + { 0x100074c9, 0x47 }, + { 0x100074ca, 0x37 }, + { 0x100074cb, 0x54 }, + { 0x100074cc, 0x93 }, + { 0x100074cd, 0xf7 }, + { 0x100074ce, 0xf7 }, + { 0x100074cf, 0x0f }, + { 0x100074d0, 0x93 }, + { 0x100074d1, 0xe7 }, + { 0x100074d2, 0x07 }, + { 0x100074d3, 0x02 }, + { 0x100074d4, 0xa3 }, + { 0x100074d5, 0x01 }, + { 0x100074d6, 0xf7 }, + { 0x100074d7, 0x54 }, + { 0x100074d8, 0x83 }, + { 0x100074d9, 0x47 }, + { 0x100074da, 0x37 }, + { 0x100074db, 0x54 }, + { 0x100074dc, 0x93 }, + { 0x100074dd, 0xf7 }, + { 0x100074de, 0xf7 }, + { 0x100074df, 0x0d }, + { 0x100074e0, 0xa3 }, + { 0x100074e1, 0x01 }, + { 0x100074e2, 0xf7 }, + { 0x100074e3, 0x54 }, + { 0x100074e4, 0x23 }, + { 0x100074e5, 0xac }, + { 0x100074e6, 0x01 }, + { 0x100074e7, 0x56 }, + { 0x100074e8, 0x37 }, + { 0x100074e9, 0xd4 }, + { 0x100074ea, 0x00 }, + { 0x100074eb, 0x00 }, + { 0x100074ec, 0x83 }, + { 0x100074ed, 0x47 }, + { 0x100074ee, 0xd4 }, + { 0x100074ef, 0x47 }, + { 0x100074f0, 0x93 }, + { 0x100074f1, 0xf7 }, + { 0x100074f2, 0x17 }, + { 0x100074f3, 0x00 }, + { 0x100074f4, 0x63 }, + { 0x100074f5, 0x80 }, + { 0x100074f6, 0x07 }, + { 0x100074f7, 0x06 }, + { 0x100074f8, 0x37 }, + { 0x100074f9, 0xd7 }, + { 0x100074fa, 0x00 }, + { 0x100074fb, 0x10 }, + { 0x100074fc, 0x83 }, + { 0x100074fd, 0x47 }, + { 0x100074fe, 0x77 }, + { 0x100074ff, 0xd9 }, + { 0x10007500, 0x93 }, + { 0x10007501, 0x87 }, + { 0x10007502, 0x17 }, + { 0x10007503, 0x00 }, + { 0x10007504, 0x93 }, + { 0x10007505, 0xf7 }, + { 0x10007506, 0xf7 }, + { 0x10007507, 0x0f }, + { 0x10007508, 0xa3 }, + { 0x10007509, 0x0b }, + { 0x1000750a, 0xf7 }, + { 0x1000750b, 0xd8 }, + { 0x1000750c, 0x03 }, + { 0x1000750d, 0x47 }, + { 0x1000750e, 0x77 }, + { 0x1000750f, 0xd9 }, + { 0x10007510, 0x83 }, + { 0x10007511, 0x47 }, + { 0x10007512, 0xc4 }, + { 0x10007513, 0x47 }, + { 0x10007514, 0x13 }, + { 0x10007515, 0x77 }, + { 0x10007516, 0xf7 }, + { 0x10007517, 0x0f }, + { 0x10007518, 0x93 }, + { 0x10007519, 0xf7 }, + { 0x1000751a, 0xf7 }, + { 0x1000751b, 0x0f }, + { 0x1000751c, 0x63 }, + { 0x1000751d, 0x6c }, + { 0x1000751e, 0xf7 }, + { 0x1000751f, 0x02 }, + { 0x10007520, 0xb7 }, + { 0x10007521, 0xf4 }, + { 0x10007522, 0x00 }, + { 0x10007523, 0x00 }, + { 0x10007524, 0x93 }, + { 0x10007525, 0x05 }, + { 0x10007526, 0x00 }, + { 0x10007527, 0x01 }, + { 0x10007528, 0x13 }, + { 0x10007529, 0x85 }, + { 0x1000752a, 0x34 }, + { 0x1000752b, 0x52 }, + { 0x1000752c, 0xef }, + { 0x1000752d, 0xa0 }, + { 0x1000752e, 0x8f }, + { 0x1000752f, 0xc6 }, + { 0x10007530, 0x93 }, + { 0x10007531, 0x05 }, + { 0x10007532, 0x00 }, + { 0x10007533, 0x00 }, + { 0x10007534, 0x13 }, + { 0x10007535, 0x85 }, + { 0x10007536, 0x54 }, + { 0x10007537, 0x10 }, + { 0x10007538, 0xef }, + { 0x10007539, 0xa0 }, + { 0x1000753a, 0xcf }, + { 0x1000753b, 0xc5 }, + { 0x1000753c, 0x93 }, + { 0x1000753d, 0x05 }, + { 0x1000753e, 0x00 }, + { 0x1000753f, 0x00 }, + { 0x10007540, 0x13 }, + { 0x10007541, 0x85 }, + { 0x10007542, 0x74 }, + { 0x10007543, 0x10 }, + { 0x10007544, 0xef }, + { 0x10007545, 0xa0 }, + { 0x10007546, 0x0f }, + { 0x10007547, 0xc5 }, + { 0x10007548, 0x83 }, + { 0x10007549, 0x47 }, + { 0x1000754a, 0xd4 }, + { 0x1000754b, 0x47 }, + { 0x1000754c, 0x93 }, + { 0x1000754d, 0xf7 }, + { 0x1000754e, 0xe7 }, + { 0x1000754f, 0x0f }, + { 0x10007550, 0xa3 }, + { 0x10007551, 0x0e }, + { 0x10007552, 0xf4 }, + { 0x10007553, 0x46 }, + { 0x10007554, 0x83 }, + { 0x10007555, 0x20 }, + { 0x10007556, 0xc1 }, + { 0x10007557, 0x00 }, + { 0x10007558, 0x03 }, + { 0x10007559, 0x24 }, + { 0x1000755a, 0x81 }, + { 0x1000755b, 0x00 }, + { 0x1000755c, 0x83 }, + { 0x1000755d, 0x24 }, + { 0x1000755e, 0x41 }, + { 0x1000755f, 0x00 }, + { 0x10007560, 0x13 }, + { 0x10007561, 0x01 }, + { 0x10007562, 0x01 }, + { 0x10007563, 0x01 }, + { 0x10007564, 0x67 }, + { 0x10007565, 0x80 }, + { 0x10007566, 0x00 }, + { 0x10007567, 0x00 }, + { 0x10007568, 0x13 }, + { 0x10007569, 0x01 }, + { 0x1000756a, 0x01 }, + { 0x1000756b, 0xff }, + { 0x1000756c, 0x23 }, + { 0x1000756d, 0x24 }, + { 0x1000756e, 0x81 }, + { 0x1000756f, 0x00 }, + { 0x10007570, 0x23 }, + { 0x10007571, 0x26 }, + { 0x10007572, 0x11 }, + { 0x10007573, 0x00 }, + { 0x10007574, 0x23 }, + { 0x10007575, 0x22 }, + { 0x10007576, 0x91 }, + { 0x10007577, 0x00 }, + { 0x10007578, 0x37 }, + { 0x10007579, 0xd4 }, + { 0x1000757a, 0x00 }, + { 0x1000757b, 0x00 }, + { 0x1000757c, 0x83 }, + { 0x1000757d, 0x47 }, + { 0x1000757e, 0x04 }, + { 0x1000757f, 0x54 }, + { 0x10007580, 0x93 }, + { 0x10007581, 0x97 }, + { 0x10007582, 0x87 }, + { 0x10007583, 0x01 }, + { 0x10007584, 0x93 }, + { 0x10007585, 0xd7 }, + { 0x10007586, 0x87 }, + { 0x10007587, 0x41 }, + { 0x10007588, 0x63 }, + { 0x10007589, 0xd0 }, + { 0x1000758a, 0x07 }, + { 0x1000758b, 0x06 }, + { 0x1000758c, 0xb7 }, + { 0x1000758d, 0xf4 }, + { 0x1000758e, 0x00 }, + { 0x1000758f, 0x00 }, + { 0x10007590, 0x93 }, + { 0x10007591, 0x05 }, + { 0x10007592, 0x60 }, + { 0x10007593, 0x01 }, + { 0x10007594, 0x13 }, + { 0x10007595, 0x85 }, + { 0x10007596, 0x34 }, + { 0x10007597, 0x52 }, + { 0x10007598, 0xef }, + { 0x10007599, 0xa0 }, + { 0x1000759a, 0xcf }, + { 0x1000759b, 0xbf }, + { 0x1000759c, 0x93 }, + { 0x1000759d, 0x05 }, + { 0x1000759e, 0x00 }, + { 0x1000759f, 0x04 }, + { 0x100075a0, 0x13 }, + { 0x100075a1, 0x85 }, + { 0x100075a2, 0x54 }, + { 0x100075a3, 0x10 }, + { 0x100075a4, 0xef }, + { 0x100075a5, 0xa0 }, + { 0x100075a6, 0x0f }, + { 0x100075a7, 0xbf }, + { 0x100075a8, 0x93 }, + { 0x100075a9, 0x05 }, + { 0x100075aa, 0x00 }, + { 0x100075ab, 0x04 }, + { 0x100075ac, 0x13 }, + { 0x100075ad, 0x85 }, + { 0x100075ae, 0x74 }, + { 0x100075af, 0x10 }, + { 0x100075b0, 0xef }, + { 0x100075b1, 0xa0 }, + { 0x100075b2, 0x4f }, + { 0x100075b3, 0xbe }, + { 0x100075b4, 0x83 }, + { 0x100075b5, 0x47 }, + { 0x100075b6, 0xd4 }, + { 0x100075b7, 0x47 }, + { 0x100075b8, 0x37 }, + { 0x100075b9, 0xd7 }, + { 0x100075ba, 0x00 }, + { 0x100075bb, 0x10 }, + { 0x100075bc, 0x93 }, + { 0x100075bd, 0xf7 }, + { 0x100075be, 0xf7 }, + { 0x100075bf, 0x0f }, + { 0x100075c0, 0x93 }, + { 0x100075c1, 0xe7 }, + { 0x100075c2, 0x17 }, + { 0x100075c3, 0x00 }, + { 0x100075c4, 0xa3 }, + { 0x100075c5, 0x0e }, + { 0x100075c6, 0xf4 }, + { 0x100075c7, 0x46 }, + { 0x100075c8, 0xa3 }, + { 0x100075c9, 0x0b }, + { 0x100075ca, 0x07 }, + { 0x100075cb, 0xd8 }, + { 0x100075cc, 0x83 }, + { 0x100075cd, 0x47 }, + { 0x100075ce, 0x87 }, + { 0x100075cf, 0xd9 }, + { 0x100075d0, 0x93 }, + { 0x100075d1, 0x87 }, + { 0x100075d2, 0x17 }, + { 0x100075d3, 0x00 }, + { 0x100075d4, 0x93 }, + { 0x100075d5, 0xf7 }, + { 0x100075d6, 0xf7 }, + { 0x100075d7, 0x0f }, + { 0x100075d8, 0x23 }, + { 0x100075d9, 0x0c }, + { 0x100075da, 0xf7 }, + { 0x100075db, 0xd8 }, + { 0x100075dc, 0x83 }, + { 0x100075dd, 0x47 }, + { 0x100075de, 0x04 }, + { 0x100075df, 0x54 }, + { 0x100075e0, 0x93 }, + { 0x100075e1, 0xf7 }, + { 0x100075e2, 0xf7 }, + { 0x100075e3, 0x07 }, + { 0x100075e4, 0x23 }, + { 0x100075e5, 0x00 }, + { 0x100075e6, 0xf4 }, + { 0x100075e7, 0x54 }, + { 0x100075e8, 0x83 }, + { 0x100075e9, 0x20 }, + { 0x100075ea, 0xc1 }, + { 0x100075eb, 0x00 }, + { 0x100075ec, 0x03 }, + { 0x100075ed, 0x24 }, + { 0x100075ee, 0x81 }, + { 0x100075ef, 0x00 }, + { 0x100075f0, 0x83 }, + { 0x100075f1, 0x24 }, + { 0x100075f2, 0x41 }, + { 0x100075f3, 0x00 }, + { 0x100075f4, 0x13 }, + { 0x100075f5, 0x01 }, + { 0x100075f6, 0x01 }, + { 0x100075f7, 0x01 }, + { 0x100075f8, 0x67 }, + { 0x100075f9, 0x80 }, + { 0x100075fa, 0x00 }, + { 0x100075fb, 0x00 }, + { 0x100075fc, 0x13 }, + { 0x100075fd, 0x01 }, + { 0x100075fe, 0x01 }, + { 0x100075ff, 0xff }, + { 0x10007600, 0x23 }, + { 0x10007601, 0x24 }, + { 0x10007602, 0x81 }, + { 0x10007603, 0x00 }, + { 0x10007604, 0x37 }, + { 0x10007605, 0xd4 }, + { 0x10007606, 0x00 }, + { 0x10007607, 0x00 }, + { 0x10007608, 0x83 }, + { 0x10007609, 0x27 }, + { 0x1000760a, 0x04 }, + { 0x1000760b, 0x53 }, + { 0x1000760c, 0x23 }, + { 0x1000760d, 0x22 }, + { 0x1000760e, 0x91 }, + { 0x1000760f, 0x00 }, + { 0x10007610, 0xb7 }, + { 0x10007611, 0x04 }, + { 0x10007612, 0x00 }, + { 0x10007613, 0x40 }, + { 0x10007614, 0x23 }, + { 0x10007615, 0x26 }, + { 0x10007616, 0x11 }, + { 0x10007617, 0x00 }, + { 0x10007618, 0xb3 }, + { 0x10007619, 0xf7 }, + { 0x1000761a, 0x97 }, + { 0x1000761b, 0x00 }, + { 0x1000761c, 0x63 }, + { 0x1000761d, 0x86 }, + { 0x1000761e, 0x07 }, + { 0x1000761f, 0x00 }, + { 0x10007620, 0xef }, + { 0x10007621, 0xd0 }, + { 0x10007622, 0x5f }, + { 0x10007623, 0xc2 }, + { 0x10007624, 0x23 }, + { 0x10007625, 0x28 }, + { 0x10007626, 0x94 }, + { 0x10007627, 0x52 }, + { 0x10007628, 0x83 }, + { 0x10007629, 0x20 }, + { 0x1000762a, 0xc1 }, + { 0x1000762b, 0x00 }, + { 0x1000762c, 0x03 }, + { 0x1000762d, 0x24 }, + { 0x1000762e, 0x81 }, + { 0x1000762f, 0x00 }, + { 0x10007630, 0x83 }, + { 0x10007631, 0x24 }, + { 0x10007632, 0x41 }, + { 0x10007633, 0x00 }, + { 0x10007634, 0x13 }, + { 0x10007635, 0x01 }, + { 0x10007636, 0x01 }, + { 0x10007637, 0x01 }, + { 0x10007638, 0x67 }, + { 0x10007639, 0x80 }, + { 0x1000763a, 0x00 }, + { 0x1000763b, 0x00 }, + { 0x1000763c, 0x37 }, + { 0x1000763d, 0xc7 }, + { 0x1000763e, 0x00 }, + { 0x1000763f, 0x00 }, + { 0x10007640, 0x83 }, + { 0x10007641, 0x27 }, + { 0x10007642, 0xc7 }, + { 0x10007643, 0x5f }, + { 0x10007644, 0x23 }, + { 0x10007645, 0xa2 }, + { 0x10007646, 0xf1 }, + { 0x10007647, 0x42 }, + { 0x10007648, 0xb7 }, + { 0x10007649, 0x06 }, + { 0x1000764a, 0x00 }, + { 0x1000764b, 0x10 }, + { 0x1000764c, 0xb3 }, + { 0x1000764d, 0xf7 }, + { 0x1000764e, 0xd7 }, + { 0x1000764f, 0x00 }, + { 0x10007650, 0x63 }, + { 0x10007651, 0x80 }, + { 0x10007652, 0x07 }, + { 0x10007653, 0x0a }, + { 0x10007654, 0x83 }, + { 0x10007655, 0x47 }, + { 0x10007656, 0x07 }, + { 0x10007657, 0x56 }, + { 0x10007658, 0x93 }, + { 0x10007659, 0xf7 }, + { 0x1000765a, 0x87 }, + { 0x1000765b, 0x01 }, + { 0x1000765c, 0x63 }, + { 0x1000765d, 0x8a }, + { 0x1000765e, 0x07 }, + { 0x1000765f, 0x08 }, + { 0x10007660, 0x83 }, + { 0x10007661, 0x47 }, + { 0x10007662, 0x17 }, + { 0x10007663, 0x08 }, + { 0x10007664, 0x93 }, + { 0x10007665, 0xf7 }, + { 0x10007666, 0x47 }, + { 0x10007667, 0x00 }, + { 0x10007668, 0x63 }, + { 0x10007669, 0x84 }, + { 0x1000766a, 0x07 }, + { 0x1000766b, 0x08 }, + { 0x1000766c, 0x13 }, + { 0x1000766d, 0x01 }, + { 0x1000766e, 0x01 }, + { 0x1000766f, 0xff }, + { 0x10007670, 0x23 }, + { 0x10007671, 0x26 }, + { 0x10007672, 0x11 }, + { 0x10007673, 0x00 }, + { 0x10007674, 0xb7 }, + { 0x10007675, 0xc7 }, + { 0x10007676, 0xc2 }, + { 0x10007677, 0x3f }, + { 0x10007678, 0x03 }, + { 0x10007679, 0xa7 }, + { 0x1000767a, 0x07 }, + { 0x1000767b, 0xfc }, + { 0x1000767c, 0x63 }, + { 0x1000767d, 0x10 }, + { 0x1000767e, 0x05 }, + { 0x1000767f, 0x06 }, + { 0x10007680, 0x13 }, + { 0x10007681, 0x67 }, + { 0x10007682, 0x07 }, + { 0x10007683, 0x20 }, + { 0x10007684, 0x23 }, + { 0x10007685, 0xa0 }, + { 0x10007686, 0xe7 }, + { 0x10007687, 0xfc }, + { 0x10007688, 0x03 }, + { 0x10007689, 0xa7 }, + { 0x1000768a, 0x07 }, + { 0x1000768b, 0xfc }, + { 0x1000768c, 0x13 }, + { 0x1000768d, 0x67 }, + { 0x1000768e, 0x07 }, + { 0x1000768f, 0x40 }, + { 0x10007690, 0x23 }, + { 0x10007691, 0xa0 }, + { 0x10007692, 0xe7 }, + { 0x10007693, 0xfc }, + { 0x10007694, 0x37 }, + { 0x10007695, 0xc7 }, + { 0x10007696, 0xc2 }, + { 0x10007697, 0x3f }, + { 0x10007698, 0x83 }, + { 0x10007699, 0x27 }, + { 0x1000769a, 0x07 }, + { 0x1000769b, 0xfc }, + { 0x1000769c, 0x13 }, + { 0x1000769d, 0x75 }, + { 0x1000769e, 0x15 }, + { 0x1000769f, 0x00 }, + { 0x100076a0, 0x13 }, + { 0x100076a1, 0x15 }, + { 0x100076a2, 0x85 }, + { 0x100076a3, 0x00 }, + { 0x100076a4, 0x93 }, + { 0x100076a5, 0xf7 }, + { 0x100076a6, 0xf7 }, + { 0x100076a7, 0xef }, + { 0x100076a8, 0x33 }, + { 0x100076a9, 0xe5 }, + { 0x100076aa, 0xa7 }, + { 0x100076ab, 0x00 }, + { 0x100076ac, 0x23 }, + { 0x100076ad, 0x20 }, + { 0x100076ae, 0xa7 }, + { 0x100076af, 0xfc }, + { 0x100076b0, 0x93 }, + { 0x100076b1, 0x05 }, + { 0x100076b2, 0x00 }, + { 0x100076b3, 0x00 }, + { 0x100076b4, 0x13 }, + { 0x100076b5, 0x05 }, + { 0x100076b6, 0xa0 }, + { 0x100076b7, 0x00 }, + { 0x100076b8, 0xef }, + { 0x100076b9, 0xe0 }, + { 0x100076ba, 0xcf }, + { 0x100076bb, 0xb6 }, + { 0x100076bc, 0x37 }, + { 0x100076bd, 0xf7 }, + { 0x100076be, 0x00 }, + { 0x100076bf, 0x00 }, + { 0x100076c0, 0x83 }, + { 0x100076c1, 0x47 }, + { 0x100076c2, 0x57 }, + { 0x100076c3, 0x01 }, + { 0x100076c4, 0x93 }, + { 0x100076c5, 0xf7 }, + { 0x100076c6, 0xf7 }, + { 0x100076c7, 0x0f }, + { 0x100076c8, 0x93 }, + { 0x100076c9, 0xe7 }, + { 0x100076ca, 0x47 }, + { 0x100076cb, 0x00 }, + { 0x100076cc, 0xa3 }, + { 0x100076cd, 0x0a }, + { 0x100076ce, 0xf7 }, + { 0x100076cf, 0x00 }, + { 0x100076d0, 0x83 }, + { 0x100076d1, 0x20 }, + { 0x100076d2, 0xc1 }, + { 0x100076d3, 0x00 }, + { 0x100076d4, 0x13 }, + { 0x100076d5, 0x01 }, + { 0x100076d6, 0x01 }, + { 0x100076d7, 0x01 }, + { 0x100076d8, 0x67 }, + { 0x100076d9, 0x80 }, + { 0x100076da, 0x00 }, + { 0x100076db, 0x00 }, + { 0x100076dc, 0x13 }, + { 0x100076dd, 0x77 }, + { 0x100076de, 0xf7 }, + { 0x100076df, 0xdf }, + { 0x100076e0, 0x23 }, + { 0x100076e1, 0xa0 }, + { 0x100076e2, 0xe7 }, + { 0x100076e3, 0xfc }, + { 0x100076e4, 0x03 }, + { 0x100076e5, 0xa7 }, + { 0x100076e6, 0x07 }, + { 0x100076e7, 0xfc }, + { 0x100076e8, 0x13 }, + { 0x100076e9, 0x77 }, + { 0x100076ea, 0xf7 }, + { 0x100076eb, 0xbf }, + { 0x100076ec, 0x6f }, + { 0x100076ed, 0xf0 }, + { 0x100076ee, 0x5f }, + { 0x100076ef, 0xfa }, + { 0x100076f0, 0x67 }, + { 0x100076f1, 0x80 }, + { 0x100076f2, 0x00 }, + { 0x100076f3, 0x00 }, + { 0x100076f4, 0xb7 }, + { 0x100076f5, 0xc7 }, + { 0x100076f6, 0x00 }, + { 0x100076f7, 0x00 }, + { 0x100076f8, 0x03 }, + { 0x100076f9, 0xc7 }, + { 0x100076fa, 0x87 }, + { 0x100076fb, 0x59 }, + { 0x100076fc, 0x13 }, + { 0x100076fd, 0x77 }, + { 0x100076fe, 0xf7 }, + { 0x100076ff, 0x0f }, + { 0x10007700, 0x13 }, + { 0x10007701, 0x67 }, + { 0x10007702, 0x17 }, + { 0x10007703, 0x00 }, + { 0x10007704, 0x23 }, + { 0x10007705, 0x8c }, + { 0x10007706, 0xe7 }, + { 0x10007707, 0x58 }, + { 0x10007708, 0x03 }, + { 0x10007709, 0xc7 }, + { 0x1000770a, 0x77 }, + { 0x1000770b, 0x04 }, + { 0x1000770c, 0x13 }, + { 0x1000770d, 0x17 }, + { 0x1000770e, 0x87 }, + { 0x1000770f, 0x01 }, + { 0x10007710, 0x13 }, + { 0x10007711, 0x57 }, + { 0x10007712, 0x87 }, + { 0x10007713, 0x41 }, + { 0x10007714, 0x63 }, + { 0x10007715, 0x58 }, + { 0x10007716, 0x07 }, + { 0x10007717, 0x12 }, + { 0x10007718, 0x37 }, + { 0x10007719, 0xd7 }, + { 0x1000771a, 0x00 }, + { 0x1000771b, 0x00 }, + { 0x1000771c, 0x83 }, + { 0x1000771d, 0x26 }, + { 0x1000771e, 0x87 }, + { 0x1000771f, 0x53 }, + { 0x10007720, 0x37 }, + { 0x10007721, 0x06 }, + { 0x10007722, 0x00 }, + { 0x10007723, 0x40 }, + { 0x10007724, 0x93 }, + { 0x10007725, 0x05 }, + { 0x10007726, 0x80 }, + { 0x10007727, 0x01 }, + { 0x10007728, 0xb3 }, + { 0x10007729, 0xe6 }, + { 0x1000772a, 0xc6 }, + { 0x1000772b, 0x00 }, + { 0x1000772c, 0x23 }, + { 0x1000772d, 0x2c }, + { 0x1000772e, 0xd7 }, + { 0x1000772f, 0x52 }, + { 0x10007730, 0x83 }, + { 0x10007731, 0xc6 }, + { 0x10007732, 0x07 }, + { 0x10007733, 0x56 }, + { 0x10007734, 0x93 }, + { 0x10007735, 0xf6 }, + { 0x10007736, 0xf6 }, + { 0x10007737, 0x0f }, + { 0x10007738, 0x63 }, + { 0x10007739, 0x9c }, + { 0x1000773a, 0xb6 }, + { 0x1000773b, 0x0e }, + { 0x1000773c, 0x83 }, + { 0x1000773d, 0x27 }, + { 0x1000773e, 0x87 }, + { 0x1000773f, 0x53 }, + { 0x10007740, 0xb3 }, + { 0x10007741, 0xf7 }, + { 0x10007742, 0xc7 }, + { 0x10007743, 0x00 }, + { 0x10007744, 0x63 }, + { 0x10007745, 0x80 }, + { 0x10007746, 0x07 }, + { 0x10007747, 0x10 }, + { 0x10007748, 0x13 }, + { 0x10007749, 0x01 }, + { 0x1000774a, 0x01 }, + { 0x1000774b, 0xff }, + { 0x1000774c, 0x23 }, + { 0x1000774d, 0x24 }, + { 0x1000774e, 0x81 }, + { 0x1000774f, 0x00 }, + { 0x10007750, 0x83 }, + { 0x10007751, 0xa7 }, + { 0x10007752, 0x41 }, + { 0x10007753, 0x58 }, + { 0x10007754, 0x23 }, + { 0x10007755, 0x26 }, + { 0x10007756, 0x11 }, + { 0x10007757, 0x00 }, + { 0x10007758, 0x63 }, + { 0x10007759, 0x94 }, + { 0x1000775a, 0x07 }, + { 0x1000775b, 0x0c }, + { 0x1000775c, 0x83 }, + { 0x1000775d, 0x27 }, + { 0x1000775e, 0x07 }, + { 0x1000775f, 0x53 }, + { 0x10007760, 0x03 }, + { 0x10007761, 0xc6 }, + { 0x10007762, 0xb1 }, + { 0x10007763, 0x42 }, + { 0x10007764, 0x93 }, + { 0x10007765, 0xd7 }, + { 0x10007766, 0xe7 }, + { 0x10007767, 0x01 }, + { 0x10007768, 0x93 }, + { 0x10007769, 0xf7 }, + { 0x1000776a, 0x17 }, + { 0x1000776b, 0x00 }, + { 0x1000776c, 0x93 }, + { 0x1000776d, 0x06 }, + { 0x1000776e, 0x10 }, + { 0x1000776f, 0x00 }, + { 0x10007770, 0x63 }, + { 0x10007771, 0x14 }, + { 0x10007772, 0x06 }, + { 0x10007773, 0x00 }, + { 0x10007774, 0xb3 }, + { 0x10007775, 0x86 }, + { 0x10007776, 0xf6 }, + { 0x10007777, 0x40 }, + { 0x10007778, 0xa3 }, + { 0x10007779, 0x85 }, + { 0x1000777a, 0xd1 }, + { 0x1000777b, 0x42 }, + { 0x1000777c, 0x03 }, + { 0x1000777d, 0xc6 }, + { 0x1000777e, 0xa1 }, + { 0x1000777f, 0x42 }, + { 0x10007780, 0x93 }, + { 0x10007781, 0x06 }, + { 0x10007782, 0x10 }, + { 0x10007783, 0x00 }, + { 0x10007784, 0x63 }, + { 0x10007785, 0x14 }, + { 0x10007786, 0x06 }, + { 0x10007787, 0x00 }, + { 0x10007788, 0xb3 }, + { 0x10007789, 0x86 }, + { 0x1000778a, 0xf6 }, + { 0x1000778b, 0x40 }, + { 0x1000778c, 0x23 }, + { 0x1000778d, 0x85 }, + { 0x1000778e, 0xd1 }, + { 0x1000778f, 0x42 }, + { 0x10007790, 0x03 }, + { 0x10007791, 0xc6 }, + { 0x10007792, 0x91 }, + { 0x10007793, 0x42 }, + { 0x10007794, 0x93 }, + { 0x10007795, 0x06 }, + { 0x10007796, 0x10 }, + { 0x10007797, 0x00 }, + { 0x10007798, 0x63 }, + { 0x10007799, 0x14 }, + { 0x1000779a, 0x06 }, + { 0x1000779b, 0x00 }, + { 0x1000779c, 0xb3 }, + { 0x1000779d, 0x86 }, + { 0x1000779e, 0xf6 }, + { 0x1000779f, 0x40 }, + { 0x100077a0, 0xa3 }, + { 0x100077a1, 0x84 }, + { 0x100077a2, 0xd1 }, + { 0x100077a3, 0x42 }, + { 0x100077a4, 0x03 }, + { 0x100077a5, 0xc6 }, + { 0x100077a6, 0x81 }, + { 0x100077a7, 0x42 }, + { 0x100077a8, 0x93 }, + { 0x100077a9, 0x06 }, + { 0x100077aa, 0x10 }, + { 0x100077ab, 0x00 }, + { 0x100077ac, 0x63 }, + { 0x100077ad, 0x14 }, + { 0x100077ae, 0x06 }, + { 0x100077af, 0x00 }, + { 0x100077b0, 0xb3 }, + { 0x100077b1, 0x86 }, + { 0x100077b2, 0xf6 }, + { 0x100077b3, 0x40 }, + { 0x100077b4, 0x23 }, + { 0x100077b5, 0x84 }, + { 0x100077b6, 0xd1 }, + { 0x100077b7, 0x42 }, + { 0x100077b8, 0xb7 }, + { 0x100077b9, 0xd7 }, + { 0x100077ba, 0x00 }, + { 0x100077bb, 0x00 }, + { 0x100077bc, 0x83 }, + { 0x100077bd, 0xa7 }, + { 0x100077be, 0x07 }, + { 0x100077bf, 0x53 }, + { 0x100077c0, 0x37 }, + { 0x100077c1, 0x07 }, + { 0x100077c2, 0x00 }, + { 0x100077c3, 0x40 }, + { 0x100077c4, 0xb3 }, + { 0x100077c5, 0xf7 }, + { 0x100077c6, 0xe7 }, + { 0x100077c7, 0x00 }, + { 0x100077c8, 0x63 }, + { 0x100077c9, 0x8c }, + { 0x100077ca, 0x07 }, + { 0x100077cb, 0x04 }, + { 0x100077cc, 0xb7 }, + { 0x100077cd, 0x47 }, + { 0x100077ce, 0x0f }, + { 0x100077cf, 0x00 }, + { 0x100077d0, 0x93 }, + { 0x100077d1, 0x87 }, + { 0x100077d2, 0x17 }, + { 0x100077d3, 0x24 }, + { 0x100077d4, 0xb7 }, + { 0x100077d5, 0xf6 }, + { 0x100077d6, 0x00 }, + { 0x100077d7, 0x00 }, + { 0x100077d8, 0x03 }, + { 0x100077d9, 0xc7 }, + { 0x100077da, 0xf6 }, + { 0x100077db, 0x83 }, + { 0x100077dc, 0x13 }, + { 0x100077dd, 0x77 }, + { 0x100077de, 0x07 }, + { 0x100077df, 0x04 }, + { 0x100077e0, 0x63 }, + { 0x100077e1, 0x16 }, + { 0x100077e2, 0x07 }, + { 0x100077e3, 0x00 }, + { 0x100077e4, 0x93 }, + { 0x100077e5, 0x87 }, + { 0x100077e6, 0xf7 }, + { 0x100077e7, 0xff }, + { 0x100077e8, 0xe3 }, + { 0x100077e9, 0x98 }, + { 0x100077ea, 0x07 }, + { 0x100077eb, 0xfe }, + { 0x100077ec, 0x13 }, + { 0x100077ed, 0x05 }, + { 0x100077ee, 0x80 }, + { 0x100077ef, 0x3e }, + { 0x100077f0, 0x93 }, + { 0x100077f1, 0x05 }, + { 0x100077f2, 0x00 }, + { 0x100077f3, 0x00 }, + { 0x100077f4, 0xef }, + { 0x100077f5, 0xe0 }, + { 0x100077f6, 0x0f }, + { 0x100077f7, 0xa3 }, + { 0x100077f8, 0x37 }, + { 0x100077f9, 0xf7 }, + { 0x100077fa, 0x00 }, + { 0x100077fb, 0x00 }, + { 0x100077fc, 0x83 }, + { 0x100077fd, 0x47 }, + { 0x100077fe, 0xb7 }, + { 0x100077ff, 0x80 }, + { 0x10007800, 0x93 }, + { 0x10007801, 0xe7 }, + { 0x10007802, 0x07 }, + { 0x10007803, 0xf8 }, + { 0x10007804, 0x93 }, + { 0x10007805, 0xf7 }, + { 0x10007806, 0xf7 }, + { 0x10007807, 0x0f }, + { 0x10007808, 0xa3 }, + { 0x10007809, 0x05 }, + { 0x1000780a, 0xf7 }, + { 0x1000780b, 0x80 }, + { 0x1000780c, 0xb7 }, + { 0x1000780d, 0xd7 }, + { 0x1000780e, 0x00 }, + { 0x1000780f, 0x00 }, + { 0x10007810, 0x37 }, + { 0x10007811, 0x07 }, + { 0x10007812, 0x00 }, + { 0x10007813, 0x40 }, + { 0x10007814, 0x23 }, + { 0x10007815, 0xa8 }, + { 0x10007816, 0xe7 }, + { 0x10007817, 0x52 }, + { 0x10007818, 0x93 }, + { 0x10007819, 0x07 }, + { 0x1000781a, 0x10 }, + { 0x1000781b, 0x00 }, + { 0x1000781c, 0x23 }, + { 0x1000781d, 0xa2 }, + { 0x1000781e, 0xf1 }, + { 0x1000781f, 0x58 }, + { 0x10007820, 0x83 }, + { 0x10007821, 0x20 }, + { 0x10007822, 0xc1 }, + { 0x10007823, 0x00 }, + { 0x10007824, 0x03 }, + { 0x10007825, 0x24 }, + { 0x10007826, 0x81 }, + { 0x10007827, 0x00 }, + { 0x10007828, 0x13 }, + { 0x10007829, 0x01 }, + { 0x1000782a, 0x01 }, + { 0x1000782b, 0x01 }, + { 0x1000782c, 0x67 }, + { 0x1000782d, 0x80 }, + { 0x1000782e, 0x00 }, + { 0x1000782f, 0x00 }, + { 0x10007830, 0x83 }, + { 0x10007831, 0xc7 }, + { 0x10007832, 0x07 }, + { 0x10007833, 0x56 }, + { 0x10007834, 0x93 }, + { 0x10007835, 0xf7 }, + { 0x10007836, 0xf7 }, + { 0x10007837, 0x0f }, + { 0x10007838, 0x63 }, + { 0x10007839, 0x96 }, + { 0x1000783a, 0x07 }, + { 0x1000783b, 0x00 }, + { 0x1000783c, 0x23 }, + { 0x1000783d, 0xa2 }, + { 0x1000783e, 0x01 }, + { 0x1000783f, 0x58 }, + { 0x10007840, 0x67 }, + { 0x10007841, 0x80 }, + { 0x10007842, 0x00 }, + { 0x10007843, 0x00 }, + { 0x10007844, 0x67 }, + { 0x10007845, 0x80 }, + { 0x10007846, 0x00 }, + { 0x10007847, 0x00 }, + { 0x10007848, 0xb7 }, + { 0x10007849, 0xc7 }, + { 0x1000784a, 0x00 }, + { 0x1000784b, 0x00 }, + { 0x1000784c, 0x83 }, + { 0x1000784d, 0xc7 }, + { 0x1000784e, 0x07 }, + { 0x1000784f, 0x56 }, + { 0x10007850, 0x13 }, + { 0x10007851, 0x07 }, + { 0x10007852, 0x80 }, + { 0x10007853, 0x01 }, + { 0x10007854, 0x93 }, + { 0x10007855, 0xf7 }, + { 0x10007856, 0xf7 }, + { 0x10007857, 0x0f }, + { 0x10007858, 0x63 }, + { 0x10007859, 0x98 }, + { 0x1000785a, 0xe7 }, + { 0x1000785b, 0x00 }, + { 0x1000785c, 0x13 }, + { 0x1000785d, 0x05 }, + { 0x1000785e, 0x00 }, + { 0x1000785f, 0x7d }, + { 0x10007860, 0x93 }, + { 0x10007861, 0x05 }, + { 0x10007862, 0x00 }, + { 0x10007863, 0x00 }, + { 0x10007864, 0x6f }, + { 0x10007865, 0xe0 }, + { 0x10007866, 0x0f }, + { 0x10007867, 0x9c }, + { 0x10007868, 0x67 }, + { 0x10007869, 0x80 }, + { 0x1000786a, 0x00 }, + { 0x1000786b, 0x00 }, + { 0x1000786c, 0x13 }, + { 0x1000786d, 0x01 }, + { 0x1000786e, 0x01 }, + { 0x1000786f, 0xff }, + { 0x10007870, 0x23 }, + { 0x10007871, 0x26 }, + { 0x10007872, 0x11 }, + { 0x10007873, 0x00 }, + { 0x10007874, 0x23 }, + { 0x10007875, 0x24 }, + { 0x10007876, 0x81 }, + { 0x10007877, 0x00 }, + { 0x10007878, 0xef }, + { 0x10007879, 0xd0 }, + { 0x1000787a, 0x4f }, + { 0x1000787b, 0x91 }, + { 0x1000787c, 0x83 }, + { 0x1000787d, 0xc7 }, + { 0x1000787e, 0x81 }, + { 0x1000787f, 0x41 }, + { 0x10007880, 0x63 }, + { 0x10007881, 0x84 }, + { 0x10007882, 0x07 }, + { 0x10007883, 0x08 }, + { 0x10007884, 0xb7 }, + { 0x10007885, 0xd7 }, + { 0x10007886, 0x00 }, + { 0x10007887, 0x00 }, + { 0x10007888, 0x83 }, + { 0x10007889, 0xc7 }, + { 0x1000788a, 0x07 }, + { 0x1000788b, 0x47 }, + { 0x1000788c, 0x93 }, + { 0x1000788d, 0xf7 }, + { 0x1000788e, 0x07 }, + { 0x1000788f, 0x02 }, + { 0x10007890, 0x63 }, + { 0x10007891, 0x8a }, + { 0x10007892, 0x07 }, + { 0x10007893, 0x04 }, + { 0x10007894, 0x83 }, + { 0x10007895, 0xc7 }, + { 0x10007896, 0x11 }, + { 0x10007897, 0x44 }, + { 0x10007898, 0x93 }, + { 0x10007899, 0xf7 }, + { 0x1000789a, 0xd7 }, + { 0x1000789b, 0x0f }, + { 0x1000789c, 0x63 }, + { 0x1000789d, 0x90 }, + { 0x1000789e, 0x07 }, + { 0x1000789f, 0x02 }, + { 0x100078a0, 0x03 }, + { 0x100078a1, 0xc7 }, + { 0x100078a2, 0xd1 }, + { 0x100078a3, 0x58 }, + { 0x100078a4, 0xb7 }, + { 0x100078a5, 0x07 }, + { 0x100078a6, 0x00 }, + { 0x100078a7, 0x11 }, + { 0x100078a8, 0x23 }, + { 0x100078a9, 0x88 }, + { 0x100078aa, 0xe7 }, + { 0x100078ab, 0x00 }, + { 0x100078ac, 0x23 }, + { 0x100078ad, 0x88 }, + { 0x100078ae, 0xe7 }, + { 0x100078af, 0x20 }, + { 0x100078b0, 0x03 }, + { 0x100078b1, 0xc7 }, + { 0x100078b2, 0xc1 }, + { 0x100078b3, 0x58 }, + { 0x100078b4, 0x23 }, + { 0x100078b5, 0x8c }, + { 0x100078b6, 0xe7 }, + { 0x100078b7, 0x00 }, + { 0x100078b8, 0x6f }, + { 0x100078b9, 0x00 }, + { 0x100078ba, 0x80 }, + { 0x100078bb, 0x04 }, + { 0x100078bc, 0xb7 }, + { 0x100078bd, 0x07 }, + { 0x100078be, 0x00 }, + { 0x100078bf, 0x11 }, + { 0x100078c0, 0x23 }, + { 0x100078c1, 0x88 }, + { 0x100078c2, 0x07 }, + { 0x100078c3, 0x00 }, + { 0x100078c4, 0x23 }, + { 0x100078c5, 0x88 }, + { 0x100078c6, 0x07 }, + { 0x100078c7, 0x20 }, + { 0x100078c8, 0x23 }, + { 0x100078c9, 0x8c }, + { 0x100078ca, 0x07 }, + { 0x100078cb, 0x00 }, + { 0x100078cc, 0x23 }, + { 0x100078cd, 0x8c }, + { 0x100078ce, 0x07 }, + { 0x100078cf, 0x20 }, + { 0x100078d0, 0xef }, + { 0x100078d1, 0xb0 }, + { 0x100078d2, 0xcf }, + { 0x100078d3, 0xc4 }, + { 0x100078d4, 0x03 }, + { 0x100078d5, 0x24 }, + { 0x100078d6, 0x81 }, + { 0x100078d7, 0x00 }, + { 0x100078d8, 0x83 }, + { 0x100078d9, 0x20 }, + { 0x100078da, 0xc1 }, + { 0x100078db, 0x00 }, + { 0x100078dc, 0x13 }, + { 0x100078dd, 0x01 }, + { 0x100078de, 0x01 }, + { 0x100078df, 0x01 }, + { 0x100078e0, 0x6f }, + { 0x100078e1, 0xb0 }, + { 0x100078e2, 0xcf }, + { 0x100078e3, 0xcd }, + { 0x100078e4, 0x03 }, + { 0x100078e5, 0xc7 }, + { 0x100078e6, 0xd1 }, + { 0x100078e7, 0x58 }, + { 0x100078e8, 0xb7 }, + { 0x100078e9, 0x07 }, + { 0x100078ea, 0x00 }, + { 0x100078eb, 0x11 }, + { 0x100078ec, 0x23 }, + { 0x100078ed, 0x88 }, + { 0x100078ee, 0xe7 }, + { 0x100078ef, 0x00 }, + { 0x100078f0, 0x83 }, + { 0x100078f1, 0xc6 }, + { 0x100078f2, 0xc1 }, + { 0x100078f3, 0x58 }, + { 0x100078f4, 0x23 }, + { 0x100078f5, 0x88 }, + { 0x100078f6, 0xd7 }, + { 0x100078f7, 0x20 }, + { 0x100078f8, 0x23 }, + { 0x100078f9, 0x8c }, + { 0x100078fa, 0xd7 }, + { 0x100078fb, 0x00 }, + { 0x100078fc, 0x03 }, + { 0x100078fd, 0xc7 }, + { 0x100078fe, 0xc1 }, + { 0x100078ff, 0x58 }, + { 0x10007900, 0x23 }, + { 0x10007901, 0x8c }, + { 0x10007902, 0xe7 }, + { 0x10007903, 0x20 }, + { 0x10007904, 0x6f }, + { 0x10007905, 0xf0 }, + { 0x10007906, 0xdf }, + { 0x10007907, 0xfc }, + { 0x10007908, 0xb7 }, + { 0x10007909, 0x06 }, + { 0x1000790a, 0x00 }, + { 0x1000790b, 0x11 }, + { 0x1000790c, 0x03 }, + { 0x1000790d, 0xc7 }, + { 0x1000790e, 0x06 }, + { 0x1000790f, 0x21 }, + { 0x10007910, 0x03 }, + { 0x10007911, 0xc6 }, + { 0x10007912, 0xd1 }, + { 0x10007913, 0x58 }, + { 0x10007914, 0x13 }, + { 0x10007915, 0x84 }, + { 0x10007916, 0x07 }, + { 0x10007917, 0x00 }, + { 0x10007918, 0x13 }, + { 0x10007919, 0x77 }, + { 0x1000791a, 0xf7 }, + { 0x1000791b, 0x0f }, + { 0x1000791c, 0x63 }, + { 0x1000791d, 0x1a }, + { 0x1000791e, 0xe6 }, + { 0x1000791f, 0x00 }, + { 0x10007920, 0x83 }, + { 0x10007921, 0xc7 }, + { 0x10007922, 0x86 }, + { 0x10007923, 0x21 }, + { 0x10007924, 0x03 }, + { 0x10007925, 0xc7 }, + { 0x10007926, 0xc1 }, + { 0x10007927, 0x58 }, + { 0x10007928, 0x93 }, + { 0x10007929, 0xf7 }, + { 0x1000792a, 0xf7 }, + { 0x1000792b, 0x0f }, + { 0x1000792c, 0xe3 }, + { 0x1000792d, 0x02 }, + { 0x1000792e, 0xf7 }, + { 0x1000792f, 0xfa }, + { 0x10007930, 0xb7 }, + { 0x10007931, 0xc7 }, + { 0x10007932, 0x00 }, + { 0x10007933, 0x00 }, + { 0x10007934, 0x83 }, + { 0x10007935, 0xc7 }, + { 0x10007936, 0x07 }, + { 0x10007937, 0x56 }, + { 0x10007938, 0x13 }, + { 0x10007939, 0x07 }, + { 0x1000793a, 0xf0 }, + { 0x1000793b, 0x00 }, + { 0x1000793c, 0x93 }, + { 0x1000793d, 0xf7 }, + { 0x1000793e, 0xf7 }, + { 0x1000793f, 0x0f }, + { 0x10007940, 0xe3 }, + { 0x10007941, 0x78 }, + { 0x10007942, 0xf7 }, + { 0x10007943, 0xf8 }, + { 0x10007944, 0xb7 }, + { 0x10007945, 0xd7 }, + { 0x10007946, 0x00 }, + { 0x10007947, 0x00 }, + { 0x10007948, 0x83 }, + { 0x10007949, 0xc5 }, + { 0x1000794a, 0xa7 }, + { 0x1000794b, 0x47 }, + { 0x1000794c, 0x93 }, + { 0x1000794d, 0xf7 }, + { 0x1000794e, 0xf5 }, + { 0x1000794f, 0x0f }, + { 0x10007950, 0x93 }, + { 0x10007951, 0x95 }, + { 0x10007952, 0x57 }, + { 0x10007953, 0x00 }, + { 0x10007954, 0xb3 }, + { 0x10007955, 0x85 }, + { 0x10007956, 0xf5 }, + { 0x10007957, 0x40 }, + { 0x10007958, 0x93 }, + { 0x10007959, 0x95 }, + { 0x1000795a, 0x25 }, + { 0x1000795b, 0x00 }, + { 0x1000795c, 0xb3 }, + { 0x1000795d, 0x85 }, + { 0x1000795e, 0xf5 }, + { 0x1000795f, 0x00 }, + { 0x10007960, 0x13 }, + { 0x10007961, 0x95 }, + { 0x10007962, 0x35 }, + { 0x10007963, 0x00 }, + { 0x10007964, 0x93 }, + { 0x10007965, 0xd5 }, + { 0x10007966, 0xf5 }, + { 0x10007967, 0x41 }, + { 0x10007968, 0xef }, + { 0x10007969, 0xe0 }, + { 0x1000796a, 0xcf }, + { 0x1000796b, 0x8b }, + { 0x1000796c, 0x03 }, + { 0x1000796d, 0xc7 }, + { 0x1000796e, 0xd1 }, + { 0x1000796f, 0x58 }, + { 0x10007970, 0x6f }, + { 0x10007971, 0xf0 }, + { 0x10007972, 0x5f }, + { 0x10007973, 0xf3 }, + { 0x10007974, 0x13 }, + { 0x10007975, 0x01 }, + { 0x10007976, 0x01 }, + { 0x10007977, 0xfe }, + { 0x10007978, 0x23 }, + { 0x10007979, 0x2c }, + { 0x1000797a, 0x81 }, + { 0x1000797b, 0x00 }, + { 0x1000797c, 0x83 }, + { 0x1000797d, 0xc7 }, + { 0x1000797e, 0x21 }, + { 0x1000797f, 0x41 }, + { 0x10007980, 0x23 }, + { 0x10007981, 0x2e }, + { 0x10007982, 0x11 }, + { 0x10007983, 0x00 }, + { 0x10007984, 0x23 }, + { 0x10007985, 0x2a }, + { 0x10007986, 0x91 }, + { 0x10007987, 0x00 }, + { 0x10007988, 0x23 }, + { 0x10007989, 0x28 }, + { 0x1000798a, 0x21 }, + { 0x1000798b, 0x01 }, + { 0x1000798c, 0x23 }, + { 0x1000798d, 0x26 }, + { 0x1000798e, 0x31 }, + { 0x1000798f, 0x01 }, + { 0x10007990, 0x13 }, + { 0x10007991, 0x07 }, + { 0x10007992, 0x10 }, + { 0x10007993, 0x00 }, + { 0x10007994, 0x63 }, + { 0x10007995, 0x92 }, + { 0x10007996, 0xe7 }, + { 0x10007997, 0x02 }, + { 0x10007998, 0xa3 }, + { 0x10007999, 0x81 }, + { 0x1000799a, 0xf1 }, + { 0x1000799b, 0x40 }, + { 0x1000799c, 0x83 }, + { 0x1000799d, 0x20 }, + { 0x1000799e, 0xc1 }, + { 0x1000799f, 0x01 }, + { 0x100079a0, 0x03 }, + { 0x100079a1, 0x24 }, + { 0x100079a2, 0x81 }, + { 0x100079a3, 0x01 }, + { 0x100079a4, 0x83 }, + { 0x100079a5, 0x24 }, + { 0x100079a6, 0x41 }, + { 0x100079a7, 0x01 }, + { 0x100079a8, 0x03 }, + { 0x100079a9, 0x29 }, + { 0x100079aa, 0x01 }, + { 0x100079ab, 0x01 }, + { 0x100079ac, 0x83 }, + { 0x100079ad, 0x29 }, + { 0x100079ae, 0xc1 }, + { 0x100079af, 0x00 }, + { 0x100079b0, 0x13 }, + { 0x100079b1, 0x01 }, + { 0x100079b2, 0x01 }, + { 0x100079b3, 0x02 }, + { 0x100079b4, 0x67 }, + { 0x100079b5, 0x80 }, + { 0x100079b6, 0x00 }, + { 0x100079b7, 0x00 }, + { 0x100079b8, 0xe3 }, + { 0x100079b9, 0x92 }, + { 0x100079ba, 0x07 }, + { 0x100079bb, 0xfe }, + { 0x100079bc, 0x37 }, + { 0x100079bd, 0xc9 }, + { 0x100079be, 0x00 }, + { 0x100079bf, 0x00 }, + { 0x100079c0, 0x83 }, + { 0x100079c1, 0x47 }, + { 0x100079c2, 0x09 }, + { 0x100079c3, 0x56 }, + { 0x100079c4, 0x13 }, + { 0x100079c5, 0x07 }, + { 0x100079c6, 0x80 }, + { 0x100079c7, 0x01 }, + { 0x100079c8, 0x93 }, + { 0x100079c9, 0xf7 }, + { 0x100079ca, 0xf7 }, + { 0x100079cb, 0x0f }, + { 0x100079cc, 0xe3 }, + { 0x100079cd, 0x78 }, + { 0x100079ce, 0xf7 }, + { 0x100079cf, 0xfc }, + { 0x100079d0, 0x83 }, + { 0x100079d1, 0xc7 }, + { 0x100079d2, 0x31 }, + { 0x100079d3, 0x40 }, + { 0x100079d4, 0xe3 }, + { 0x100079d5, 0x84 }, + { 0x100079d6, 0x07 }, + { 0x100079d7, 0xfc }, + { 0x100079d8, 0xb7 }, + { 0x100079d9, 0xd4 }, + { 0x100079da, 0x00 }, + { 0x100079db, 0x00 }, + { 0x100079dc, 0x03 }, + { 0x100079dd, 0xc5 }, + { 0x100079de, 0x94 }, + { 0x100079df, 0x47 }, + { 0x100079e0, 0xb7 }, + { 0x100079e1, 0x15 }, + { 0x100079e2, 0x00 }, + { 0x100079e3, 0x00 }, + { 0x100079e4, 0x93 }, + { 0x100079e5, 0x85 }, + { 0x100079e6, 0x85 }, + { 0x100079e7, 0x38 }, + { 0x100079e8, 0x13 }, + { 0x100079e9, 0x75 }, + { 0x100079ea, 0xf5 }, + { 0x100079eb, 0x0f }, + { 0x100079ec, 0xef }, + { 0x100079ed, 0xe0 }, + { 0x100079ee, 0x5f }, + { 0x100079ef, 0xe0 }, + { 0x100079f0, 0x93 }, + { 0x100079f1, 0x55 }, + { 0x100079f2, 0xf5 }, + { 0x100079f3, 0x41 }, + { 0x100079f4, 0xef }, + { 0x100079f5, 0xe0 }, + { 0x100079f6, 0x0f }, + { 0x100079f7, 0x83 }, + { 0x100079f8, 0xa3 }, + { 0x100079f9, 0x81 }, + { 0x100079fa, 0x01 }, + { 0x100079fb, 0x40 }, + { 0x100079fc, 0x83 }, + { 0x100079fd, 0x27 }, + { 0x100079fe, 0xc9 }, + { 0x100079ff, 0x5f }, + { 0x10007a00, 0x37 }, + { 0x10007a01, 0x07 }, + { 0x10007a02, 0x00 }, + { 0x10007a03, 0x02 }, + { 0x10007a04, 0xb3 }, + { 0x10007a05, 0xf7 }, + { 0x10007a06, 0xe7 }, + { 0x10007a07, 0x00 }, + { 0x10007a08, 0xe3 }, + { 0x10007a09, 0x8a }, + { 0x10007a0a, 0x07 }, + { 0x10007a0b, 0xf8 }, + { 0x10007a0c, 0x03 }, + { 0x10007a0d, 0xc7 }, + { 0x10007a0e, 0x04 }, + { 0x10007a0f, 0x90 }, + { 0x10007a10, 0x93 }, + { 0x10007a11, 0x07 }, + { 0x10007a12, 0x10 }, + { 0x10007a13, 0x00 }, + { 0x10007a14, 0x13 }, + { 0x10007a15, 0x77 }, + { 0x10007a16, 0x17 }, + { 0x10007a17, 0x00 }, + { 0x10007a18, 0x63 }, + { 0x10007a19, 0x1c }, + { 0x10007a1a, 0x07 }, + { 0x10007a1b, 0x00 }, + { 0x10007a1c, 0x83 }, + { 0x10007a1d, 0xc7 }, + { 0x10007a1e, 0x34 }, + { 0x10007a1f, 0x54 }, + { 0x10007a20, 0x93 }, + { 0x10007a21, 0xf7 }, + { 0x10007a22, 0xf7 }, + { 0x10007a23, 0x0f }, + { 0x10007a24, 0x93 }, + { 0x10007a25, 0xd7 }, + { 0x10007a26, 0x17 }, + { 0x10007a27, 0x00 }, + { 0x10007a28, 0x93 }, + { 0x10007a29, 0xc7 }, + { 0x10007a2a, 0x17 }, + { 0x10007a2b, 0x00 }, + { 0x10007a2c, 0x93 }, + { 0x10007a2d, 0xf7 }, + { 0x10007a2e, 0x17 }, + { 0x10007a2f, 0x00 }, + { 0x10007a30, 0xa3 }, + { 0x10007a31, 0x85 }, + { 0x10007a32, 0xf1 }, + { 0x10007a33, 0x42 }, + { 0x10007a34, 0x37 }, + { 0x10007a35, 0xd6 }, + { 0x10007a36, 0x00 }, + { 0x10007a37, 0x00 }, + { 0x10007a38, 0x03 }, + { 0x10007a39, 0x47 }, + { 0x10007a3a, 0x06 }, + { 0x10007a3b, 0x90 }, + { 0x10007a3c, 0x93 }, + { 0x10007a3d, 0x06 }, + { 0x10007a3e, 0x10 }, + { 0x10007a3f, 0x00 }, + { 0x10007a40, 0x13 }, + { 0x10007a41, 0x77 }, + { 0x10007a42, 0x27 }, + { 0x10007a43, 0x00 }, + { 0x10007a44, 0x63 }, + { 0x10007a45, 0x18 }, + { 0x10007a46, 0x07 }, + { 0x10007a47, 0x00 }, + { 0x10007a48, 0x03 }, + { 0x10007a49, 0x47 }, + { 0x10007a4a, 0x36 }, + { 0x10007a4b, 0x54 }, + { 0x10007a4c, 0x13 }, + { 0x10007a4d, 0x77 }, + { 0x10007a4e, 0x17 }, + { 0x10007a4f, 0x00 }, + { 0x10007a50, 0xb3 }, + { 0x10007a51, 0x86 }, + { 0x10007a52, 0xe6 }, + { 0x10007a53, 0x40 }, + { 0x10007a54, 0x23 }, + { 0x10007a55, 0x85 }, + { 0x10007a56, 0xd1 }, + { 0x10007a57, 0x42 }, + { 0x10007a58, 0xb7 }, + { 0x10007a59, 0xd5 }, + { 0x10007a5a, 0x00 }, + { 0x10007a5b, 0x00 }, + { 0x10007a5c, 0x03 }, + { 0x10007a5d, 0xc6 }, + { 0x10007a5e, 0x05 }, + { 0x10007a5f, 0x92 }, + { 0x10007a60, 0x13 }, + { 0x10007a61, 0x07 }, + { 0x10007a62, 0x10 }, + { 0x10007a63, 0x00 }, + { 0x10007a64, 0x13 }, + { 0x10007a65, 0x76 }, + { 0x10007a66, 0x16 }, + { 0x10007a67, 0x00 }, + { 0x10007a68, 0x63 }, + { 0x10007a69, 0x1c }, + { 0x10007a6a, 0x06 }, + { 0x10007a6b, 0x00 }, + { 0x10007a6c, 0x03 }, + { 0x10007a6d, 0xc7 }, + { 0x10007a6e, 0x35 }, + { 0x10007a6f, 0x54 }, + { 0x10007a70, 0x13 }, + { 0x10007a71, 0x77 }, + { 0x10007a72, 0xf7 }, + { 0x10007a73, 0x0f }, + { 0x10007a74, 0x13 }, + { 0x10007a75, 0x57 }, + { 0x10007a76, 0x37 }, + { 0x10007a77, 0x00 }, + { 0x10007a78, 0x13 }, + { 0x10007a79, 0x47 }, + { 0x10007a7a, 0x17 }, + { 0x10007a7b, 0x00 }, + { 0x10007a7c, 0x13 }, + { 0x10007a7d, 0x77 }, + { 0x10007a7e, 0x17 }, + { 0x10007a7f, 0x00 }, + { 0x10007a80, 0xa3 }, + { 0x10007a81, 0x84 }, + { 0x10007a82, 0xe1 }, + { 0x10007a83, 0x42 }, + { 0x10007a84, 0xb7 }, + { 0x10007a85, 0xd5 }, + { 0x10007a86, 0x00 }, + { 0x10007a87, 0x00 }, + { 0x10007a88, 0x03 }, + { 0x10007a89, 0xc6 }, + { 0x10007a8a, 0x05 }, + { 0x10007a8b, 0x92 }, + { 0x10007a8c, 0x13 }, + { 0x10007a8d, 0x07 }, + { 0x10007a8e, 0x10 }, + { 0x10007a8f, 0x00 }, + { 0x10007a90, 0x13 }, + { 0x10007a91, 0x76 }, + { 0x10007a92, 0x26 }, + { 0x10007a93, 0x00 }, + { 0x10007a94, 0x63 }, + { 0x10007a95, 0x1c }, + { 0x10007a96, 0x06 }, + { 0x10007a97, 0x00 }, + { 0x10007a98, 0x03 }, + { 0x10007a99, 0xc7 }, + { 0x10007a9a, 0x35 }, + { 0x10007a9b, 0x54 }, + { 0x10007a9c, 0x13 }, + { 0x10007a9d, 0x77 }, + { 0x10007a9e, 0xf7 }, + { 0x10007a9f, 0x0f }, + { 0x10007aa0, 0x13 }, + { 0x10007aa1, 0x57 }, + { 0x10007aa2, 0x27 }, + { 0x10007aa3, 0x00 }, + { 0x10007aa4, 0x13 }, + { 0x10007aa5, 0x47 }, + { 0x10007aa6, 0x17 }, + { 0x10007aa7, 0x00 }, + { 0x10007aa8, 0x13 }, + { 0x10007aa9, 0x77 }, + { 0x10007aaa, 0x17 }, + { 0x10007aab, 0x00 }, + { 0x10007aac, 0x23 }, + { 0x10007aad, 0x84 }, + { 0x10007aae, 0xe1 }, + { 0x10007aaf, 0x42 }, + { 0x10007ab0, 0x63 }, + { 0x10007ab1, 0x84 }, + { 0x10007ab2, 0x07 }, + { 0x10007ab3, 0x00 }, + { 0x10007ab4, 0xe3 }, + { 0x10007ab5, 0x94 }, + { 0x10007ab6, 0x06 }, + { 0x10007ab7, 0xee }, + { 0x10007ab8, 0xef }, + { 0x10007ab9, 0x90 }, + { 0x10007aba, 0x0f }, + { 0x10007abb, 0x86 }, + { 0x10007abc, 0xef }, + { 0x10007abd, 0xd0 }, + { 0x10007abe, 0x0f }, + { 0x10007abf, 0x97 }, + { 0x10007ac0, 0x37 }, + { 0x10007ac1, 0x15 }, + { 0x10007ac2, 0x00 }, + { 0x10007ac3, 0x00 }, + { 0x10007ac4, 0x13 }, + { 0x10007ac5, 0x05 }, + { 0x10007ac6, 0x85 }, + { 0x10007ac7, 0xbb }, + { 0x10007ac8, 0x93 }, + { 0x10007ac9, 0x05 }, + { 0x10007aca, 0x00 }, + { 0x10007acb, 0x00 }, + { 0x10007acc, 0xef }, + { 0x10007acd, 0xd0 }, + { 0x10007ace, 0x9f }, + { 0x10007acf, 0xf5 }, + { 0x10007ad0, 0xb7 }, + { 0x10007ad1, 0xd7 }, + { 0x10007ad2, 0x00 }, + { 0x10007ad3, 0x00 }, + { 0x10007ad4, 0x83 }, + { 0x10007ad5, 0xc7 }, + { 0x10007ad6, 0x07 }, + { 0x10007ad7, 0x47 }, + { 0x10007ad8, 0x93 }, + { 0x10007ad9, 0xf7 }, + { 0x10007ada, 0x47 }, + { 0x10007adb, 0x00 }, + { 0x10007adc, 0xe3 }, + { 0x10007add, 0x80 }, + { 0x10007ade, 0x07 }, + { 0x10007adf, 0xec }, + { 0x10007ae0, 0xef }, + { 0x10007ae1, 0x80 }, + { 0x10007ae2, 0xdf }, + { 0x10007ae3, 0xf4 }, + { 0x10007ae4, 0x23 }, + { 0x10007ae5, 0x89 }, + { 0x10007ae6, 0xa1 }, + { 0x10007ae7, 0x40 }, + { 0x10007ae8, 0x6f }, + { 0x10007ae9, 0xf0 }, + { 0x10007aea, 0x5f }, + { 0x10007aeb, 0xeb }, + { 0x10007aec, 0x00 }, + { 0x10007aed, 0x00 }, + { 0x10007aee, 0x00 }, + { 0x10007aef, 0x00 }, + { 0x3fc2bf83, 0x00 }, + { 0x3fc2bf82, 0x00 }, + { 0x3fc2bf81, 0x00 }, + { 0x3fc2bf80, 0x00 }, + { 0x3fc2bfc7, 0x00 }, + { 0x3fc2bfc6, 0x00 }, + { 0x3fc2bfc5, 0x00 }, + { 0x3fc2bfc4, 0x00 }, + { 0x3fc2bfc3, 0x00 }, + { 0x3fc2bfc2, 0x00 }, + { 0x3fc2bfc1, 0x00 }, + { 0x3fc2bfc0, 0x03 }, + { 0x0000d486, 0x43 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x00 }, + { 0x1000db00, 0x04 }, + { 0x1000db01, 0x00 }, + { 0x1000db02, 0x11 }, + { 0x1000db03, 0x00 }, + { 0x1000db04, 0x00 }, + { 0x1000db05, 0x82 }, + { 0x1000db06, 0x04 }, + { 0x1000db07, 0xf1 }, + { 0x1000db08, 0x00 }, + { 0x1000db09, 0x00 }, + { 0x1000db0a, 0x40 }, + { 0x1000db0b, 0x02 }, + { 0x1000db0c, 0xf2 }, + { 0x1000db0d, 0x00 }, + { 0x1000db0e, 0x00 }, + { 0x1000db0f, 0xe0 }, + { 0x1000db10, 0x00 }, + { 0x1000db11, 0x10 }, + { 0x1000db12, 0x00 }, + { 0x1000db13, 0x00 }, + { 0x1000db14, 0x45 }, + { 0x0000d540, 0x01 }, + { 0x0000c081, 0xfc }, + { 0x0000f01e, 0x80 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, +}; + /* * The 'patch code' is written to the patch code area. * The patch code area is used for SDCA register expansion flexibility. @@ -1418,12 +3439,13 @@ static const struct reg_sequence rt1320_patch_code_write[] = { static const struct reg_default rt1320_reg_defaults[] = { { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_FU21, RT1320_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_FU21, RT1320_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE27, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PPU21, RT1320_SDCA_CTL_POSTURE_NUMBER, 0), 0x00 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS113, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS14, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x0b }, { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS21, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, - { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE27, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, - { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_ACTUAL_POWER_STATE, 0), 0x03 }, }; static const struct reg_default rt1320_mbq_defaults[] = { @@ -1484,21 +3506,36 @@ static bool rt1320_readable_register(struct device *dev, unsigned int reg) case 0xde00 ... 0xde09: case 0xdf00 ... 0xdf1b: case 0xe000 ... 0xe847: + case 0xf01e: case 0xf717 ... 0xf719: case 0xf720 ... 0xf723: + case 0x1000cd91 ... 0x1000cd96: case 0x1000f008: + case 0x1000f021: case 0x3fe2e000 ... 0x3fe2e003: - case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT0, RT1320_SDCA_CTL_FUNC_STATUS, 0): + case 0x3fc2ab80 ... 0x3fc2abd4: + /* 0x41000189/0x4100018a */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_FU21, RT1320_SDCA_CTL_FU_MUTE, CH_01): case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_FU21, RT1320_SDCA_CTL_FU_MUTE, CH_02): - case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PPU21, RT1320_SDCA_CTL_POSTURE_NUMBER, 0): - case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0): + /* 0x41001388 */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE27, RT1320_SDCA_CTL_REQ_POWER_STATE, 0): + /* 0x41001988 */ + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0): + /* 0x41080000 */ + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT0, RT1320_SDCA_CTL_FUNC_STATUS, 0): + /* 0x41080200 */ + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PPU21, RT1320_SDCA_CTL_POSTURE_NUMBER, 0): + /* 0x41080900 */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS113, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0): + /* 0x41080980 */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS14, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0): + /* 0x41081080 */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_CS21, RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX, 0): + /* 0x41081480/0x41081488 */ case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_SAPU, RT1320_SDCA_CTL_SAPU_PROTECTION_MODE, 0): case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_SAPU, RT1320_SDCA_CTL_SAPU_PROTECTION_STATUS, 0): + /* 0x41081980 */ + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_ACTUAL_POWER_STATE, 0): return true; default: return false; @@ -1508,6 +3545,9 @@ static bool rt1320_readable_register(struct device *dev, unsigned int reg) static bool rt1320_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { + case 0xc000: + case 0xc003: + case 0xc081: case 0xc402 ... 0xc406: case 0xc48c ... 0xc48f: case 0xc560: @@ -1545,16 +3585,21 @@ static bool rt1320_volatile_register(struct device *dev, unsigned int reg) case 0xde02: case 0xdf14 ... 0xdf1b: case 0xe83c ... 0xe847: + case 0xf01e: case 0xf717 ... 0xf719: case 0xf720 ... 0xf723: case 0x10000000 ... 0x10007fff: case 0x1000c000 ... 0x1000dfff: case 0x1000f008: - case 0x3fc2bfc4 ... 0x3fc2bfc7: + case 0x1000f021: + case 0x3fc2ab80 ... 0x3fc2abd4: + case 0x3fc2bf80 ... 0x3fc2bf83: + case 0x3fc2bfc0 ... 0x3fc2bfc7: case 0x3fe2e000 ... 0x3fe2e003: case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT0, RT1320_SDCA_CTL_FUNC_STATUS, 0): case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_SAPU, RT1320_SDCA_CTL_SAPU_PROTECTION_MODE, 0): case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_SAPU, RT1320_SDCA_CTL_SAPU_PROTECTION_STATUS, 0): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_ACTUAL_POWER_STATE, 0): return true; default: return false; @@ -1577,7 +3622,7 @@ static const struct regmap_config rt1320_sdw_regmap = { .val_bits = 8, .readable_reg = rt1320_readable_register, .volatile_reg = rt1320_volatile_register, - .max_register = 0x41081488, + .max_register = 0x41081980, .reg_defaults = rt1320_reg_defaults, .num_reg_defaults = ARRAY_SIZE(rt1320_reg_defaults), .cache_type = REGCACHE_MAPLE, @@ -1663,6 +3708,63 @@ static int rt1320_read_prop(struct sdw_slave *slave) return 0; } +static int rt1320_pde_transition_delay(struct rt1320_sdw_priv *rt1320, unsigned char ps) +{ + unsigned int delay = 1000, val; + + pm_runtime_mark_last_busy(&rt1320->sdw_slave->dev); + + /* waiting for Actual PDE becomes to PS0/PS3 */ + while (delay) { + regmap_read(rt1320->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, + RT1320_SDCA_CTL_ACTUAL_POWER_STATE, 0), &val); + if (val == ps) + break; + + usleep_range(1000, 1500); + delay--; + } + if (!delay) { + dev_warn(&rt1320->sdw_slave->dev, "%s PDE to %s is NOT ready", __func__, ps?"PS3":"PS0"); + return -ETIMEDOUT; + } + + return 0; +} + +static void rt1320_vc_preset(struct rt1320_sdw_priv *rt1320) +{ + struct sdw_slave *slave = rt1320->sdw_slave; + unsigned int i, reg, val, delay, retry, tmp; + + regmap_multi_reg_write(rt1320->regmap, rt1320_vc_blind_write, ARRAY_SIZE(rt1320_vc_blind_write)); + + for (i = 0; i < ARRAY_SIZE(rt1320_vc_patch_code_write); i++) { + reg = rt1320_vc_patch_code_write[i].reg; + val = rt1320_vc_patch_code_write[i].def; + delay = rt1320_vc_patch_code_write[i].delay_us; + + if ((reg == SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0)) && + (val == 0x00)) { + retry = 200; + while (retry) { + regmap_read(rt1320->regmap, RT1320_KR0_INT_READY, &tmp); + dev_dbg(&slave->dev, "%s, RT1320_KR0_INT_READY=0x%x\n", __func__, tmp); + if (tmp == 0x1f) + break; + usleep_range(1000, 1500); + retry--; + } + if (!retry) + dev_warn(&slave->dev, "%s MCU is NOT ready!", __func__); + } + regmap_write(rt1320->regmap, reg, val); + if (delay) + usleep_range(delay, delay + 1000); + } +} + static int rt1320_io_init(struct device *dev, struct sdw_slave *slave) { struct rt1320_sdw_priv *rt1320 = dev_get_drvdata(dev); @@ -1696,16 +3798,20 @@ static int rt1320_io_init(struct device *dev, struct sdw_slave *slave) dev_dbg(dev, "%s amp func_status=0x%x\n", __func__, amp_func_status); /* initialization write */ - if ((amp_func_status & FUNCTION_NEEDS_INITIALIZATION) || (!rt1320->first_hw_init)) { - regmap_multi_reg_write(rt1320->regmap, rt1320_blind_write, ARRAY_SIZE(rt1320_blind_write)); - regmap_multi_reg_write(rt1320->regmap, rt1320_patch_code_write, - ARRAY_SIZE(rt1320_patch_code_write)); + if ((amp_func_status & FUNCTION_NEEDS_INITIALIZATION)) { + if (rt1320->version_id < RT1320_VC) { + regmap_multi_reg_write(rt1320->regmap, rt1320_blind_write, ARRAY_SIZE(rt1320_blind_write)); + regmap_multi_reg_write(rt1320->regmap, rt1320_patch_code_write, + ARRAY_SIZE(rt1320_patch_code_write)); + } else if (rt1320->version_id == RT1320_VC) { + rt1320_vc_preset(rt1320); + } regmap_write(rt1320->regmap, SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT0, RT1320_SDCA_CTL_FUNC_STATUS, 0), FUNCTION_NEEDS_INITIALIZATION); } - if (!rt1320->first_hw_init) { + if (!rt1320->first_hw_init && rt1320->version_id == RT1320_VA) { regmap_write(rt1320->regmap, SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, RT1320_SDCA_CTL_REQ_POWER_STATE, 0), 0); regmap_read(rt1320->regmap, RT1320_HIFI_VER_0, &val); @@ -1776,14 +3882,14 @@ static int rt1320_pde23_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMU: regmap_write(rt1320->regmap, SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, - RT1320_SDCA_CTL_REQ_POWER_STATE, 0), - ps0); + RT1320_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + rt1320_pde_transition_delay(rt1320, ps0); break; case SND_SOC_DAPM_PRE_PMD: regmap_write(rt1320->regmap, SDW_SDCA_CTL(FUNC_NUM_AMP, RT1320_SDCA_ENT_PDE23, - RT1320_SDCA_CTL_REQ_POWER_STATE, 0), - ps3); + RT1320_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + rt1320_pde_transition_delay(rt1320, ps3); break; default: break; @@ -1799,7 +3905,7 @@ static int rt1320_set_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt1320_sdw_priv *rt1320 = snd_soc_component_get_drvdata(component); - unsigned int read_l, read_r, gain_l_val, gain_r_val; + unsigned int gain_l_val, gain_r_val; unsigned int lvalue, rvalue; const unsigned int interval_offset = 0xc0; @@ -1828,12 +3934,7 @@ static int rt1320_set_gain_put(struct snd_kcontrol *kcontrol, /* Rch */ regmap_write(rt1320->mbq_regmap, mc->rreg, gain_r_val); - regmap_read(rt1320->mbq_regmap, mc->reg, &read_l); - regmap_read(rt1320->mbq_regmap, mc->rreg, &read_r); - if (read_r == gain_r_val && read_l == gain_l_val) - return 1; - - return -EIO; + return 1; } static int rt1320_set_gain_get(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/codecs/rt1320-sdw.h b/sound/soc/codecs/rt1320-sdw.h index b23228e74568..1fbc1fcd71cf 100644 --- a/sound/soc/codecs/rt1320-sdw.h +++ b/sound/soc/codecs/rt1320-sdw.h @@ -18,6 +18,7 @@ #define RT1320_DEV_VERSION_ID_1 0xc404 #define RT1320_KR0_STATUS_CNT 0x1000f008 +#define RT1320_KR0_INT_READY 0x1000f021 #define RT1320_HIFI_VER_0 0x3fe2e000 #define RT1320_HIFI_VER_1 0x3fe2e001 #define RT1320_HIFI_VER_2 0x3fe2e002 @@ -43,6 +44,7 @@ /* RT1320 SDCA control */ #define RT1320_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 #define RT1320_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT1320_SDCA_CTL_ACTUAL_POWER_STATE 0x10 #define RT1320_SDCA_CTL_FU_MUTE 0x01 #define RT1320_SDCA_CTL_FU_VOLUME 0x02 #define RT1320_SDCA_CTL_SAPU_PROTECTION_MODE 0x10 @@ -76,6 +78,7 @@ enum { enum rt1320_version_id { RT1320_VA, RT1320_VB, + RT1320_VC, }; #define RT1320_VER_B_ID 0x07392238 From dc70fd02404398388f3471a57b9f4e26c0eeba5e Mon Sep 17 00:00:00 2001 From: Hongbo Li Date: Sat, 31 Aug 2024 17:51:49 +0800 Subject: [PATCH 290/410] ASoC: adi: Use str_enabled_disabled() helper Use str_enabled_disabled() helper instead of open coding the same. Signed-off-by: Hongbo Li Link: https://patch.msgid.link/20240831095149.4161729-1-lihongbo22@huawei.com Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 7b2563075743..4107eddc9ca8 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -264,8 +264,8 @@ static int axi_i2s_probe(struct platform_device *pdev) goto err_clk_disable; dev_info(&pdev->dev, "probed, capture %s, playback %s\n", - i2s->has_capture ? "enabled" : "disabled", - i2s->has_playback ? "enabled" : "disabled"); + str_enabled_disabled(i2s->has_capture), + str_enabled_disabled(i2s->has_playback)); return 0; From a14e9323267d8f20bdb5a1cebc4abc5abd80cfb2 Mon Sep 17 00:00:00 2001 From: tangbin Date: Tue, 3 Sep 2024 17:03:01 +0800 Subject: [PATCH 291/410] ASoC: loongson: remove unnecessary assignment in i2s_resume() In this function, the assignment ret is unnecessary, thus remove it. Signed-off-by: tangbin Link: https://patch.msgid.link/20240903090301.6192-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_i2s.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c index d45228a3a558..3b9786076501 100644 --- a/sound/soc/loongson/loongson_i2s.c +++ b/sound/soc/loongson/loongson_i2s.c @@ -255,13 +255,10 @@ static int i2s_suspend(struct device *dev) static int i2s_resume(struct device *dev) { struct loongson_i2s *i2s = dev_get_drvdata(dev); - int ret; regcache_cache_only(i2s->regmap, false); regcache_mark_dirty(i2s->regmap); - ret = regcache_sync(i2s->regmap); - - return ret; + return regcache_sync(i2s->regmap); } const struct dev_pm_ops loongson_i2s_pm = { From ea8f615b399988abd801fa52a5eb631d826d0ba4 Mon Sep 17 00:00:00 2001 From: Heiko Stuebner Date: Fri, 30 Aug 2024 22:38:17 +0200 Subject: [PATCH 292/410] ASoC: dt-bindings: realtek,rt5616: document mclk clock Both devicetrees and driver implementation already use the specified mclk in the field, so at least document the clock too, similarly to other Realtek codec. This has the nice additional effect of getting rid of dtbscheck warning. Signed-off-by: Heiko Stuebner Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240830203819.1972536-2-heiko@sntech.de Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/realtek,rt5616.yaml | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml index 248320804e5f..754111f2e70a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -30,6 +30,14 @@ properties: reg: maxItems: 1 + clocks: + items: + - description: Master clock to the CODEC + + clock-names: + items: + - const: mclk + required: - compatible - reg From 92ff90cffbeef9b1a4983017555a194f8bc83f77 Mon Sep 17 00:00:00 2001 From: Heiko Stuebner Date: Fri, 30 Aug 2024 22:38:18 +0200 Subject: [PATCH 293/410] ASoC: dt-bindings: realtek,rt5616: Document audio graph port The codec can be used in conjunction with an audio-graph-card to provide an endpoint for binding with the other side of the audio link. Document the 'port' property that is used for this to prevent dtbscheck errors like: rockchip/rk3588-nanopc-t6-lts.dtb: codec@1b: Unevaluated properties are not allowed ('port' was unexpected) from schema $id: http://devicetree.org/schemas/sound/realtek,rt5616.yaml# Signed-off-by: Heiko Stuebner Reviewed-by: Krzysztof Kozlowski Link: https://patch.msgid.link/20240830203819.1972536-3-heiko@sntech.de Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/realtek,rt5616.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml index 754111f2e70a..29071044c66e 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -38,6 +38,10 @@ properties: items: - const: mclk + port: + $ref: audio-graph-port.yaml# + unevaluatedProperties: false + required: - compatible - reg From cd60dec8994cf0626faf80a67be9350ae335f7e9 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:16 +0530 Subject: [PATCH 294/410] ASoC: amd: acp: Refactor TDM slots selction based on acp revision id Refactor TDM slots selection based on acp revision id. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-2-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 40 +++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 97258b4cf89b..5d1d27078626 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -95,9 +95,11 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas { struct device *dev = dai->component->dev; struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); + struct acp_chip_info *chip; struct acp_stream *stream; int slot_len, no_of_slots; + chip = dev_get_platdata(dev); switch (slot_width) { case SLOT_WIDTH_8: slot_len = 8; @@ -116,15 +118,23 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas return -EINVAL; } - switch (slots) { - case 1 ... 7: - no_of_slots = slots; - break; - case 8: - no_of_slots = 0; + switch (chip->acp_rev) { + case ACP3X_DEV: + case ACP6X_DEV: + switch (slots) { + case 1 ... 7: + no_of_slots = slots; + break; + case 8: + no_of_slots = 0; + break; + default: + dev_err(dev, "Unsupported slots %d\n", slots); + return -EINVAL; + } break; default: - dev_err(dev, "Unsupported slots %d\n", slots); + dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev); return -EINVAL; } @@ -132,12 +142,20 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas spin_lock_irq(&adata->acp_lock); list_for_each_entry(stream, &adata->stream_list, list) { - if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) - adata->tdm_tx_fmt[stream->dai_id - 1] = + switch (chip->acp_rev) { + case ACP3X_DEV: + case ACP6X_DEV: + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 15) | (slot_len << 18); - else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) - adata->tdm_rx_fmt[stream->dai_id - 1] = + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 15) | (slot_len << 18); + break; + default: + dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev); + return -EINVAL; + } } spin_unlock_irq(&adata->acp_lock); return 0; From 093184a3fe443f31a13ffa5d8d6d29ae68a4335d Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:17 +0530 Subject: [PATCH 295/410] ASoC: amd: acp: Refactor I2S dai driver All I2S instances are connected to different powertile form acp6.0 onwards, refactor dai driver to support all I2S instances for all acp platforms. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-3-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 49 ++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 32 +++++------ sound/soc/amd/acp/amd.h | 16 +++--- sound/soc/amd/acp/chip_offset_byte.h | 77 ++++++++++++++------------- 4 files changed, 90 insertions(+), 84 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 5d1d27078626..aca99020120a 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -339,16 +339,16 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { switch (dai->driver->id) { case I2S_BT_INSTANCE: - water_val = ACP_BT_TX_INTR_WATERMARK_SIZE; + water_val = ACP_BT_TX_INTR_WATERMARK_SIZE(adata); reg_val = ACP_BTTDM_ITER; ier_val = ACP_BTTDM_IER; - buf_reg = ACP_BT_TX_RINGBUFSIZE; + buf_reg = ACP_BT_TX_RINGBUFSIZE(adata); break; case I2S_SP_INSTANCE: - water_val = ACP_I2S_TX_INTR_WATERMARK_SIZE; + water_val = ACP_I2S_TX_INTR_WATERMARK_SIZE(adata); reg_val = ACP_I2STDM_ITER; ier_val = ACP_I2STDM_IER; - buf_reg = ACP_I2S_TX_RINGBUFSIZE; + buf_reg = ACP_I2S_TX_RINGBUFSIZE(adata); break; case I2S_HS_INSTANCE: water_val = ACP_HS_TX_INTR_WATERMARK_SIZE; @@ -363,16 +363,16 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct } else { switch (dai->driver->id) { case I2S_BT_INSTANCE: - water_val = ACP_BT_RX_INTR_WATERMARK_SIZE; + water_val = ACP_BT_RX_INTR_WATERMARK_SIZE(adata); reg_val = ACP_BTTDM_IRER; ier_val = ACP_BTTDM_IER; - buf_reg = ACP_BT_RX_RINGBUFSIZE; + buf_reg = ACP_BT_RX_RINGBUFSIZE(adata); break; case I2S_SP_INSTANCE: - water_val = ACP_I2S_RX_INTR_WATERMARK_SIZE; + water_val = ACP_I2S_RX_INTR_WATERMARK_SIZE(adata); reg_val = ACP_I2STDM_IRER; ier_val = ACP_I2STDM_IER; - buf_reg = ACP_I2S_RX_RINGBUFSIZE; + buf_reg = ACP_I2S_RX_RINGBUFSIZE(adata); break; case I2S_HS_INSTANCE: water_val = ACP_HS_RX_INTR_WATERMARK_SIZE; @@ -385,6 +385,7 @@ static int acp_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct return -EINVAL; } } + writel(period_bytes, adata->acp_base + water_val); writel(buf_size, adata->acp_base + buf_reg); if (rsrc->soc_mclk) @@ -463,43 +464,43 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - reg_dma_size = ACP_I2S_TX_DMA_SIZE; + reg_dma_size = ACP_I2S_TX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + SP_PB_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_I2S_TX_FIFOADDR; - reg_fifo_size = ACP_I2S_TX_FIFOSIZE; + reg_fifo_addr = ACP_I2S_TX_FIFOADDR(adata); + reg_fifo_size = ACP_I2S_TX_FIFOSIZE(adata); phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR(adata)); } else { - reg_dma_size = ACP_I2S_RX_DMA_SIZE; + reg_dma_size = ACP_I2S_RX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + SP_CAPT_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_I2S_RX_FIFOADDR; - reg_fifo_size = ACP_I2S_RX_FIFOSIZE; + reg_fifo_addr = ACP_I2S_RX_FIFOADDR(adata); + reg_fifo_size = ACP_I2S_RX_FIFOSIZE(adata); phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_I2S_RX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_I2S_RX_RINGBUFADDR(adata)); } break; case I2S_BT_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - reg_dma_size = ACP_BT_TX_DMA_SIZE; + reg_dma_size = ACP_BT_TX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + BT_PB_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_BT_TX_FIFOADDR; - reg_fifo_size = ACP_BT_TX_FIFOSIZE; + reg_fifo_addr = ACP_BT_TX_FIFOADDR(adata); + reg_fifo_size = ACP_BT_TX_FIFOSIZE(adata); phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR(adata)); } else { - reg_dma_size = ACP_BT_RX_DMA_SIZE; + reg_dma_size = ACP_BT_RX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + BT_CAPT_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_BT_RX_FIFOADDR; - reg_fifo_size = ACP_BT_RX_FIFOSIZE; + reg_fifo_addr = ACP_BT_RX_FIFOADDR(adata); + reg_fifo_size = ACP_BT_RX_FIFOSIZE(adata); phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR(adata)); } break; case I2S_HS_INSTANCE: diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 3cc083fac837..be01b178172e 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -113,40 +113,40 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - reg_dma_size = ACP_I2S_TX_DMA_SIZE; + reg_dma_size = ACP_I2S_TX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + SP_PB_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_I2S_TX_FIFOADDR; - reg_fifo_size = ACP_I2S_TX_FIFOSIZE; + reg_fifo_addr = ACP_I2S_TX_FIFOADDR(adata); + reg_fifo_size = ACP_I2S_TX_FIFOSIZE(adata); phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR(adata)); } else { - reg_dma_size = ACP_I2S_RX_DMA_SIZE; + reg_dma_size = ACP_I2S_RX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + SP_CAPT_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_I2S_RX_FIFOADDR; - reg_fifo_size = ACP_I2S_RX_FIFOSIZE; + reg_fifo_addr = ACP_I2S_RX_FIFOADDR(adata); + reg_fifo_size = ACP_I2S_RX_FIFOSIZE(adata); phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_I2S_RX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_I2S_RX_RINGBUFADDR(adata)); } break; case I2S_BT_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - reg_dma_size = ACP_BT_TX_DMA_SIZE; + reg_dma_size = ACP_BT_TX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + BT_PB_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_BT_TX_FIFOADDR; - reg_fifo_size = ACP_BT_TX_FIFOSIZE; + reg_fifo_addr = ACP_BT_TX_FIFOADDR(adata); + reg_fifo_size = ACP_BT_TX_FIFOSIZE(adata); phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR(adata)); } else { - reg_dma_size = ACP_BT_RX_DMA_SIZE; + reg_dma_size = ACP_BT_RX_DMA_SIZE(adata); acp_fifo_addr = rsrc->sram_pte_offset + BT_CAPT_FIFO_ADDR_OFFSET; - reg_fifo_addr = ACP_BT_RX_FIFOADDR; - reg_fifo_size = ACP_BT_RX_FIFOSIZE; + reg_fifo_addr = ACP_BT_RX_FIFOADDR(adata); + reg_fifo_size = ACP_BT_RX_FIFOSIZE(adata); phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; - writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR); + writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR(adata)); } break; case I2S_HS_INSTANCE: diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 4f0cd7dd04e5..9fb4e234d518 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -259,12 +259,12 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int if (direction == SNDRV_PCM_STREAM_PLAYBACK) { switch (dai_id) { case I2S_BT_INSTANCE: - high = readl(adata->acp_base + ACP_BT_TX_LINEARPOSITIONCNTR_HIGH); - low = readl(adata->acp_base + ACP_BT_TX_LINEARPOSITIONCNTR_LOW); + high = readl(adata->acp_base + ACP_BT_TX_LINEARPOSITIONCNTR_HIGH(adata)); + low = readl(adata->acp_base + ACP_BT_TX_LINEARPOSITIONCNTR_LOW(adata)); break; case I2S_SP_INSTANCE: - high = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_HIGH); - low = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_LOW); + high = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_HIGH(adata)); + low = readl(adata->acp_base + ACP_I2S_TX_LINEARPOSITIONCNTR_LOW(adata)); break; case I2S_HS_INSTANCE: high = readl(adata->acp_base + ACP_HS_TX_LINEARPOSITIONCNTR_HIGH); @@ -277,12 +277,12 @@ static inline u64 acp_get_byte_count(struct acp_dev_data *adata, int dai_id, int } else { switch (dai_id) { case I2S_BT_INSTANCE: - high = readl(adata->acp_base + ACP_BT_RX_LINEARPOSITIONCNTR_HIGH); - low = readl(adata->acp_base + ACP_BT_RX_LINEARPOSITIONCNTR_LOW); + high = readl(adata->acp_base + ACP_BT_RX_LINEARPOSITIONCNTR_HIGH(adata)); + low = readl(adata->acp_base + ACP_BT_RX_LINEARPOSITIONCNTR_LOW(adata)); break; case I2S_SP_INSTANCE: - high = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH); - low = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_LOW); + high = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH(adata)); + low = readl(adata->acp_base + ACP_I2S_RX_LINEARPOSITIONCNTR_LOW(adata)); break; case I2S_HS_INSTANCE: high = readl(adata->acp_base + ACP_HS_RX_LINEARPOSITIONCNTR_HIGH); diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index 18da734c0e9e..97b8b49f1e64 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -32,42 +32,47 @@ /* Registers from ACP_AUDIO_BUFFERS block */ -#define ACP_I2S_RX_RINGBUFADDR 0x2000 -#define ACP_I2S_RX_RINGBUFSIZE 0x2004 -#define ACP_I2S_RX_LINKPOSITIONCNTR 0x2008 -#define ACP_I2S_RX_FIFOADDR 0x200C -#define ACP_I2S_RX_FIFOSIZE 0x2010 -#define ACP_I2S_RX_DMA_SIZE 0x2014 -#define ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH 0x2018 -#define ACP_I2S_RX_LINEARPOSITIONCNTR_LOW 0x201C -#define ACP_I2S_RX_INTR_WATERMARK_SIZE 0x2020 -#define ACP_I2S_TX_RINGBUFADDR 0x2024 -#define ACP_I2S_TX_RINGBUFSIZE 0x2028 -#define ACP_I2S_TX_LINKPOSITIONCNTR 0x202C -#define ACP_I2S_TX_FIFOADDR 0x2030 -#define ACP_I2S_TX_FIFOSIZE 0x2034 -#define ACP_I2S_TX_DMA_SIZE 0x2038 -#define ACP_I2S_TX_LINEARPOSITIONCNTR_HIGH 0x203C -#define ACP_I2S_TX_LINEARPOSITIONCNTR_LOW 0x2040 -#define ACP_I2S_TX_INTR_WATERMARK_SIZE 0x2044 -#define ACP_BT_RX_RINGBUFADDR 0x2048 -#define ACP_BT_RX_RINGBUFSIZE 0x204C -#define ACP_BT_RX_LINKPOSITIONCNTR 0x2050 -#define ACP_BT_RX_FIFOADDR 0x2054 -#define ACP_BT_RX_FIFOSIZE 0x2058 -#define ACP_BT_RX_DMA_SIZE 0x205C -#define ACP_BT_RX_LINEARPOSITIONCNTR_HIGH 0x2060 -#define ACP_BT_RX_LINEARPOSITIONCNTR_LOW 0x2064 -#define ACP_BT_RX_INTR_WATERMARK_SIZE 0x2068 -#define ACP_BT_TX_RINGBUFADDR 0x206C -#define ACP_BT_TX_RINGBUFSIZE 0x2070 -#define ACP_BT_TX_LINKPOSITIONCNTR 0x2074 -#define ACP_BT_TX_FIFOADDR 0x2078 -#define ACP_BT_TX_FIFOSIZE 0x207C -#define ACP_BT_TX_DMA_SIZE 0x2080 -#define ACP_BT_TX_LINEARPOSITIONCNTR_HIGH 0x2084 -#define ACP_BT_TX_LINEARPOSITIONCNTR_LOW 0x2088 -#define ACP_BT_TX_INTR_WATERMARK_SIZE 0x208C +#define ACP_I2S_REG_ADDR(acp_adata, addr) \ + ((addr) + (acp_adata->rsrc->irqp_used * \ + acp_adata->rsrc->irq_reg_offset)) + +#define ACP_I2S_RX_RINGBUFADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2000) +#define ACP_I2S_RX_RINGBUFSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2004) +#define ACP_I2S_RX_LINKPOSITIONCNTR(adata) ACP_I2S_REG_ADDR(adata, 0x2008) +#define ACP_I2S_RX_FIFOADDR(adata) ACP_I2S_REG_ADDR(adata, 0x200C) +#define ACP_I2S_RX_FIFOSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2010) +#define ACP_I2S_RX_DMA_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2014) +#define ACP_I2S_RX_LINEARPOSITIONCNTR_HIGH(adata) ACP_I2S_REG_ADDR(adata, 0x2018) +#define ACP_I2S_RX_LINEARPOSITIONCNTR_LOW(adata) ACP_I2S_REG_ADDR(adata, 0x201C) +#define ACP_I2S_RX_INTR_WATERMARK_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2020) +#define ACP_I2S_TX_RINGBUFADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2024) +#define ACP_I2S_TX_RINGBUFSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2028) +#define ACP_I2S_TX_LINKPOSITIONCNTR(adata) ACP_I2S_REG_ADDR(adata, 0x202C) +#define ACP_I2S_TX_FIFOADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2030) +#define ACP_I2S_TX_FIFOSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2034) +#define ACP_I2S_TX_DMA_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2038) +#define ACP_I2S_TX_LINEARPOSITIONCNTR_HIGH(adata) ACP_I2S_REG_ADDR(adata, 0x203C) +#define ACP_I2S_TX_LINEARPOSITIONCNTR_LOW(adata) ACP_I2S_REG_ADDR(adata, 0x2040) +#define ACP_I2S_TX_INTR_WATERMARK_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2044) +#define ACP_BT_RX_RINGBUFADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2048) +#define ACP_BT_RX_RINGBUFSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x204C) +#define ACP_BT_RX_LINKPOSITIONCNTR(adata) ACP_I2S_REG_ADDR(adata, 0x2050) +#define ACP_BT_RX_FIFOADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2054) +#define ACP_BT_RX_FIFOSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2058) +#define ACP_BT_RX_DMA_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x205C) +#define ACP_BT_RX_LINEARPOSITIONCNTR_HIGH(adata) ACP_I2S_REG_ADDR(adata, 0x2060) +#define ACP_BT_RX_LINEARPOSITIONCNTR_LOW(adata) ACP_I2S_REG_ADDR(adata, 0x2064) +#define ACP_BT_RX_INTR_WATERMARK_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2068) +#define ACP_BT_TX_RINGBUFADDR(adata) ACP_I2S_REG_ADDR(adata, 0x206C) +#define ACP_BT_TX_RINGBUFSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2070) +#define ACP_BT_TX_LINKPOSITIONCNTR(adata) ACP_I2S_REG_ADDR(adata, 0x2074) +#define ACP_BT_TX_FIFOADDR(adata) ACP_I2S_REG_ADDR(adata, 0x2078) +#define ACP_BT_TX_FIFOSIZE(adata) ACP_I2S_REG_ADDR(adata, 0x207C) +#define ACP_BT_TX_DMA_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x2080) +#define ACP_BT_TX_LINEARPOSITIONCNTR_HIGH(adata) ACP_I2S_REG_ADDR(adata, 0x2084) +#define ACP_BT_TX_LINEARPOSITIONCNTR_LOW(adata) ACP_I2S_REG_ADDR(adata, 0x2088) +#define ACP_BT_TX_INTR_WATERMARK_SIZE(adata) ACP_I2S_REG_ADDR(adata, 0x208C) + #define ACP_HS_RX_RINGBUFADDR 0x3A90 #define ACP_HS_RX_RINGBUFSIZE 0x3A94 #define ACP_HS_RX_LINKPOSITIONCNTR 0x3A98 From 973e9edea93943924638040f9c6eb849163bd421 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:18 +0530 Subject: [PATCH 296/410] ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform Update pcm hardware capabilities based on acp revision id. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-4-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-platform.c | 61 +++++++++++++++++++++++++++++--- 1 file changed, 56 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 4f409cd09c11..238b4f648f44 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -68,6 +68,46 @@ static const struct snd_pcm_hardware acp_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +static const struct snd_pcm_hardware acp6x_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 32, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp6x_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 32, + .rates = SNDRV_PCM_RATE_8000_192000, + .rate_min = 8000, + .rate_max = 192000, + .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + int acp_machine_select(struct acp_dev_data *adata) { struct snd_soc_acpi_mach *mach; @@ -183,6 +223,7 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_chip_info *chip; struct acp_stream *stream; int ret; @@ -191,11 +232,21 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = acp_pcm_hardware_playback; - else - runtime->hw = acp_pcm_hardware_capture; + chip = dev_get_platdata(dev); + switch (chip->acp_rev) { + case ACP63_DEV: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp6x_pcm_hardware_playback; + else + runtime->hw = acp6x_pcm_hardware_capture; + break; + default: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + runtime->hw = acp_pcm_hardware_playback; + else + runtime->hw = acp_pcm_hardware_capture; + break; + } ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, DMA_SIZE); if (ret) { From 13aeb56e6dacaa392cde42df86096dfdbeef5ac4 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:19 +0530 Subject: [PATCH 297/410] ASoC: amd: acp: Add I2S TDM support for acp6.3 platform Add slots selection and 32-channels TDM support for acp6.3 platform. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-5-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 56 +++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index aca99020120a..eceffc69bf3e 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -133,6 +133,19 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas return -EINVAL; } break; + case ACP63_DEV: + switch (slots) { + case 1 ... 31: + no_of_slots = slots; + break; + case 32: + no_of_slots = 0; + break; + default: + dev_err(dev, "Unsupported slots %d\n", slots); + return -EINVAL; + } + break; default: dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev); return -EINVAL; @@ -152,6 +165,14 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas adata->tdm_rx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 15) | (slot_len << 18); break; + case ACP63_DEV: + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 13) | (slot_len << 18); + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 13) | (slot_len << 18); + break; default: dev_err(dev, "Unknown chip revision %d\n", chip->acp_rev); return -EINVAL; @@ -314,6 +335,41 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ default: return -EINVAL; } + + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 48000: + case 96000: + case 192000: + switch (params_channels(params)) { + case 2: + break; + case 4: + bclk_div_val = bclk_div_val >> 1; + lrclk_div_val = lrclk_div_val << 1; + break; + case 8: + bclk_div_val = bclk_div_val >> 2; + lrclk_div_val = lrclk_div_val << 2; + break; + case 16: + bclk_div_val = bclk_div_val >> 3; + lrclk_div_val = lrclk_div_val << 3; + break; + case 32: + bclk_div_val = bclk_div_val >> 4; + lrclk_div_val = lrclk_div_val << 4; + break; + default: + dev_err(dev, "Unsupported channels %#x\n", + params_channels(params)); + } + break; + default: + break; + } adata->lrclk_div = lrclk_div_val; adata->bclk_div = bclk_div_val; } From 7a040cc5579708a327b2d82caa26ab9a02623343 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:20 +0530 Subject: [PATCH 298/410] ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform Update pcm hardware capabilities for acp7.0 platform. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-6-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 238b4f648f44..55573bdd5ef4 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -235,6 +235,7 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs chip = dev_get_platdata(dev); switch (chip->acp_rev) { case ACP63_DEV: + case ACP70_DEV: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp6x_pcm_hardware_playback; else From fb2eaec6a38d853437c4a83b0c7168ef77536094 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:21 +0530 Subject: [PATCH 299/410] ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform Add I2S master clock generation support for acp7.0 platforms. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-7-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp70.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 81636faa22cd..a9be059889b7 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -25,6 +25,9 @@ #define DRV_NAME "acp_asoc_acp70" +#define CLK7_CLK0_DFS_CNTL_N1 0X0006C1A4 +#define CLK0_DIVIDER 0X19 + static struct acp_resource rsrc = { .offset = 0, .no_of_ctrls = 2, @@ -134,12 +137,27 @@ static struct snd_soc_dai_driver acp70_dai[] = { }, }; +static int acp70_i2s_master_clock_generate(struct acp_dev_data *adata) +{ + struct pci_dev *smn_dev; + + smn_dev = pci_get_device(PCI_VENDOR_ID_AMD, 0x1507, NULL); + if (!smn_dev) + return -ENODEV; + + /* Set clk7 DFS clock divider register value to get mclk as 196.608MHz*/ + smn_write(smn_dev, CLK7_CLK0_DFS_CNTL_N1, CLK0_DIVIDER); + + return 0; +} + static int acp_acp70_audio_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct acp_chip_info *chip; struct acp_dev_data *adata; struct resource *res; + int ret; chip = dev_get_platdata(&pdev->dev); if (!chip || !chip->base) { @@ -191,6 +209,12 @@ static int acp_acp70_audio_probe(struct platform_device *pdev) acp_machine_select(adata); dev_set_drvdata(dev, adata); + + ret = acp70_i2s_master_clock_generate(adata); + if (ret) { + dev_err(&pdev->dev, "Failed to set I2S master clock as 196.608MHz\n"); + return ret; + } acp_enable_interrupts(adata); acp_platform_register(dev); pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); From 13073ed06a9f0234033d31e0ccba1e3167d500a3 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:22 +0530 Subject: [PATCH 300/410] ASoC: amd: acp: Set i2s clock for acp7.0 platform Set i2s bclk and lrclk for acp7.0 platform. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-8-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index eceffc69bf3e..31475a4dbced 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -60,6 +60,7 @@ static inline void acp_set_i2s_clk(struct acp_dev_data *adata, int dai_id) switch (chip->acp_rev) { case ACP63_DEV: + case ACP70_DEV: val |= FIELD_PREP(ACP63_LRCLK_DIV_FIELD, adata->lrclk_div); val |= FIELD_PREP(ACP63_BCLK_DIV_FIELD, adata->bclk_div); break; From b24df4fa40cc236ccea3de74d9cdb25c2906013b Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:23 +0530 Subject: [PATCH 301/410] ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver Modify max channels and max sample rate support in the dai driver for acp7.0 platform. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-9-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp70.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index a9be059889b7..a662c7db7506 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -52,23 +52,23 @@ static struct snd_soc_dai_driver acp70_dai[] = { .id = I2S_SP_INSTANCE, .playback = { .stream_name = "I2S SP Playback", - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 8, + .channels_max = 32, .rate_min = 8000, - .rate_max = 96000, + .rate_max = 192000, }, .capture = { .stream_name = "I2S SP Capture", - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 2, + .channels_max = 32, .rate_min = 8000, - .rate_max = 48000, + .rate_max = 192000, }, .ops = &asoc_acp_cpu_dai_ops, }, @@ -77,23 +77,23 @@ static struct snd_soc_dai_driver acp70_dai[] = { .id = I2S_BT_INSTANCE, .playback = { .stream_name = "I2S BT Playback", - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 8, + .channels_max = 32, .rate_min = 8000, - .rate_max = 96000, + .rate_max = 192000, }, .capture = { .stream_name = "I2S BT Capture", - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 2, + .channels_max = 32, .rate_min = 8000, - .rate_max = 48000, + .rate_max = 192000, }, .ops = &asoc_acp_cpu_dai_ops, }, @@ -102,23 +102,23 @@ static struct snd_soc_dai_driver acp70_dai[] = { .id = I2S_HS_INSTANCE, .playback = { .stream_name = "I2S HS Playback", - .rates = SNDRV_PCM_RATE_8000_96000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 8, + .channels_max = 32, .rate_min = 8000, - .rate_max = 96000, + .rate_max = 192000, }, .capture = { .stream_name = "I2S HS Capture", - .rates = SNDRV_PCM_RATE_8000_48000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .channels_min = 2, - .channels_max = 8, + .channels_max = 32, .rate_min = 8000, - .rate_max = 48000, + .rate_max = 192000, }, .ops = &asoc_acp_cpu_dai_ops, }, From b80d5a0c875fa0a708be24260d8f160d31273b1b Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:24 +0530 Subject: [PATCH 302/410] ASoC: amd: acp: Add I2S TDM support for acp7.0 platform Add acp70 revision id to support I2S TDM. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-10-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 31475a4dbced..2a23c52eb512 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -135,6 +135,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas } break; case ACP63_DEV: + case ACP70_DEV: switch (slots) { case 1 ... 31: no_of_slots = slots; @@ -167,6 +168,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas FRM_LEN | (slots << 15) | (slot_len << 18); break; case ACP63_DEV: + case ACP70_DEV: if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) adata->tdm_tx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 13) | (slot_len << 18); From f6f7d25b11033bdbf25d800572c003fced21a035 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:25 +0530 Subject: [PATCH 303/410] ASoC: amd: acp: Add pte configuration for ACP7.0 platform Add page table entry configurations to support higher sample rate streams with multiple channels for ACP7.0 platforms. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-11-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 33 +++++++++++++---- sound/soc/amd/acp/acp-pdm.c | 7 +++- sound/soc/amd/acp/acp-platform.c | 53 +++++++++++++++++++++++----- sound/soc/amd/acp/acp70.c | 4 +-- sound/soc/amd/acp/amd.h | 8 +++++ sound/soc/amd/acp/chip_offset_byte.h | 7 ++++ 6 files changed, 94 insertions(+), 18 deletions(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 2a23c52eb512..74f2ad62e596 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -514,12 +514,14 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d { struct device *dev = dai->component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_chip_info *chip; struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size = 0, reg_fifo_size = 0, reg_fifo_addr = 0; u32 phy_addr = 0, acp_fifo_addr = 0, ext_int_ctrl; unsigned int dir = substream->stream; + chip = dev_get_platdata(dev); switch (dai->driver->id) { case I2S_SP_INSTANCE: if (dir == SNDRV_PCM_STREAM_PLAYBACK) { @@ -529,7 +531,10 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d reg_fifo_addr = ACP_I2S_TX_FIFOADDR(adata); reg_fifo_size = ACP_I2S_TX_FIFOSIZE(adata); - phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_SP_TX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_I2S_TX_RINGBUFADDR(adata)); } else { reg_dma_size = ACP_I2S_RX_DMA_SIZE(adata); @@ -537,7 +542,11 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d SP_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_RX_FIFOADDR(adata); reg_fifo_size = ACP_I2S_RX_FIFOSIZE(adata); - phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; + + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_SP_RX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_I2S_RX_RINGBUFADDR(adata)); } break; @@ -549,7 +558,10 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d reg_fifo_addr = ACP_BT_TX_FIFOADDR(adata); reg_fifo_size = ACP_BT_TX_FIFOSIZE(adata); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_BT_TX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_BT_TX_RINGBUFADDR(adata)); } else { reg_dma_size = ACP_BT_RX_DMA_SIZE(adata); @@ -558,7 +570,10 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d reg_fifo_addr = ACP_BT_RX_FIFOADDR(adata); reg_fifo_size = ACP_BT_RX_FIFOSIZE(adata); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_BT_RX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_BT_RX_RINGBUFADDR(adata)); } break; @@ -570,7 +585,10 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d reg_fifo_addr = ACP_HS_TX_FIFOADDR; reg_fifo_size = ACP_HS_TX_FIFOSIZE; - phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_HS_TX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_HS_TX_RINGBUFADDR); } else { reg_dma_size = ACP_HS_RX_DMA_SIZE; @@ -579,7 +597,10 @@ static int acp_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_d reg_fifo_addr = ACP_HS_RX_FIFOADDR; reg_fifo_size = ACP_HS_RX_FIFOSIZE; - phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_DEV) + phy_addr = ACP7x_I2S_HS_RX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, adata->acp_base + ACP_HS_RX_RINGBUFADDR); } break; diff --git a/sound/soc/amd/acp/acp-pdm.c b/sound/soc/amd/acp/acp-pdm.c index bb79269c2fc1..22dd8988d005 100644 --- a/sound/soc/amd/acp/acp-pdm.c +++ b/sound/soc/amd/acp/acp-pdm.c @@ -31,9 +31,11 @@ static int acp_dmic_prepare(struct snd_pcm_substream *substream, struct acp_stream *stream = substream->runtime->private_data; struct device *dev = dai->component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); + struct acp_chip_info *chip; u32 physical_addr, size_dmic, period_bytes; unsigned int dmic_ctrl; + chip = dev_get_platdata(dev); /* Enable default DMIC clk */ writel(PDM_CLK_FREQ_MASK, adata->acp_base + ACP_WOV_CLK_CTRL); dmic_ctrl = readl(adata->acp_base + ACP_WOV_MISC_CTRL); @@ -45,7 +47,10 @@ static int acp_dmic_prepare(struct snd_pcm_substream *substream, size_dmic = frames_to_bytes(substream->runtime, substream->runtime->buffer_size); - physical_addr = stream->reg_offset + MEM_WINDOW_START; + if (chip->acp_rev >= ACP70_DEV) + physical_addr = ACP7x_DMIC_MEM_WINDOW_START; + else + physical_addr = stream->reg_offset + MEM_WINDOW_START; /* Init DMIC Ring buffer */ writel(physical_addr, adata->acp_base + ACP_WOV_RX_RINGBUFADDR); diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 55573bdd5ef4..6ef9baeed74a 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -177,17 +177,20 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream *stream) { struct acp_resource *rsrc = adata->rsrc; - u32 pte_reg, pte_size, reg_val; + u32 reg_val; - /* Use ATU base Group5 */ - pte_reg = ACPAXI2AXI_ATU_BASE_ADDR_GRP_5; - pte_size = ACPAXI2AXI_ATU_PAGE_SIZE_GRP_5; + reg_val = rsrc->sram_pte_offset; stream->reg_offset = 0x02000000; - /* Group Enable */ - reg_val = rsrc->sram_pte_offset; - writel(reg_val | BIT(31), adata->acp_base + pte_reg); - writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + pte_size); + writel((reg_val + GRP1_OFFSET) | BIT(31), adata->acp_base + ACPAXI2AXI_ATU_BASE_ADDR_GRP_1); + writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + ACPAXI2AXI_ATU_PAGE_SIZE_GRP_1); + + writel((reg_val + GRP2_OFFSET) | BIT(31), adata->acp_base + ACPAXI2AXI_ATU_BASE_ADDR_GRP_2); + writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + ACPAXI2AXI_ATU_PAGE_SIZE_GRP_2); + + writel(reg_val | BIT(31), adata->acp_base + ACPAXI2AXI_ATU_BASE_ADDR_GRP_5); + writel(PAGE_SIZE_4K_ENABLE, adata->acp_base + ACPAXI2AXI_ATU_PAGE_SIZE_GRP_5); + writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } EXPORT_SYMBOL_NS_GPL(config_pte_for_stream, SND_SOC_ACP_COMMON); @@ -201,7 +204,39 @@ void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int s u32 low, high, val; u16 page_idx; - val = stream->pte_offset; + switch (adata->platform) { + case ACP70: + switch (stream->dai_id) { + case I2S_SP_INSTANCE: + if (stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + val = 0x0; + else + val = 0x1000; + break; + case I2S_BT_INSTANCE: + if (stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + val = 0x2000; + else + val = 0x3000; + break; + case I2S_HS_INSTANCE: + if (stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + val = 0x4000; + else + val = 0x5000; + break; + case DMIC_INSTANCE: + val = 0x6000; + break; + default: + dev_err(adata->dev, "Invalid dai id %x\n", stream->dai_id); + break; + } + break; + default: + val = stream->pte_offset; + break; + } for (page_idx = 0; page_idx < num_pages; page_idx++) { /* Load the low address of page int ACP SRAM through SRBM */ diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index a662c7db7506..3b3bd15964fc 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -34,8 +34,8 @@ static struct acp_resource rsrc = { .irqp_used = 1, .soc_mclk = true, .irq_reg_offset = 0x1a00, - .scratch_reg_offset = 0x12800, - .sram_pte_offset = 0x03802800, + .scratch_reg_offset = 0x10000, + .sram_pte_offset = 0x03800000, }; static struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_acp_machines[] = { diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 9fb4e234d518..854269fea875 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -62,6 +62,14 @@ #define I2S_HS_TX_MEM_WINDOW_START 0x40A0000 #define I2S_HS_RX_MEM_WINDOW_START 0x40C0000 +#define ACP7x_I2S_SP_TX_MEM_WINDOW_START 0x4000000 +#define ACP7x_I2S_SP_RX_MEM_WINDOW_START 0x4200000 +#define ACP7x_I2S_BT_TX_MEM_WINDOW_START 0x4400000 +#define ACP7x_I2S_BT_RX_MEM_WINDOW_START 0x4600000 +#define ACP7x_I2S_HS_TX_MEM_WINDOW_START 0x4800000 +#define ACP7x_I2S_HS_RX_MEM_WINDOW_START 0x4A00000 +#define ACP7x_DMIC_MEM_WINDOW_START 0x4C00000 + #define SP_PB_FIFO_ADDR_OFFSET 0x500 #define SP_CAPT_FIFO_ADDR_OFFSET 0x700 #define BT_PB_FIFO_ADDR_OFFSET 0x900 diff --git a/sound/soc/amd/acp/chip_offset_byte.h b/sound/soc/amd/acp/chip_offset_byte.h index 97b8b49f1e64..117ea63e85c6 100644 --- a/sound/soc/amd/acp/chip_offset_byte.h +++ b/sound/soc/amd/acp/chip_offset_byte.h @@ -12,9 +12,16 @@ #define _ACP_IP_OFFSET_HEADER #define ACPAXI2AXI_ATU_CTRL 0xC40 +#define ACPAXI2AXI_ATU_PAGE_SIZE_GRP_1 0xC00 +#define ACPAXI2AXI_ATU_BASE_ADDR_GRP_1 0xC04 +#define ACPAXI2AXI_ATU_PAGE_SIZE_GRP_2 0xC08 +#define ACPAXI2AXI_ATU_BASE_ADDR_GRP_2 0xC0C #define ACPAXI2AXI_ATU_PAGE_SIZE_GRP_5 0xC20 #define ACPAXI2AXI_ATU_BASE_ADDR_GRP_5 0xC24 +#define GRP1_OFFSET 0x0 +#define GRP2_OFFSET 0x4000 + #define ACP_PGFSM_CONTROL 0x141C #define ACP_PGFSM_STATUS 0x1420 #define ACP_SOFT_RESET 0x1000 From 1150c18ba35376518b6ed9af3e96c671336fa5c7 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:26 +0530 Subject: [PATCH 304/410] ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform Add i2s master generation support for acp7.1 platform based on pci device id. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-12-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp70.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 3b3bd15964fc..1b0b59a22924 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -140,8 +140,17 @@ static struct snd_soc_dai_driver acp70_dai[] = { static int acp70_i2s_master_clock_generate(struct acp_dev_data *adata) { struct pci_dev *smn_dev; + u32 device_id; + + if (adata->platform == ACP70) + device_id = 0x1507; + else if (adata->platform == ACP71) + device_id = 0x1122; + else + return -ENODEV; + + smn_dev = pci_get_device(PCI_VENDOR_ID_AMD, device_id, NULL); - smn_dev = pci_get_device(PCI_VENDOR_ID_AMD, 0x1507, NULL); if (!smn_dev) return -ENODEV; From 3f600592fa0ca1599326e20aa22e845de5fc8ac5 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Tue, 3 Sep 2024 17:04:27 +0530 Subject: [PATCH 305/410] ASoC: amd: acp: Add I2S TDM support for acp7.1 platform Add acp71 revision id to support i2s/tdm mode. Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20240903113427.182997-13-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-i2s.c | 3 +++ sound/soc/amd/acp/acp-platform.c | 2 ++ 2 files changed, 5 insertions(+) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 74f2ad62e596..56ce9e4b6acc 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -61,6 +61,7 @@ static inline void acp_set_i2s_clk(struct acp_dev_data *adata, int dai_id) switch (chip->acp_rev) { case ACP63_DEV: case ACP70_DEV: + case ACP71_DEV: val |= FIELD_PREP(ACP63_LRCLK_DIV_FIELD, adata->lrclk_div); val |= FIELD_PREP(ACP63_BCLK_DIV_FIELD, adata->bclk_div); break; @@ -136,6 +137,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas break; case ACP63_DEV: case ACP70_DEV: + case ACP71_DEV: switch (slots) { case 1 ... 31: no_of_slots = slots; @@ -169,6 +171,7 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas break; case ACP63_DEV: case ACP70_DEV: + case ACP71_DEV: if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) adata->tdm_tx_fmt[stream->dai_id - 1] = FRM_LEN | (slots << 13) | (slot_len << 18); diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index 6ef9baeed74a..ae63b2e693ab 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -206,6 +206,7 @@ void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int s switch (adata->platform) { case ACP70: + case ACP71: switch (stream->dai_id) { case I2S_SP_INSTANCE: if (stream->dir == SNDRV_PCM_STREAM_PLAYBACK) @@ -271,6 +272,7 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs switch (chip->acp_rev) { case ACP63_DEV: case ACP70_DEV: + case ACP71_DEV: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp6x_pcm_hardware_playback; else From c032044e9672408c534d64a6df2b1ba14449e948 Mon Sep 17 00:00:00 2001 From: Cyan Nyan Date: Tue, 3 Sep 2024 19:52:29 +0900 Subject: [PATCH 306/410] ALSA: usb-audio: Add quirk for RME Digiface USB Add trivial support for audio streaming on the RME Digiface USB. Binds only to the first interface to allow userspace to directly drive the complex I/O and matrix mixer controls. Signed-off-by: Cyan Nyan [Lina: Added 2x/4x sample rate support & boot/format quirks] Co-developed-by: Asahi Lina Signed-off-by: Asahi Lina Link: https://patch.msgid.link/20240903-rme-digiface-v2-1-71b06c912e97@asahilina.net Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 171 ++++++++++++++++++++++++++++++++++++++- sound/usb/quirks.c | 58 +++++++++++++ 2 files changed, 228 insertions(+), 1 deletion(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 8d22de8bc2a9..631b9ab80f6c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3604,6 +3604,175 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, - +{ + /* Only claim interface 0 */ + .match_flags = USB_DEVICE_ID_MATCH_VENDOR | + USB_DEVICE_ID_MATCH_PRODUCT | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_NUMBER, + .idVendor = 0x2a39, + .idProduct = 0x3f8c, + .bInterfaceClass = USB_CLASS_VENDOR_SPEC, + .bInterfaceNumber = 0, + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + /* + * Three modes depending on sample rate band, + * with different channel counts for in/out + */ + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 34, // outputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x02, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 32000, + .rate_max = 48000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 32000, 44100, 48000, + }, + .sync_ep = 0x81, + .sync_iface = 0, + .sync_altsetting = 1, + .sync_ep_idx = 0, + .implicit_fb = 1, + }, + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 18, // outputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x02, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 64000, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 64000, 88200, 96000, + }, + .sync_ep = 0x81, + .sync_iface = 0, + .sync_altsetting = 1, + .sync_ep_idx = 0, + .implicit_fb = 1, + }, + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 10, // outputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x02, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_KNOT | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 128000, + .rate_max = 192000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 128000, 176400, 192000, + }, + .sync_ep = 0x81, + .sync_iface = 0, + .sync_altsetting = 1, + .sync_ep_idx = 0, + .implicit_fb = 1, + }, + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 32, // inputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 32000, + .rate_max = 48000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 32000, 44100, 48000, + } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 16, // inputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 64000, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 64000, 88200, 96000, + } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels = 8, // inputs + .fmt_bits = 24, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_KNOT | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000, + .rate_min = 128000, + .rate_max = 192000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 128000, 176400, 192000, + } + } + }, + QUIRK_COMPOSITE_END + } + } +}, #undef USB_DEVICE_VENDOR_SPEC #undef USB_AUDIO_DEVICE diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 53c69f3069c4..f62631b54e10 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1389,6 +1389,27 @@ static int snd_usb_motu_m_series_boot_quirk(struct usb_device *dev) return 0; } +static int snd_usb_rme_digiface_boot_quirk(struct usb_device *dev) +{ + /* Disable mixer, internal clock, all outputs ADAT, 48kHz, TMS off */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 16, 0x40, 0x2410, 0x7fff, NULL, 0); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 18, 0x40, 0x0104, 0xffff, NULL, 0); + + /* Disable loopback for all inputs */ + for (int ch = 0; ch < 32; ch++) + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 22, 0x40, 0x400, ch, NULL, 0); + + /* Unity gain for all outputs */ + for (int ch = 0; ch < 34; ch++) + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 21, 0x40, 0x9000, 0x100 + ch, NULL, 0); + + return 0; +} + /* * Setup quirks */ @@ -1616,6 +1637,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, get_iface_desc(intf->altsetting)->bInterfaceNumber < 3) return snd_usb_motu_microbookii_boot_quirk(dev); break; + case USB_ID(0x2a39, 0x3f8c): /* RME Digiface USB */ + return snd_usb_rme_digiface_boot_quirk(dev); } return 0; @@ -1771,6 +1794,38 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs, dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); } +static const int rme_digiface_rate_table[] = { + 32000, 44100, 48000, 0, + 64000, 88200, 96000, 0, + 128000, 176400, 192000, 0, +}; + +static int rme_digiface_set_format_quirk(struct snd_usb_substream *subs) +{ + unsigned int cur_rate = subs->data_endpoint->cur_rate; + u16 val; + int speed_mode; + int id; + + for (id = 0; id < ARRAY_SIZE(rme_digiface_rate_table); id++) { + if (rme_digiface_rate_table[id] == cur_rate) + break; + } + + if (id >= ARRAY_SIZE(rme_digiface_rate_table)) + return -EINVAL; + + /* 2, 3, 4 for 1x, 2x, 4x */ + speed_mode = (id >> 2) + 2; + val = (id << 3) | (speed_mode << 12); + + /* Set the sample rate */ + snd_usb_ctl_msg(subs->stream->chip->dev, + usb_sndctrlpipe(subs->stream->chip->dev, 0), + 16, 0x40, val, 0x7078, NULL, 0); + return 0; +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1795,6 +1850,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x0dba, 0x5000): mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ break; + case USB_ID(0x2a39, 0x3f8c): /* RME Digiface USB */ + rme_digiface_set_format_quirk(subs); + break; } } From 611a96f6acf2e74fe28cb90908a9c183862348ce Mon Sep 17 00:00:00 2001 From: Asahi Lina Date: Tue, 3 Sep 2024 19:52:30 +0900 Subject: [PATCH 307/410] ALSA: usb-audio: Add mixer quirk for RME Digiface USB Implement sync, output format, and input status mixer controls, to allow the interface to be used as a straight ADAT/SPDIF (+ Headphones) I/O interface. This does not implement the matrix mixer, output gain controls, or input level meter feedback. The full mixer interface is only really usable using a dedicated userspace control app (there are too many mixer nodes for alsamixer to be usable), so for now we leave it up to userspace to directly control these features using raw USB control messages. This is similar to how it's done with some FireWire interfaces (ffado-mixer). Signed-off-by: Asahi Lina Link: https://patch.msgid.link/20240903-rme-digiface-v2-2-71b06c912e97@asahilina.net Signed-off-by: Takashi Iwai --- sound/usb/mixer_quirks.c | 413 +++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 1 + 2 files changed, 414 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 5d6792f51f4c..2a9594f34dac 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -14,6 +14,7 @@ * Przemek Rudy (prudy1@o2.pl) */ +#include #include #include #include @@ -3087,6 +3088,415 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * RME Digiface USB + */ + +#define RME_DIGIFACE_READ_STATUS 17 +#define RME_DIGIFACE_STATUS_REG0L 0 +#define RME_DIGIFACE_STATUS_REG0H 1 +#define RME_DIGIFACE_STATUS_REG1L 2 +#define RME_DIGIFACE_STATUS_REG1H 3 +#define RME_DIGIFACE_STATUS_REG2L 4 +#define RME_DIGIFACE_STATUS_REG2H 5 +#define RME_DIGIFACE_STATUS_REG3L 6 +#define RME_DIGIFACE_STATUS_REG3H 7 + +#define RME_DIGIFACE_CTL_REG1 16 +#define RME_DIGIFACE_CTL_REG2 18 + +/* Reg is overloaded, 0-7 for status halfwords or 16 or 18 for control registers */ +#define RME_DIGIFACE_REGISTER(reg, mask) (((reg) << 16) | (mask)) +#define RME_DIGIFACE_INVERT BIT(31) + +/* Nonconst helpers */ +#define field_get(_mask, _reg) (((_reg) & (_mask)) >> (ffs(_mask) - 1)) +#define field_prep(_mask, _val) (((_val) << (ffs(_mask) - 1)) & (_mask)) + +static int snd_rme_digiface_write_reg(struct snd_kcontrol *kcontrol, int item, u16 mask, u16 val) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + struct usb_device *dev = chip->dev; + int err; + + err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + item, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + val, mask, NULL, 0); + if (err < 0) + dev_err(&dev->dev, + "unable to issue control set request %d (ret = %d)", + item, err); + return err; +} + +static int snd_rme_digiface_read_status(struct snd_kcontrol *kcontrol, u32 status[4]) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct snd_usb_audio *chip = list->mixer->chip; + struct usb_device *dev = chip->dev; + __le32 buf[4]; + int err; + + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + RME_DIGIFACE_READ_STATUS, + USB_DIR_IN | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + 0, 0, + buf, sizeof(buf)); + if (err < 0) { + dev_err(&dev->dev, + "unable to issue status read request (ret = %d)", + err); + } else { + for (int i = 0; i < ARRAY_SIZE(buf); i++) + status[i] = le32_to_cpu(buf[i]); + } + return err; +} + +static int snd_rme_digiface_get_status_val(struct snd_kcontrol *kcontrol) +{ + int err; + u32 status[4]; + bool invert = kcontrol->private_value & RME_DIGIFACE_INVERT; + u8 reg = (kcontrol->private_value >> 16) & 0xff; + u16 mask = kcontrol->private_value & 0xffff; + u16 val; + + err = snd_rme_digiface_read_status(kcontrol, status); + if (err < 0) + return err; + + switch (reg) { + /* Status register halfwords */ + case RME_DIGIFACE_STATUS_REG0L ... RME_DIGIFACE_STATUS_REG3H: + break; + case RME_DIGIFACE_CTL_REG1: /* Control register 1, present in halfword 3L */ + reg = RME_DIGIFACE_STATUS_REG3L; + break; + case RME_DIGIFACE_CTL_REG2: /* Control register 2, present in halfword 3H */ + reg = RME_DIGIFACE_STATUS_REG3H; + break; + default: + return -EINVAL; + } + + if (reg & 1) + val = status[reg >> 1] >> 16; + else + val = status[reg >> 1] & 0xffff; + + if (invert) + val ^= mask; + + return field_get(mask, val); +} + +static int snd_rme_digiface_rate_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int freq = snd_rme_digiface_get_status_val(kcontrol); + + if (freq < 0) + return freq; + if (freq >= ARRAY_SIZE(snd_rme_rate_table)) + return -EIO; + + ucontrol->value.integer.value[0] = snd_rme_rate_table[freq]; + return 0; +} + +static int snd_rme_digiface_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int val = snd_rme_digiface_get_status_val(kcontrol); + + if (val < 0) + return val; + + ucontrol->value.enumerated.item[0] = val; + return 0; +} + +static int snd_rme_digiface_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + bool invert = kcontrol->private_value & RME_DIGIFACE_INVERT; + u8 reg = (kcontrol->private_value >> 16) & 0xff; + u16 mask = kcontrol->private_value & 0xffff; + u16 val = field_prep(mask, ucontrol->value.enumerated.item[0]); + + if (invert) + val ^= mask; + + return snd_rme_digiface_write_reg(kcontrol, reg, mask, val); +} + +static int snd_rme_digiface_current_sync_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int ret = snd_rme_digiface_enum_get(kcontrol, ucontrol); + + /* 7 means internal for current sync */ + if (ucontrol->value.enumerated.item[0] == 7) + ucontrol->value.enumerated.item[0] = 0; + + return ret; +} + +static int snd_rme_digiface_sync_state_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + u32 status[4]; + int err; + bool valid, sync; + + err = snd_rme_digiface_read_status(kcontrol, status); + if (err < 0) + return err; + + valid = status[0] & BIT(kcontrol->private_value); + sync = status[0] & BIT(5 + kcontrol->private_value); + + if (!valid) + ucontrol->value.enumerated.item[0] = SND_RME_CLOCK_NOLOCK; + else if (!sync) + ucontrol->value.enumerated.item[0] = SND_RME_CLOCK_LOCK; + else + ucontrol->value.enumerated.item[0] = SND_RME_CLOCK_SYNC; + return 0; +} + + +static int snd_rme_digiface_format_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const format[] = { + "ADAT", "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(format), format); +} + + +static int snd_rme_digiface_sync_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const sync_sources[] = { + "Internal", "Input 1", "Input 2", "Input 3", "Input 4" + }; + + return snd_ctl_enum_info(uinfo, 1, + ARRAY_SIZE(sync_sources), sync_sources); +} + +static int snd_rme_digiface_rate_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 200000; + uinfo->value.integer.step = 0; + return 0; +} + +static const struct snd_kcontrol_new snd_rme_digiface_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1 Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_digiface_sync_state_get, + .private_value = 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG0H, BIT(0)) | + RME_DIGIFACE_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 1 Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG1L, GENMASK(3, 0)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 2 Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_digiface_sync_state_get, + .private_value = 1, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 2 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG0L, BIT(13)) | + RME_DIGIFACE_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 2 Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG1L, GENMASK(7, 4)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 3 Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_digiface_sync_state_get, + .private_value = 2, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 3 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG0L, BIT(14)) | + RME_DIGIFACE_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 3 Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG1L, GENMASK(11, 8)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 4 Sync", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_sync_state_info, + .get = snd_rme_digiface_sync_state_get, + .private_value = 3, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 4 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG0L, GENMASK(15, 12)) | + RME_DIGIFACE_INVERT, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input 4 Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG1L, GENMASK(3, 0)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .put = snd_rme_digiface_enum_put, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG2, BIT(0)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 2 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .put = snd_rme_digiface_enum_put, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG2, BIT(1)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .put = snd_rme_digiface_enum_put, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG2, BIT(3)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 4 Format", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_rme_digiface_format_info, + .get = snd_rme_digiface_enum_get, + .put = snd_rme_digiface_enum_put, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG2, BIT(4)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_rme_digiface_sync_source_info, + .get = snd_rme_digiface_enum_get, + .put = snd_rme_digiface_enum_put, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG1, GENMASK(2, 0)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Sync Source", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_digiface_sync_source_info, + .get = snd_rme_digiface_current_sync_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG0L, GENMASK(12, 10)), + }, + { + /* + * This is writeable, but it is only set by the PCM rate. + * Mixer apps currently need to drive the mixer using raw USB requests, + * so they can also change this that way to configure the rate for + * stand-alone operation when the PCM is closed. + */ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "System Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_CTL_REG1, GENMASK(6, 3)), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Current Rate", + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .info = snd_rme_rate_info, + .get = snd_rme_digiface_rate_get, + .private_value = RME_DIGIFACE_REGISTER(RME_DIGIFACE_STATUS_REG1H, GENMASK(7, 4)), + } +}; + +static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + + for (i = 0; i < ARRAY_SIZE(snd_rme_digiface_controls); ++i) { + err = add_single_ctl_with_resume(mixer, 0, + NULL, + &snd_rme_digiface_controls[i], + NULL); + if (err < 0) + return err; + } + + return 0; +} + /* * Pioneer DJ DJM Mixers * @@ -3645,6 +4055,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2a39, 0x3f8c): /* RME Digiface USB */ + err = snd_rme_digiface_controls_create(mixer); + break; case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ err = snd_djm_controls_create(mixer, SND_DJM_250MK2_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 631b9ab80f6c..24c981c9b240 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3620,6 +3620,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), * Three modes depending on sample rate band, * with different channel counts for in/out */ + { QUIRK_DATA_STANDARD_MIXER(0) }, { QUIRK_DATA_AUDIOFORMAT(0) { .formats = SNDRV_PCM_FMTBIT_S32_LE, From ceb3ca2876243e3ea02f78b3d488b1f2d734de49 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:30 +0200 Subject: [PATCH 308/410] ASoC: dt-bindings: mediatek,mt8365-afe: Add audio afe document Add MT8365 audio front-end bindings Reviewed-by: Krzysztof Kozlowski Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-1-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- .../bindings/sound/mediatek,mt8365-afe.yaml | 130 ++++++++++++++++++ 1 file changed, 130 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml new file mode 100644 index 000000000000..45ad56d37234 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml @@ -0,0 +1,130 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8365-afe.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek Audio Front End PCM controller for MT8365 + +maintainers: + - Alexandre Mergnat + +properties: + compatible: + const: mediatek,mt8365-afe-pcm + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: 26M clock + - description: mux for audio clock + - description: audio i2s0 mck + - description: audio i2s1 mck + - description: audio i2s2 mck + - description: audio i2s3 mck + - description: engen 1 clock + - description: engen 2 clock + - description: audio 1 clock + - description: audio 2 clock + - description: mux for i2s0 + - description: mux for i2s1 + - description: mux for i2s2 + - description: mux for i2s3 + + clock-names: + items: + - const: top_clk26m_clk + - const: top_audio_sel + - const: audio_i2s0_m + - const: audio_i2s1_m + - const: audio_i2s2_m + - const: audio_i2s3_m + - const: engen1 + - const: engen2 + - const: aud1 + - const: aud2 + - const: i2s0_m_sel + - const: i2s1_m_sel + - const: i2s2_m_sel + - const: i2s3_m_sel + + interrupts: + maxItems: 1 + + power-domains: + maxItems: 1 + + mediatek,dmic-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + description: + Indicates how many data pins are used to transmit two channels of PDM + signal. 1 means two wires, 0 means one wire. Default value is 0. + enum: + - 0 # one wire + - 1 # two wires + +required: + - compatible + - reg + - clocks + - clock-names + - interrupts + - power-domains + +additionalProperties: false + +examples: + - | + #include + #include + #include + #include + + soc { + #address-cells = <2>; + #size-cells = <2>; + + audio-controller@11220000 { + compatible = "mediatek,mt8365-afe-pcm"; + reg = <0 0x11220000 0 0x1000>; + #sound-dai-cells = <0>; + clocks = <&clk26m>, + <&topckgen CLK_TOP_AUDIO_SEL>, + <&topckgen CLK_TOP_AUD_I2S0_M>, + <&topckgen CLK_TOP_AUD_I2S1_M>, + <&topckgen CLK_TOP_AUD_I2S2_M>, + <&topckgen CLK_TOP_AUD_I2S3_M>, + <&topckgen CLK_TOP_AUD_ENGEN1_SEL>, + <&topckgen CLK_TOP_AUD_ENGEN2_SEL>, + <&topckgen CLK_TOP_AUD_1_SEL>, + <&topckgen CLK_TOP_AUD_2_SEL>, + <&topckgen CLK_TOP_APLL_I2S0_SEL>, + <&topckgen CLK_TOP_APLL_I2S1_SEL>, + <&topckgen CLK_TOP_APLL_I2S2_SEL>, + <&topckgen CLK_TOP_APLL_I2S3_SEL>; + clock-names = "top_clk26m_clk", + "top_audio_sel", + "audio_i2s0_m", + "audio_i2s1_m", + "audio_i2s2_m", + "audio_i2s3_m", + "engen1", + "engen2", + "aud1", + "aud2", + "i2s0_m_sel", + "i2s1_m_sel", + "i2s2_m_sel", + "i2s3_m_sel"; + interrupts = ; + power-domains = <&spm MT8365_POWER_DOMAIN_AUDIO>; + mediatek,dmic-mode = <1>; + }; + }; + +... From 76d80dcdd55f70b28930edb97b96ee375e1cce5a Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:31 +0200 Subject: [PATCH 309/410] ASoC: dt-bindings: mediatek,mt8365-mt6357: Add audio sound card document Add soundcard bindings for the MT8365 SoC with the MT6357 audio codec. Reviewed-by: AngeloGioacchino Del Regno Reviewed-by: Krzysztof Kozlowski Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-2-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- .../sound/mediatek,mt8365-mt6357.yaml | 107 ++++++++++++++++++ 1 file changed, 107 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml diff --git a/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml b/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml new file mode 100644 index 000000000000..ff9ebb63a05f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml @@ -0,0 +1,107 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mediatek,mt8365-mt6357.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: MediaTek MT8365 ASoC sound card + +maintainers: + - Alexandre Mergnat + +properties: + compatible: + const: mediatek,mt8365-mt6357 + + pinctrl-names: + minItems: 1 + items: + - const: default + - const: dmic + - const: miso_off + - const: miso_on + - const: mosi_off + - const: mosi_on + + mediatek,platform: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of MT8365 ASoC platform. + +patternProperties: + "^dai-link-[0-9]+$": + type: object + description: + Container for dai-link level properties and CODEC sub-nodes. + + properties: + codec: + type: object + description: Holds subnode which indicates codec dai. + + properties: + sound-dai: + maxItems: 1 + description: phandle of the codec DAI + + additionalProperties: false + + link-name: + description: Indicates dai-link name and PCM stream name + enum: + - I2S_IN_BE + - I2S_OUT_BE + - PCM1_BE + - PDM1_BE + - PDM2_BE + - PDM3_BE + - PDM4_BE + - SPDIF_IN_BE + - SPDIF_OUT_BE + - TDM_IN_BE + - TDM_OUT_BE + + sound-dai: + maxItems: 1 + description: phandle of the CPU DAI + + required: + - link-name + - sound-dai + + additionalProperties: false + +required: + - compatible + - pinctrl-names + - mediatek,platform + +additionalProperties: false + +examples: + - | + sound { + compatible = "mediatek,mt8365-mt6357"; + pinctrl-names = "default", + "dmic", + "miso_off", + "miso_on", + "mosi_off", + "mosi_on"; + pinctrl-0 = <&aud_default_pins>; + pinctrl-1 = <&aud_dmic_pins>; + pinctrl-2 = <&aud_miso_off_pins>; + pinctrl-3 = <&aud_miso_on_pins>; + pinctrl-4 = <&aud_mosi_off_pins>; + pinctrl-5 = <&aud_mosi_on_pins>; + mediatek,platform = <&afe>; + + /* hdmi interface */ + dai-link-0 { + link-name = "I2S_OUT_BE"; + sound-dai = <&afe>; + + codec { + sound-dai = <&it66121hdmitx>; + }; + }; + }; From 761cab667898d86c04867948f1b7aec1090be796 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:32 +0200 Subject: [PATCH 310/410] dt-bindings: mfd: mediatek: Add codec property for MT6357 PMIC Add the audio codec sub-device. This sub-device is used to set the optional voltage values according to the hardware. The properties are: - Setup of microphone bias voltage. - Setup of the speaker pin pull-down. Also, add the audio power supply property which is dedicated for the audio codec sub-device. Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-3-6518d953a141@baylibre.com Reviewed-by: Krzysztof Kozlowski Signed-off-by: Mark Brown --- .../bindings/mfd/mediatek,mt6357.yaml | 21 +++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml b/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml index 37423c2e0fdf..b67fbe0e7a63 100644 --- a/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml +++ b/Documentation/devicetree/bindings/mfd/mediatek,mt6357.yaml @@ -37,6 +37,24 @@ properties: "#interrupt-cells": const: 2 + mediatek,hp-pull-down: + description: + Earphone driver positive output stage short to + the audio reference ground. + type: boolean + + mediatek,micbias0-microvolt: + description: Selects MIC Bias 0 output voltage. + enum: [1700000, 1800000, 1900000, 2000000, + 2100000, 2500000, 2600000, 2700000] + default: 1700000 + + mediatek,micbias1-microvolt: + description: Selects MIC Bias 1 output voltage. + enum: [1700000, 1800000, 1900000, 2000000, + 2100000, 2500000, 2600000, 2700000] + default: 1700000 + regulators: type: object $ref: /schemas/regulator/mediatek,mt6357-regulator.yaml @@ -83,6 +101,9 @@ examples: interrupt-controller; #interrupt-cells = <2>; + mediatek,micbias0-microvolt = <1700000>; + mediatek,micbias1-microvolt = <1700000>; + regulators { mt6357_vproc_reg: buck-vproc { regulator-name = "vproc"; From 38c7c9ddc74033406461d64e541bbc8268e77f73 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:33 +0200 Subject: [PATCH 311/410] ASoC: mediatek: mt8365: Add common header Add header files for register definition and structure. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-4-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-common.h | 449 ++++++++ sound/soc/mediatek/mt8365/mt8365-reg.h | 991 ++++++++++++++++++ 2 files changed, 1440 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-common.h create mode 100644 sound/soc/mediatek/mt8365/mt8365-reg.h diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-common.h b/sound/soc/mediatek/mt8365/mt8365-afe-common.h new file mode 100644 index 000000000000..1fa87e54a57f --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-afe-common.h @@ -0,0 +1,449 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * MediaTek 8365 audio driver common definitions + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#ifndef _MT8365_AFE_COMMON_H_ +#define _MT8365_AFE_COMMON_H_ + +#include +#include +#include +#include +#include +#include "../common/mtk-base-afe.h" +#include "mt8365-reg.h" + +enum { + MT8365_AFE_MEMIF_DL1, + MT8365_AFE_MEMIF_DL2, + MT8365_AFE_MEMIF_TDM_OUT, + /* + * MT8365_AFE_MEMIF_SPDIF_OUT, + */ + MT8365_AFE_MEMIF_AWB, + MT8365_AFE_MEMIF_VUL, + MT8365_AFE_MEMIF_VUL2, + MT8365_AFE_MEMIF_VUL3, + MT8365_AFE_MEMIF_TDM_IN, + /* + * MT8365_AFE_MEMIF_SPDIF_IN, + */ + MT8365_AFE_MEMIF_NUM, + MT8365_AFE_BACKEND_BASE = MT8365_AFE_MEMIF_NUM, + MT8365_AFE_IO_TDM_OUT = MT8365_AFE_BACKEND_BASE, + MT8365_AFE_IO_TDM_IN, + MT8365_AFE_IO_I2S, + MT8365_AFE_IO_2ND_I2S, + MT8365_AFE_IO_PCM1, + MT8365_AFE_IO_VIRTUAL_DL_SRC, + MT8365_AFE_IO_VIRTUAL_TDM_OUT_SRC, + MT8365_AFE_IO_VIRTUAL_FM, + MT8365_AFE_IO_DMIC, + MT8365_AFE_IO_INT_ADDA, + MT8365_AFE_IO_GASRC1, + MT8365_AFE_IO_GASRC2, + MT8365_AFE_IO_TDM_ASRC, + MT8365_AFE_IO_HW_GAIN1, + MT8365_AFE_IO_HW_GAIN2, + MT8365_AFE_BACKEND_END, + MT8365_AFE_BACKEND_NUM = (MT8365_AFE_BACKEND_END - + MT8365_AFE_BACKEND_BASE), +}; + +enum { + MT8365_AFE_IRQ1, + MT8365_AFE_IRQ2, + MT8365_AFE_IRQ3, + MT8365_AFE_IRQ4, + MT8365_AFE_IRQ5, + MT8365_AFE_IRQ6, + MT8365_AFE_IRQ7, + MT8365_AFE_IRQ8, + MT8365_AFE_IRQ9, + MT8365_AFE_IRQ10, + MT8365_AFE_IRQ_NUM, +}; + +enum { + MT8365_TOP_CG_AFE, + MT8365_TOP_CG_I2S_IN, + MT8365_TOP_CG_22M, + MT8365_TOP_CG_24M, + MT8365_TOP_CG_INTDIR_CK, + MT8365_TOP_CG_APLL2_TUNER, + MT8365_TOP_CG_APLL_TUNER, + MT8365_TOP_CG_SPDIF, + MT8365_TOP_CG_TDM_OUT, + MT8365_TOP_CG_TDM_IN, + MT8365_TOP_CG_ADC, + MT8365_TOP_CG_DAC, + MT8365_TOP_CG_DAC_PREDIS, + MT8365_TOP_CG_TML, + MT8365_TOP_CG_I2S1_BCLK, + MT8365_TOP_CG_I2S2_BCLK, + MT8365_TOP_CG_I2S3_BCLK, + MT8365_TOP_CG_I2S4_BCLK, + MT8365_TOP_CG_DMIC0_ADC, + MT8365_TOP_CG_DMIC1_ADC, + MT8365_TOP_CG_DMIC2_ADC, + MT8365_TOP_CG_DMIC3_ADC, + MT8365_TOP_CG_CONNSYS_I2S_ASRC, + MT8365_TOP_CG_GENERAL1_ASRC, + MT8365_TOP_CG_GENERAL2_ASRC, + MT8365_TOP_CG_TDM_ASRC, + MT8365_TOP_CG_NUM +}; + +enum { + MT8365_CLK_TOP_AUD_SEL, + MT8365_CLK_AUD_I2S0_M, + MT8365_CLK_AUD_I2S1_M, + MT8365_CLK_AUD_I2S2_M, + MT8365_CLK_AUD_I2S3_M, + MT8365_CLK_ENGEN1, + MT8365_CLK_ENGEN2, + MT8365_CLK_AUD1, + MT8365_CLK_AUD2, + MT8365_CLK_I2S0_M_SEL, + MT8365_CLK_I2S1_M_SEL, + MT8365_CLK_I2S2_M_SEL, + MT8365_CLK_I2S3_M_SEL, + MT8365_CLK_CLK26M, + MT8365_CLK_NUM +}; + +enum { + MT8365_AFE_APLL1 = 0, + MT8365_AFE_APLL2, + MT8365_AFE_APLL_NUM, +}; + +enum { + MT8365_AFE_1ST_I2S = 0, + MT8365_AFE_2ND_I2S, + MT8365_AFE_I2S_SETS, +}; + +enum { + MT8365_AFE_I2S_SEPARATE_CLOCK = 0, + MT8365_AFE_I2S_SHARED_CLOCK, +}; + +enum { + MT8365_AFE_TDM_OUT_I2S = 0, + MT8365_AFE_TDM_OUT_TDM, + MT8365_AFE_TDM_OUT_I2S_32BITS, +}; + +enum mt8365_afe_tdm_ch_start { + AFE_TDM_CH_START_O28_O29 = 0, + AFE_TDM_CH_START_O30_O31, + AFE_TDM_CH_START_O32_O33, + AFE_TDM_CH_START_O34_O35, + AFE_TDM_CH_ZERO, +}; + +enum { + MT8365_PCM_FORMAT_I2S = 0, + MT8365_PCM_FORMAT_EIAJ, + MT8365_PCM_FORMAT_PCMA, + MT8365_PCM_FORMAT_PCMB, +}; + +enum { + MT8365_FS_8K = 0, + MT8365_FS_11D025K, + MT8365_FS_12K, + MT8365_FS_384K, + MT8365_FS_16K, + MT8365_FS_22D05K, + MT8365_FS_24K, + MT8365_FS_130K, + MT8365_FS_32K, + MT8365_FS_44D1K, + MT8365_FS_48K, + MT8365_FS_88D2K, + MT8365_FS_96K, + MT8365_FS_176D4K, + MT8365_FS_192K, +}; + +enum { + FS_8000HZ = 0, /* 0000b */ + FS_11025HZ = 1, /* 0001b */ + FS_12000HZ = 2, /* 0010b */ + FS_384000HZ = 3, /* 0011b */ + FS_16000HZ = 4, /* 0100b */ + FS_22050HZ = 5, /* 0101b */ + FS_24000HZ = 6, /* 0110b */ + FS_130000HZ = 7, /* 0111b */ + FS_32000HZ = 8, /* 1000b */ + FS_44100HZ = 9, /* 1001b */ + FS_48000HZ = 10, /* 1010b */ + FS_88200HZ = 11, /* 1011b */ + FS_96000HZ = 12, /* 1100b */ + FS_176400HZ = 13, /* 1101b */ + FS_192000HZ = 14, /* 1110b */ + FS_260000HZ = 15, /* 1111b */ +}; + +enum { + MT8365_AFE_DEBUGFS_AFE, + MT8365_AFE_DEBUGFS_MEMIF, + MT8365_AFE_DEBUGFS_IRQ, + MT8365_AFE_DEBUGFS_CONN, + MT8365_AFE_DEBUGFS_DBG, + MT8365_AFE_DEBUGFS_NUM, +}; + +enum { + MT8365_AFE_IRQ_DIR_MCU = 0, + MT8365_AFE_IRQ_DIR_DSP, + MT8365_AFE_IRQ_DIR_BOTH, +}; + +/* MCLK */ +enum { + MT8365_I2S0_MCK = 0, + MT8365_I2S3_MCK, + MT8365_MCK_NUM, +}; + +struct mt8365_fe_dai_data { + bool use_sram; + unsigned int sram_phy_addr; + void __iomem *sram_vir_addr; + unsigned int sram_size; +}; + +struct mt8365_be_dai_data { + bool prepared[SNDRV_PCM_STREAM_LAST + 1]; + unsigned int fmt_mode; +}; + +#define MT8365_CLK_26M 26000000 +#define MT8365_CLK_24M 24000000 +#define MT8365_CLK_22M 22000000 +#define MT8365_CM_UPDATA_CNT_SET 8 + +enum mt8365_cm_num { + MT8365_CM1 = 0, + MT8365_CM2, + MT8365_CM_NUM, +}; + +enum mt8365_cm2_mux_in { + MT8365_FROM_GASRC1 = 1, + MT8365_FROM_GASRC2, + MT8365_FROM_TDM_ASRC, + MT8365_CM_MUX_NUM, +}; + +enum cm2_mux_conn_in { + GENERAL2_ASRC_OUT_LCH = 0, + GENERAL2_ASRC_OUT_RCH = 1, + TDM_IN_CH0 = 2, + TDM_IN_CH1 = 3, + TDM_IN_CH2 = 4, + TDM_IN_CH3 = 5, + TDM_IN_CH4 = 6, + TDM_IN_CH5 = 7, + TDM_IN_CH6 = 8, + TDM_IN_CH7 = 9, + GENERAL1_ASRC_OUT_LCH = 10, + GENERAL1_ASRC_OUT_RCH = 11, + TDM_OUT_ASRC_CH0 = 12, + TDM_OUT_ASRC_CH1 = 13, + TDM_OUT_ASRC_CH2 = 14, + TDM_OUT_ASRC_CH3 = 15, + TDM_OUT_ASRC_CH4 = 16, + TDM_OUT_ASRC_CH5 = 17, + TDM_OUT_ASRC_CH6 = 18, + TDM_OUT_ASRC_CH7 = 19 +}; + +struct mt8365_cm_ctrl_reg { + unsigned int con0; + unsigned int con1; + unsigned int con2; + unsigned int con3; + unsigned int con4; +}; + +struct mt8365_control_data { + bool bypass_cm1; + bool bypass_cm2; + unsigned int loopback_type; +}; + +enum dmic_input_mode { + DMIC_MODE_3P25M = 0, + DMIC_MODE_1P625M, + DMIC_MODE_812P5K, + DMIC_MODE_406P25K, +}; + +enum iir_mode { + IIR_MODE0 = 0, + IIR_MODE1, + IIR_MODE2, + IIR_MODE3, + IIR_MODE4, + IIR_MODE5, +}; + +enum { + MT8365_GASRC1 = 0, + MT8365_GASRC2, + MT8365_GASRC_NUM, + MT8365_TDM_ASRC1 = MT8365_GASRC_NUM, + MT8365_TDM_ASRC2, + MT8365_TDM_ASRC3, + MT8365_TDM_ASRC4, + MT8365_TDM_ASRC_NUM, +}; + +struct mt8365_gasrc_ctrl_reg { + unsigned int con0; + unsigned int con2; + unsigned int con3; + unsigned int con4; + unsigned int con5; + unsigned int con6; + unsigned int con9; + unsigned int con10; + unsigned int con12; + unsigned int con13; +}; + +struct mt8365_gasrc_data { + bool duplex; + bool tx_mode; + bool cali_on; + bool tdm_asrc_out_cm2; + bool iir_on; +}; + +struct mt8365_afe_private { + struct clk *clocks[MT8365_CLK_NUM]; + struct regmap *topckgen; + struct mt8365_fe_dai_data fe_data[MT8365_AFE_MEMIF_NUM]; + struct mt8365_be_dai_data be_data[MT8365_AFE_BACKEND_NUM]; + struct mt8365_control_data ctrl_data; + struct mt8365_gasrc_data gasrc_data[MT8365_TDM_ASRC_NUM]; + int afe_on_ref_cnt; + int top_cg_ref_cnt[MT8365_TOP_CG_NUM]; + void __iomem *afe_sram_vir_addr; + unsigned int afe_sram_phy_addr; + unsigned int afe_sram_size; + /* locks */ + spinlock_t afe_ctrl_lock; + struct mutex afe_clk_mutex; /* Protect & sync APLL TUNER registers access*/ +#ifdef CONFIG_DEBUG_FS + struct dentry *debugfs_dentry[MT8365_AFE_DEBUGFS_NUM]; +#endif + int apll_tuner_ref_cnt[MT8365_AFE_APLL_NUM]; + unsigned int tdm_out_mode; + unsigned int cm2_mux_input; + + /* dai */ + bool dai_on[MT8365_AFE_BACKEND_END]; + void *dai_priv[MT8365_AFE_BACKEND_END]; +}; + +static inline u32 rx_frequency_palette(unsigned int fs) +{ + /* * + * A = (26M / fs) * 64 + * B = 8125 / A + * return = DEC2HEX(B * 2^23) + */ + switch (fs) { + case FS_8000HZ: return 0x050000; + case FS_11025HZ: return 0x06E400; + case FS_12000HZ: return 0x078000; + case FS_16000HZ: return 0x0A0000; + case FS_22050HZ: return 0x0DC800; + case FS_24000HZ: return 0x0F0000; + case FS_32000HZ: return 0x140000; + case FS_44100HZ: return 0x1B9000; + case FS_48000HZ: return 0x1E0000; + case FS_88200HZ: return 0x372000; + case FS_96000HZ: return 0x3C0000; + case FS_176400HZ: return 0x6E4000; + case FS_192000HZ: return 0x780000; + default: return 0x0; + } +} + +static inline u32 AutoRstThHi(unsigned int fs) +{ + switch (fs) { + case FS_8000HZ: return 0x36000; + case FS_11025HZ: return 0x27000; + case FS_12000HZ: return 0x24000; + case FS_16000HZ: return 0x1B000; + case FS_22050HZ: return 0x14000; + case FS_24000HZ: return 0x12000; + case FS_32000HZ: return 0x0D800; + case FS_44100HZ: return 0x09D00; + case FS_48000HZ: return 0x08E00; + case FS_88200HZ: return 0x04E00; + case FS_96000HZ: return 0x04800; + case FS_176400HZ: return 0x02700; + case FS_192000HZ: return 0x02400; + default: return 0x0; + } +} + +static inline u32 AutoRstThLo(unsigned int fs) +{ + switch (fs) { + case FS_8000HZ: return 0x30000; + case FS_11025HZ: return 0x23000; + case FS_12000HZ: return 0x20000; + case FS_16000HZ: return 0x18000; + case FS_22050HZ: return 0x11000; + case FS_24000HZ: return 0x0FE00; + case FS_32000HZ: return 0x0BE00; + case FS_44100HZ: return 0x08A00; + case FS_48000HZ: return 0x07F00; + case FS_88200HZ: return 0x04500; + case FS_96000HZ: return 0x04000; + case FS_176400HZ: return 0x02300; + case FS_192000HZ: return 0x02000; + default: return 0x0; + } +} + +bool mt8365_afe_clk_group_48k(int sample_rate); +bool mt8365_afe_rate_supported(unsigned int rate, unsigned int id); +bool mt8365_afe_channel_supported(unsigned int channel, unsigned int id); + +int mt8365_dai_i2s_register(struct mtk_base_afe *afe); +int mt8365_dai_set_priv(struct mtk_base_afe *afe, + int id, + int priv_size, + const void *priv_data); + +int mt8365_afe_fs_timing(unsigned int rate); + +void mt8365_afe_set_i2s_out_enable(struct mtk_base_afe *afe, bool enable); +int mt8365_afe_set_i2s_out(struct mtk_base_afe *afe, unsigned int rate, int bit_width); + +int mt8365_dai_adda_register(struct mtk_base_afe *afe); +int mt8365_dai_enable_adda_on(struct mtk_base_afe *afe); +int mt8365_dai_disable_adda_on(struct mtk_base_afe *afe); + +int mt8365_dai_dmic_register(struct mtk_base_afe *afe); + +int mt8365_dai_pcm_register(struct mtk_base_afe *afe); + +int mt8365_dai_tdm_register(struct mtk_base_afe *afe); + +#endif diff --git a/sound/soc/mediatek/mt8365/mt8365-reg.h b/sound/soc/mediatek/mt8365/mt8365-reg.h new file mode 100644 index 000000000000..b7334c2e64ed --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-reg.h @@ -0,0 +1,991 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * MediaTek 8365 audio driver reg definition + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#ifndef _MT8365_REG_H_ +#define _MT8365_REG_H_ + +#define AUDIO_TOP_CON0 (0x0000) +#define AUDIO_TOP_CON1 (0x0004) +#define AUDIO_TOP_CON2 (0x0008) +#define AUDIO_TOP_CON3 (0x000c) + +#define AFE_DAC_CON0 (0x0010) +#define AFE_DAC_CON1 (0x0014) +#define AFE_I2S_CON (0x0018) +#define AFE_CONN0 (0x0020) +#define AFE_CONN1 (0x0024) +#define AFE_CONN2 (0x0028) +#define AFE_CONN3 (0x002c) +#define AFE_CONN4 (0x0030) +#define AFE_I2S_CON1 (0x0034) +#define AFE_I2S_CON2 (0x0038) +#define AFE_MRGIF_CON (0x003c) +#define AFE_DL1_BASE (0x0040) +#define AFE_DL1_CUR (0x0044) +#define AFE_DL1_END (0x0048) +#define AFE_I2S_CON3 (0x004c) +#define AFE_DL2_BASE (0x0050) +#define AFE_DL2_CUR (0x0054) +#define AFE_DL2_END (0x0058) +#define AFE_CONN5 (0x005c) +#define AFE_AWB_BASE (0x0070) +#define AFE_AWB_END (0x0078) +#define AFE_AWB_CUR (0x007c) +#define AFE_VUL_BASE (0x0080) +#define AFE_VUL_END (0x0088) +#define AFE_VUL_CUR (0x008c) +#define AFE_CONN6 (0x00bc) +#define AFE_MEMIF_MSB (0x00cc) +#define AFE_MEMIF_MON0 (0x00d0) +#define AFE_MEMIF_MON1 (0x00d4) +#define AFE_MEMIF_MON2 (0x00d8) +#define AFE_MEMIF_MON3 (0x00dc) +#define AFE_MEMIF_MON4 (0x00e0) +#define AFE_MEMIF_MON5 (0x00e4) +#define AFE_MEMIF_MON6 (0x00e8) +#define AFE_MEMIF_MON7 (0x00ec) +#define AFE_MEMIF_MON8 (0x00f0) +#define AFE_MEMIF_MON9 (0x00f4) +#define AFE_MEMIF_MON10 (0x00f8) +#define AFE_MEMIF_MON11 (0x00fc) +#define AFE_ADDA_DL_SRC2_CON0 (0x0108) +#define AFE_ADDA_DL_SRC2_CON1 (0x010c) +#define AFE_ADDA_UL_SRC_CON0 (0x0114) +#define AFE_ADDA_UL_SRC_CON1 (0x0118) +#define AFE_ADDA_TOP_CON0 (0x0120) +#define AFE_ADDA_UL_DL_CON0 (0x0124) +#define AFE_ADDA_SRC_DEBUG (0x012c) +#define AFE_ADDA_SRC_DEBUG_MON0 (0x0130) +#define AFE_ADDA_SRC_DEBUG_MON1 (0x0134) +#define AFE_ADDA_UL_SRC_MON0 (0x0148) +#define AFE_ADDA_UL_SRC_MON1 (0x014c) +#define AFE_SRAM_BOUND (0x0170) +#define AFE_SECURE_CON (0x0174) +#define AFE_SECURE_CONN0 (0x0178) +#define AFE_SIDETONE_DEBUG (0x01d0) +#define AFE_SIDETONE_MON (0x01d4) +#define AFE_SIDETONE_CON0 (0x01e0) +#define AFE_SIDETONE_COEFF (0x01e4) +#define AFE_SIDETONE_CON1 (0x01e8) +#define AFE_SIDETONE_GAIN (0x01ec) +#define AFE_SGEN_CON0 (0x01f0) +#define AFE_SINEGEN_CON_TDM (0x01f8) +#define AFE_SINEGEN_CON_TDM_IN (0x01fc) +#define AFE_TOP_CON0 (0x0200) +#define AFE_BUS_CFG (0x0240) +#define AFE_BUS_MON0 (0x0244) +#define AFE_ADDA_PREDIS_CON0 (0x0260) +#define AFE_ADDA_PREDIS_CON1 (0x0264) +#define AFE_CONN_MON0 (0x0280) +#define AFE_CONN_MON1 (0x0284) +#define AFE_CONN_MON2 (0x0288) +#define AFE_CONN_MON3 (0x028c) +#define AFE_ADDA_IIR_COEF_02_01 (0x0290) +#define AFE_ADDA_IIR_COEF_04_03 (0x0294) +#define AFE_ADDA_IIR_COEF_06_05 (0x0298) +#define AFE_ADDA_IIR_COEF_08_07 (0x029c) +#define AFE_ADDA_IIR_COEF_10_09 (0x02a0) +#define AFE_VUL_D2_BASE (0x0350) +#define AFE_VUL_D2_END (0x0358) +#define AFE_VUL_D2_CUR (0x035c) +#define AFE_HDMI_OUT_CON0 (0x0370) +#define AFE_HDMI_OUT_BASE (0x0374) +#define AFE_HDMI_OUT_CUR (0x0378) +#define AFE_HDMI_OUT_END (0x037c) +#define AFE_SPDIF_OUT_CON0 (0x0380) +#define AFE_SPDIF_OUT_BASE (0x0384) +#define AFE_SPDIF_OUT_CUR (0x0388) +#define AFE_SPDIF_OUT_END (0x038c) +#define AFE_HDMI_CONN0 (0x0390) +#define AFE_HDMI_CONN1 (0x0398) +#define AFE_CONN_TDMIN_CON (0x039c) +#define AFE_IRQ_MCU_CON (0x03a0) +#define AFE_IRQ_MCU_STATUS (0x03a4) +#define AFE_IRQ_MCU_CLR (0x03a8) +#define AFE_IRQ_MCU_CNT1 (0x03ac) +#define AFE_IRQ_MCU_CNT2 (0x03b0) +#define AFE_IRQ_MCU_EN (0x03b4) +#define AFE_IRQ_MCU_MON2 (0x03b8) +#define AFE_IRQ_MCU_CNT5 (0x03bc) +#define AFE_IRQ1_MCU_CNT_MON (0x03c0) +#define AFE_IRQ2_MCU_CNT_MON (0x03c4) +#define AFE_IRQ1_MCU_EN_CNT_MON (0x03c8) +#define AFE_IRQ5_MCU_CNT_MON (0x03cc) +#define AFE_MEMIF_MINLEN (0x03d0) +#define AFE_MEMIF_MAXLEN (0x03d4) +#define AFE_MEMIF_PBUF_SIZE (0x03d8) +#define AFE_IRQ_MCU_CNT7 (0x03dc) +#define AFE_IRQ7_MCU_CNT_MON (0x03e0) +#define AFE_MEMIF_PBUF2_SIZE (0x03ec) +#define AFE_APLL_TUNER_CFG (0x03f0) +#define AFE_APLL_TUNER_CFG1 (0x03f4) +#define AFE_IRQ_MCU_CON2 (0x03f8) +#define IRQ13_MCU_CNT (0x0408) +#define IRQ13_MCU_CNT_MON (0x040c) +#define AFE_GAIN1_CON0 (0x0410) +#define AFE_GAIN1_CON1 (0x0414) +#define AFE_GAIN1_CON2 (0x0418) +#define AFE_GAIN1_CON3 (0x041c) +#define AFE_GAIN2_CON0 (0x0428) +#define AFE_GAIN2_CON1 (0x042c) +#define AFE_GAIN2_CON2 (0x0430) +#define AFE_GAIN2_CON3 (0x0434) +#define AFE_GAIN2_CUR (0x043c) +#define AFE_CONN11 (0x0448) +#define AFE_CONN12 (0x044c) +#define AFE_CONN13 (0x0450) +#define AFE_CONN14 (0x0454) +#define AFE_CONN15 (0x0458) +#define AFE_CONN16 (0x045c) +#define AFE_CONN7 (0x0460) +#define AFE_CONN8 (0x0464) +#define AFE_CONN9 (0x0468) +#define AFE_CONN10 (0x046c) +#define AFE_CONN21 (0x0470) +#define AFE_CONN22 (0x0474) +#define AFE_CONN23 (0x0478) +#define AFE_CONN24 (0x047c) +#define AFE_IEC_CFG (0x0480) +#define AFE_IEC_NSNUM (0x0484) +#define AFE_IEC_BURST_INFO (0x0488) +#define AFE_IEC_BURST_LEN (0x048c) +#define AFE_IEC_NSADR (0x0490) +#define AFE_CONN_RS (0x0494) +#define AFE_CONN_DI (0x0498) +#define AFE_IEC_CHL_STAT0 (0x04a0) +#define AFE_IEC_CHL_STAT1 (0x04a4) +#define AFE_IEC_CHR_STAT0 (0x04a8) +#define AFE_IEC_CHR_STAT1 (0x04ac) +#define AFE_CONN25 (0x04b0) +#define AFE_CONN26 (0x04b4) +#define FPGA_CFG2 (0x04b8) +#define FPGA_CFG3 (0x04bc) +#define FPGA_CFG0 (0x04c0) +#define FPGA_CFG1 (0x04c4) +#define AFE_SRAM_DELSEL_CON0 (0x04f0) +#define AFE_SRAM_DELSEL_CON1 (0x04f4) +#define AFE_SRAM_DELSEL_CON2 (0x04f8) +#define FPGA_CFG4 (0x04fc) +#define AFE_TDM_GASRC4_ASRC_2CH_CON0 (0x0500) +#define AFE_TDM_GASRC4_ASRC_2CH_CON1 (0x0504) +#define AFE_TDM_GASRC4_ASRC_2CH_CON2 (0x0508) +#define AFE_TDM_GASRC4_ASRC_2CH_CON3 (0x050c) +#define AFE_TDM_GASRC4_ASRC_2CH_CON4 (0x0510) +#define AFE_TDM_GASRC4_ASRC_2CH_CON5 (0x0514) +#define AFE_TDM_GASRC4_ASRC_2CH_CON6 (0x0518) +#define AFE_TDM_GASRC4_ASRC_2CH_CON7 (0x051c) +#define AFE_TDM_GASRC4_ASRC_2CH_CON8 (0x0520) +#define AFE_TDM_GASRC4_ASRC_2CH_CON9 (0x0524) +#define AFE_TDM_GASRC4_ASRC_2CH_CON10 (0x0528) +#define AFE_TDM_GASRC4_ASRC_2CH_CON12 (0x0530) +#define AFE_TDM_GASRC4_ASRC_2CH_CON13 (0x0534) +#define PCM_INTF_CON2 (0x0538) +#define PCM2_INTF_CON (0x053c) +#define AFE_APB_MON (0x0540) +#define AFE_CONN34 (0x0544) +#define AFE_TDM_CON1 (0x0548) +#define AFE_TDM_CON2 (0x054c) +#define PCM_INTF_CON1 (0x0550) +#define AFE_SECURE_MASK_CONN47_1 (0x0554) +#define AFE_SECURE_MASK_CONN48_1 (0x0558) +#define AFE_SECURE_MASK_CONN49_1 (0x055c) +#define AFE_SECURE_MASK_CONN50_1 (0x0560) +#define AFE_SECURE_MASK_CONN51_1 (0x0564) +#define AFE_SECURE_MASK_CONN52_1 (0x0568) +#define AFE_SECURE_MASK_CONN53_1 (0x056c) +#define AFE_SE_SECURE_CON (0x0570) +#define AFE_TDM_IN_CON1 (0x0588) +#define AFE_TDM_IN_CON2 (0x058c) +#define AFE_TDM_IN_MON1 (0x0590) +#define AFE_TDM_IN_MON2 (0x0594) +#define AFE_TDM_IN_MON3 (0x0598) +#define AFE_DMIC0_UL_SRC_CON0 (0x05b4) +#define AFE_DMIC0_UL_SRC_CON1 (0x05b8) +#define AFE_DMIC0_SRC_DEBUG (0x05bc) +#define AFE_DMIC0_SRC_DEBUG_MON0 (0x05c0) +#define AFE_DMIC0_UL_SRC_MON0 (0x05c8) +#define AFE_DMIC0_UL_SRC_MON1 (0x05cc) +#define AFE_DMIC0_IIR_COEF_02_01 (0x05d0) +#define AFE_DMIC0_IIR_COEF_04_03 (0x05d4) +#define AFE_DMIC0_IIR_COEF_06_05 (0x05d8) +#define AFE_DMIC0_IIR_COEF_08_07 (0x05dc) +#define AFE_DMIC0_IIR_COEF_10_09 (0x05e0) +#define AFE_DMIC1_UL_SRC_CON0 (0x0620) +#define AFE_DMIC1_UL_SRC_CON1 (0x0624) +#define AFE_DMIC1_SRC_DEBUG (0x0628) +#define AFE_DMIC1_SRC_DEBUG_MON0 (0x062c) +#define AFE_DMIC1_UL_SRC_MON0 (0x0634) +#define AFE_DMIC1_UL_SRC_MON1 (0x0638) +#define AFE_DMIC1_IIR_COEF_02_01 (0x063c) +#define AFE_DMIC1_IIR_COEF_04_03 (0x0640) +#define AFE_DMIC1_IIR_COEF_06_05 (0x0644) +#define AFE_DMIC1_IIR_COEF_08_07 (0x0648) +#define AFE_DMIC1_IIR_COEF_10_09 (0x064c) +#define AFE_SECURE_MASK_CONN39_1 (0x068c) +#define AFE_SECURE_MASK_CONN40_1 (0x0690) +#define AFE_SECURE_MASK_CONN41_1 (0x0694) +#define AFE_SECURE_MASK_CONN42_1 (0x0698) +#define AFE_SECURE_MASK_CONN43_1 (0x069c) +#define AFE_SECURE_MASK_CONN44_1 (0x06a0) +#define AFE_SECURE_MASK_CONN45_1 (0x06a4) +#define AFE_SECURE_MASK_CONN46_1 (0x06a8) +#define AFE_TDM_GASRC1_ASRC_2CH_CON0 (0x06c0) +#define AFE_TDM_GASRC1_ASRC_2CH_CON1 (0x06c4) +#define AFE_TDM_GASRC1_ASRC_2CH_CON2 (0x06c8) +#define AFE_TDM_GASRC1_ASRC_2CH_CON3 (0x06cc) +#define AFE_TDM_GASRC1_ASRC_2CH_CON4 (0x06d0) +#define AFE_TDM_GASRC1_ASRC_2CH_CON5 (0x06d4) +#define AFE_TDM_GASRC1_ASRC_2CH_CON6 (0x06d8) +#define AFE_TDM_GASRC1_ASRC_2CH_CON7 (0x06dc) +#define AFE_TDM_GASRC1_ASRC_2CH_CON8 (0x06e0) +#define AFE_TDM_GASRC1_ASRC_2CH_CON9 (0x06e4) +#define AFE_TDM_GASRC1_ASRC_2CH_CON10 (0x06e8) +#define AFE_TDM_GASRC1_ASRC_2CH_CON12 (0x06f0) +#define AFE_TDM_GASRC1_ASRC_2CH_CON13 (0x06f4) +#define AFE_TDM_ASRC_CON0 (0x06f8) +#define AFE_TDM_GASRC2_ASRC_2CH_CON0 (0x0700) +#define AFE_TDM_GASRC2_ASRC_2CH_CON1 (0x0704) +#define AFE_TDM_GASRC2_ASRC_2CH_CON2 (0x0708) +#define AFE_TDM_GASRC2_ASRC_2CH_CON3 (0x070c) +#define AFE_TDM_GASRC2_ASRC_2CH_CON4 (0x0710) +#define AFE_TDM_GASRC2_ASRC_2CH_CON5 (0x0714) +#define AFE_TDM_GASRC2_ASRC_2CH_CON6 (0x0718) +#define AFE_TDM_GASRC2_ASRC_2CH_CON7 (0x071c) +#define AFE_TDM_GASRC2_ASRC_2CH_CON8 (0x0720) +#define AFE_TDM_GASRC2_ASRC_2CH_CON9 (0x0724) +#define AFE_TDM_GASRC2_ASRC_2CH_CON10 (0x0728) +#define AFE_TDM_GASRC2_ASRC_2CH_CON12 (0x0730) +#define AFE_TDM_GASRC2_ASRC_2CH_CON13 (0x0734) +#define AFE_TDM_GASRC3_ASRC_2CH_CON0 (0x0740) +#define AFE_TDM_GASRC3_ASRC_2CH_CON1 (0x0744) +#define AFE_TDM_GASRC3_ASRC_2CH_CON2 (0x0748) +#define AFE_TDM_GASRC3_ASRC_2CH_CON3 (0x074c) +#define AFE_TDM_GASRC3_ASRC_2CH_CON4 (0x0750) +#define AFE_TDM_GASRC3_ASRC_2CH_CON5 (0x0754) +#define AFE_TDM_GASRC3_ASRC_2CH_CON6 (0x0758) +#define AFE_TDM_GASRC3_ASRC_2CH_CON7 (0x075c) +#define AFE_TDM_GASRC3_ASRC_2CH_CON8 (0x0760) +#define AFE_TDM_GASRC3_ASRC_2CH_CON9 (0x0764) +#define AFE_TDM_GASRC3_ASRC_2CH_CON10 (0x0768) +#define AFE_TDM_GASRC3_ASRC_2CH_CON12 (0x0770) +#define AFE_TDM_GASRC3_ASRC_2CH_CON13 (0x0774) +#define AFE_DMIC2_UL_SRC_CON0 (0x0780) +#define AFE_DMIC2_UL_SRC_CON1 (0x0784) +#define AFE_DMIC2_SRC_DEBUG (0x0788) +#define AFE_DMIC2_SRC_DEBUG_MON0 (0x078c) +#define AFE_DMIC2_UL_SRC_MON0 (0x0794) +#define AFE_DMIC2_UL_SRC_MON1 (0x0798) +#define AFE_DMIC2_IIR_COEF_02_01 (0x079c) +#define AFE_DMIC2_IIR_COEF_04_03 (0x07a0) +#define AFE_DMIC2_IIR_COEF_06_05 (0x07a4) +#define AFE_DMIC2_IIR_COEF_08_07 (0x07a8) +#define AFE_DMIC2_IIR_COEF_10_09 (0x07ac) +#define AFE_DMIC3_UL_SRC_CON0 (0x07ec) +#define AFE_DMIC3_UL_SRC_CON1 (0x07f0) +#define AFE_DMIC3_SRC_DEBUG (0x07f4) +#define AFE_DMIC3_SRC_DEBUG_MON0 (0x07f8) +#define AFE_DMIC3_UL_SRC_MON0 (0x0800) +#define AFE_DMIC3_UL_SRC_MON1 (0x0804) +#define AFE_DMIC3_IIR_COEF_02_01 (0x0808) +#define AFE_DMIC3_IIR_COEF_04_03 (0x080c) +#define AFE_DMIC3_IIR_COEF_06_05 (0x0810) +#define AFE_DMIC3_IIR_COEF_08_07 (0x0814) +#define AFE_DMIC3_IIR_COEF_10_09 (0x0818) +#define AFE_SECURE_MASK_CONN25_1 (0x0858) +#define AFE_SECURE_MASK_CONN26_1 (0x085c) +#define AFE_SECURE_MASK_CONN27_1 (0x0860) +#define AFE_SECURE_MASK_CONN28_1 (0x0864) +#define AFE_SECURE_MASK_CONN29_1 (0x0868) +#define AFE_SECURE_MASK_CONN30_1 (0x086c) +#define AFE_SECURE_MASK_CONN31_1 (0x0870) +#define AFE_SECURE_MASK_CONN32_1 (0x0874) +#define AFE_SECURE_MASK_CONN33_1 (0x0878) +#define AFE_SECURE_MASK_CONN34_1 (0x087c) +#define AFE_SECURE_MASK_CONN35_1 (0x0880) +#define AFE_SECURE_MASK_CONN36_1 (0x0884) +#define AFE_SECURE_MASK_CONN37_1 (0x0888) +#define AFE_SECURE_MASK_CONN38_1 (0x088c) +#define AFE_IRQ_MCU_SCP_EN (0x0890) +#define AFE_IRQ_MCU_DSP_EN (0x0894) +#define AFE_IRQ3_MCU_CNT_MON (0x0898) +#define AFE_IRQ4_MCU_CNT_MON (0x089c) +#define AFE_IRQ8_MCU_CNT_MON (0x08a0) +#define AFE_IRQ_MCU_CNT3 (0x08a4) +#define AFE_IRQ_MCU_CNT4 (0x08a8) +#define AFE_IRQ_MCU_CNT8 (0x08ac) +#define AFE_IRQ_MCU_CNT11 (0x08b0) +#define AFE_IRQ_MCU_CNT12 (0x08b4) +#define AFE_IRQ11_MCU_CNT_MON (0x08b8) +#define AFE_IRQ12_MCU_CNT_MON (0x08bc) +#define AFE_VUL3_BASE (0x08c0) +#define AFE_VUL3_CUR (0x08c4) +#define AFE_VUL3_END (0x08c8) +#define AFE_VUL3_BASE_MSB (0x08d0) +#define AFE_VUL3_END_MSB (0x08d4) +#define AFE_IRQ10_MCU_CNT_MON (0x08d8) +#define AFE_IRQ_MCU_CNT10 (0x08dc) +#define AFE_IRQ_ACC1_CNT (0x08e0) +#define AFE_IRQ_ACC2_CNT (0x08e4) +#define AFE_IRQ_ACC1_CNT_MON1 (0x08e8) +#define AFE_IRQ_ACC2_CNT_MON (0x08ec) +#define AFE_TSF_CON (0x08f0) +#define AFE_TSF_MON (0x08f4) +#define AFE_IRQ_ACC1_CNT_MON2 (0x08f8) +#define AFE_SPDIFIN_CFG0 (0x0900) +#define AFE_SPDIFIN_CFG1 (0x0904) +#define AFE_SPDIFIN_CHSTS1 (0x0908) +#define AFE_SPDIFIN_CHSTS2 (0x090c) +#define AFE_SPDIFIN_CHSTS3 (0x0910) +#define AFE_SPDIFIN_CHSTS4 (0x0914) +#define AFE_SPDIFIN_CHSTS5 (0x0918) +#define AFE_SPDIFIN_CHSTS6 (0x091c) +#define AFE_SPDIFIN_DEBUG1 (0x0920) +#define AFE_SPDIFIN_DEBUG2 (0x0924) +#define AFE_SPDIFIN_DEBUG3 (0x0928) +#define AFE_SPDIFIN_DEBUG4 (0x092c) +#define AFE_SPDIFIN_EC (0x0930) +#define AFE_SPDIFIN_CKLOCK_CFG (0x0934) +#define AFE_SPDIFIN_BR (0x093c) +#define AFE_SPDIFIN_BR_DBG1 (0x0940) +#define AFE_SPDIFIN_INT_EXT (0x0948) +#define AFE_SPDIFIN_INT_EXT2 (0x094c) +#define SPDIFIN_FREQ_INFO (0x0950) +#define SPDIFIN_FREQ_INFO_2 (0x0954) +#define SPDIFIN_FREQ_INFO_3 (0x0958) +#define SPDIFIN_FREQ_STATUS (0x095c) +#define SPDIFIN_USERCODE1 (0x0960) +#define SPDIFIN_USERCODE2 (0x0964) +#define SPDIFIN_USERCODE3 (0x0968) +#define SPDIFIN_USERCODE4 (0x096c) +#define SPDIFIN_USERCODE5 (0x0970) +#define SPDIFIN_USERCODE6 (0x0974) +#define SPDIFIN_USERCODE7 (0x0978) +#define SPDIFIN_USERCODE8 (0x097c) +#define SPDIFIN_USERCODE9 (0x0980) +#define SPDIFIN_USERCODE10 (0x0984) +#define SPDIFIN_USERCODE11 (0x0988) +#define SPDIFIN_USERCODE12 (0x098c) +#define SPDIFIN_MEMIF_CON0 (0x0990) +#define SPDIFIN_BASE_ADR (0x0994) +#define SPDIFIN_END_ADR (0x0998) +#define SPDIFIN_APLL_TUNER_CFG (0x09a0) +#define SPDIFIN_APLL_TUNER_CFG1 (0x09a4) +#define SPDIFIN_APLL2_TUNER_CFG (0x09a8) +#define SPDIFIN_APLL2_TUNER_CFG1 (0x09ac) +#define SPDIFIN_TYPE_DET (0x09b0) +#define MPHONE_MULTI_CON0 (0x09b4) +#define SPDIFIN_CUR_ADR (0x09b8) +#define AFE_SINEGEN_CON_SPDIFIN (0x09bc) +#define AFE_HDMI_IN_2CH_CON0 (0x09c0) +#define AFE_HDMI_IN_2CH_BASE (0x09c4) +#define AFE_HDMI_IN_2CH_END (0x09c8) +#define AFE_HDMI_IN_2CH_CUR (0x09cc) +#define AFE_MEMIF_BUF_MON0 (0x09d0) +#define AFE_MEMIF_BUF_MON1 (0x09d4) +#define AFE_MEMIF_BUF_MON2 (0x09d8) +#define AFE_MEMIF_BUF_MON3 (0x09dc) +#define AFE_MEMIF_BUF_MON6 (0x09e8) +#define AFE_MEMIF_BUF_MON7 (0x09ec) +#define AFE_MEMIF_BUF_MON8 (0x09f0) +#define AFE_MEMIF_BUF_MON10 (0x09f8) +#define AFE_MEMIF_BUF_MON11 (0x09fc) +#define SYSTOP_STC_CONFIG (0x0a00) +#define AUDIO_STC_STATUS (0x0a04) +#define SYSTOP_W_STC_H (0x0a08) +#define SYSTOP_W_STC_L (0x0a0c) +#define SYSTOP_R_STC_H (0x0a10) +#define SYSTOP_R_STC_L (0x0a14) +#define AUDIO_W_STC_H (0x0a18) +#define AUDIO_W_STC_L (0x0a1c) +#define AUDIO_R_STC_H (0x0a20) +#define AUDIO_R_STC_L (0x0a24) +#define SYSTOP_W_STC2_H (0x0a28) +#define SYSTOP_W_STC2_L (0x0a2c) +#define SYSTOP_R_STC2_H (0x0a30) +#define SYSTOP_R_STC2_L (0x0a34) +#define AUDIO_W_STC2_H (0x0a38) +#define AUDIO_W_STC2_L (0x0a3c) +#define AUDIO_R_STC2_H (0x0a40) +#define AUDIO_R_STC2_L (0x0a44) + +#define AFE_CONN17 (0x0a48) +#define AFE_CONN18 (0x0a4c) +#define AFE_CONN19 (0x0a50) +#define AFE_CONN20 (0x0a54) +#define AFE_CONN27 (0x0a58) +#define AFE_CONN28 (0x0a5c) +#define AFE_CONN29 (0x0a60) +#define AFE_CONN30 (0x0a64) +#define AFE_CONN31 (0x0a68) +#define AFE_CONN32 (0x0a6c) +#define AFE_CONN33 (0x0a70) +#define AFE_CONN35 (0x0a74) +#define AFE_CONN36 (0x0a78) +#define AFE_CONN37 (0x0a7c) +#define AFE_CONN38 (0x0a80) +#define AFE_CONN39 (0x0a84) +#define AFE_CONN40 (0x0a88) +#define AFE_CONN41 (0x0a8c) +#define AFE_CONN42 (0x0a90) +#define AFE_CONN44 (0x0a94) +#define AFE_CONN45 (0x0a98) +#define AFE_CONN46 (0x0a9c) +#define AFE_CONN47 (0x0aa0) +#define AFE_CONN_24BIT (0x0aa4) +#define AFE_CONN0_1 (0x0aa8) +#define AFE_CONN1_1 (0x0aac) +#define AFE_CONN2_1 (0x0ab0) +#define AFE_CONN3_1 (0x0ab4) +#define AFE_CONN4_1 (0x0ab8) +#define AFE_CONN5_1 (0x0abc) +#define AFE_CONN6_1 (0x0ac0) +#define AFE_CONN7_1 (0x0ac4) +#define AFE_CONN8_1 (0x0ac8) +#define AFE_CONN9_1 (0x0acc) +#define AFE_CONN10_1 (0x0ad0) +#define AFE_CONN11_1 (0x0ad4) +#define AFE_CONN12_1 (0x0ad8) +#define AFE_CONN13_1 (0x0adc) +#define AFE_CONN14_1 (0x0ae0) +#define AFE_CONN15_1 (0x0ae4) +#define AFE_CONN16_1 (0x0ae8) +#define AFE_CONN17_1 (0x0aec) +#define AFE_CONN18_1 (0x0af0) +#define AFE_CONN19_1 (0x0af4) +#define AFE_CONN43 (0x0af8) +#define AFE_CONN43_1 (0x0afc) +#define AFE_CONN21_1 (0x0b00) +#define AFE_CONN22_1 (0x0b04) +#define AFE_CONN23_1 (0x0b08) +#define AFE_CONN24_1 (0x0b0c) +#define AFE_CONN25_1 (0x0b10) +#define AFE_CONN26_1 (0x0b14) +#define AFE_CONN27_1 (0x0b18) +#define AFE_CONN28_1 (0x0b1c) +#define AFE_CONN29_1 (0x0b20) +#define AFE_CONN30_1 (0x0b24) +#define AFE_CONN31_1 (0x0b28) +#define AFE_CONN32_1 (0x0b2c) +#define AFE_CONN33_1 (0x0b30) +#define AFE_CONN34_1 (0x0b34) +#define AFE_CONN35_1 (0x0b38) +#define AFE_CONN36_1 (0x0b3c) +#define AFE_CONN37_1 (0x0b40) +#define AFE_CONN38_1 (0x0b44) +#define AFE_CONN39_1 (0x0b48) +#define AFE_CONN40_1 (0x0b4c) +#define AFE_CONN41_1 (0x0b50) +#define AFE_CONN42_1 (0x0b54) +#define AFE_CONN44_1 (0x0b58) +#define AFE_CONN45_1 (0x0b5c) +#define AFE_CONN46_1 (0x0b60) +#define AFE_CONN47_1 (0x0b64) +#define AFE_CONN_RS_1 (0x0b68) +#define AFE_CONN_DI_1 (0x0b6c) +#define AFE_CONN_24BIT_1 (0x0b70) +#define AFE_GAIN1_CUR (0x0b78) +#define AFE_CONN20_1 (0x0b7c) +#define AFE_DL1_BASE_MSB (0x0b80) +#define AFE_DL1_END_MSB (0x0b84) +#define AFE_DL2_BASE_MSB (0x0b88) +#define AFE_DL2_END_MSB (0x0b8c) +#define AFE_AWB_BASE_MSB (0x0b90) +#define AFE_AWB_END_MSB (0x0b94) +#define AFE_VUL_BASE_MSB (0x0ba0) +#define AFE_VUL_END_MSB (0x0ba4) +#define AFE_VUL_D2_BASE_MSB (0x0ba8) +#define AFE_VUL_D2_END_MSB (0x0bac) +#define AFE_HDMI_OUT_BASE_MSB (0x0bb8) +#define AFE_HDMI_OUT_END_MSB (0x0bbc) +#define AFE_HDMI_IN_2CH_BASE_MSB (0x0bc0) +#define AFE_HDMI_IN_2CH_END_MSB (0x0bc4) +#define AFE_SPDIF_OUT_BASE_MSB (0x0bc8) +#define AFE_SPDIF_OUT_END_MSB (0x0bcc) +#define SPDIFIN_BASE_MSB (0x0bd0) +#define SPDIFIN_END_MSB (0x0bd4) +#define AFE_DL1_CUR_MSB (0x0bd8) +#define AFE_DL2_CUR_MSB (0x0bdc) +#define AFE_AWB_CUR_MSB (0x0be8) +#define AFE_VUL_CUR_MSB (0x0bf8) +#define AFE_VUL_D2_CUR_MSB (0x0c04) +#define AFE_HDMI_OUT_CUR_MSB (0x0c0c) +#define AFE_HDMI_IN_2CH_CUR_MSB (0x0c10) +#define AFE_SPDIF_OUT_CUR_MSB (0x0c14) +#define SPDIFIN_CUR_MSB (0x0c18) +#define AFE_CONN_REG (0x0c20) +#define AFE_SECURE_MASK_CONN14_1 (0x0c24) +#define AFE_SECURE_MASK_CONN15_1 (0x0c28) +#define AFE_SECURE_MASK_CONN16_1 (0x0c2c) +#define AFE_SECURE_MASK_CONN17_1 (0x0c30) +#define AFE_SECURE_MASK_CONN18_1 (0x0c34) +#define AFE_SECURE_MASK_CONN19_1 (0x0c38) +#define AFE_SECURE_MASK_CONN20_1 (0x0c3c) +#define AFE_SECURE_MASK_CONN21_1 (0x0c40) +#define AFE_SECURE_MASK_CONN22_1 (0x0c44) +#define AFE_SECURE_MASK_CONN23_1 (0x0c48) +#define AFE_SECURE_MASK_CONN24_1 (0x0c4c) +#define AFE_ADDA_DL_SDM_DCCOMP_CON (0x0c50) +#define AFE_ADDA_DL_SDM_TEST (0x0c54) +#define AFE_ADDA_DL_DC_COMP_CFG0 (0x0c58) +#define AFE_ADDA_DL_DC_COMP_CFG1 (0x0c5c) +#define AFE_ADDA_DL_SDM_FIFO_MON (0x0c60) +#define AFE_ADDA_DL_SRC_LCH_MON (0x0c64) +#define AFE_ADDA_DL_SRC_RCH_MON (0x0c68) +#define AFE_ADDA_DL_SDM_OUT_MON (0x0c6c) +#define AFE_ADDA_DL_SDM_DITHER_CON (0x0c70) + +#define AFE_VUL3_CUR_MSB (0x0c78) +#define AFE_ASRC_2CH_CON0 (0x0c80) +#define AFE_ASRC_2CH_CON1 (0x0c84) +#define AFE_ASRC_2CH_CON2 (0x0c88) +#define AFE_ASRC_2CH_CON3 (0x0c8c) +#define AFE_ASRC_2CH_CON4 (0x0c90) +#define AFE_ASRC_2CH_CON5 (0x0c94) +#define AFE_ASRC_2CH_CON6 (0x0c98) +#define AFE_ASRC_2CH_CON7 (0x0c9c) +#define AFE_ASRC_2CH_CON8 (0x0ca0) +#define AFE_ASRC_2CH_CON9 (0x0ca4) +#define AFE_ASRC_2CH_CON10 (0x0ca8) +#define AFE_ASRC_2CH_CON12 (0x0cb0) +#define AFE_ASRC_2CH_CON13 (0x0cb4) + +#define AFE_PCM_TX_ASRC_2CH_CON0 (0x0cc0) +#define AFE_PCM_TX_ASRC_2CH_CON1 (0x0cc4) +#define AFE_PCM_TX_ASRC_2CH_CON2 (0x0cc8) +#define AFE_PCM_TX_ASRC_2CH_CON3 (0x0ccc) +#define AFE_PCM_TX_ASRC_2CH_CON4 (0x0cd0) +#define AFE_PCM_TX_ASRC_2CH_CON5 (0x0cd4) +#define AFE_PCM_TX_ASRC_2CH_CON6 (0x0cd8) +#define AFE_PCM_TX_ASRC_2CH_CON7 (0x0cdc) +#define AFE_PCM_TX_ASRC_2CH_CON8 (0x0ce0) +#define AFE_PCM_TX_ASRC_2CH_CON9 (0x0ce4) +#define AFE_PCM_TX_ASRC_2CH_CON10 (0x0ce8) +#define AFE_PCM_TX_ASRC_2CH_CON12 (0x0cf0) +#define AFE_PCM_TX_ASRC_2CH_CON13 (0x0cf4) +#define AFE_PCM_RX_ASRC_2CH_CON0 (0x0d00) +#define AFE_PCM_RX_ASRC_2CH_CON1 (0x0d04) +#define AFE_PCM_RX_ASRC_2CH_CON2 (0x0d08) +#define AFE_PCM_RX_ASRC_2CH_CON3 (0x0d0c) +#define AFE_PCM_RX_ASRC_2CH_CON4 (0x0d10) +#define AFE_PCM_RX_ASRC_2CH_CON5 (0x0d14) +#define AFE_PCM_RX_ASRC_2CH_CON6 (0x0d18) +#define AFE_PCM_RX_ASRC_2CH_CON7 (0x0d1c) +#define AFE_PCM_RX_ASRC_2CH_CON8 (0x0d20) +#define AFE_PCM_RX_ASRC_2CH_CON9 (0x0d24) +#define AFE_PCM_RX_ASRC_2CH_CON10 (0x0d28) +#define AFE_PCM_RX_ASRC_2CH_CON12 (0x0d30) +#define AFE_PCM_RX_ASRC_2CH_CON13 (0x0d34) + +#define AFE_ADDA_PREDIS_CON2 (0x0d40) +#define AFE_ADDA_PREDIS_CON3 (0x0d44) +#define AFE_SECURE_MASK_CONN4_1 (0x0d48) +#define AFE_SECURE_MASK_CONN5_1 (0x0d4c) +#define AFE_SECURE_MASK_CONN6_1 (0x0d50) +#define AFE_SECURE_MASK_CONN7_1 (0x0d54) +#define AFE_SECURE_MASK_CONN8_1 (0x0d58) +#define AFE_SECURE_MASK_CONN9_1 (0x0d5c) +#define AFE_SECURE_MASK_CONN10_1 (0x0d60) +#define AFE_SECURE_MASK_CONN11_1 (0x0d64) +#define AFE_SECURE_MASK_CONN12_1 (0x0d68) +#define AFE_SECURE_MASK_CONN13_1 (0x0d6c) +#define AFE_MEMIF_MON12 (0x0d70) +#define AFE_MEMIF_MON13 (0x0d74) +#define AFE_MEMIF_MON14 (0x0d78) +#define AFE_MEMIF_MON15 (0x0d7c) +#define AFE_SECURE_MASK_CONN42 (0x0dbc) +#define AFE_SECURE_MASK_CONN43 (0x0dc0) +#define AFE_SECURE_MASK_CONN44 (0x0dc4) +#define AFE_SECURE_MASK_CONN45 (0x0dc8) +#define AFE_SECURE_MASK_CONN46 (0x0dcc) +#define AFE_HD_ENGEN_ENABLE (0x0dd0) +#define AFE_SECURE_MASK_CONN47 (0x0dd4) +#define AFE_SECURE_MASK_CONN48 (0x0dd8) +#define AFE_SECURE_MASK_CONN49 (0x0ddc) +#define AFE_SECURE_MASK_CONN50 (0x0de0) +#define AFE_SECURE_MASK_CONN51 (0x0de4) +#define AFE_SECURE_MASK_CONN52 (0x0de8) +#define AFE_SECURE_MASK_CONN53 (0x0dec) +#define AFE_SECURE_MASK_CONN0_1 (0x0df0) +#define AFE_SECURE_MASK_CONN1_1 (0x0df4) +#define AFE_SECURE_MASK_CONN2_1 (0x0df8) +#define AFE_SECURE_MASK_CONN3_1 (0x0dfc) + +#define AFE_ADDA_MTKAIF_CFG0 (0x0e00) +#define AFE_ADDA_MTKAIF_SYNCWORD_CFG (0x0e14) +#define AFE_ADDA_MTKAIF_RX_CFG0 (0x0e20) +#define AFE_ADDA_MTKAIF_RX_CFG1 (0x0e24) +#define AFE_ADDA_MTKAIF_RX_CFG2 (0x0e28) +#define AFE_ADDA_MTKAIF_MON0 (0x0e34) +#define AFE_ADDA_MTKAIF_MON1 (0x0e38) +#define AFE_AUD_PAD_TOP (0x0e40) + +#define AFE_CM1_CON4 (0x0e48) +#define AFE_CM2_CON4 (0x0e4c) +#define AFE_CM1_CON0 (0x0e50) +#define AFE_CM1_CON1 (0x0e54) +#define AFE_CM1_CON2 (0x0e58) +#define AFE_CM1_CON3 (0x0e5c) +#define AFE_CM2_CON0 (0x0e60) +#define AFE_CM2_CON1 (0x0e64) +#define AFE_CM2_CON2 (0x0e68) +#define AFE_CM2_CON3 (0x0e6c) +#define AFE_CM2_CONN0 (0x0e70) +#define AFE_CM2_CONN1 (0x0e74) +#define AFE_CM2_CONN2 (0x0e78) + +#define AFE_GENERAL1_ASRC_2CH_CON0 (0x0e80) +#define AFE_GENERAL1_ASRC_2CH_CON1 (0x0e84) +#define AFE_GENERAL1_ASRC_2CH_CON2 (0x0e88) +#define AFE_GENERAL1_ASRC_2CH_CON3 (0x0e8c) +#define AFE_GENERAL1_ASRC_2CH_CON4 (0x0e90) +#define AFE_GENERAL1_ASRC_2CH_CON5 (0x0e94) +#define AFE_GENERAL1_ASRC_2CH_CON6 (0x0e98) +#define AFE_GENERAL1_ASRC_2CH_CON7 (0x0e9c) +#define AFE_GENERAL1_ASRC_2CH_CON8 (0x0ea0) +#define AFE_GENERAL1_ASRC_2CH_CON9 (0x0ea4) +#define AFE_GENERAL1_ASRC_2CH_CON10 (0x0ea8) +#define AFE_GENERAL1_ASRC_2CH_CON12 (0x0eb0) +#define AFE_GENERAL1_ASRC_2CH_CON13 (0x0eb4) +#define GENERAL_ASRC_MODE (0x0eb8) +#define GENERAL_ASRC_EN_ON (0x0ebc) + +#define AFE_CONN48 (0x0ec0) +#define AFE_CONN49 (0x0ec4) +#define AFE_CONN50 (0x0ec8) +#define AFE_CONN51 (0x0ecc) +#define AFE_CONN52 (0x0ed0) +#define AFE_CONN53 (0x0ed4) +#define AFE_CONN48_1 (0x0ee0) +#define AFE_CONN49_1 (0x0ee4) +#define AFE_CONN50_1 (0x0ee8) +#define AFE_CONN51_1 (0x0eec) +#define AFE_CONN52_1 (0x0ef0) +#define AFE_CONN53_1 (0x0ef4) + +#define AFE_GENERAL2_ASRC_2CH_CON0 (0x0f00) +#define AFE_GENERAL2_ASRC_2CH_CON1 (0x0f04) +#define AFE_GENERAL2_ASRC_2CH_CON2 (0x0f08) +#define AFE_GENERAL2_ASRC_2CH_CON3 (0x0f0c) +#define AFE_GENERAL2_ASRC_2CH_CON4 (0x0f10) +#define AFE_GENERAL2_ASRC_2CH_CON5 (0x0f14) +#define AFE_GENERAL2_ASRC_2CH_CON6 (0x0f18) +#define AFE_GENERAL2_ASRC_2CH_CON7 (0x0f1c) +#define AFE_GENERAL2_ASRC_2CH_CON8 (0x0f20) +#define AFE_GENERAL2_ASRC_2CH_CON9 (0x0f24) +#define AFE_GENERAL2_ASRC_2CH_CON10 (0x0f28) +#define AFE_GENERAL2_ASRC_2CH_CON12 (0x0f30) +#define AFE_GENERAL2_ASRC_2CH_CON13 (0x0f34) + +#define AFE_SECURE_MASK_CONN28 (0x0f48) +#define AFE_SECURE_MASK_CONN29 (0x0f4c) +#define AFE_SECURE_MASK_CONN30 (0x0f50) +#define AFE_SECURE_MASK_CONN31 (0x0f54) +#define AFE_SECURE_MASK_CONN32 (0x0f58) +#define AFE_SECURE_MASK_CONN33 (0x0f5c) +#define AFE_SECURE_MASK_CONN34 (0x0f60) +#define AFE_SECURE_MASK_CONN35 (0x0f64) +#define AFE_SECURE_MASK_CONN36 (0x0f68) +#define AFE_SECURE_MASK_CONN37 (0x0f6c) +#define AFE_SECURE_MASK_CONN38 (0x0f70) +#define AFE_SECURE_MASK_CONN39 (0x0f74) +#define AFE_SECURE_MASK_CONN40 (0x0f78) +#define AFE_SECURE_MASK_CONN41 (0x0f7c) +#define AFE_SIDEBAND0 (0x0f80) +#define AFE_SIDEBAND1 (0x0f84) +#define AFE_SECURE_SIDEBAND0 (0x0f88) +#define AFE_SECURE_SIDEBAND1 (0x0f8c) +#define AFE_SECURE_MASK_CONN0 (0x0f90) +#define AFE_SECURE_MASK_CONN1 (0x0f94) +#define AFE_SECURE_MASK_CONN2 (0x0f98) +#define AFE_SECURE_MASK_CONN3 (0x0f9c) +#define AFE_SECURE_MASK_CONN4 (0x0fa0) +#define AFE_SECURE_MASK_CONN5 (0x0fa4) +#define AFE_SECURE_MASK_CONN6 (0x0fa8) +#define AFE_SECURE_MASK_CONN7 (0x0fac) +#define AFE_SECURE_MASK_CONN8 (0x0fb0) +#define AFE_SECURE_MASK_CONN9 (0x0fb4) +#define AFE_SECURE_MASK_CONN10 (0x0fb8) +#define AFE_SECURE_MASK_CONN11 (0x0fbc) +#define AFE_SECURE_MASK_CONN12 (0x0fc0) +#define AFE_SECURE_MASK_CONN13 (0x0fc4) +#define AFE_SECURE_MASK_CONN14 (0x0fc8) +#define AFE_SECURE_MASK_CONN15 (0x0fcc) +#define AFE_SECURE_MASK_CONN16 (0x0fd0) +#define AFE_SECURE_MASK_CONN17 (0x0fd4) +#define AFE_SECURE_MASK_CONN18 (0x0fd8) +#define AFE_SECURE_MASK_CONN19 (0x0fdc) +#define AFE_SECURE_MASK_CONN20 (0x0fe0) +#define AFE_SECURE_MASK_CONN21 (0x0fe4) +#define AFE_SECURE_MASK_CONN22 (0x0fe8) +#define AFE_SECURE_MASK_CONN23 (0x0fec) +#define AFE_SECURE_MASK_CONN24 (0x0ff0) +#define AFE_SECURE_MASK_CONN25 (0x0ff4) +#define AFE_SECURE_MASK_CONN26 (0x0ff8) +#define AFE_SECURE_MASK_CONN27 (0x0ffc) + +#define MAX_REGISTER AFE_SECURE_MASK_CONN27 + +#define AFE_IRQ_STATUS_BITS 0x3ff + +/* AUDIO_TOP_CON0 (0x0000) */ +#define AUD_TCON0_PDN_TML BIT(27) +#define AUD_TCON0_PDN_DAC_PREDIS BIT(26) +#define AUD_TCON0_PDN_DAC BIT(25) +#define AUD_TCON0_PDN_ADC BIT(24) +#define AUD_TCON0_PDN_TDM_IN BIT(23) +#define AUD_TCON0_PDN_TDM_OUT BIT(22) +#define AUD_TCON0_PDN_SPDIF BIT(21) +#define AUD_TCON0_PDN_APLL_TUNER BIT(19) +#define AUD_TCON0_PDN_APLL2_TUNER BIT(18) +#define AUD_TCON0_PDN_INTDIR BIT(15) +#define AUD_TCON0_PDN_24M BIT(9) +#define AUD_TCON0_PDN_22M BIT(8) +#define AUD_TCON0_PDN_I2S_IN BIT(6) +#define AUD_TCON0_PDN_AFE BIT(2) + +/* AUDIO_TOP_CON1 (0x0004) */ +#define AUD_TCON1_PDN_TDM_ASRC BIT(15) +#define AUD_TCON1_PDN_GENERAL2_ASRC BIT(14) +#define AUD_TCON1_PDN_GENERAL1_ASRC BIT(13) +#define AUD_TCON1_PDN_CONNSYS_I2S_ASRC BIT(12) +#define AUD_TCON1_PDN_DMIC3_ADC BIT(11) +#define AUD_TCON1_PDN_DMIC2_ADC BIT(10) +#define AUD_TCON1_PDN_DMIC1_ADC BIT(9) +#define AUD_TCON1_PDN_DMIC0_ADC BIT(8) +#define AUD_TCON1_PDN_I2S4_BCLK BIT(7) +#define AUD_TCON1_PDN_I2S3_BCLK BIT(6) +#define AUD_TCON1_PDN_I2S2_BCLK BIT(5) +#define AUD_TCON1_PDN_I2S1_BCLK BIT(4) + +/* AUDIO_TOP_CON3 (0x000C) */ +#define AUD_TCON3_HDMI_BCK_INV BIT(3) + +/* AFE_I2S_CON (0x0018) */ +#define AFE_I2S_CON_PHASE_SHIFT_FIX BIT(31) +#define AFE_I2S_CON_FROM_IO_MUX BIT(28) +#define AFE_I2S_CON_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON_RATE_MASK GENMASK(11, 8) +#define AFE_I2S_CON_FORMAT_I2S BIT(3) +#define AFE_I2S_CON_SRC_SLAVE BIT(2) + +/* AFE_ASRC_2CH_CON0 */ +#define ONE_HEART BIT(31) +#define CHSET_STR_CLR BIT(4) +#define COEFF_SRAM_CTRL BIT(1) +#define ASM_ON BIT(0) + +/* CON2 */ +#define O16BIT BIT(19) +#define CLR_IIR_HISTORY BIT(17) +#define IS_MONO BIT(16) +#define IIR_EN BIT(11) +#define IIR_STAGE_MASK GENMASK(10, 8) + +/* CON5 */ +#define CALI_CYCLE_MASK GENMASK(31, 16) +#define CALI_64_CYCLE FIELD_PREP(CALI_CYCLE_MASK, 0x3F) +#define CALI_96_CYCLE FIELD_PREP(CALI_CYCLE_MASK, 0x5F) +#define CALI_441_CYCLE FIELD_PREP(CALI_CYCLE_MASK, 0x1B8) + +#define CALI_AUTORST BIT(15) +#define AUTO_TUNE_FREQ5 BIT(12) +#define COMP_FREQ_RES BIT(11) + +#define CALI_SEL_MASK GENMASK(9, 8) +#define CALI_SEL_00 FIELD_PREP(CALI_SEL_MASK, 0) +#define CALI_SEL_01 FIELD_PREP(CALI_SEL_MASK, 1) + +#define CALI_BP_DGL BIT(7) /* Bypass the deglitch circuit */ +#define AUTO_TUNE_FREQ4 BIT(3) +#define CALI_AUTO_RESTART BIT(2) +#define CALI_USE_FREQ_OUT BIT(1) +#define CALI_ON BIT(0) + +#define AFE_I2S_CON_WLEN_32BIT BIT(1) +#define AFE_I2S_CON_EN BIT(0) + +#define AFE_CONN3_I03_O03_S BIT(3) +#define AFE_CONN4_I04_O04_S BIT(4) +#define AFE_CONN4_I03_O04_S BIT(3) + +/* AFE_I2S_CON1 (0x0034) */ +#define AFE_I2S_CON1_I2S2_TO_PAD BIT(18) +#define AFE_I2S_CON1_TDMOUT_TO_PAD (0 << 18) +#define AFE_I2S_CON1_RATE GENMASK(11, 8) +#define AFE_I2S_CON1_FORMAT_I2S BIT(3) +#define AFE_I2S_CON1_WLEN_32BIT BIT(1) +#define AFE_I2S_CON1_EN BIT(0) + +/* AFE_I2S_CON2 (0x0038) */ +#define AFE_I2S_CON2_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON2_RATE GENMASK(11, 8) +#define AFE_I2S_CON2_FORMAT_I2S BIT(3) +#define AFE_I2S_CON2_WLEN_32BIT BIT(1) +#define AFE_I2S_CON2_EN BIT(0) + +/* AFE_I2S_CON3 (0x004C) */ +#define AFE_I2S_CON3_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON3_RATE GENMASK(11, 8) +#define AFE_I2S_CON3_FORMAT_I2S BIT(3) +#define AFE_I2S_CON3_WLEN_32BIT BIT(1) +#define AFE_I2S_CON3_EN BIT(0) + +/* AFE_ADDA_DL_SRC2_CON0 (0x0108) */ +#define AFE_ADDA_DL_SAMPLING_RATE GENMASK(31, 28) +#define AFE_ADDA_DL_8X_UPSAMPLE GENMASK(25, 24) +#define AFE_ADDA_DL_MUTE_OFF_CH1 BIT(12) +#define AFE_ADDA_DL_MUTE_OFF_CH2 BIT(11) +#define AFE_ADDA_DL_VOICE_DATA BIT(5) +#define AFE_ADDA_DL_DEGRADE_GAIN BIT(1) + +/* AFE_ADDA_UL_SRC_CON0 (0x0114) */ +#define AFE_ADDA_UL_SAMPLING_RATE GENMASK(19, 17) + +/* AFE_ADDA_UL_DL_CON0 */ +#define AFE_ADDA_UL_DL_ADDA_AFE_ON BIT(0) +#define AFE_ADDA_UL_DL_DMIC_CLKDIV_ON BIT(1) + +/* AFE_APLL_TUNER_CFG (0x03f0) */ +#define AFE_APLL_TUNER_CFG_MASK GENMASK(15, 1) +#define AFE_APLL_TUNER_CFG_EN_MASK BIT(0) + +/* AFE_APLL_TUNER_CFG1 (0x03f4) */ +#define AFE_APLL_TUNER_CFG1_MASK GENMASK(15, 1) +#define AFE_APLL_TUNER_CFG1_EN_MASK BIT(0) + +/* PCM_INTF_CON1 (0x0550) */ +#define PCM_INTF_CON1_EXT_MODEM BIT(17) +#define PCM_INTF_CON1_16BIT (0 << 16) +#define PCM_INTF_CON1_24BIT BIT(16) +#define PCM_INTF_CON1_32BCK (0 << 14) +#define PCM_INTF_CON1_64BCK BIT(14) +#define PCM_INTF_CON1_MASTER_MODE (0 << 5) +#define PCM_INTF_CON1_SLAVE_MODE BIT(5) +#define PCM_INTF_CON1_FS_MASK GENMASK(4, 3) +#define PCM_INTF_CON1_FS_8K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 0) +#define PCM_INTF_CON1_FS_16K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 1) +#define PCM_INTF_CON1_FS_32K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 2) +#define PCM_INTF_CON1_FS_48K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 3) +#define PCM_INTF_CON1_SYNC_LEN_MASK GENMASK(13, 9) +#define PCM_INTF_CON1_SYNC_LEN(x) FIELD_PREP(PCM_INTF_CON1_SYNC_LEN_MASK, ((x) - 1)) +#define PCM_INTF_CON1_FORMAT_MASK GENMASK(2, 1) +#define PCM_INTF_CON1_SYNC_OUT_INV BIT(23) +#define PCM_INTF_CON1_BCLK_OUT_INV BIT(22) +#define PCM_INTF_CON1_SYNC_IN_INV BIT(21) +#define PCM_INTF_CON1_BCLK_IN_INV BIT(20) +#define PCM_INTF_CON1_BYPASS_ASRC BIT(6) +#define PCM_INTF_CON1_EN BIT(0) +#define PCM_INTF_CON1_CONFIG_MASK (0xf3fffe) + +/* AFE_DMIC0_UL_SRC_CON0 (0x05b4) + * AFE_DMIC1_UL_SRC_CON0 (0x0620) + * AFE_DMIC2_UL_SRC_CON0 (0x0780) + * AFE_DMIC3_UL_SRC_CON0 (0x07ec) + */ +#define DMIC_TOP_CON_CK_PHASE_SEL_CH1 GENMASK(29, 27) +#define DMIC_TOP_CON_CK_PHASE_SEL_CH2 GENMASK(26, 24) +#define DMIC_TOP_CON_TWO_WIRE_MODE BIT(23) +#define DMIC_TOP_CON_CH2_ON BIT(22) +#define DMIC_TOP_CON_CH1_ON BIT(21) +#define DMIC_TOP_CON_VOICE_MODE_MASK GENMASK(19, 17) +#define DMIC_TOP_CON_VOICE_MODE_8K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 0) +#define DMIC_TOP_CON_VOICE_MODE_16K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 1) +#define DMIC_TOP_CON_VOICE_MODE_32K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 2) +#define DMIC_TOP_CON_VOICE_MODE_48K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 3) +#define DMIC_TOP_CON_LOW_POWER_MODE_MASK GENMASK(15, 14) +#define DMIC_TOP_CON_LOW_POWER_MODE(x) FIELD_PREP(DMIC_TOP_CON_LOW_POWER_MODE_MASK, (x)) +#define DMIC_TOP_CON_IIR_ON BIT(10) +#define DMIC_TOP_CON_IIR_MODE GENMASK(9, 7) +#define DMIC_TOP_CON_INPUT_MODE BIT(5) +#define DMIC_TOP_CON_SDM3_LEVEL_MODE BIT(1) +#define DMIC_TOP_CON_SRC_ON BIT(0) +#define DMIC_TOP_CON_SDM3_DE_SELECT (0 << 1) +#define DMIC_TOP_CON_CONFIG_MASK (0x3f8ed7a6) + +/* AFE_CONN_24BIT (0x0AA4) */ +#define AFE_CONN_24BIT_O10 BIT(10) +#define AFE_CONN_24BIT_O09 BIT(9) +#define AFE_CONN_24BIT_O06 BIT(6) +#define AFE_CONN_24BIT_O05 BIT(5) +#define AFE_CONN_24BIT_O04 BIT(4) +#define AFE_CONN_24BIT_O03 BIT(3) +#define AFE_CONN_24BIT_O02 BIT(2) +#define AFE_CONN_24BIT_O01 BIT(1) +#define AFE_CONN_24BIT_O00 BIT(0) + +/* AFE_HD_ENGEN_ENABLE */ +#define AFE_22M_PLL_EN BIT(0) +#define AFE_24M_PLL_EN BIT(1) + +/* AFE_GAIN1_CON0 (0x0410) */ +#define AFE_GAIN1_CON0_EN_MASK GENMASK(0, 0) +#define AFE_GAIN1_CON0_MODE_MASK GENMASK(7, 4) +#define AFE_GAIN1_CON0_SAMPLE_PER_STEP_MASK GENMASK(15, 8) + +/* AFE_GAIN1_CON1 (0x0414) */ +#define AFE_GAIN1_CON1_MASK GENMASK(19, 0) + +/* AFE_GAIN1_CUR (0x0B78) */ +#define AFE_GAIN1_CUR_MASK GENMASK(19, 0) + +/* AFE_CM1_CON0 (0x0e50) */ +/* AFE_CM2_CON0 (0x0e60) */ +#define CM_AFE_CM_CH_NUM_MASK GENMASK(3, 0) +#define CM_AFE_CM_CH_NUM(x) FIELD_PREP(CM_AFE_CM_CH_NUM_MASK, ((x) - 1)) +#define CM_AFE_CM_ON BIT(4) +#define CM_AFE_CM_START_DATA_MASK GENMASK(11, 8) + +#define CM_AFE_CM1_VUL_SEL BIT(12) +#define CM_AFE_CM1_IN_MODE_MASK GENMASK(19, 16) +#define CM_AFE_CM2_TDM_SEL BIT(12) +#define CM_AFE_CM2_CLK_SEL BIT(13) +#define CM_AFE_CM2_GASRC1_OUT_SEL BIT(17) +#define CM_AFE_CM2_GASRC2_OUT_SEL BIT(16) + +/* AFE_CM2_CONN* */ +#define CM2_AFE_CM2_CONN_CFG1(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG1_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG1_MASK GENMASK(4, 0) +#define CM2_AFE_CM2_CONN_CFG2(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG2_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG2_MASK GENMASK(9, 5) +#define CM2_AFE_CM2_CONN_CFG3(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG3_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG3_MASK GENMASK(14, 10) +#define CM2_AFE_CM2_CONN_CFG4(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG4_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG4_MASK GENMASK(19, 15) +#define CM2_AFE_CM2_CONN_CFG5(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG5_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG5_MASK GENMASK(24, 20) +#define CM2_AFE_CM2_CONN_CFG6(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG6_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG6_MASK GENMASK(29, 25) +#define CM2_AFE_CM2_CONN_CFG7(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG7_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG7_MASK GENMASK(4, 0) +#define CM2_AFE_CM2_CONN_CFG8(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG8_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG8_MASK GENMASK(9, 5) +#define CM2_AFE_CM2_CONN_CFG9(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG9_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG9_MASK GENMASK(14, 10) +#define CM2_AFE_CM2_CONN_CFG10(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG10_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG10_MASK GENMASK(19, 15) +#define CM2_AFE_CM2_CONN_CFG11(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG11_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG11_MASK GENMASK(24, 20) +#define CM2_AFE_CM2_CONN_CFG12(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG12_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG12_MASK GENMASK(29, 25) +#define CM2_AFE_CM2_CONN_CFG13(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG13_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG13_MASK GENMASK(4, 0) +#define CM2_AFE_CM2_CONN_CFG14(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG14_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG14_MASK GENMASK(9, 5) +#define CM2_AFE_CM2_CONN_CFG15(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG15_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG15_MASK GENMASK(14, 10) +#define CM2_AFE_CM2_CONN_CFG16(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG16_MASK, (x)) +#define CM2_AFE_CM2_CONN_CFG16_MASK GENMASK(19, 15) + +/* AFE_CM1_CON* */ +#define CM_AFE_CM_UPDATE_CNT1_MASK GENMASK(15, 0) +#define CM_AFE_CM_UPDATE_CNT1(x) FIELD_PREP(CM_AFE_CM_UPDATE_CNT1_MASK, (x)) +#define CM_AFE_CM_UPDATE_CNT2_MASK GENMASK(31, 16) +#define CM_AFE_CM_UPDATE_CNT2(x) FIELD_PREP(CM_AFE_CM_UPDATE_CNT2_MASK, (x)) + +#endif From ef307b40b7f0042d54f020bccb3e728ced292282 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:34 +0200 Subject: [PATCH 312/410] ASoC: mediatek: mt8365: Add audio clock control support Add audio clock wrapper and audio tuner control. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-5-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-clk.c | 421 +++++++++++++++++++++ sound/soc/mediatek/mt8365/mt8365-afe-clk.h | 32 ++ 2 files changed, 453 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.h diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-clk.c b/sound/soc/mediatek/mt8365/mt8365-afe-clk.c new file mode 100644 index 000000000000..300d1f0ae660 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-afe-clk.c @@ -0,0 +1,421 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 AFE clock control + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include "mt8365-afe-clk.h" +#include "mt8365-afe-common.h" +#include "mt8365-reg.h" +#include "../common/mtk-base-afe.h" +#include +#include + +static const char *aud_clks[MT8365_CLK_NUM] = { + [MT8365_CLK_TOP_AUD_SEL] = "top_audio_sel", + [MT8365_CLK_AUD_I2S0_M] = "audio_i2s0_m", + [MT8365_CLK_AUD_I2S1_M] = "audio_i2s1_m", + [MT8365_CLK_AUD_I2S2_M] = "audio_i2s2_m", + [MT8365_CLK_AUD_I2S3_M] = "audio_i2s3_m", + [MT8365_CLK_ENGEN1] = "engen1", + [MT8365_CLK_ENGEN2] = "engen2", + [MT8365_CLK_AUD1] = "aud1", + [MT8365_CLK_AUD2] = "aud2", + [MT8365_CLK_I2S0_M_SEL] = "i2s0_m_sel", + [MT8365_CLK_I2S1_M_SEL] = "i2s1_m_sel", + [MT8365_CLK_I2S2_M_SEL] = "i2s2_m_sel", + [MT8365_CLK_I2S3_M_SEL] = "i2s3_m_sel", + [MT8365_CLK_CLK26M] = "top_clk26m_clk", +}; + +int mt8365_afe_init_audio_clk(struct mtk_base_afe *afe) +{ + size_t i; + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + for (i = 0; i < ARRAY_SIZE(aud_clks); i++) { + afe_priv->clocks[i] = devm_clk_get(afe->dev, aud_clks[i]); + if (IS_ERR(afe_priv->clocks[i])) { + dev_err(afe->dev, "%s devm_clk_get %s fail\n", + __func__, aud_clks[i]); + return PTR_ERR(afe_priv->clocks[i]); + } + } + return 0; +} + +void mt8365_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk) +{ + if (clk) + clk_disable_unprepare(clk); +} + +int mt8365_afe_set_clk_rate(struct mtk_base_afe *afe, struct clk *clk, + unsigned int rate) +{ + int ret; + + if (clk) { + ret = clk_set_rate(clk, rate); + if (ret) { + dev_err(afe->dev, "Failed to set rate\n"); + return ret; + } + } + return 0; +} + +int mt8365_afe_set_clk_parent(struct mtk_base_afe *afe, struct clk *clk, + struct clk *parent) +{ + int ret; + + if (clk && parent) { + ret = clk_set_parent(clk, parent); + if (ret) { + dev_err(afe->dev, "Failed to set parent\n"); + return ret; + } + } + return 0; +} + +static unsigned int get_top_cg_reg(unsigned int cg_type) +{ + switch (cg_type) { + case MT8365_TOP_CG_AFE: + case MT8365_TOP_CG_I2S_IN: + case MT8365_TOP_CG_22M: + case MT8365_TOP_CG_24M: + case MT8365_TOP_CG_INTDIR_CK: + case MT8365_TOP_CG_APLL2_TUNER: + case MT8365_TOP_CG_APLL_TUNER: + case MT8365_TOP_CG_SPDIF: + case MT8365_TOP_CG_TDM_OUT: + case MT8365_TOP_CG_TDM_IN: + case MT8365_TOP_CG_ADC: + case MT8365_TOP_CG_DAC: + case MT8365_TOP_CG_DAC_PREDIS: + case MT8365_TOP_CG_TML: + return AUDIO_TOP_CON0; + case MT8365_TOP_CG_I2S1_BCLK: + case MT8365_TOP_CG_I2S2_BCLK: + case MT8365_TOP_CG_I2S3_BCLK: + case MT8365_TOP_CG_I2S4_BCLK: + case MT8365_TOP_CG_DMIC0_ADC: + case MT8365_TOP_CG_DMIC1_ADC: + case MT8365_TOP_CG_DMIC2_ADC: + case MT8365_TOP_CG_DMIC3_ADC: + case MT8365_TOP_CG_CONNSYS_I2S_ASRC: + case MT8365_TOP_CG_GENERAL1_ASRC: + case MT8365_TOP_CG_GENERAL2_ASRC: + case MT8365_TOP_CG_TDM_ASRC: + return AUDIO_TOP_CON1; + default: + return 0; + } +} + +static unsigned int get_top_cg_mask(unsigned int cg_type) +{ + switch (cg_type) { + case MT8365_TOP_CG_AFE: + return AUD_TCON0_PDN_AFE; + case MT8365_TOP_CG_I2S_IN: + return AUD_TCON0_PDN_I2S_IN; + case MT8365_TOP_CG_22M: + return AUD_TCON0_PDN_22M; + case MT8365_TOP_CG_24M: + return AUD_TCON0_PDN_24M; + case MT8365_TOP_CG_INTDIR_CK: + return AUD_TCON0_PDN_INTDIR; + case MT8365_TOP_CG_APLL2_TUNER: + return AUD_TCON0_PDN_APLL2_TUNER; + case MT8365_TOP_CG_APLL_TUNER: + return AUD_TCON0_PDN_APLL_TUNER; + case MT8365_TOP_CG_SPDIF: + return AUD_TCON0_PDN_SPDIF; + case MT8365_TOP_CG_TDM_OUT: + return AUD_TCON0_PDN_TDM_OUT; + case MT8365_TOP_CG_TDM_IN: + return AUD_TCON0_PDN_TDM_IN; + case MT8365_TOP_CG_ADC: + return AUD_TCON0_PDN_ADC; + case MT8365_TOP_CG_DAC: + return AUD_TCON0_PDN_DAC; + case MT8365_TOP_CG_DAC_PREDIS: + return AUD_TCON0_PDN_DAC_PREDIS; + case MT8365_TOP_CG_TML: + return AUD_TCON0_PDN_TML; + case MT8365_TOP_CG_I2S1_BCLK: + return AUD_TCON1_PDN_I2S1_BCLK; + case MT8365_TOP_CG_I2S2_BCLK: + return AUD_TCON1_PDN_I2S2_BCLK; + case MT8365_TOP_CG_I2S3_BCLK: + return AUD_TCON1_PDN_I2S3_BCLK; + case MT8365_TOP_CG_I2S4_BCLK: + return AUD_TCON1_PDN_I2S4_BCLK; + case MT8365_TOP_CG_DMIC0_ADC: + return AUD_TCON1_PDN_DMIC0_ADC; + case MT8365_TOP_CG_DMIC1_ADC: + return AUD_TCON1_PDN_DMIC1_ADC; + case MT8365_TOP_CG_DMIC2_ADC: + return AUD_TCON1_PDN_DMIC2_ADC; + case MT8365_TOP_CG_DMIC3_ADC: + return AUD_TCON1_PDN_DMIC3_ADC; + case MT8365_TOP_CG_CONNSYS_I2S_ASRC: + return AUD_TCON1_PDN_CONNSYS_I2S_ASRC; + case MT8365_TOP_CG_GENERAL1_ASRC: + return AUD_TCON1_PDN_GENERAL1_ASRC; + case MT8365_TOP_CG_GENERAL2_ASRC: + return AUD_TCON1_PDN_GENERAL2_ASRC; + case MT8365_TOP_CG_TDM_ASRC: + return AUD_TCON1_PDN_TDM_ASRC; + default: + return 0; + } +} + +static unsigned int get_top_cg_on_val(unsigned int cg_type) +{ + return 0; +} + +static unsigned int get_top_cg_off_val(unsigned int cg_type) +{ + return get_top_cg_mask(cg_type); +} + +int mt8365_afe_enable_top_cg(struct mtk_base_afe *afe, unsigned int cg_type) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + unsigned int reg = get_top_cg_reg(cg_type); + unsigned int mask = get_top_cg_mask(cg_type); + unsigned int val = get_top_cg_on_val(cg_type); + unsigned long flags; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + afe_priv->top_cg_ref_cnt[cg_type]++; + if (afe_priv->top_cg_ref_cnt[cg_type] == 1) + regmap_update_bits(afe->regmap, reg, mask, val); + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +int mt8365_afe_disable_top_cg(struct mtk_base_afe *afe, unsigned int cg_type) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + unsigned int reg = get_top_cg_reg(cg_type); + unsigned int mask = get_top_cg_mask(cg_type); + unsigned int val = get_top_cg_off_val(cg_type); + unsigned long flags; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + afe_priv->top_cg_ref_cnt[cg_type]--; + if (afe_priv->top_cg_ref_cnt[cg_type] == 0) + regmap_update_bits(afe->regmap, reg, mask, val); + else if (afe_priv->top_cg_ref_cnt[cg_type] < 0) + afe_priv->top_cg_ref_cnt[cg_type] = 0; + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +int mt8365_afe_enable_main_clk(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + clk_prepare_enable(afe_priv->clocks[MT8365_CLK_TOP_AUD_SEL]); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_AFE); + mt8365_afe_enable_afe_on(afe); + + return 0; +} + +int mt8365_afe_disable_main_clk(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + mt8365_afe_disable_afe_on(afe); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_AFE); + mt8365_afe_disable_clk(afe, afe_priv->clocks[MT8365_CLK_TOP_AUD_SEL]); + + return 0; +} + +int mt8365_afe_emi_clk_on(struct mtk_base_afe *afe) +{ + return 0; +} + +int mt8365_afe_emi_clk_off(struct mtk_base_afe *afe) +{ + return 0; +} + +int mt8365_afe_enable_afe_on(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + unsigned long flags; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + afe_priv->afe_on_ref_cnt++; + if (afe_priv->afe_on_ref_cnt == 1) + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +int mt8365_afe_disable_afe_on(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + unsigned long flags; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + afe_priv->afe_on_ref_cnt--; + if (afe_priv->afe_on_ref_cnt == 0) + regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x0); + else if (afe_priv->afe_on_ref_cnt < 0) + afe_priv->afe_on_ref_cnt = 0; + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +int mt8365_afe_hd_engen_enable(struct mtk_base_afe *afe, bool apll1) +{ + if (apll1) + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_22M_PLL_EN, AFE_22M_PLL_EN); + else + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_24M_PLL_EN, AFE_24M_PLL_EN); + + return 0; +} + +int mt8365_afe_hd_engen_disable(struct mtk_base_afe *afe, bool apll1) +{ + if (apll1) + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_22M_PLL_EN, ~AFE_22M_PLL_EN); + else + regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, + AFE_24M_PLL_EN, ~AFE_24M_PLL_EN); + + return 0; +} + +int mt8365_afe_enable_apll_tuner_cfg(struct mtk_base_afe *afe, unsigned int apll) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + mutex_lock(&afe_priv->afe_clk_mutex); + + afe_priv->apll_tuner_ref_cnt[apll]++; + if (afe_priv->apll_tuner_ref_cnt[apll] != 1) { + mutex_unlock(&afe_priv->afe_clk_mutex); + return 0; + } + + if (apll == MT8365_AFE_APLL1) { + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG, + AFE_APLL_TUNER_CFG_MASK, 0x432); + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG, + AFE_APLL_TUNER_CFG_EN_MASK, 0x1); + } else { + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG1, + AFE_APLL_TUNER_CFG1_MASK, 0x434); + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG1, + AFE_APLL_TUNER_CFG1_EN_MASK, 0x1); + } + + mutex_unlock(&afe_priv->afe_clk_mutex); + return 0; +} + +int mt8365_afe_disable_apll_tuner_cfg(struct mtk_base_afe *afe, unsigned int apll) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + mutex_lock(&afe_priv->afe_clk_mutex); + + afe_priv->apll_tuner_ref_cnt[apll]--; + if (afe_priv->apll_tuner_ref_cnt[apll] == 0) { + if (apll == MT8365_AFE_APLL1) + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG, + AFE_APLL_TUNER_CFG_EN_MASK, 0x0); + else + regmap_update_bits(afe->regmap, AFE_APLL_TUNER_CFG1, + AFE_APLL_TUNER_CFG1_EN_MASK, 0x0); + + } else if (afe_priv->apll_tuner_ref_cnt[apll] < 0) { + afe_priv->apll_tuner_ref_cnt[apll] = 0; + } + + mutex_unlock(&afe_priv->afe_clk_mutex); + return 0; +} + +int mt8365_afe_enable_apll_associated_cfg(struct mtk_base_afe *afe, unsigned int apll) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + if (apll == MT8365_AFE_APLL1) { + if (clk_prepare_enable(afe_priv->clocks[MT8365_CLK_ENGEN1])) { + dev_info(afe->dev, "%s Failed to enable ENGEN1 clk\n", + __func__); + return 0; + } + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_22M); + mt8365_afe_hd_engen_enable(afe, true); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_APLL_TUNER); + mt8365_afe_enable_apll_tuner_cfg(afe, MT8365_AFE_APLL1); + } else { + if (clk_prepare_enable(afe_priv->clocks[MT8365_CLK_ENGEN2])) { + dev_info(afe->dev, "%s Failed to enable ENGEN2 clk\n", + __func__); + return 0; + } + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_24M); + mt8365_afe_hd_engen_enable(afe, false); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_APLL2_TUNER); + mt8365_afe_enable_apll_tuner_cfg(afe, MT8365_AFE_APLL2); + } + + return 0; +} + +int mt8365_afe_disable_apll_associated_cfg(struct mtk_base_afe *afe, unsigned int apll) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + if (apll == MT8365_AFE_APLL1) { + mt8365_afe_disable_apll_tuner_cfg(afe, MT8365_AFE_APLL1); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_APLL_TUNER); + mt8365_afe_hd_engen_disable(afe, true); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_22M); + clk_disable_unprepare(afe_priv->clocks[MT8365_CLK_ENGEN1]); + } else { + mt8365_afe_disable_apll_tuner_cfg(afe, MT8365_AFE_APLL2); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_APLL2_TUNER); + mt8365_afe_hd_engen_disable(afe, false); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_24M); + clk_disable_unprepare(afe_priv->clocks[MT8365_CLK_ENGEN2]); + } + + return 0; +} diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-clk.h b/sound/soc/mediatek/mt8365/mt8365-afe-clk.h new file mode 100644 index 000000000000..a6fa653f2183 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-afe-clk.h @@ -0,0 +1,32 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * MediaTek 8365 AFE clock control definitions + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#ifndef _MT8365_AFE_UTILS_H_ +#define _MT8365_AFE_UTILS_H_ + +struct mtk_base_afe; +struct clk; + +int mt8365_afe_init_audio_clk(struct mtk_base_afe *afe); +void mt8365_afe_disable_clk(struct mtk_base_afe *afe, struct clk *clk); +int mt8365_afe_set_clk_rate(struct mtk_base_afe *afe, struct clk *clk, unsigned int rate); +int mt8365_afe_set_clk_parent(struct mtk_base_afe *afe, struct clk *clk, struct clk *parent); +int mt8365_afe_enable_top_cg(struct mtk_base_afe *afe, unsigned int cg_type); +int mt8365_afe_disable_top_cg(struct mtk_base_afe *afe, unsigned int cg_type); +int mt8365_afe_enable_main_clk(struct mtk_base_afe *afe); +int mt8365_afe_disable_main_clk(struct mtk_base_afe *afe); +int mt8365_afe_emi_clk_on(struct mtk_base_afe *afe); +int mt8365_afe_emi_clk_off(struct mtk_base_afe *afe); +int mt8365_afe_enable_afe_on(struct mtk_base_afe *afe); +int mt8365_afe_disable_afe_on(struct mtk_base_afe *afe); +int mt8365_afe_enable_apll_tuner_cfg(struct mtk_base_afe *afe, unsigned int apll); +int mt8365_afe_disable_apll_tuner_cfg(struct mtk_base_afe *afe, unsigned int apll); +int mt8365_afe_enable_apll_associated_cfg(struct mtk_base_afe *afe, unsigned int apll); +int mt8365_afe_disable_apll_associated_cfg(struct mtk_base_afe *afe, unsigned int apll); +#endif From 402bbb13a195caa83b3279ebecdabfb11ddee084 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:35 +0200 Subject: [PATCH 313/410] ASoC: mediatek: mt8365: Add I2S DAI support Add I2S Device Audio Interface support for MT8365 SoC. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-6-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 850 +++++++++++++++++++++ 1 file changed, 850 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-i2s.c diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c new file mode 100644 index 000000000000..5003fe5e5ccf --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c @@ -0,0 +1,850 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 ALSA SoC Audio DAI I2S Control + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include +#include +#include +#include "mt8365-afe-clk.h" +#include "mt8365-afe-common.h" + +#define IIR_RATIOVER 9 +#define IIR_INV_COEF 10 +#define IIR_NO_NEED 11 + +struct mtk_afe_i2s_priv { + bool adda_link; + int i2s_out_on_ref_cnt; + int id; + int low_jitter_en; + int mclk_id; + int share_i2s_id; + unsigned int clk_id_in; + unsigned int clk_id_in_m_sel; + unsigned int clk_id_out; + unsigned int clk_id_out_m_sel; + unsigned int clk_in_mult; + unsigned int clk_out_mult; + unsigned int config_val_in; + unsigned int config_val_out; + unsigned int dynamic_bck; + unsigned int reg_off_in; + unsigned int reg_off_out; +}; + +/* This enum is merely for mtk_afe_i2s_priv declare */ +enum { + DAI_I2S0 = 0, + DAI_I2S3, + DAI_I2S_NUM, +}; + +static const struct mtk_afe_i2s_priv mt8365_i2s_priv[DAI_I2S_NUM] = { + [DAI_I2S0] = { + .id = MT8365_AFE_IO_I2S, + .mclk_id = MT8365_I2S0_MCK, + .share_i2s_id = -1, + .clk_id_in = MT8365_CLK_AUD_I2S2_M, + .clk_id_out = MT8365_CLK_AUD_I2S1_M, + .clk_id_in_m_sel = MT8365_CLK_I2S2_M_SEL, + .clk_id_out_m_sel = MT8365_CLK_I2S1_M_SEL, + .clk_in_mult = 256, + .clk_out_mult = 256, + .adda_link = true, + .config_val_out = AFE_I2S_CON1_I2S2_TO_PAD, + .reg_off_in = AFE_I2S_CON2, + .reg_off_out = AFE_I2S_CON1, + }, + [DAI_I2S3] = { + .id = MT8365_AFE_IO_2ND_I2S, + .mclk_id = MT8365_I2S3_MCK, + .share_i2s_id = -1, + .clk_id_in = MT8365_CLK_AUD_I2S0_M, + .clk_id_out = MT8365_CLK_AUD_I2S3_M, + .clk_id_in_m_sel = MT8365_CLK_I2S0_M_SEL, + .clk_id_out_m_sel = MT8365_CLK_I2S3_M_SEL, + .clk_in_mult = 256, + .clk_out_mult = 256, + .adda_link = false, + .config_val_in = AFE_I2S_CON_FROM_IO_MUX, + .reg_off_in = AFE_I2S_CON, + .reg_off_out = AFE_I2S_CON3, + }, +}; + +static const u32 *get_iir_coef(unsigned int input_fs, + unsigned int output_fs, unsigned int *count) +{ + static const u32 IIR_COEF_48_TO_44p1[30] = { + 0x061fb0, 0x0bd256, 0x061fb0, 0xe3a3e6, 0xf0a300, 0x000003, + 0x0e416d, 0x1bb577, 0x0e416d, 0xe59178, 0xf23637, 0x000003, + 0x0c7d72, 0x189060, 0x0c7d72, 0xe96f09, 0xf505b2, 0x000003, + 0x126054, 0x249143, 0x126054, 0xe1fc0c, 0xf4b20a, 0x000002, + 0x000000, 0x323c85, 0x323c85, 0xf76d4e, 0x000000, 0x000002, + }; + + static const u32 IIR_COEF_44p1_TO_32[42] = { + 0x0a6074, 0x0d237a, 0x0a6074, 0xdd8d6c, 0xe0b3f6, 0x000002, + 0x0e41f8, 0x128d48, 0x0e41f8, 0xefc14e, 0xf12d7a, 0x000003, + 0x0cfa60, 0x11e89c, 0x0cfa60, 0xf1b09e, 0xf27205, 0x000003, + 0x15b69c, 0x20e7e4, 0x15b69c, 0xea799a, 0xe9314a, 0x000002, + 0x0f79e2, 0x1a7064, 0x0f79e2, 0xf65e4a, 0xf03d8e, 0x000002, + 0x10c34f, 0x1ffe4b, 0x10c34f, 0x0bbecb, 0xf2bc4b, 0x000001, + 0x000000, 0x23b063, 0x23b063, 0x07335f, 0x000000, 0x000002, + }; + + static const u32 IIR_COEF_48_TO_32[42] = { + 0x0a2a9b, 0x0a2f05, 0x0a2a9b, 0xe73873, 0xe0c525, 0x000002, + 0x0dd4ad, 0x0e765a, 0x0dd4ad, 0xf49808, 0xf14844, 0x000003, + 0x18a8cd, 0x1c40d0, 0x18a8cd, 0xed2aab, 0xe542ec, 0x000002, + 0x13e044, 0x1a47c4, 0x13e044, 0xf44aed, 0xe9acc7, 0x000002, + 0x1abd9c, 0x2a5429, 0x1abd9c, 0xff3441, 0xe0fc5f, 0x000001, + 0x0d86db, 0x193e2e, 0x0d86db, 0x1a6f15, 0xf14507, 0x000001, + 0x000000, 0x1f820c, 0x1f820c, 0x0a1b1f, 0x000000, 0x000002, + }; + + static const u32 IIR_COEF_32_TO_16[48] = { + 0x122893, 0xffadd4, 0x122893, 0x0bc205, 0xc0ee1c, 0x000001, + 0x1bab8a, 0x00750d, 0x1bab8a, 0x06a983, 0xe18a5c, 0x000002, + 0x18f68e, 0x02706f, 0x18f68e, 0x0886a9, 0xe31bcb, 0x000002, + 0x149c05, 0x054487, 0x149c05, 0x0bec31, 0xe5973e, 0x000002, + 0x0ea303, 0x07f24a, 0x0ea303, 0x115ff9, 0xe967b6, 0x000002, + 0x0823fd, 0x085531, 0x0823fd, 0x18d5b4, 0xee8d21, 0x000002, + 0x06888e, 0x0acbbb, 0x06888e, 0x40b55c, 0xe76dce, 0x000001, + 0x000000, 0x2d31a9, 0x2d31a9, 0x23ba4f, 0x000000, 0x000001, + }; + + static const u32 IIR_COEF_96_TO_44p1[48] = { + 0x08b543, 0xfd80f4, 0x08b543, 0x0e2332, 0xe06ed0, 0x000002, + 0x1b6038, 0xf90e7e, 0x1b6038, 0x0ec1ac, 0xe16f66, 0x000002, + 0x188478, 0xfbb921, 0x188478, 0x105859, 0xe2e596, 0x000002, + 0x13eff3, 0xffa707, 0x13eff3, 0x13455c, 0xe533b7, 0x000002, + 0x0dc239, 0x03d458, 0x0dc239, 0x17f120, 0xe8b617, 0x000002, + 0x0745f1, 0x05d790, 0x0745f1, 0x1e3d75, 0xed5f18, 0x000002, + 0x05641f, 0x085e2b, 0x05641f, 0x48efd0, 0xe3e9c8, 0x000001, + 0x000000, 0x28f632, 0x28f632, 0x273905, 0x000000, 0x000001, + }; + + static const u32 IIR_COEF_44p1_TO_16[48] = { + 0x0998fb, 0xf7f925, 0x0998fb, 0x1e54a0, 0xe06605, 0x000002, + 0x0d828e, 0xf50f97, 0x0d828e, 0x0f41b5, 0xf0a999, 0x000003, + 0x17ebeb, 0xee30d8, 0x17ebeb, 0x1f48ca, 0xe2ae88, 0x000002, + 0x12fab5, 0xf46ddc, 0x12fab5, 0x20cc51, 0xe4d068, 0x000002, + 0x0c7ac6, 0xfbd00e, 0x0c7ac6, 0x2337da, 0xe8028c, 0x000002, + 0x060ddc, 0x015b3e, 0x060ddc, 0x266754, 0xec21b6, 0x000002, + 0x0407b5, 0x04f827, 0x0407b5, 0x52e3d0, 0xe0149f, 0x000001, + 0x000000, 0x1f9521, 0x1f9521, 0x2ac116, 0x000000, 0x000001, + }; + + static const u32 IIR_COEF_48_TO_16[48] = { + 0x0955ff, 0xf6544a, 0x0955ff, 0x2474e5, 0xe062e6, 0x000002, + 0x0d4180, 0xf297f4, 0x0d4180, 0x12415b, 0xf0a3b0, 0x000003, + 0x0ba079, 0xf4f0b0, 0x0ba079, 0x1285d3, 0xf1488b, 0x000003, + 0x12247c, 0xf1033c, 0x12247c, 0x2625be, 0xe48e0d, 0x000002, + 0x0b98e0, 0xf96d1a, 0x0b98e0, 0x27e79c, 0xe7798a, 0x000002, + 0x055e3b, 0xffed09, 0x055e3b, 0x2a2e2d, 0xeb2854, 0x000002, + 0x01a934, 0x01ca03, 0x01a934, 0x2c4fea, 0xee93ab, 0x000002, + 0x000000, 0x1c46c5, 0x1c46c5, 0x2d37dc, 0x000000, 0x000001, + }; + + static const u32 IIR_COEF_96_TO_16[48] = { + 0x0805a1, 0xf21ae3, 0x0805a1, 0x3840bb, 0xe02a2e, 0x000002, + 0x0d5dd8, 0xe8f259, 0x0d5dd8, 0x1c0af6, 0xf04700, 0x000003, + 0x0bb422, 0xec08d9, 0x0bb422, 0x1bfccc, 0xf09216, 0x000003, + 0x08fde6, 0xf108be, 0x08fde6, 0x1bf096, 0xf10ae0, 0x000003, + 0x0ae311, 0xeeeda3, 0x0ae311, 0x37c646, 0xe385f5, 0x000002, + 0x044089, 0xfa7242, 0x044089, 0x37a785, 0xe56526, 0x000002, + 0x00c75c, 0xffb947, 0x00c75c, 0x378ba3, 0xe72c5f, 0x000002, + 0x000000, 0x0ef76e, 0x0ef76e, 0x377fda, 0x000000, 0x000001, + }; + + static const struct { + const u32 *coef; + unsigned int cnt; + } iir_coef_tbl_list[8] = { + /* 0: 0.9188 */ + { IIR_COEF_48_TO_44p1, ARRAY_SIZE(IIR_COEF_48_TO_44p1) }, + /* 1: 0.7256 */ + { IIR_COEF_44p1_TO_32, ARRAY_SIZE(IIR_COEF_44p1_TO_32) }, + /* 2: 0.6667 */ + { IIR_COEF_48_TO_32, ARRAY_SIZE(IIR_COEF_48_TO_32) }, + /* 3: 0.5 */ + { IIR_COEF_32_TO_16, ARRAY_SIZE(IIR_COEF_32_TO_16) }, + /* 4: 0.4594 */ + { IIR_COEF_96_TO_44p1, ARRAY_SIZE(IIR_COEF_96_TO_44p1) }, + /* 5: 0.3628 */ + { IIR_COEF_44p1_TO_16, ARRAY_SIZE(IIR_COEF_44p1_TO_16) }, + /* 6: 0.3333 */ + { IIR_COEF_48_TO_16, ARRAY_SIZE(IIR_COEF_48_TO_16) }, + /* 7: 0.1667 */ + { IIR_COEF_96_TO_16, ARRAY_SIZE(IIR_COEF_96_TO_16) }, + }; + + static const u32 freq_new_index[16] = { + 0, 1, 2, 99, 3, 4, 5, 99, 6, 7, 8, 9, 10, 11, 12, 99 + }; + + static const u32 iir_coef_tbl_matrix[13][13] = { + {/*0*/ + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*1*/ + 1, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*2*/ + 2, 0, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*3*/ + 3, IIR_INV_COEF, IIR_INV_COEF, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*4*/ + 5, 3, IIR_INV_COEF, 2, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*5*/ + 6, 4, 3, 2, 0, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED + }, + {/*6*/ + IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, 3, IIR_INV_COEF, + IIR_INV_COEF, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*7*/ + IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, 5, 3, + IIR_INV_COEF, 1, IIR_NO_NEED, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*8*/ + 7, IIR_INV_COEF, IIR_INV_COEF, 6, 4, 3, 2, 0, IIR_NO_NEED, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*9*/ + IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, + IIR_INV_COEF, IIR_INV_COEF, 5, 3, IIR_INV_COEF, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + {/*10*/ + IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, 7, IIR_INV_COEF, + IIR_INV_COEF, 6, 4, 3, 0, + IIR_NO_NEED, IIR_NO_NEED, IIR_NO_NEED + }, + { /*11*/ + IIR_RATIOVER, IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, + IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, IIR_INV_COEF, + IIR_INV_COEF, 3, IIR_INV_COEF, IIR_NO_NEED, IIR_NO_NEED + }, + {/*12*/ + IIR_RATIOVER, IIR_RATIOVER, IIR_INV_COEF, IIR_INV_COEF, + IIR_INV_COEF, IIR_INV_COEF, 7, IIR_INV_COEF, + IIR_INV_COEF, 4, 3, 0, IIR_NO_NEED + }, + }; + + const u32 *coef = NULL; + unsigned int cnt = 0; + u32 i = freq_new_index[input_fs]; + u32 j = freq_new_index[output_fs]; + + if (i < 13 && j < 13) { + u32 k = iir_coef_tbl_matrix[i][j]; + + if (k >= IIR_NO_NEED) { + } else if (k == IIR_RATIOVER) { + } else if (k == IIR_INV_COEF) { + } else { + coef = iir_coef_tbl_list[k].coef; + cnt = iir_coef_tbl_list[k].cnt; + } + } + *count = cnt; + return coef; +} + +static int mt8365_dai_set_config(struct mtk_base_afe *afe, + struct mtk_afe_i2s_priv *i2s_data, + bool is_input, unsigned int rate, + int bit_width) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = + &afe_priv->be_data[i2s_data->id - MT8365_AFE_BACKEND_BASE]; + unsigned int val, reg_off; + int fs = mt8365_afe_fs_timing(rate); + + if (fs < 0) + return -EINVAL; + + val = AFE_I2S_CON_LOW_JITTER_CLK | AFE_I2S_CON_FORMAT_I2S; + val |= FIELD_PREP(AFE_I2S_CON_RATE_MASK, fs); + + if (is_input) { + reg_off = i2s_data->reg_off_in; + if (i2s_data->adda_link) + val |= i2s_data->config_val_in; + } else { + reg_off = i2s_data->reg_off_out; + val |= i2s_data->config_val_in; + } + + /* 1:bck=32lrck(16bit) or bck=64lrck(32bit) 0:fix bck=64lrck */ + if (i2s_data->dynamic_bck) { + if (bit_width > 16) + val |= AFE_I2S_CON_WLEN_32BIT; + else + val &= ~(u32)AFE_I2S_CON_WLEN_32BIT; + } else { + val |= AFE_I2S_CON_WLEN_32BIT; + } + + if ((be->fmt_mode & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFM) { + val |= AFE_I2S_CON_SRC_SLAVE; + val &= ~(u32)AFE_I2S_CON_FROM_IO_MUX;//from consys + } + + regmap_update_bits(afe->regmap, reg_off, ~(u32)AFE_I2S_CON_EN, val); + + if (i2s_data->adda_link && is_input) + regmap_update_bits(afe->regmap, AFE_ADDA_TOP_CON0, 0x1, 0x1); + + return 0; +} + +int mt8365_afe_set_i2s_out(struct mtk_base_afe *afe, + unsigned int rate, int bit_width) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_data = + afe_priv->dai_priv[MT8365_AFE_IO_I2S]; + + return mt8365_dai_set_config(afe, i2s_data, false, rate, bit_width); +} + +static int mt8365_afe_set_2nd_i2s_asrc(struct mtk_base_afe *afe, + unsigned int rate_in, + unsigned int rate_out, + unsigned int width, + unsigned int mono, + int o16bit, int tracking) +{ + int ifs, ofs = 0; + unsigned int val = 0; + unsigned int mask = 0; + const u32 *coef; + u32 iir_stage; + unsigned int coef_count = 0; + + ifs = mt8365_afe_fs_timing(rate_in); + + if (ifs < 0) + return -EINVAL; + + ofs = mt8365_afe_fs_timing(rate_out); + + if (ofs < 0) + return -EINVAL; + + val = FIELD_PREP(O16BIT, o16bit) | FIELD_PREP(IS_MONO, mono); + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, + O16BIT | IS_MONO, val); + + coef = get_iir_coef(ifs, ofs, &coef_count); + iir_stage = ((u32)coef_count / 6) - 1; + + if (coef) { + unsigned int i; + + /* CPU control IIR coeff SRAM */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + COEFF_SRAM_CTRL, COEFF_SRAM_CTRL); + + /* set to 0, IIR coeff SRAM addr */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON13, + 0xffffffff, 0x0); + + for (i = 0; i < coef_count; ++i) + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON12, + 0xffffffff, coef[i]); + + /* disable IIR coeff SRAM access */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + COEFF_SRAM_CTRL, + (unsigned long)~COEFF_SRAM_CTRL); + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, + CLR_IIR_HISTORY | IIR_EN | IIR_STAGE_MASK, + CLR_IIR_HISTORY | IIR_EN | + FIELD_PREP(IIR_STAGE_MASK, iir_stage)); + } else { + /* disable IIR */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, + IIR_EN, (unsigned long)~IIR_EN); + } + + /* CON3 setting (RX OFS) */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON3, + 0x00FFFFFF, rx_frequency_palette(ofs)); + /* CON4 setting (RX IFS) */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON4, + 0x00FFFFFF, rx_frequency_palette(ifs)); + + /* CON5 setting */ + if (tracking) { + val = CALI_64_CYCLE | + CALI_AUTORST | + AUTO_TUNE_FREQ5 | + COMP_FREQ_RES | + CALI_BP_DGL | + CALI_AUTO_RESTART | + CALI_USE_FREQ_OUT | + CALI_SEL_01; + + mask = CALI_CYCLE_MASK | + CALI_AUTORST | + AUTO_TUNE_FREQ5 | + COMP_FREQ_RES | + CALI_SEL_MASK | + CALI_BP_DGL | + AUTO_TUNE_FREQ4 | + CALI_AUTO_RESTART | + CALI_USE_FREQ_OUT | + CALI_ON; + + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON5, + mask, val); + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON5, + CALI_ON, CALI_ON); + } else { + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON5, + 0xffffffff, 0x0); + } + /* CON6 setting fix 8125 */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON6, + 0x0000ffff, 0x1FBD); + /* CON9 setting (RX IFS) */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON9, + 0x000fffff, AutoRstThHi(ifs)); + /* CON10 setting (RX IFS) */ + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON10, + 0x000fffff, AutoRstThLo(ifs)); + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + CHSET_STR_CLR, CHSET_STR_CLR); + + return 0; +} + +static int mt8365_afe_set_2nd_i2s_asrc_enable(struct mtk_base_afe *afe, + bool enable) +{ + if (enable) + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + ASM_ON, ASM_ON); + else + regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, + ASM_ON, (unsigned long)~ASM_ON); + return 0; +} + +void mt8365_afe_set_i2s_out_enable(struct mtk_base_afe *afe, bool enable) +{ + int i; + unsigned long flags; + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_data; + + for (i = 0; i < DAI_I2S_NUM; i++) { + if (mt8365_i2s_priv[i].adda_link) + i2s_data = afe_priv->dai_priv[mt8365_i2s_priv[i].id]; + } + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + if (enable) { + i2s_data->i2s_out_on_ref_cnt++; + if (i2s_data->i2s_out_on_ref_cnt == 1) + regmap_update_bits(afe->regmap, AFE_I2S_CON1, + 0x1, enable); + } else { + i2s_data->i2s_out_on_ref_cnt--; + if (i2s_data->i2s_out_on_ref_cnt == 0) + regmap_update_bits(afe->regmap, AFE_I2S_CON1, + 0x1, enable); + else if (i2s_data->i2s_out_on_ref_cnt < 0) + i2s_data->i2s_out_on_ref_cnt = 0; + } + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); +} + +static void mt8365_dai_set_enable(struct mtk_base_afe *afe, + struct mtk_afe_i2s_priv *i2s_data, + bool is_input, bool enable) +{ + unsigned int reg_off; + + if (is_input) { + reg_off = i2s_data->reg_off_in; + } else { + if (i2s_data->adda_link) { + mt8365_afe_set_i2s_out_enable(afe, enable); + return; + } + reg_off = i2s_data->reg_off_out; + } + regmap_update_bits(afe->regmap, reg_off, + 0x1, enable); +} + +static int mt8365_dai_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_data = afe_priv->dai_priv[dai->id]; + struct mt8365_be_dai_data *be = &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + bool i2s_in_slave = + (substream->stream == SNDRV_PCM_STREAM_CAPTURE) && + ((be->fmt_mode & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFM); + + mt8365_afe_enable_main_clk(afe); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + clk_prepare_enable(afe_priv->clocks[i2s_data->clk_id_out]); + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && !i2s_in_slave) + clk_prepare_enable(afe_priv->clocks[i2s_data->clk_id_in]); + + if (i2s_in_slave) + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_I2S_IN); + + return 0; +} + +static void mt8365_dai_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_data = afe_priv->dai_priv[dai->id]; + struct mt8365_be_dai_data *be = &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + bool reset_i2s_out_change = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool reset_i2s_in_change = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + bool i2s_in_slave = + (substream->stream == SNDRV_PCM_STREAM_CAPTURE) && + ((be->fmt_mode & SND_SOC_DAIFMT_MASTER_MASK) == + SND_SOC_DAIFMT_CBM_CFM); + + if (be->prepared[substream->stream]) { + if (reset_i2s_out_change) + mt8365_dai_set_enable(afe, i2s_data, false, false); + + if (reset_i2s_in_change) + mt8365_dai_set_enable(afe, i2s_data, true, false); + + if (substream->runtime->rate % 8000) + mt8365_afe_disable_apll_associated_cfg(afe, MT8365_AFE_APLL1); + else + mt8365_afe_disable_apll_associated_cfg(afe, MT8365_AFE_APLL2); + + if (reset_i2s_out_change) + be->prepared[SNDRV_PCM_STREAM_PLAYBACK] = false; + + if (reset_i2s_in_change) + be->prepared[SNDRV_PCM_STREAM_CAPTURE] = false; + } + + if (reset_i2s_out_change) + mt8365_afe_disable_clk(afe, + afe_priv->clocks[i2s_data->clk_id_out]); + + if (reset_i2s_in_change && !i2s_in_slave) + mt8365_afe_disable_clk(afe, + afe_priv->clocks[i2s_data->clk_id_in]); + + if (i2s_in_slave) + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_I2S_IN); + + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_dai_i2s_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mtk_afe_i2s_priv *i2s_data = afe_priv->dai_priv[dai->id]; + struct mt8365_be_dai_data *be = &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + bool apply_i2s_out_change = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + bool apply_i2s_in_change = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned int rate = substream->runtime->rate; + int bit_width = snd_pcm_format_width(substream->runtime->format); + int ret; + + if (be->prepared[substream->stream]) { + dev_info(afe->dev, "%s '%s' prepared already\n", + __func__, snd_pcm_stream_str(substream)); + return 0; + } + + if (apply_i2s_out_change) { + ret = mt8365_dai_set_config(afe, i2s_data, false, rate, bit_width); + if (ret) + return ret; + } + + if (apply_i2s_in_change) { + if ((be->fmt_mode & SND_SOC_DAIFMT_MASTER_MASK) + == SND_SOC_DAIFMT_CBM_CFM) { + ret = mt8365_afe_set_2nd_i2s_asrc(afe, 32000, rate, + (unsigned int)bit_width, + 0, 0, 1); + if (ret < 0) + return ret; + } + ret = mt8365_dai_set_config(afe, i2s_data, true, rate, bit_width); + if (ret) + return ret; + } + + if (rate % 8000) + mt8365_afe_enable_apll_associated_cfg(afe, MT8365_AFE_APLL1); + else + mt8365_afe_enable_apll_associated_cfg(afe, MT8365_AFE_APLL2); + + if (apply_i2s_out_change) { + mt8365_afe_set_clk_parent(afe, + afe_priv->clocks[i2s_data->clk_id_out_m_sel], + ((rate % 8000) ? + afe_priv->clocks[MT8365_CLK_AUD1] : + afe_priv->clocks[MT8365_CLK_AUD2])); + + mt8365_afe_set_clk_rate(afe, + afe_priv->clocks[i2s_data->clk_id_out], + rate * i2s_data->clk_out_mult); + + mt8365_dai_set_enable(afe, i2s_data, false, true); + be->prepared[SNDRV_PCM_STREAM_PLAYBACK] = true; + } + + if (apply_i2s_in_change) { + mt8365_afe_set_clk_parent(afe, + afe_priv->clocks[i2s_data->clk_id_in_m_sel], + ((rate % 8000) ? + afe_priv->clocks[MT8365_CLK_AUD1] : + afe_priv->clocks[MT8365_CLK_AUD2])); + + mt8365_afe_set_clk_rate(afe, + afe_priv->clocks[i2s_data->clk_id_in], + rate * i2s_data->clk_in_mult); + + mt8365_dai_set_enable(afe, i2s_data, true, true); + + if ((be->fmt_mode & SND_SOC_DAIFMT_MASTER_MASK) + == SND_SOC_DAIFMT_CBM_CFM) + mt8365_afe_set_2nd_i2s_asrc_enable(afe, true); + + be->prepared[SNDRV_PCM_STREAM_CAPTURE] = true; + } + return 0; +} + +static int mt8365_afe_2nd_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + unsigned int width_val = params_width(params) > 16 ? + (AFE_CONN_24BIT_O00 | AFE_CONN_24BIT_O01) : 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_update_bits(afe->regmap, AFE_CONN_24BIT, + AFE_CONN_24BIT_O00 | AFE_CONN_24BIT_O01, width_val); + + return 0; +} + +static int mt8365_afe_2nd_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + + be->fmt_mode = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + be->fmt_mode |= SND_SOC_DAIFMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + be->fmt_mode |= SND_SOC_DAIFMT_LEFT_J; + break; + default: + dev_err(afe->dev, "invalid audio format for 2nd i2s!\n"); + return -EINVAL; + } + + if (((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_NF) && + ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_NB_IF) && + ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_IB_NF) && + ((fmt & SND_SOC_DAIFMT_INV_MASK) != SND_SOC_DAIFMT_IB_IF)) { + dev_err(afe->dev, "invalid audio format for 2nd i2s!\n"); + return -EINVAL; + } + + be->fmt_mode |= (fmt & SND_SOC_DAIFMT_INV_MASK); + + if (((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)) + be->fmt_mode |= (fmt & SND_SOC_DAIFMT_MASTER_MASK); + + return 0; +} + +static const struct snd_soc_dai_ops mt8365_afe_i2s_ops = { + .startup = mt8365_dai_i2s_startup, + .shutdown = mt8365_dai_i2s_shutdown, + .prepare = mt8365_dai_i2s_prepare, +}; + +static const struct snd_soc_dai_ops mt8365_afe_2nd_i2s_ops = { + .startup = mt8365_dai_i2s_startup, + .shutdown = mt8365_dai_i2s_shutdown, + .hw_params = mt8365_afe_2nd_i2s_hw_params, + .prepare = mt8365_dai_i2s_prepare, + .set_fmt = mt8365_afe_2nd_i2s_set_fmt, +}; + +static struct snd_soc_dai_driver mtk_dai_i2s_driver[] = { + { + .name = "I2S", + .id = MT8365_AFE_IO_I2S, + .playback = { + .stream_name = "I2S Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "I2S Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_i2s_ops, + }, { + .name = "2ND I2S", + .id = MT8365_AFE_IO_2ND_I2S, + .playback = { + .stream_name = "2ND I2S Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "2ND I2S Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_2nd_i2s_ops, + } +}; + +/* low jitter control */ +static const char * const mt8365_i2s_hd_str[] = { + "Normal", "Low_Jitter" +}; + +static SOC_ENUM_SINGLE_EXT_DECL(mt8365_i2s_enum, mt8365_i2s_hd_str); + +static const char * const fmi2sin_text[] = { + "OPEN", "FM_2ND_I2S_IN" +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(fmi2sin_enum, fmi2sin_text); + +static const struct snd_kcontrol_new fmi2sin_mux = + SOC_DAPM_ENUM("FM 2ND I2S Source", fmi2sin_enum); + +static const struct snd_kcontrol_new i2s_o03_o04_enable_ctl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +static const struct snd_soc_dapm_widget mtk_dai_i2s_widgets[] = { + SND_SOC_DAPM_SWITCH("I2S O03_O04", SND_SOC_NOPM, 0, 0, + &i2s_o03_o04_enable_ctl), + SND_SOC_DAPM_MUX("FM 2ND I2S Mux", SND_SOC_NOPM, 0, 0, &fmi2sin_mux), + SND_SOC_DAPM_INPUT("2ND I2S In"), +}; + +static const struct snd_soc_dapm_route mtk_dai_i2s_routes[] = { + {"I2S O03_O04", "Switch", "O03"}, + {"I2S O03_O04", "Switch", "O04"}, + {"I2S Playback", NULL, "I2S O03_O04"}, + {"2ND I2S Playback", NULL, "O00"}, + {"2ND I2S Playback", NULL, "O01"}, + {"2ND I2S Capture", NULL, "2ND I2S In"}, + {"FM 2ND I2S Mux", "FM_2ND_I2S_IN", "2ND I2S Capture"}, +}; + +static int mt8365_dai_i2s_set_priv(struct mtk_base_afe *afe) +{ + int i, ret; + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + for (i = 0; i < DAI_I2S_NUM; i++) { + ret = mt8365_dai_set_priv(afe, mt8365_i2s_priv[i].id, + sizeof(*afe_priv), + &mt8365_i2s_priv[i]); + if (ret) + return ret; + } + return 0; +} + +int mt8365_dai_i2s_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mtk_dai_i2s_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_i2s_driver); + dai->dapm_widgets = mtk_dai_i2s_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_i2s_widgets); + dai->dapm_routes = mtk_dai_i2s_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_i2s_routes); + + /* set all dai i2s private data */ + return mt8365_dai_i2s_set_priv(afe); +} From 7c58c88e524180e8439acdfc44872325e7f6d33d Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:36 +0200 Subject: [PATCH 314/410] ASoC: mediatek: mt8365: Add ADDA DAI support Add ADDA Device Audio Interface support for MT8365 SoC. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-7-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-adda.c | 311 ++++++++++++++++++++ 1 file changed, 311 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-adda.c diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-adda.c b/sound/soc/mediatek/mt8365/mt8365-dai-adda.c new file mode 100644 index 000000000000..a04c24bbfcff --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-dai-adda.c @@ -0,0 +1,311 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 ALSA SoC Audio DAI ADDA Control + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include +#include +#include +#include "mt8365-afe-clk.h" +#include "mt8365-afe-common.h" +#include "../common/mtk-dai-adda-common.h" + +static int adda_afe_on_ref_cnt; + +/* DAI Drivers */ + +static int mt8365_dai_set_adda_out(struct mtk_base_afe *afe, unsigned int rate) +{ + unsigned int val; + + if (rate == 8000 || rate == 16000) + val = AFE_ADDA_DL_VOICE_DATA; + else + val = 0; + + val |= FIELD_PREP(AFE_ADDA_DL_SAMPLING_RATE, + mtk_adda_dl_rate_transform(afe, rate)); + val |= AFE_ADDA_DL_8X_UPSAMPLE | + AFE_ADDA_DL_MUTE_OFF_CH1 | + AFE_ADDA_DL_MUTE_OFF_CH2 | + AFE_ADDA_DL_DEGRADE_GAIN; + + regmap_update_bits(afe->regmap, AFE_ADDA_PREDIS_CON0, 0xffffffff, 0); + regmap_update_bits(afe->regmap, AFE_ADDA_PREDIS_CON1, 0xffffffff, 0); + regmap_update_bits(afe->regmap, AFE_ADDA_DL_SRC2_CON0, 0xffffffff, val); + /* SA suggest apply -0.3db to audio/speech path */ + regmap_update_bits(afe->regmap, AFE_ADDA_DL_SRC2_CON1, + 0xffffffff, 0xf74f0000); + /* SA suggest use default value for sdm */ + regmap_update_bits(afe->regmap, AFE_ADDA_DL_SDM_DCCOMP_CON, + 0xffffffff, 0x0700701e); + + return 0; +} + +static int mt8365_dai_set_adda_in(struct mtk_base_afe *afe, unsigned int rate) +{ + unsigned int val; + + val = FIELD_PREP(AFE_ADDA_UL_SAMPLING_RATE, + mtk_adda_ul_rate_transform(afe, rate)); + regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0, + AFE_ADDA_UL_SAMPLING_RATE, val); + /* Using Internal ADC */ + regmap_update_bits(afe->regmap, AFE_ADDA_TOP_CON0, 0x1, 0x0); + + return 0; +} + +int mt8365_dai_enable_adda_on(struct mtk_base_afe *afe) +{ + unsigned long flags; + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + adda_afe_on_ref_cnt++; + if (adda_afe_on_ref_cnt == 1) + regmap_update_bits(afe->regmap, AFE_ADDA_UL_DL_CON0, + AFE_ADDA_UL_DL_ADDA_AFE_ON, + AFE_ADDA_UL_DL_ADDA_AFE_ON); + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +int mt8365_dai_disable_adda_on(struct mtk_base_afe *afe) +{ + unsigned long flags; + struct mt8365_afe_private *afe_priv = afe->platform_priv; + + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); + + adda_afe_on_ref_cnt--; + if (adda_afe_on_ref_cnt == 0) + regmap_update_bits(afe->regmap, AFE_ADDA_UL_DL_CON0, + AFE_ADDA_UL_DL_ADDA_AFE_ON, + ~AFE_ADDA_UL_DL_ADDA_AFE_ON); + else if (adda_afe_on_ref_cnt < 0) { + adda_afe_on_ref_cnt = 0; + dev_warn(afe->dev, "Abnormal adda_on ref count. Force it to 0\n"); + } + + spin_unlock_irqrestore(&afe_priv->afe_ctrl_lock, flags); + + return 0; +} + +static void mt8365_dai_set_adda_out_enable(struct mtk_base_afe *afe, + bool enable) +{ + regmap_update_bits(afe->regmap, AFE_ADDA_DL_SRC2_CON0, 0x1, enable); + + if (enable) + mt8365_dai_enable_adda_on(afe); + else + mt8365_dai_disable_adda_on(afe); +} + +static void mt8365_dai_set_adda_in_enable(struct mtk_base_afe *afe, bool enable) +{ + if (enable) { + regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0, 0x1, 0x1); + mt8365_dai_enable_adda_on(afe); + /* enable aud_pad_top fifo */ + regmap_update_bits(afe->regmap, AFE_AUD_PAD_TOP, + 0xffffffff, 0x31); + } else { + /* disable aud_pad_top fifo */ + regmap_update_bits(afe->regmap, AFE_AUD_PAD_TOP, + 0xffffffff, 0x30); + regmap_update_bits(afe->regmap, AFE_ADDA_UL_SRC_CON0, 0x1, 0x0); + /* de suggest disable ADDA_UL_SRC at least wait 125us */ + usleep_range(150, 300); + mt8365_dai_disable_adda_on(afe); + } +} + +static int mt8365_dai_int_adda_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + unsigned int stream = substream->stream; + + mt8365_afe_enable_main_clk(afe); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DAC); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DAC_PREDIS); + } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_ADC); + } + + return 0; +} + +static void mt8365_dai_int_adda_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = + &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + unsigned int stream = substream->stream; + + if (be->prepared[stream]) { + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mt8365_dai_set_adda_out_enable(afe, false); + mt8365_afe_set_i2s_out_enable(afe, false); + } else { + mt8365_dai_set_adda_in_enable(afe, false); + } + be->prepared[stream] = false; + } + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DAC_PREDIS); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DAC); + } else if (stream == SNDRV_PCM_STREAM_CAPTURE) { + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_ADC); + } + + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_dai_int_adda_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = + &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + unsigned int rate = substream->runtime->rate; + int bit_width = snd_pcm_format_width(substream->runtime->format); + int ret; + + dev_info(afe->dev, "%s '%s' rate = %u\n", __func__, + snd_pcm_stream_str(substream), rate); + + if (be->prepared[substream->stream]) { + dev_info(afe->dev, "%s '%s' prepared already\n", + __func__, snd_pcm_stream_str(substream)); + return 0; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = mt8365_dai_set_adda_out(afe, rate); + if (ret) + return ret; + + ret = mt8365_afe_set_i2s_out(afe, rate, bit_width); + if (ret) + return ret; + + mt8365_dai_set_adda_out_enable(afe, true); + mt8365_afe_set_i2s_out_enable(afe, true); + } else { + ret = mt8365_dai_set_adda_in(afe, rate); + if (ret) + return ret; + + mt8365_dai_set_adda_in_enable(afe, true); + } + be->prepared[substream->stream] = true; + return 0; +} + +static const struct snd_soc_dai_ops mt8365_afe_int_adda_ops = { + .startup = mt8365_dai_int_adda_startup, + .shutdown = mt8365_dai_int_adda_shutdown, + .prepare = mt8365_dai_int_adda_prepare, +}; + +static struct snd_soc_dai_driver mtk_dai_adda_driver[] = { + { + .name = "INT ADDA", + .id = MT8365_AFE_IO_INT_ADDA, + .playback = { + .stream_name = "INT ADDA Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "INT ADDA Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_int_adda_ops, + } +}; + +/* DAI Controls */ + +static const struct snd_kcontrol_new mtk_adda_dl_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH1 Switch", AFE_CONN3, + 10, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_adda_dl_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("GAIN1_OUT_CH2 Switch", AFE_CONN4, + 11, 1, 0), +}; + +static const struct snd_kcontrol_new int_adda_o03_o04_enable_ctl = + SOC_DAPM_SINGLE_VIRT("Switch", 1); + +/* DAI widget */ + +static const struct snd_soc_dapm_widget mtk_dai_adda_widgets[] = { + SND_SOC_DAPM_SWITCH("INT ADDA O03_O04", SND_SOC_NOPM, 0, 0, + &int_adda_o03_o04_enable_ctl), + /* inter-connections */ + SND_SOC_DAPM_MIXER("ADDA_DL_CH1", SND_SOC_NOPM, 0, 0, + mtk_adda_dl_ch1_mix, + ARRAY_SIZE(mtk_adda_dl_ch1_mix)), + SND_SOC_DAPM_MIXER("ADDA_DL_CH2", SND_SOC_NOPM, 0, 0, + mtk_adda_dl_ch2_mix, + ARRAY_SIZE(mtk_adda_dl_ch2_mix)), +}; + +/* DAI route */ + +static const struct snd_soc_dapm_route mtk_dai_adda_routes[] = { + {"INT ADDA O03_O04", "Switch", "O03"}, + {"INT ADDA O03_O04", "Switch", "O04"}, + {"INT ADDA Playback", NULL, "INT ADDA O03_O04"}, + {"INT ADDA Playback", NULL, "ADDA_DL_CH1"}, + {"INT ADDA Playback", NULL, "ADDA_DL_CH2"}, + {"AIN Mux", "INT ADC", "INT ADDA Capture"}, + {"ADDA_DL_CH1", "GAIN1_OUT_CH1", "Hostless FM DL"}, + {"ADDA_DL_CH2", "GAIN1_OUT_CH2", "Hostless FM DL"}, +}; + +int mt8365_dai_adda_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_adda_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_adda_driver); + dai->dapm_widgets = mtk_dai_adda_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_adda_widgets); + dai->dapm_routes = mtk_dai_adda_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_adda_routes); + return 0; +} From 1c50ec75ce6c0c6b5736499393e522f73e19d0cf Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:37 +0200 Subject: [PATCH 315/410] ASoC: mediatek: mt8365: Add DMIC DAI support Add Digital Micro Device Audio Interface support for MT8365 SoC. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-8-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 340 ++++++++++++++++++++ 1 file changed, 340 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-dmic.c diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c new file mode 100644 index 000000000000..a3bf54751420 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c @@ -0,0 +1,340 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 ALSA SoC Audio DAI DMIC Control + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include +#include +#include +#include "mt8365-afe-clk.h" +#include "mt8365-afe-common.h" + +struct mt8365_dmic_data { + bool two_wire_mode; + unsigned int clk_phase_sel_ch1; + unsigned int clk_phase_sel_ch2; + bool iir_on; + unsigned int irr_mode; + unsigned int dmic_mode; + unsigned int dmic_channel; +}; + +static int get_chan_reg(unsigned int channel) +{ + switch (channel) { + case 8: + fallthrough; + case 7: + return AFE_DMIC3_UL_SRC_CON0; + case 6: + fallthrough; + case 5: + return AFE_DMIC2_UL_SRC_CON0; + case 4: + fallthrough; + case 3: + return AFE_DMIC1_UL_SRC_CON0; + case 2: + fallthrough; + case 1: + return AFE_DMIC0_UL_SRC_CON0; + default: + return -EINVAL; + } +} + +/* DAI Drivers */ + +static void audio_dmic_adda_enable(struct mtk_base_afe *afe) +{ + mt8365_dai_enable_adda_on(afe); + regmap_update_bits(afe->regmap, AFE_ADDA_UL_DL_CON0, + AFE_ADDA_UL_DL_DMIC_CLKDIV_ON, + AFE_ADDA_UL_DL_DMIC_CLKDIV_ON); +} + +static void audio_dmic_adda_disable(struct mtk_base_afe *afe) +{ + regmap_update_bits(afe->regmap, AFE_ADDA_UL_DL_CON0, + AFE_ADDA_UL_DL_DMIC_CLKDIV_ON, + ~AFE_ADDA_UL_DL_DMIC_CLKDIV_ON); + mt8365_dai_disable_adda_on(afe); +} + +static void mt8365_dai_enable_dmic(struct mtk_base_afe *afe, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_dmic_data *dmic_data = afe_priv->dai_priv[MT8365_AFE_IO_DMIC]; + unsigned int val_mask; + int reg = get_chan_reg(dmic_data->dmic_channel); + + if (reg < 0) + return; + + /* val and mask will be always same to enable */ + val_mask = DMIC_TOP_CON_CH1_ON | + DMIC_TOP_CON_CH2_ON | + DMIC_TOP_CON_SRC_ON; + + regmap_update_bits(afe->regmap, reg, val_mask, val_mask); +} + +static void mt8365_dai_disable_dmic(struct mtk_base_afe *afe, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_dmic_data *dmic_data = afe_priv->dai_priv[MT8365_AFE_IO_DMIC]; + unsigned int mask; + int reg = get_chan_reg(dmic_data->dmic_channel); + + if (reg < 0) + return; + + dev_dbg(afe->dev, "%s dmic_channel %d\n", __func__, dmic_data->dmic_channel); + + mask = DMIC_TOP_CON_CH1_ON | + DMIC_TOP_CON_CH2_ON | + DMIC_TOP_CON_SRC_ON | + DMIC_TOP_CON_SDM3_LEVEL_MODE; + + /* Set all masked values to 0 */ + regmap_update_bits(afe->regmap, reg, mask, 0); +} + +static const struct reg_sequence mt8365_dmic_iir_coeff[] = { + { AFE_DMIC0_IIR_COEF_02_01, 0x00000000 }, + { AFE_DMIC0_IIR_COEF_04_03, 0x00003FB8 }, + { AFE_DMIC0_IIR_COEF_06_05, 0x3FB80000 }, + { AFE_DMIC0_IIR_COEF_08_07, 0x3FB80000 }, + { AFE_DMIC0_IIR_COEF_10_09, 0x0000C048 }, + { AFE_DMIC1_IIR_COEF_02_01, 0x00000000 }, + { AFE_DMIC1_IIR_COEF_04_03, 0x00003FB8 }, + { AFE_DMIC1_IIR_COEF_06_05, 0x3FB80000 }, + { AFE_DMIC1_IIR_COEF_08_07, 0x3FB80000 }, + { AFE_DMIC1_IIR_COEF_10_09, 0x0000C048 }, + { AFE_DMIC2_IIR_COEF_02_01, 0x00000000 }, + { AFE_DMIC2_IIR_COEF_04_03, 0x00003FB8 }, + { AFE_DMIC2_IIR_COEF_06_05, 0x3FB80000 }, + { AFE_DMIC2_IIR_COEF_08_07, 0x3FB80000 }, + { AFE_DMIC2_IIR_COEF_10_09, 0x0000C048 }, + { AFE_DMIC3_IIR_COEF_02_01, 0x00000000 }, + { AFE_DMIC3_IIR_COEF_04_03, 0x00003FB8 }, + { AFE_DMIC3_IIR_COEF_06_05, 0x3FB80000 }, + { AFE_DMIC3_IIR_COEF_08_07, 0x3FB80000 }, + { AFE_DMIC3_IIR_COEF_10_09, 0x0000C048 }, +}; + +static int mt8365_dai_load_dmic_iir_coeff_table(struct mtk_base_afe *afe) +{ + return regmap_multi_reg_write(afe->regmap, + mt8365_dmic_iir_coeff, + ARRAY_SIZE(mt8365_dmic_iir_coeff)); +} + +static int mt8365_dai_configure_dmic(struct mtk_base_afe *afe, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_dmic_data *dmic_data = afe_priv->dai_priv[MT8365_AFE_IO_DMIC]; + bool two_wire_mode = dmic_data->two_wire_mode; + unsigned int clk_phase_sel_ch1 = dmic_data->clk_phase_sel_ch1; + unsigned int clk_phase_sel_ch2 = dmic_data->clk_phase_sel_ch2; + unsigned int val = 0; + unsigned int rate = dai->rate; + int reg = get_chan_reg(dai->channels); + + if (reg < 0) + return -EINVAL; + + dmic_data->dmic_channel = dai->channels; + + val |= DMIC_TOP_CON_SDM3_LEVEL_MODE; + + if (two_wire_mode) { + val |= DMIC_TOP_CON_TWO_WIRE_MODE; + } else { + val |= FIELD_PREP(DMIC_TOP_CON_CK_PHASE_SEL_CH1, + clk_phase_sel_ch1); + val |= FIELD_PREP(DMIC_TOP_CON_CK_PHASE_SEL_CH2, + clk_phase_sel_ch2); + } + + switch (rate) { + case 48000: + val |= DMIC_TOP_CON_VOICE_MODE_48K; + break; + case 32000: + val |= DMIC_TOP_CON_VOICE_MODE_32K; + break; + case 16000: + val |= DMIC_TOP_CON_VOICE_MODE_16K; + break; + case 8000: + val |= DMIC_TOP_CON_VOICE_MODE_8K; + break; + default: + return -EINVAL; + } + + regmap_update_bits(afe->regmap, reg, DMIC_TOP_CON_CONFIG_MASK, val); + + return 0; +} + +static int mt8365_dai_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + mt8365_afe_enable_main_clk(afe); + + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DMIC0_ADC); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DMIC1_ADC); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DMIC2_ADC); + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_DMIC3_ADC); + + audio_dmic_adda_enable(afe); + + return 0; +} + +static void mt8365_dai_dmic_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + mt8365_dai_disable_dmic(afe, substream, dai); + audio_dmic_adda_disable(afe); + /* HW Request delay 125us before CG off */ + usleep_range(125, 300); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DMIC3_ADC); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DMIC2_ADC); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DMIC1_ADC); + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_DMIC0_ADC); + + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_dai_dmic_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + mt8365_dai_configure_dmic(afe, substream, dai); + mt8365_dai_enable_dmic(afe, substream, dai); + + return 0; +} + +static const struct snd_soc_dai_ops mt8365_afe_dmic_ops = { + .startup = mt8365_dai_dmic_startup, + .shutdown = mt8365_dai_dmic_shutdown, + .prepare = mt8365_dai_dmic_prepare, +}; + +static struct snd_soc_dai_driver mtk_dai_dmic_driver[] = { + { + .name = "DMIC", + .id = MT8365_AFE_IO_DMIC, + .capture = { + .stream_name = "DMIC Capture", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_dmic_ops, + } +}; + +/* DAI Controls */ + +/* Values for 48kHz mode */ +static const char * const iir_mode_src[] = { + "SW custom", "5Hz", "10Hz", "25Hz", "50Hz", "65Hz" +}; + +static SOC_ENUM_SINGLE_DECL(iir_mode, AFE_DMIC0_UL_SRC_CON0, 7, iir_mode_src); + +static const struct snd_kcontrol_new mtk_dai_dmic_controls[] = { + SOC_SINGLE("DMIC IIR Switch", AFE_DMIC0_UL_SRC_CON0, DMIC_TOP_CON_IIR_ON, 1, 0), + SOC_ENUM("DMIC IIR Mode", iir_mode), +}; + +/* DAI widget */ + +static const struct snd_soc_dapm_widget mtk_dai_dmic_widgets[] = { + SND_SOC_DAPM_INPUT("DMIC In"), +}; + +/* DAI route */ + +static const struct snd_soc_dapm_route mtk_dai_dmic_routes[] = { + {"I14", NULL, "DMIC Capture"}, + {"I15", NULL, "DMIC Capture"}, + {"I16", NULL, "DMIC Capture"}, + {"I17", NULL, "DMIC Capture"}, + {"I18", NULL, "DMIC Capture"}, + {"I19", NULL, "DMIC Capture"}, + {"I20", NULL, "DMIC Capture"}, + {"I21", NULL, "DMIC Capture"}, + {"DMIC Capture", NULL, "DMIC In"}, +}; + +static int init_dmic_priv_data(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_dmic_data *dmic_priv; + struct device_node *np = afe->dev->of_node; + unsigned int temps[4]; + int ret; + + dmic_priv = devm_kzalloc(afe->dev, sizeof(*dmic_priv), GFP_KERNEL); + if (!dmic_priv) + return -ENOMEM; + + ret = of_property_read_u32_array(np, "mediatek,dmic-mode", + &temps[0], + 1); + if (ret == 0) + dmic_priv->two_wire_mode = !!temps[0]; + + if (!dmic_priv->two_wire_mode) { + dmic_priv->clk_phase_sel_ch1 = 0; + dmic_priv->clk_phase_sel_ch2 = 4; + } + + afe_priv->dai_priv[MT8365_AFE_IO_DMIC] = dmic_priv; + return 0; +} + +int mt8365_dai_dmic_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_dmic_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_dmic_driver); + dai->controls = mtk_dai_dmic_controls; + dai->num_controls = ARRAY_SIZE(mtk_dai_dmic_controls); + dai->dapm_widgets = mtk_dai_dmic_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_dmic_widgets); + dai->dapm_routes = mtk_dai_dmic_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_dmic_routes); + return init_dmic_priv_data(afe); +} From 5097c0c8634d703e3c59cfb89831b7db9dc46339 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:38 +0200 Subject: [PATCH 316/410] ASoC: mediatek: mt8365: Add PCM DAI support Add Pulse Code Modulation Device Audio Interface support for MT8365 SoC. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-9-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 293 +++++++++++++++++++++ 1 file changed, 293 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-pcm.c diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c b/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c new file mode 100644 index 000000000000..f85ec07249c3 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-dai-pcm.c @@ -0,0 +1,293 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 ALSA SoC Audio DAI PCM Control + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include +#include +#include +#include "mt8365-afe-clk.h" +#include "mt8365-afe-common.h" + +struct mt8365_pcm_intf_data { + bool slave_mode; + bool lrck_inv; + bool bck_inv; + unsigned int format; +}; + +/* DAI Drivers */ + +static void mt8365_dai_enable_pcm1(struct mtk_base_afe *afe) +{ + regmap_update_bits(afe->regmap, PCM_INTF_CON1, + PCM_INTF_CON1_EN, PCM_INTF_CON1_EN); +} + +static void mt8365_dai_disable_pcm1(struct mtk_base_afe *afe) +{ + regmap_update_bits(afe->regmap, PCM_INTF_CON1, + PCM_INTF_CON1_EN, 0x0); +} + +static int mt8365_dai_configure_pcm1(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_pcm_intf_data *pcm_priv = afe_priv->dai_priv[MT8365_AFE_IO_PCM1]; + bool slave_mode = pcm_priv->slave_mode; + bool lrck_inv = pcm_priv->lrck_inv; + bool bck_inv = pcm_priv->bck_inv; + unsigned int fmt = pcm_priv->format; + unsigned int bit_width = dai->sample_bits; + unsigned int val = 0; + + if (!slave_mode) { + val |= PCM_INTF_CON1_MASTER_MODE | + PCM_INTF_CON1_BYPASS_ASRC; + + if (lrck_inv) + val |= PCM_INTF_CON1_SYNC_OUT_INV; + if (bck_inv) + val |= PCM_INTF_CON1_BCLK_OUT_INV; + } else { + val |= PCM_INTF_CON1_SLAVE_MODE; + + if (lrck_inv) + val |= PCM_INTF_CON1_SYNC_IN_INV; + if (bck_inv) + val |= PCM_INTF_CON1_BCLK_IN_INV; + + /* TODO: add asrc setting */ + } + + val |= FIELD_PREP(PCM_INTF_CON1_FORMAT_MASK, fmt); + + if (fmt == MT8365_PCM_FORMAT_PCMA || + fmt == MT8365_PCM_FORMAT_PCMB) + val |= PCM_INTF_CON1_SYNC_LEN(1); + else + val |= PCM_INTF_CON1_SYNC_LEN(bit_width); + + switch (substream->runtime->rate) { + case 48000: + val |= PCM_INTF_CON1_FS_48K; + break; + case 32000: + val |= PCM_INTF_CON1_FS_32K; + break; + case 16000: + val |= PCM_INTF_CON1_FS_16K; + break; + case 8000: + val |= PCM_INTF_CON1_FS_8K; + break; + default: + return -EINVAL; + } + + if (bit_width > 16) + val |= PCM_INTF_CON1_24BIT | PCM_INTF_CON1_64BCK; + else + val |= PCM_INTF_CON1_16BIT | PCM_INTF_CON1_32BCK; + + val |= PCM_INTF_CON1_EXT_MODEM; + + regmap_update_bits(afe->regmap, PCM_INTF_CON1, + PCM_INTF_CON1_CONFIG_MASK, val); + + return 0; +} + +static int mt8365_dai_pcm1_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + if (snd_soc_dai_active(dai)) + return 0; + + mt8365_afe_enable_main_clk(afe); + + return 0; +} + +static void mt8365_dai_pcm1_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + if (snd_soc_dai_active(dai)) + return; + + mt8365_dai_disable_pcm1(afe); + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_dai_pcm1_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + int ret; + + if ((snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_PLAYBACK) + + snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_CAPTURE)) > 1) { + dev_info(afe->dev, "%s '%s' active(%u-%u) already\n", + __func__, snd_pcm_stream_str(substream), + snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_PLAYBACK), + snd_soc_dai_stream_active(dai, SNDRV_PCM_STREAM_CAPTURE)); + return 0; + } + + ret = mt8365_dai_configure_pcm1(substream, dai); + if (ret) + return ret; + + mt8365_dai_enable_pcm1(afe); + + return 0; +} + +static int mt8365_dai_pcm1_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_pcm_intf_data *pcm_priv = afe_priv->dai_priv[MT8365_AFE_IO_PCM1]; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + pcm_priv->format = MT8365_PCM_FORMAT_I2S; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + pcm_priv->bck_inv = false; + pcm_priv->lrck_inv = false; + break; + case SND_SOC_DAIFMT_NB_IF: + pcm_priv->bck_inv = false; + pcm_priv->lrck_inv = true; + break; + case SND_SOC_DAIFMT_IB_NF: + pcm_priv->bck_inv = true; + pcm_priv->lrck_inv = false; + break; + case SND_SOC_DAIFMT_IB_IF: + pcm_priv->bck_inv = true; + pcm_priv->lrck_inv = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + pcm_priv->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + pcm_priv->slave_mode = false; + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops mt8365_dai_pcm1_ops = { + .startup = mt8365_dai_pcm1_startup, + .shutdown = mt8365_dai_pcm1_shutdown, + .prepare = mt8365_dai_pcm1_prepare, + .set_fmt = mt8365_dai_pcm1_set_fmt, +}; + +static struct snd_soc_dai_driver mtk_dai_pcm_driver[] = { + { + .name = "PCM1", + .id = MT8365_AFE_IO_PCM1, + .playback = { + .stream_name = "PCM1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "PCM1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | + SNDRV_PCM_RATE_16000 | + SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_dai_pcm1_ops, + .symmetric_rate = 1, + .symmetric_sample_bits = 1, + } +}; + +/* DAI widget */ + +static const struct snd_soc_dapm_widget mtk_dai_pcm_widgets[] = { + SND_SOC_DAPM_OUTPUT("PCM1 Out"), + SND_SOC_DAPM_INPUT("PCM1 In"), +}; + +/* DAI route */ + +static const struct snd_soc_dapm_route mtk_dai_pcm_routes[] = { + {"PCM1 Playback", NULL, "O07"}, + {"PCM1 Playback", NULL, "O08"}, + {"PCM1 Out", NULL, "PCM1 Playback"}, + + {"I09", NULL, "PCM1 Capture"}, + {"I22", NULL, "PCM1 Capture"}, + {"PCM1 Capture", NULL, "PCM1 In"}, +}; + +static int init_pcmif_priv_data(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_pcm_intf_data *pcmif_priv; + + pcmif_priv = devm_kzalloc(afe->dev, sizeof(struct mt8365_pcm_intf_data), + GFP_KERNEL); + if (!pcmif_priv) + return -ENOMEM; + + afe_priv->dai_priv[MT8365_AFE_IO_PCM1] = pcmif_priv; + return 0; +} + +int mt8365_dai_pcm_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + dai->dai_drivers = mtk_dai_pcm_driver; + dai->num_dai_drivers = ARRAY_SIZE(mtk_dai_pcm_driver); + dai->dapm_widgets = mtk_dai_pcm_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mtk_dai_pcm_widgets); + dai->dapm_routes = mtk_dai_pcm_routes; + dai->num_dapm_routes = ARRAY_SIZE(mtk_dai_pcm_routes); + return init_pcmif_priv_data(afe); +} From 1bf6dbd75f7603dd026660bebf324f812200dc1b Mon Sep 17 00:00:00 2001 From: Nicolas Belin Date: Mon, 22 Jul 2024 08:53:39 +0200 Subject: [PATCH 317/410] ASoc: mediatek: mt8365: Add a specific soundcard for EVK Add a specific soundcard for mt8365-evk. It supports audio jack in/out, dmics, the amic and lineout. Signed-off-by: Nicolas Belin Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-10-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 345 ++++++++++++++++++++++ 1 file changed, 345 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-mt6357.c diff --git a/sound/soc/mediatek/mt8365/mt8365-mt6357.c b/sound/soc/mediatek/mt8365/mt8365-mt6357.c new file mode 100644 index 000000000000..fef76118f801 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-mt6357.c @@ -0,0 +1,345 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek MT8365 Sound Card driver + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Nicolas Belin + */ + +#include +#include +#include +#include +#include "mt8365-afe-common.h" +#include +#include "../common/mtk-soc-card.h" +#include "../common/mtk-soundcard-driver.h" + +enum pinctrl_pin_state { + PIN_STATE_DEFAULT, + PIN_STATE_DMIC, + PIN_STATE_MISO_OFF, + PIN_STATE_MISO_ON, + PIN_STATE_MOSI_OFF, + PIN_STATE_MOSI_ON, + PIN_STATE_MAX +}; + +static const char * const mt8365_mt6357_pin_str[PIN_STATE_MAX] = { + "default", + "dmic", + "miso_off", + "miso_on", + "mosi_off", + "mosi_on", +}; + +struct mt8365_mt6357_priv { + struct pinctrl *pinctrl; + struct pinctrl_state *pin_states[PIN_STATE_MAX]; +}; + +enum { + /* FE */ + DAI_LINK_DL1_PLAYBACK = 0, + DAI_LINK_DL2_PLAYBACK, + DAI_LINK_AWB_CAPTURE, + DAI_LINK_VUL_CAPTURE, + /* BE */ + DAI_LINK_2ND_I2S_INTF, + DAI_LINK_DMIC, + DAI_LINK_INT_ADDA, + DAI_LINK_NUM +}; + +static const struct snd_soc_dapm_widget mt8365_mt6357_widgets[] = { + SND_SOC_DAPM_OUTPUT("HDMI Out"), +}; + +static const struct snd_soc_dapm_route mt8365_mt6357_routes[] = { + {"HDMI Out", NULL, "2ND I2S Playback"}, + {"DMIC In", NULL, "MICBIAS0"}, +}; + +static int mt8365_mt6357_int_adda_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8365_mt6357_priv *priv = snd_soc_card_get_drvdata(rtd->card); + int ret = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (IS_ERR(priv->pin_states[PIN_STATE_MOSI_ON])) + return ret; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_MOSI_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (IS_ERR(priv->pin_states[PIN_STATE_MISO_ON])) + return ret; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_MISO_ON]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + } + + return 0; +} + +static void mt8365_mt6357_int_adda_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct mt8365_mt6357_priv *priv = snd_soc_card_get_drvdata(rtd->card); + int ret = 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (IS_ERR(priv->pin_states[PIN_STATE_MOSI_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_MOSI_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + } + + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (IS_ERR(priv->pin_states[PIN_STATE_MISO_OFF])) + return; + + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[PIN_STATE_MISO_OFF]); + if (ret) + dev_err(rtd->card->dev, "%s failed to select state %d\n", + __func__, ret); + } +} + +static const struct snd_soc_ops mt8365_mt6357_int_adda_ops = { + .startup = mt8365_mt6357_int_adda_startup, + .shutdown = mt8365_mt6357_int_adda_shutdown, +}; + +SND_SOC_DAILINK_DEFS(playback1, + DAILINK_COMP_ARRAY(COMP_CPU("DL1")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(playback2, + DAILINK_COMP_ARRAY(COMP_CPU("DL2")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(awb_capture, + DAILINK_COMP_ARRAY(COMP_CPU("AWB")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(vul, + DAILINK_COMP_ARRAY(COMP_CPU("VUL")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +SND_SOC_DAILINK_DEFS(i2s3, + DAILINK_COMP_ARRAY(COMP_CPU("2ND I2S")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(dmic, + DAILINK_COMP_ARRAY(COMP_CPU("DMIC")), + DAILINK_COMP_ARRAY(COMP_DUMMY()), + DAILINK_COMP_ARRAY(COMP_EMPTY())); +SND_SOC_DAILINK_DEFS(primary_codec, + DAILINK_COMP_ARRAY(COMP_CPU("INT ADDA")), + DAILINK_COMP_ARRAY(COMP_CODEC("mt6357-sound", "mt6357-snd-codec-aif1")), + DAILINK_COMP_ARRAY(COMP_EMPTY())); + +/* Digital audio interface glue - connects codec <---> CPU */ +static struct snd_soc_dai_link mt8365_mt6357_dais[] = { + /* Front End DAI links */ + [DAI_LINK_DL1_PLAYBACK] = { + .name = "DL1_FE", + .stream_name = "MultiMedia1_PLayback", + .id = DAI_LINK_DL1_PLAYBACK, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_merged_rate = 1, + SND_SOC_DAILINK_REG(playback1), + }, + [DAI_LINK_DL2_PLAYBACK] = { + .name = "DL2_FE", + .stream_name = "MultiMedia2_PLayback", + .id = DAI_LINK_DL2_PLAYBACK, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_merged_rate = 1, + SND_SOC_DAILINK_REG(playback2), + }, + [DAI_LINK_AWB_CAPTURE] = { + .name = "AWB_FE", + .stream_name = "DL1_AWB_Record", + .id = DAI_LINK_AWB_CAPTURE, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dynamic = 1, + .dpcm_capture = 1, + .dpcm_merged_rate = 1, + SND_SOC_DAILINK_REG(awb_capture), + }, + [DAI_LINK_VUL_CAPTURE] = { + .name = "VUL_FE", + .stream_name = "MultiMedia1_Capture", + .id = DAI_LINK_VUL_CAPTURE, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dynamic = 1, + .dpcm_capture = 1, + .dpcm_merged_rate = 1, + SND_SOC_DAILINK_REG(vul), + }, + /* Back End DAI links */ + [DAI_LINK_2ND_I2S_INTF] = { + .name = "I2S_OUT_BE", + .no_pcm = 1, + .id = DAI_LINK_2ND_I2S_INTF, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(i2s3), + }, + [DAI_LINK_DMIC] = { + .name = "DMIC_BE", + .no_pcm = 1, + .id = DAI_LINK_DMIC, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(dmic), + }, + [DAI_LINK_INT_ADDA] = { + .name = "MTK_Codec", + .no_pcm = 1, + .id = DAI_LINK_INT_ADDA, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &mt8365_mt6357_int_adda_ops, + SND_SOC_DAILINK_REG(primary_codec), + }, +}; + +static int mt8365_mt6357_gpio_probe(struct snd_soc_card *card) +{ + struct mt8365_mt6357_priv *priv = snd_soc_card_get_drvdata(card); + int ret, i; + + priv->pinctrl = devm_pinctrl_get(card->dev); + if (IS_ERR(priv->pinctrl)) { + ret = PTR_ERR(priv->pinctrl); + return dev_err_probe(card->dev, ret, + "Failed to get pinctrl\n"); + } + + for (i = PIN_STATE_DEFAULT ; i < PIN_STATE_MAX ; i++) { + priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, + mt8365_mt6357_pin_str[i]); + if (IS_ERR(priv->pin_states[i])) { + ret = PTR_ERR(priv->pin_states[i]); + dev_warn(card->dev, "No pin state for %s\n", + mt8365_mt6357_pin_str[i]); + } else { + ret = pinctrl_select_state(priv->pinctrl, + priv->pin_states[i]); + if (ret) { + dev_err_probe(card->dev, ret, + "Failed to select pin state %s\n", + mt8365_mt6357_pin_str[i]); + return ret; + } + } + } + return 0; +} + +static struct snd_soc_card mt8365_mt6357_soc_card = { + .name = "mt8365-evk", + .owner = THIS_MODULE, + .dai_link = mt8365_mt6357_dais, + .num_links = ARRAY_SIZE(mt8365_mt6357_dais), + .dapm_widgets = mt8365_mt6357_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt8365_mt6357_widgets), + .dapm_routes = mt8365_mt6357_routes, + .num_dapm_routes = ARRAY_SIZE(mt8365_mt6357_routes), +}; + +static int mt8365_mt6357_dev_probe(struct mtk_soc_card_data *soc_card_data, bool legacy) +{ + struct mtk_platform_card_data *card_data = soc_card_data->card_data; + struct snd_soc_card *card = card_data->card; + struct device *dev = card->dev; + struct device_node *platform_node; + struct mt8365_mt6357_priv *mach_priv; + int i, ret; + + card->dev = dev; + ret = parse_dai_link_info(card); + if (ret) + goto err; + + mach_priv = devm_kzalloc(dev, sizeof(*mach_priv), + GFP_KERNEL); + if (!mach_priv) + return -ENOMEM; + soc_card_data->mach_priv = mach_priv; + snd_soc_card_set_drvdata(card, soc_card_data); + mt8365_mt6357_gpio_probe(card); + return 0; + +err: + clean_card_reference(card); + return ret; +} + +static const struct mtk_soundcard_pdata mt8365_mt6357_card = { + .card_name = "mt8365-mt6357", + .card_data = &(struct mtk_platform_card_data) { + .card = &mt8365_mt6357_soc_card, + }, + .soc_probe = mt8365_mt6357_dev_probe +}; + +static const struct of_device_id mt8365_mt6357_dt_match[] = { + { .compatible = "mediatek,mt8365-mt6357", .data = &mt8365_mt6357_card }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, mt8365_mt6357_dt_match); + +static struct platform_driver mt8365_mt6357_driver = { + .driver = { + .name = "mt8365_mt6357", + .of_match_table = mt8365_mt6357_dt_match, + .pm = &snd_soc_pm_ops, + }, + .probe = mtk_soundcard_common_probe, +}; + +module_platform_driver(mt8365_mt6357_driver); + +/* Module information */ +MODULE_DESCRIPTION("MT8365 EVK SoC machine driver"); +MODULE_AUTHOR("Nicolas Belin "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform: mt8365_mt6357"); From e1991d102bc2abb32331c462f8f3e77059c69578 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Mon, 22 Jul 2024 08:53:40 +0200 Subject: [PATCH 318/410] ASoC: mediatek: mt8365: Add the AFE driver support Add a driver for the Audio Front End (AFE) PCM to provide Audio Uplink (UL) and Downlink (DL) paths. Use the ALSA SoC Dynamic Audio Power Management to add widget and kcontrol supports. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v7-11-6518d953a141@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2275 ++++++++++++++++++++ 1 file changed, 2275 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-pcm.c diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c new file mode 100644 index 000000000000..df6dd8c5bbe5 --- /dev/null +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -0,0 +1,2275 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MediaTek 8365 ALSA SoC AFE platform driver + * + * Copyright (c) 2024 MediaTek Inc. + * Authors: Jia Zeng + * Alexandre Mergnat + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include "mt8365-afe-common.h" +#include "mt8365-afe-clk.h" +#include "mt8365-reg.h" +#include "../common/mtk-base-afe.h" +#include "../common/mtk-afe-platform-driver.h" +#include "../common/mtk-afe-fe-dai.h" + +#define AFE_BASE_END_OFFSET 8 + +static unsigned int mCM2Input; + +static const unsigned int mt8365_afe_backup_list[] = { + AUDIO_TOP_CON0, + AFE_CONN0, + AFE_CONN1, + AFE_CONN3, + AFE_CONN4, + AFE_CONN5, + AFE_CONN6, + AFE_CONN7, + AFE_CONN8, + AFE_CONN9, + AFE_CONN10, + AFE_CONN11, + AFE_CONN12, + AFE_CONN13, + AFE_CONN14, + AFE_CONN15, + AFE_CONN16, + AFE_CONN17, + AFE_CONN18, + AFE_CONN19, + AFE_CONN20, + AFE_CONN21, + AFE_CONN26, + AFE_CONN27, + AFE_CONN28, + AFE_CONN29, + AFE_CONN30, + AFE_CONN31, + AFE_CONN32, + AFE_CONN33, + AFE_CONN34, + AFE_CONN35, + AFE_CONN36, + AFE_CONN_24BIT, + AFE_CONN_24BIT_1, + AFE_DAC_CON0, + AFE_DAC_CON1, + AFE_DL1_BASE, + AFE_DL1_END, + AFE_DL2_BASE, + AFE_DL2_END, + AFE_VUL_BASE, + AFE_VUL_END, + AFE_AWB_BASE, + AFE_AWB_END, + AFE_VUL3_BASE, + AFE_VUL3_END, + AFE_HDMI_OUT_BASE, + AFE_HDMI_OUT_END, + AFE_HDMI_IN_2CH_BASE, + AFE_HDMI_IN_2CH_END, + AFE_ADDA_UL_DL_CON0, + AFE_ADDA_DL_SRC2_CON0, + AFE_ADDA_DL_SRC2_CON1, + AFE_I2S_CON, + AFE_I2S_CON1, + AFE_I2S_CON2, + AFE_I2S_CON3, + AFE_ADDA_UL_SRC_CON0, + AFE_AUD_PAD_TOP, + AFE_HD_ENGEN_ENABLE, +}; + +static const struct snd_pcm_hardware mt8365_afe_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .buffer_bytes_max = 256 * 1024, + .period_bytes_min = 512, + .period_bytes_max = 128 * 1024, + .periods_min = 2, + .periods_max = 256, + .fifo_size = 0, +}; + +struct mt8365_afe_rate { + unsigned int rate; + unsigned int reg_val; +}; + +static const struct mt8365_afe_rate mt8365_afe_fs_rates[] = { + { .rate = 8000, .reg_val = MT8365_FS_8K }, + { .rate = 11025, .reg_val = MT8365_FS_11D025K }, + { .rate = 12000, .reg_val = MT8365_FS_12K }, + { .rate = 16000, .reg_val = MT8365_FS_16K }, + { .rate = 22050, .reg_val = MT8365_FS_22D05K }, + { .rate = 24000, .reg_val = MT8365_FS_24K }, + { .rate = 32000, .reg_val = MT8365_FS_32K }, + { .rate = 44100, .reg_val = MT8365_FS_44D1K }, + { .rate = 48000, .reg_val = MT8365_FS_48K }, + { .rate = 88200, .reg_val = MT8365_FS_88D2K }, + { .rate = 96000, .reg_val = MT8365_FS_96K }, + { .rate = 176400, .reg_val = MT8365_FS_176D4K }, + { .rate = 192000, .reg_val = MT8365_FS_192K }, +}; + +int mt8365_afe_fs_timing(unsigned int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(mt8365_afe_fs_rates); i++) + if (mt8365_afe_fs_rates[i].rate == rate) + return mt8365_afe_fs_rates[i].reg_val; + + return -EINVAL; +} + +bool mt8365_afe_rate_supported(unsigned int rate, unsigned int id) +{ + switch (id) { + case MT8365_AFE_IO_TDM_IN: + if (rate >= 8000 && rate <= 192000) + return true; + break; + case MT8365_AFE_IO_DMIC: + if (rate >= 8000 && rate <= 48000) + return true; + break; + default: + break; + } + + return false; +} + +bool mt8365_afe_channel_supported(unsigned int channel, unsigned int id) +{ + switch (id) { + case MT8365_AFE_IO_TDM_IN: + if (channel >= 1 && channel <= 8) + return true; + break; + case MT8365_AFE_IO_DMIC: + if (channel >= 1 && channel <= 8) + return true; + break; + default: + break; + } + + return false; +} + +bool mt8365_afe_clk_group_44k(int sample_rate) +{ + if (sample_rate == 11025 || + sample_rate == 22050 || + sample_rate == 44100 || + sample_rate == 88200 || + sample_rate == 176400) + return true; + else + return false; +} + +bool mt8365_afe_clk_group_48k(int sample_rate) +{ + return (!mt8365_afe_clk_group_44k(sample_rate)); +} + +int mt8365_dai_set_priv(struct mtk_base_afe *afe, int id, + int priv_size, const void *priv_data) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + void *temp_data; + + temp_data = devm_kzalloc(afe->dev, priv_size, GFP_KERNEL); + if (!temp_data) + return -ENOMEM; + + if (priv_data) + memcpy(temp_data, priv_data, priv_size); + + afe_priv->dai_priv[id] = temp_data; + + return 0; +} + +static int mt8365_afe_irq_direction_enable(struct mtk_base_afe *afe, + int irq_id, int direction) +{ + struct mtk_base_afe_irq *irq; + + if (irq_id >= MT8365_AFE_IRQ_NUM) + return -1; + + irq = &afe->irqs[irq_id]; + + if (direction == MT8365_AFE_IRQ_DIR_MCU) { + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_DSP_EN, + (1 << irq->irq_data->irq_clr_shift), + 0); + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, + (1 << irq->irq_data->irq_clr_shift), + (1 << irq->irq_data->irq_clr_shift)); + } else if (direction == MT8365_AFE_IRQ_DIR_DSP) { + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_DSP_EN, + (1 << irq->irq_data->irq_clr_shift), + (1 << irq->irq_data->irq_clr_shift)); + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, + (1 << irq->irq_data->irq_clr_shift), + 0); + } else { + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_DSP_EN, + (1 << irq->irq_data->irq_clr_shift), + (1 << irq->irq_data->irq_clr_shift)); + regmap_update_bits(afe->regmap, AFE_IRQ_MCU_EN, + (1 << irq->irq_data->irq_clr_shift), + (1 << irq->irq_data->irq_clr_shift)); + } + return 0; +} + +static int mt8365_memif_fs(struct snd_pcm_substream *substream, + unsigned int rate) +{ + return mt8365_afe_fs_timing(rate); +} + +static int mt8365_irq_fs(struct snd_pcm_substream *substream, + unsigned int rate) +{ + return mt8365_memif_fs(substream, rate); +} + +static const struct mt8365_cm_ctrl_reg cm_ctrl_reg[MT8365_CM_NUM] = { + [MT8365_CM1] = { + .con0 = AFE_CM1_CON0, + .con1 = AFE_CM1_CON1, + .con2 = AFE_CM1_CON2, + .con3 = AFE_CM1_CON3, + .con4 = AFE_CM1_CON4, + }, + [MT8365_CM2] = { + .con0 = AFE_CM2_CON0, + .con1 = AFE_CM2_CON1, + .con2 = AFE_CM2_CON2, + .con3 = AFE_CM2_CON3, + .con4 = AFE_CM2_CON4, + } +}; + +static int mt8365_afe_cm2_mux_conn(struct mtk_base_afe *afe) +{ + struct mt8365_afe_private *afe_priv = afe->platform_priv; + unsigned int input = afe_priv->cm2_mux_input; + + /* TDM_IN interconnect to CM2 */ + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG1_MASK, + CM2_AFE_CM2_CONN_CFG1(TDM_IN_CH0)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG2_MASK, + CM2_AFE_CM2_CONN_CFG2(TDM_IN_CH1)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG3_MASK, + CM2_AFE_CM2_CONN_CFG3(TDM_IN_CH2)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG4_MASK, + CM2_AFE_CM2_CONN_CFG4(TDM_IN_CH3)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG5_MASK, + CM2_AFE_CM2_CONN_CFG5(TDM_IN_CH4)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN0, + CM2_AFE_CM2_CONN_CFG6_MASK, + CM2_AFE_CM2_CONN_CFG6(TDM_IN_CH5)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG7_MASK, + CM2_AFE_CM2_CONN_CFG7(TDM_IN_CH6)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG8_MASK, + CM2_AFE_CM2_CONN_CFG8(TDM_IN_CH7)); + + /* ref data interconnect to CM2 */ + if (input == MT8365_FROM_GASRC1) { + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG9_MASK, + CM2_AFE_CM2_CONN_CFG9(GENERAL1_ASRC_OUT_LCH)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG10_MASK, + CM2_AFE_CM2_CONN_CFG10(GENERAL1_ASRC_OUT_RCH)); + } else if (input == MT8365_FROM_GASRC2) { + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG9_MASK, + CM2_AFE_CM2_CONN_CFG9(GENERAL2_ASRC_OUT_LCH)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG10_MASK, + CM2_AFE_CM2_CONN_CFG10(GENERAL2_ASRC_OUT_RCH)); + } else if (input == MT8365_FROM_TDM_ASRC) { + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG9_MASK, + CM2_AFE_CM2_CONN_CFG9(TDM_OUT_ASRC_CH0)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG10_MASK, + CM2_AFE_CM2_CONN_CFG10(TDM_OUT_ASRC_CH1)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG11_MASK, + CM2_AFE_CM2_CONN_CFG11(TDM_OUT_ASRC_CH2)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN1, + CM2_AFE_CM2_CONN_CFG12_MASK, + CM2_AFE_CM2_CONN_CFG12(TDM_OUT_ASRC_CH3)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN2, + CM2_AFE_CM2_CONN_CFG13_MASK, + CM2_AFE_CM2_CONN_CFG13(TDM_OUT_ASRC_CH4)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN2, + CM2_AFE_CM2_CONN_CFG14_MASK, + CM2_AFE_CM2_CONN_CFG14(TDM_OUT_ASRC_CH5)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN2, + CM2_AFE_CM2_CONN_CFG15_MASK, + CM2_AFE_CM2_CONN_CFG15(TDM_OUT_ASRC_CH6)); + regmap_update_bits(afe->regmap, AFE_CM2_CONN2, + CM2_AFE_CM2_CONN_CFG16_MASK, + CM2_AFE_CM2_CONN_CFG16(TDM_OUT_ASRC_CH7)); + } else { + dev_err(afe->dev, "%s wrong CM2 input %d\n", __func__, input); + return -1; + } + + return 0; +} + +static int mt8365_afe_get_cm_update_cnt(struct mtk_base_afe *afe, + enum mt8365_cm_num cmNum, + unsigned int rate, unsigned int channel) +{ + unsigned int total_cnt, div_cnt, ch_pair, best_cnt; + unsigned int ch_update_cnt[MT8365_CM_UPDATA_CNT_SET]; + int i; + + /* calculate cm update cnt + * total_cnt = clk / fs, clk is 26m or 24m or 22m + * div_cnt = total_cnt / ch_pair, max ch 16ch ,2ch is a set + * best_cnt < div_cnt ,we set best_cnt = div_cnt -10 + * ch01 = best_cnt, ch23 = 2* ch01_up_cnt + * ch45 = 3* ch01_up_cnt ...ch1415 = 8* ch01_up_cnt + */ + + if (cmNum == MT8365_CM1) { + total_cnt = MT8365_CLK_26M / rate; + } else if (cmNum == MT8365_CM2) { + if (mt8365_afe_clk_group_48k(rate)) + total_cnt = MT8365_CLK_24M / rate; + else + total_cnt = MT8365_CLK_22M / rate; + } else { + return -1; + } + + if (channel % 2) + ch_pair = (channel / 2) + 1; + else + ch_pair = channel / 2; + + div_cnt = total_cnt / ch_pair; + best_cnt = div_cnt - 10; + + if (best_cnt <= 0) + return -1; + + for (i = 0; i < ch_pair; i++) + ch_update_cnt[i] = (i + 1) * best_cnt; + + switch (channel) { + case 16: + fallthrough; + case 15: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con4, + CM_AFE_CM_UPDATE_CNT2_MASK, + CM_AFE_CM_UPDATE_CNT2(ch_update_cnt[7])); + fallthrough; + case 14: + fallthrough; + case 13: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con4, + CM_AFE_CM_UPDATE_CNT1_MASK, + CM_AFE_CM_UPDATE_CNT1(ch_update_cnt[6])); + fallthrough; + case 12: + fallthrough; + case 11: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con3, + CM_AFE_CM_UPDATE_CNT2_MASK, + CM_AFE_CM_UPDATE_CNT2(ch_update_cnt[5])); + fallthrough; + case 10: + fallthrough; + case 9: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con3, + CM_AFE_CM_UPDATE_CNT1_MASK, + CM_AFE_CM_UPDATE_CNT1(ch_update_cnt[4])); + fallthrough; + case 8: + fallthrough; + case 7: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con2, + CM_AFE_CM_UPDATE_CNT2_MASK, + CM_AFE_CM_UPDATE_CNT2(ch_update_cnt[3])); + fallthrough; + case 6: + fallthrough; + case 5: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con2, + CM_AFE_CM_UPDATE_CNT1_MASK, + CM_AFE_CM_UPDATE_CNT1(ch_update_cnt[2])); + fallthrough; + case 4: + fallthrough; + case 3: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con1, + CM_AFE_CM_UPDATE_CNT2_MASK, + CM_AFE_CM_UPDATE_CNT2(ch_update_cnt[1])); + fallthrough; + case 2: + fallthrough; + case 1: + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con1, + CM_AFE_CM_UPDATE_CNT1_MASK, + CM_AFE_CM_UPDATE_CNT1(ch_update_cnt[0])); + break; + default: + return -1; + } + + return 0; +} + +static int mt8365_afe_configure_cm(struct mtk_base_afe *afe, + enum mt8365_cm_num cmNum, + unsigned int channels, + unsigned int rate) +{ + unsigned int val, mask; + unsigned int fs = mt8365_afe_fs_timing(rate); + + val = FIELD_PREP(CM_AFE_CM_CH_NUM_MASK, (channels - 1)) | + FIELD_PREP(CM_AFE_CM_START_DATA_MASK, 0); + + mask = CM_AFE_CM_CH_NUM_MASK | + CM_AFE_CM_START_DATA_MASK; + + if (cmNum == MT8365_CM1) { + val |= FIELD_PREP(CM_AFE_CM1_IN_MODE_MASK, fs); + + mask |= CM_AFE_CM1_VUL_SEL | + CM_AFE_CM1_IN_MODE_MASK; + } else if (cmNum == MT8365_CM2) { + if (mt8365_afe_clk_group_48k(rate)) + val |= FIELD_PREP(CM_AFE_CM2_CLK_SEL, 0); + else + val |= FIELD_PREP(CM_AFE_CM2_CLK_SEL, 1); + + val |= FIELD_PREP(CM_AFE_CM2_TDM_SEL, 1); + + mask |= CM_AFE_CM2_TDM_SEL | + CM_AFE_CM1_IN_MODE_MASK | + CM_AFE_CM2_CLK_SEL; + + mt8365_afe_cm2_mux_conn(afe); + } else { + return -1; + } + + regmap_update_bits(afe->regmap, cm_ctrl_reg[cmNum].con0, mask, val); + + mt8365_afe_get_cm_update_cnt(afe, cmNum, rate, channels); + + return 0; +} + +int mt8365_afe_fe_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_pcm_runtime *runtime = substream->runtime; + int memif_num = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mtk_base_afe_memif *memif = &afe->memif[memif_num]; + int ret; + + memif->substream = substream; + + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); + + snd_soc_set_runtime_hwparams(substream, afe->mtk_afe_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + dev_err(afe->dev, "snd_pcm_hw_constraint_integer failed\n"); + + mt8365_afe_enable_main_clk(afe); + return ret; +} + +static void mt8365_afe_fe_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + int memif_num = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mtk_base_afe_memif *memif = &afe->memif[memif_num]; + + memif->substream = NULL; + + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_afe_fe_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_control_data *ctrl_data = &afe_priv->ctrl_data; + int dai_id = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mtk_base_afe_memif *memif = &afe->memif[dai_id]; + struct mt8365_fe_dai_data *fe_data = &afe_priv->fe_data[dai_id]; + size_t request_size = params_buffer_bytes(params); + unsigned int channels = params_channels(params); + unsigned int rate = params_rate(params); + unsigned int base_end_offset = 8; + int ret, fs; + + dev_info(afe->dev, "%s %s period = %d rate = %d channels = %d\n", + __func__, memif->data->name, params_period_size(params), + rate, channels); + + if (dai_id == MT8365_AFE_MEMIF_VUL2) { + if (!ctrl_data->bypass_cm1) + /* configure cm1 */ + mt8365_afe_configure_cm(afe, MT8365_CM1, + channels, rate); + else + regmap_update_bits(afe->regmap, AFE_CM1_CON0, + CM_AFE_CM1_VUL_SEL, + CM_AFE_CM1_VUL_SEL); + } else if (dai_id == MT8365_AFE_MEMIF_TDM_IN) { + if (!ctrl_data->bypass_cm2) + /* configure cm2 */ + mt8365_afe_configure_cm(afe, MT8365_CM2, + channels, rate); + else + regmap_update_bits(afe->regmap, AFE_CM2_CON0, + CM_AFE_CM2_TDM_SEL, + ~CM_AFE_CM2_TDM_SEL); + + base_end_offset = 4; + } + + if (request_size > fe_data->sram_size) { + ret = snd_pcm_lib_malloc_pages(substream, request_size); + if (ret < 0) { + dev_err(afe->dev, + "%s %s malloc pages %zu bytes failed %d\n", + __func__, memif->data->name, request_size, ret); + return ret; + } + + fe_data->use_sram = false; + + mt8365_afe_emi_clk_on(afe); + } else { + struct snd_dma_buffer *dma_buf = &substream->dma_buffer; + + dma_buf->dev.type = SNDRV_DMA_TYPE_DEV; + dma_buf->dev.dev = substream->pcm->card->dev; + dma_buf->area = (unsigned char *)fe_data->sram_vir_addr; + dma_buf->addr = fe_data->sram_phy_addr; + dma_buf->bytes = request_size; + snd_pcm_set_runtime_buffer(substream, dma_buf); + + fe_data->use_sram = true; + } + + memif->phys_buf_addr = lower_32_bits(substream->runtime->dma_addr); + memif->buffer_size = substream->runtime->dma_bytes; + + /* start */ + regmap_write(afe->regmap, memif->data->reg_ofs_base, + memif->phys_buf_addr); + /* end */ + regmap_write(afe->regmap, + memif->data->reg_ofs_base + base_end_offset, + memif->phys_buf_addr + memif->buffer_size - 1); + + /* set channel */ + if (memif->data->mono_shift >= 0) { + unsigned int mono = (params_channels(params) == 1) ? 1 : 0; + + if (memif->data->mono_reg < 0) + dev_info(afe->dev, "%s mono_reg is NULL\n", __func__); + else + regmap_update_bits(afe->regmap, memif->data->mono_reg, + 1 << memif->data->mono_shift, + mono << memif->data->mono_shift); + } + + /* set rate */ + if (memif->data->fs_shift < 0) + return 0; + + fs = afe->memif_fs(substream, params_rate(params)); + + if (fs < 0) + return -EINVAL; + + if (memif->data->fs_reg < 0) + dev_info(afe->dev, "%s fs_reg is NULL\n", __func__); + else + regmap_update_bits(afe->regmap, memif->data->fs_reg, + memif->data->fs_maskbit << memif->data->fs_shift, + fs << memif->data->fs_shift); + + return 0; +} + +static int mt8365_afe_fe_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + int dai_id = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mtk_base_afe_memif *memif = &afe->memif[dai_id]; + struct mt8365_fe_dai_data *fe_data = &afe_priv->fe_data[dai_id]; + int ret = 0; + + if (fe_data->use_sram) { + snd_pcm_set_runtime_buffer(substream, NULL); + } else { + ret = snd_pcm_lib_free_pages(substream); + + mt8365_afe_emi_clk_off(afe); + } + + return ret; +} + +static int mt8365_afe_fe_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + int dai_id = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mtk_base_afe_memif *memif = &afe->memif[dai_id]; + + /* set format */ + if (memif->data->hd_reg >= 0) { + switch (substream->runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(afe->regmap, memif->data->hd_reg, + 3 << memif->data->hd_shift, + 0 << memif->data->hd_shift); + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(afe->regmap, memif->data->hd_reg, + 3 << memif->data->hd_shift, + 3 << memif->data->hd_shift); + + if (dai_id == MT8365_AFE_MEMIF_TDM_IN) { + regmap_update_bits(afe->regmap, + memif->data->hd_reg, + 3 << memif->data->hd_shift, + 1 << memif->data->hd_shift); + regmap_update_bits(afe->regmap, + memif->data->hd_reg, + 1 << memif->data->hd_align_mshift, + 1 << memif->data->hd_align_mshift); + } + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(afe->regmap, memif->data->hd_reg, + 3 << memif->data->hd_shift, + 1 << memif->data->hd_shift); + break; + default: + return -EINVAL; + } + } + + mt8365_afe_irq_direction_enable(afe, memif->irq_usage, + MT8365_AFE_IRQ_DIR_MCU); + + return 0; +} + +int mt8365_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + int dai_id = snd_soc_rtd_to_cpu(rtd, 0)->id; + struct mt8365_control_data *ctrl_data = &afe_priv->ctrl_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + /* enable channel merge */ + if (dai_id == MT8365_AFE_MEMIF_VUL2 && + !ctrl_data->bypass_cm1) { + regmap_update_bits(afe->regmap, AFE_CM1_CON0, + CM_AFE_CM_ON, CM_AFE_CM_ON); + } else if (dai_id == MT8365_AFE_MEMIF_TDM_IN && + !ctrl_data->bypass_cm2) { + regmap_update_bits(afe->regmap, AFE_CM2_CON0, + CM_AFE_CM_ON, CM_AFE_CM_ON); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + /* disable channel merge */ + if (dai_id == MT8365_AFE_MEMIF_VUL2 && + !ctrl_data->bypass_cm1) { + regmap_update_bits(afe->regmap, AFE_CM1_CON0, + CM_AFE_CM_ON, ~CM_AFE_CM_ON); + } else if (dai_id == MT8365_AFE_MEMIF_TDM_IN && + !ctrl_data->bypass_cm2) { + regmap_update_bits(afe->regmap, AFE_CM2_CON0, + CM_AFE_CM_ON, ~CM_AFE_CM_ON); + } + break; + default: + break; + } + + return mtk_afe_fe_trigger(substream, cmd, dai); +} + +static int mt8365_afe_hw_gain1_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + + mt8365_afe_enable_main_clk(afe); + return 0; +} + +static void mt8365_afe_hw_gain1_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = + &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + + if (be->prepared[substream->stream]) { + regmap_update_bits(afe->regmap, AFE_GAIN1_CON0, + AFE_GAIN1_CON0_EN_MASK, 0); + be->prepared[substream->stream] = false; + } + mt8365_afe_disable_main_clk(afe); +} + +static int mt8365_afe_hw_gain1_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + struct mt8365_be_dai_data *be = + &afe_priv->be_data[dai->id - MT8365_AFE_BACKEND_BASE]; + + int fs; + unsigned int val1 = 0, val2 = 0; + + if (be->prepared[substream->stream]) { + dev_info(afe->dev, "%s prepared already\n", __func__); + return 0; + } + + fs = mt8365_afe_fs_timing(substream->runtime->rate); + regmap_update_bits(afe->regmap, AFE_GAIN1_CON0, + AFE_GAIN1_CON0_MODE_MASK, (unsigned int)fs << 4); + + regmap_read(afe->regmap, AFE_GAIN1_CON1, &val1); + regmap_read(afe->regmap, AFE_GAIN1_CUR, &val2); + if ((val1 & AFE_GAIN1_CON1_MASK) != (val2 & AFE_GAIN1_CUR_MASK)) + regmap_update_bits(afe->regmap, AFE_GAIN1_CUR, + AFE_GAIN1_CUR_MASK, val1); + + regmap_update_bits(afe->regmap, AFE_GAIN1_CON0, + AFE_GAIN1_CON0_EN_MASK, 1); + be->prepared[substream->stream] = true; + + return 0; +} + +static const struct snd_pcm_hardware mt8365_hostless_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .period_bytes_min = 256, + .period_bytes_max = 4 * 48 * 1024, + .periods_min = 2, + .periods_max = 256, + .buffer_bytes_max = 8 * 48 * 1024, + .fifo_size = 0, +}; + +/* dai ops */ +static int mtk_dai_hostless_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + snd_soc_set_runtime_hwparams(substream, &mt8365_hostless_hardware); + + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + dev_err(afe->dev, "snd_pcm_hw_constraint_integer failed\n"); + return ret; +} + +/* FE DAIs */ +static const struct snd_soc_dai_ops mt8365_afe_fe_dai_ops = { + .startup = mt8365_afe_fe_startup, + .shutdown = mt8365_afe_fe_shutdown, + .hw_params = mt8365_afe_fe_hw_params, + .hw_free = mt8365_afe_fe_hw_free, + .prepare = mt8365_afe_fe_prepare, + .trigger = mt8365_afe_fe_trigger, +}; + +static const struct snd_soc_dai_ops mt8365_dai_hostless_ops = { + .startup = mtk_dai_hostless_startup, +}; + +static const struct snd_soc_dai_ops mt8365_afe_hw_gain1_ops = { + .startup = mt8365_afe_hw_gain1_startup, + .shutdown = mt8365_afe_hw_gain1_shutdown, + .prepare = mt8365_afe_hw_gain1_prepare, +}; + +static struct snd_soc_dai_driver mt8365_memif_dai_driver[] = { + /* FE DAIs: memory intefaces to CPU */ + { + .name = "DL1", + .id = MT8365_AFE_MEMIF_DL1, + .playback = { + .stream_name = "DL1", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "DL2", + .id = MT8365_AFE_MEMIF_DL2, + .playback = { + .stream_name = "DL2", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "TDM_OUT", + .id = MT8365_AFE_MEMIF_TDM_OUT, + .playback = { + .stream_name = "TDM_OUT", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "AWB", + .id = MT8365_AFE_MEMIF_AWB, + .capture = { + .stream_name = "AWB", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "VUL", + .id = MT8365_AFE_MEMIF_VUL, + .capture = { + .stream_name = "VUL", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "VUL2", + .id = MT8365_AFE_MEMIF_VUL2, + .capture = { + .stream_name = "VUL2", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "VUL3", + .id = MT8365_AFE_MEMIF_VUL3, + .capture = { + .stream_name = "VUL3", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "TDM_IN", + .id = MT8365_AFE_MEMIF_TDM_IN, + .capture = { + .stream_name = "TDM_IN", + .channels_min = 1, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_fe_dai_ops, + }, { + .name = "Hostless FM DAI", + .id = MT8365_AFE_IO_VIRTUAL_FM, + .playback = { + .stream_name = "Hostless FM DL", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Hostless FM UL", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_dai_hostless_ops, + }, { + .name = "HW_GAIN1", + .id = MT8365_AFE_IO_HW_GAIN1, + .playback = { + .stream_name = "HW Gain 1 In", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "HW Gain 1 Out", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &mt8365_afe_hw_gain1_ops, + .symmetric_rate = 1, + .symmetric_channels = 1, + .symmetric_sample_bits = 1, + }, +}; + +static const struct snd_kcontrol_new mt8365_afe_o00_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN0, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN0, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o01_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN1, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN1, 8, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o03_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN3, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN3, 7, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I00 Switch", AFE_CONN3, 0, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I10 Switch", AFE_CONN3, 10, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o04_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN4, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN4, 8, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I01 Switch", AFE_CONN4, 1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I11 Switch", AFE_CONN4, 11, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o05_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I00 Switch", AFE_CONN5, 0, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN5, 3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN5, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN5, 7, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I09 Switch", AFE_CONN5, 9, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I14 Switch", AFE_CONN5, 14, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I16 Switch", AFE_CONN5, 16, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN5, 18, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN5, 20, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN5, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I10L Switch", AFE_CONN5, 10, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o06_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I01 Switch", AFE_CONN6, 1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN6, 4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN6, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN6, 8, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I22 Switch", AFE_CONN6, 22, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I15 Switch", AFE_CONN6, 15, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN6, 17, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN6, 19, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN6, 21, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN6, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I11L Switch", AFE_CONN6, 11, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o07_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN7, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN7, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o08_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN8, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN8, 8, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o09_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I00 Switch", AFE_CONN9, 0, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN9, 3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I09 Switch", AFE_CONN9, 9, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I14 Switch", AFE_CONN9, 14, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I16 Switch", AFE_CONN9, 16, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN9, 18, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN9, 20, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o10_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I01 Switch", AFE_CONN10, 1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN10, 4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I22 Switch", AFE_CONN10, 22, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I15 Switch", AFE_CONN10, 15, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN10, 17, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN10, 19, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN10, 21, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o11_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I00 Switch", AFE_CONN11, 0, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN11, 3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I09 Switch", AFE_CONN11, 9, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I14 Switch", AFE_CONN11, 14, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I16 Switch", AFE_CONN11, 16, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN11, 18, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN11, 20, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o12_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I01 Switch", AFE_CONN12, 1, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN12, 4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I22 Switch", AFE_CONN12, 22, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I15 Switch", AFE_CONN12, 15, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN12, 17, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN12, 19, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN12, 21, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o13_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I00 Switch", AFE_CONN13, 0, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o14_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I01 Switch", AFE_CONN14, 1, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o15_mix[] = { +}; + +static const struct snd_kcontrol_new mt8365_afe_o16_mix[] = { +}; + +static const struct snd_kcontrol_new mt8365_afe_o17_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I03 Switch", AFE_CONN17, 3, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I14 Switch", AFE_CONN17, 14, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o18_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN18, 4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I15 Switch", AFE_CONN18, 15, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN18, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN18, 25, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o19_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I04 Switch", AFE_CONN19, 4, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I16 Switch", AFE_CONN19, 16, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN19, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN19, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN19, 25, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I26 Switch", AFE_CONN19, 26, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o20_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I17 Switch", AFE_CONN20, 17, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN20, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I26 Switch", AFE_CONN20, 26, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o21_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I18 Switch", AFE_CONN21, 18, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN21, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN21, 25, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o22_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I19 Switch", AFE_CONN22, 19, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN22, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I26 Switch", AFE_CONN22, 26, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o23_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I20 Switch", AFE_CONN23, 20, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN23, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN23, 25, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o24_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I21 Switch", AFE_CONN24, 21, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN24, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I26 Switch", AFE_CONN24, 26, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN24, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN24, 25, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o25_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I27 Switch", AFE_CONN25, 27, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I23 Switch", AFE_CONN25, 23, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I25 Switch", AFE_CONN25, 25, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o26_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I28 Switch", AFE_CONN26, 28, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I24 Switch", AFE_CONN26, 24, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I26 Switch", AFE_CONN26, 26, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o27_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN27, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN27, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o28_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN28, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN28, 8, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o29_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I05 Switch", AFE_CONN29, 5, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I07 Switch", AFE_CONN29, 7, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o30_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I06 Switch", AFE_CONN30, 6, 1, 0), + SOC_DAPM_SINGLE_AUTODISABLE("I08 Switch", AFE_CONN30, 8, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o31_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I29 Switch", AFE_CONN31, 29, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o32_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I30 Switch", AFE_CONN32, 30, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o33_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I31 Switch", AFE_CONN33, 31, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o34_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I32 Switch", AFE_CONN34_1, 0, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o35_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I33 Switch", AFE_CONN35_1, 1, 1, 0), +}; + +static const struct snd_kcontrol_new mt8365_afe_o36_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("I34 Switch", AFE_CONN36_1, 2, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_gain1_in_ch1_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("CONNSYS_I2S_CH1 Switch", AFE_CONN13, + 0, 1, 0), +}; + +static const struct snd_kcontrol_new mtk_hw_gain1_in_ch2_mix[] = { + SOC_DAPM_SINGLE_AUTODISABLE("CONNSYS_I2S_CH2 Switch", AFE_CONN14, + 1, 1, 0), +}; + +static int mt8365_afe_cm2_io_input_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = mCM2Input; + + return 0; +} + +static int mt8365_afe_cm2_io_input_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct snd_soc_component *comp = snd_soc_dapm_to_component(dapm); + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(comp); + struct mt8365_afe_private *afe_priv = afe->platform_priv; + int ret; + + mCM2Input = ucontrol->value.enumerated.item[0]; + + afe_priv->cm2_mux_input = mCM2Input; + ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return ret; +} + +static const char * const fmhwgain_text[] = { + "OPEN", "FM_HW_GAIN_IO" +}; + +static const char * const ain_text[] = { + "INT ADC", "EXT ADC", +}; + +static const char * const vul2_in_input_text[] = { + "VUL2_IN_FROM_O17O18", "VUL2_IN_FROM_CM1", +}; + +static const char * const mt8365_afe_cm2_mux_text[] = { + "OPEN", "FROM_GASRC1_OUT", "FROM_GASRC2_OUT", "FROM_TDM_ASRC_OUT", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(fmhwgain_enum, fmhwgain_text); +static SOC_ENUM_SINGLE_DECL(ain_enum, AFE_ADDA_TOP_CON0, 0, ain_text); +static SOC_ENUM_SINGLE_VIRT_DECL(vul2_in_input_enum, vul2_in_input_text); +static SOC_ENUM_SINGLE_VIRT_DECL(mt8365_afe_cm2_mux_input_enum, + mt8365_afe_cm2_mux_text); + +static const struct snd_kcontrol_new fmhwgain_mux = + SOC_DAPM_ENUM("FM HW Gain Source", fmhwgain_enum); + +static const struct snd_kcontrol_new ain_mux = + SOC_DAPM_ENUM("AIN Source", ain_enum); + +static const struct snd_kcontrol_new vul2_in_input_mux = + SOC_DAPM_ENUM("VUL2 Input", vul2_in_input_enum); + +static const struct snd_kcontrol_new mt8365_afe_cm2_mux_input_mux = + SOC_DAPM_ENUM_EXT("CM2_MUX Source", mt8365_afe_cm2_mux_input_enum, + mt8365_afe_cm2_io_input_mux_get, + mt8365_afe_cm2_io_input_mux_put); + +static const struct snd_soc_dapm_widget mt8365_memif_widgets[] = { + /* inter-connections */ + SND_SOC_DAPM_MIXER("I00", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I01", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I03", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I04", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I05", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I06", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I07", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I08", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I05L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I06L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I07L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I08L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I09", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I10", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I11", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I10L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I11L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I12", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I13", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I14", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I15", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I16", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I17", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I18", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I19", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I20", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I21", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I22", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I23", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I24", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I25", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I26", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I27", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I28", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I29", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I30", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I31", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I32", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I33", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("I34", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("O00", SND_SOC_NOPM, 0, 0, + mt8365_afe_o00_mix, ARRAY_SIZE(mt8365_afe_o00_mix)), + SND_SOC_DAPM_MIXER("O01", SND_SOC_NOPM, 0, 0, + mt8365_afe_o01_mix, ARRAY_SIZE(mt8365_afe_o01_mix)), + SND_SOC_DAPM_MIXER("O03", SND_SOC_NOPM, 0, 0, + mt8365_afe_o03_mix, ARRAY_SIZE(mt8365_afe_o03_mix)), + SND_SOC_DAPM_MIXER("O04", SND_SOC_NOPM, 0, 0, + mt8365_afe_o04_mix, ARRAY_SIZE(mt8365_afe_o04_mix)), + SND_SOC_DAPM_MIXER("O05", SND_SOC_NOPM, 0, 0, + mt8365_afe_o05_mix, ARRAY_SIZE(mt8365_afe_o05_mix)), + SND_SOC_DAPM_MIXER("O06", SND_SOC_NOPM, 0, 0, + mt8365_afe_o06_mix, ARRAY_SIZE(mt8365_afe_o06_mix)), + SND_SOC_DAPM_MIXER("O07", SND_SOC_NOPM, 0, 0, + mt8365_afe_o07_mix, ARRAY_SIZE(mt8365_afe_o07_mix)), + SND_SOC_DAPM_MIXER("O08", SND_SOC_NOPM, 0, 0, + mt8365_afe_o08_mix, ARRAY_SIZE(mt8365_afe_o08_mix)), + SND_SOC_DAPM_MIXER("O09", SND_SOC_NOPM, 0, 0, + mt8365_afe_o09_mix, ARRAY_SIZE(mt8365_afe_o09_mix)), + SND_SOC_DAPM_MIXER("O10", SND_SOC_NOPM, 0, 0, + mt8365_afe_o10_mix, ARRAY_SIZE(mt8365_afe_o10_mix)), + SND_SOC_DAPM_MIXER("O11", SND_SOC_NOPM, 0, 0, + mt8365_afe_o11_mix, ARRAY_SIZE(mt8365_afe_o11_mix)), + SND_SOC_DAPM_MIXER("O12", SND_SOC_NOPM, 0, 0, + mt8365_afe_o12_mix, ARRAY_SIZE(mt8365_afe_o12_mix)), + SND_SOC_DAPM_MIXER("O13", SND_SOC_NOPM, 0, 0, + mt8365_afe_o13_mix, ARRAY_SIZE(mt8365_afe_o13_mix)), + SND_SOC_DAPM_MIXER("O14", SND_SOC_NOPM, 0, 0, + mt8365_afe_o14_mix, ARRAY_SIZE(mt8365_afe_o14_mix)), + SND_SOC_DAPM_MIXER("O15", SND_SOC_NOPM, 0, 0, + mt8365_afe_o15_mix, ARRAY_SIZE(mt8365_afe_o15_mix)), + SND_SOC_DAPM_MIXER("O16", SND_SOC_NOPM, 0, 0, + mt8365_afe_o16_mix, ARRAY_SIZE(mt8365_afe_o16_mix)), + SND_SOC_DAPM_MIXER("O17", SND_SOC_NOPM, 0, 0, + mt8365_afe_o17_mix, ARRAY_SIZE(mt8365_afe_o17_mix)), + SND_SOC_DAPM_MIXER("O18", SND_SOC_NOPM, 0, 0, + mt8365_afe_o18_mix, ARRAY_SIZE(mt8365_afe_o18_mix)), + SND_SOC_DAPM_MIXER("O19", SND_SOC_NOPM, 0, 0, + mt8365_afe_o19_mix, ARRAY_SIZE(mt8365_afe_o19_mix)), + SND_SOC_DAPM_MIXER("O20", SND_SOC_NOPM, 0, 0, + mt8365_afe_o20_mix, ARRAY_SIZE(mt8365_afe_o20_mix)), + SND_SOC_DAPM_MIXER("O21", SND_SOC_NOPM, 0, 0, + mt8365_afe_o21_mix, ARRAY_SIZE(mt8365_afe_o21_mix)), + SND_SOC_DAPM_MIXER("O22", SND_SOC_NOPM, 0, 0, + mt8365_afe_o22_mix, ARRAY_SIZE(mt8365_afe_o22_mix)), + SND_SOC_DAPM_MIXER("O23", SND_SOC_NOPM, 0, 0, + mt8365_afe_o23_mix, ARRAY_SIZE(mt8365_afe_o23_mix)), + SND_SOC_DAPM_MIXER("O24", SND_SOC_NOPM, 0, 0, + mt8365_afe_o24_mix, ARRAY_SIZE(mt8365_afe_o24_mix)), + SND_SOC_DAPM_MIXER("O25", SND_SOC_NOPM, 0, 0, + mt8365_afe_o25_mix, ARRAY_SIZE(mt8365_afe_o25_mix)), + SND_SOC_DAPM_MIXER("O26", SND_SOC_NOPM, 0, 0, + mt8365_afe_o26_mix, ARRAY_SIZE(mt8365_afe_o26_mix)), + SND_SOC_DAPM_MIXER("O27", SND_SOC_NOPM, 0, 0, + mt8365_afe_o27_mix, ARRAY_SIZE(mt8365_afe_o27_mix)), + SND_SOC_DAPM_MIXER("O28", SND_SOC_NOPM, 0, 0, + mt8365_afe_o28_mix, ARRAY_SIZE(mt8365_afe_o28_mix)), + SND_SOC_DAPM_MIXER("O29", SND_SOC_NOPM, 0, 0, + mt8365_afe_o29_mix, ARRAY_SIZE(mt8365_afe_o29_mix)), + SND_SOC_DAPM_MIXER("O30", SND_SOC_NOPM, 0, 0, + mt8365_afe_o30_mix, ARRAY_SIZE(mt8365_afe_o30_mix)), + SND_SOC_DAPM_MIXER("O31", SND_SOC_NOPM, 0, 0, + mt8365_afe_o31_mix, ARRAY_SIZE(mt8365_afe_o31_mix)), + SND_SOC_DAPM_MIXER("O32", SND_SOC_NOPM, 0, 0, + mt8365_afe_o32_mix, ARRAY_SIZE(mt8365_afe_o32_mix)), + SND_SOC_DAPM_MIXER("O33", SND_SOC_NOPM, 0, 0, + mt8365_afe_o33_mix, ARRAY_SIZE(mt8365_afe_o33_mix)), + SND_SOC_DAPM_MIXER("O34", SND_SOC_NOPM, 0, 0, + mt8365_afe_o34_mix, ARRAY_SIZE(mt8365_afe_o34_mix)), + SND_SOC_DAPM_MIXER("O35", SND_SOC_NOPM, 0, 0, + mt8365_afe_o35_mix, ARRAY_SIZE(mt8365_afe_o35_mix)), + SND_SOC_DAPM_MIXER("O36", SND_SOC_NOPM, 0, 0, + mt8365_afe_o36_mix, ARRAY_SIZE(mt8365_afe_o36_mix)), + SND_SOC_DAPM_MIXER("CM2_Mux IO", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("CM1_IO", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("O17O18", SND_SOC_NOPM, 0, 0, NULL, 0), + /* inter-connections */ + SND_SOC_DAPM_MIXER("HW_GAIN1_IN_CH1", SND_SOC_NOPM, 0, 0, + mtk_hw_gain1_in_ch1_mix, + ARRAY_SIZE(mtk_hw_gain1_in_ch1_mix)), + SND_SOC_DAPM_MIXER("HW_GAIN1_IN_CH2", SND_SOC_NOPM, 0, 0, + mtk_hw_gain1_in_ch2_mix, + ARRAY_SIZE(mtk_hw_gain1_in_ch2_mix)), + + SND_SOC_DAPM_INPUT("DL Source"), + + SND_SOC_DAPM_MUX("CM2_Mux_IO Input Mux", SND_SOC_NOPM, 0, 0, + &mt8365_afe_cm2_mux_input_mux), + + SND_SOC_DAPM_MUX("AIN Mux", SND_SOC_NOPM, 0, 0, &ain_mux), + SND_SOC_DAPM_MUX("VUL2 Input Mux", SND_SOC_NOPM, 0, 0, + &vul2_in_input_mux), + + SND_SOC_DAPM_MUX("FM HW Gain Mux", SND_SOC_NOPM, 0, 0, &fmhwgain_mux), + + SND_SOC_DAPM_INPUT("HW Gain 1 Out Endpoint"), + SND_SOC_DAPM_OUTPUT("HW Gain 1 In Endpoint"), +}; + +static const struct snd_soc_dapm_route mt8365_memif_routes[] = { + /* downlink */ + {"I00", NULL, "2ND I2S Capture"}, + {"I01", NULL, "2ND I2S Capture"}, + {"I05", NULL, "DL1"}, + {"I06", NULL, "DL1"}, + {"I07", NULL, "DL2"}, + {"I08", NULL, "DL2"}, + + {"O03", "I05 Switch", "I05"}, + {"O04", "I06 Switch", "I06"}, + {"O00", "I05 Switch", "I05"}, + {"O01", "I06 Switch", "I06"}, + {"O07", "I05 Switch", "I05"}, + {"O08", "I06 Switch", "I06"}, + {"O27", "I05 Switch", "I05"}, + {"O28", "I06 Switch", "I06"}, + {"O29", "I05 Switch", "I05"}, + {"O30", "I06 Switch", "I06"}, + + {"O03", "I07 Switch", "I07"}, + {"O04", "I08 Switch", "I08"}, + {"O00", "I07 Switch", "I07"}, + {"O01", "I08 Switch", "I08"}, + {"O07", "I07 Switch", "I07"}, + {"O08", "I08 Switch", "I08"}, + + /* uplink */ + {"AWB", NULL, "O05"}, + {"AWB", NULL, "O06"}, + {"VUL", NULL, "O09"}, + {"VUL", NULL, "O10"}, + {"VUL3", NULL, "O11"}, + {"VUL3", NULL, "O12"}, + + {"AIN Mux", "EXT ADC", "I2S Capture"}, + {"I03", NULL, "AIN Mux"}, + {"I04", NULL, "AIN Mux"}, + + {"HW_GAIN1_IN_CH1", "CONNSYS_I2S_CH1", "Hostless FM DL"}, + {"HW_GAIN1_IN_CH2", "CONNSYS_I2S_CH2", "Hostless FM DL"}, + + {"HW Gain 1 In Endpoint", NULL, "HW Gain 1 In"}, + {"HW Gain 1 Out", NULL, "HW Gain 1 Out Endpoint"}, + {"HW Gain 1 In", NULL, "HW_GAIN1_IN_CH1"}, + {"HW Gain 1 In", NULL, "HW_GAIN1_IN_CH2"}, + + {"FM HW Gain Mux", "FM_HW_GAIN_IO", "HW Gain 1 Out"}, + {"Hostless FM UL", NULL, "FM HW Gain Mux"}, + {"Hostless FM UL", NULL, "FM 2ND I2S Mux"}, + + {"O05", "I05 Switch", "I05L"}, + {"O06", "I06 Switch", "I06L"}, + {"O05", "I07 Switch", "I07L"}, + {"O06", "I08 Switch", "I08L"}, + + {"O05", "I03 Switch", "I03"}, + {"O06", "I04 Switch", "I04"}, + {"O05", "I00 Switch", "I00"}, + {"O06", "I01 Switch", "I01"}, + {"O05", "I09 Switch", "I09"}, + {"O06", "I22 Switch", "I22"}, + {"O05", "I14 Switch", "I14"}, + {"O06", "I15 Switch", "I15"}, + {"O05", "I16 Switch", "I16"}, + {"O06", "I17 Switch", "I17"}, + {"O05", "I18 Switch", "I18"}, + {"O06", "I19 Switch", "I19"}, + {"O05", "I20 Switch", "I20"}, + {"O06", "I21 Switch", "I21"}, + {"O05", "I23 Switch", "I23"}, + {"O06", "I24 Switch", "I24"}, + + {"O09", "I03 Switch", "I03"}, + {"O10", "I04 Switch", "I04"}, + {"O09", "I00 Switch", "I00"}, + {"O10", "I01 Switch", "I01"}, + {"O09", "I09 Switch", "I09"}, + {"O10", "I22 Switch", "I22"}, + {"O09", "I14 Switch", "I14"}, + {"O10", "I15 Switch", "I15"}, + {"O09", "I16 Switch", "I16"}, + {"O10", "I17 Switch", "I17"}, + {"O09", "I18 Switch", "I18"}, + {"O10", "I19 Switch", "I19"}, + {"O09", "I20 Switch", "I20"}, + {"O10", "I21 Switch", "I21"}, + + {"O11", "I03 Switch", "I03"}, + {"O12", "I04 Switch", "I04"}, + {"O11", "I00 Switch", "I00"}, + {"O12", "I01 Switch", "I01"}, + {"O11", "I09 Switch", "I09"}, + {"O12", "I22 Switch", "I22"}, + {"O11", "I14 Switch", "I14"}, + {"O12", "I15 Switch", "I15"}, + {"O11", "I16 Switch", "I16"}, + {"O12", "I17 Switch", "I17"}, + {"O11", "I18 Switch", "I18"}, + {"O12", "I19 Switch", "I19"}, + {"O11", "I20 Switch", "I20"}, + {"O12", "I21 Switch", "I21"}, + + /* CM2_Mux*/ + {"CM2_Mux IO", NULL, "CM2_Mux_IO Input Mux"}, + + /* VUL2 */ + {"VUL2", NULL, "VUL2 Input Mux"}, + {"VUL2 Input Mux", "VUL2_IN_FROM_O17O18", "O17O18"}, + {"VUL2 Input Mux", "VUL2_IN_FROM_CM1", "CM1_IO"}, + + {"O17O18", NULL, "O17"}, + {"O17O18", NULL, "O18"}, + {"CM1_IO", NULL, "O17"}, + {"CM1_IO", NULL, "O18"}, + {"CM1_IO", NULL, "O19"}, + {"CM1_IO", NULL, "O20"}, + {"CM1_IO", NULL, "O21"}, + {"CM1_IO", NULL, "O22"}, + {"CM1_IO", NULL, "O23"}, + {"CM1_IO", NULL, "O24"}, + {"CM1_IO", NULL, "O25"}, + {"CM1_IO", NULL, "O26"}, + {"CM1_IO", NULL, "O31"}, + {"CM1_IO", NULL, "O32"}, + {"CM1_IO", NULL, "O33"}, + {"CM1_IO", NULL, "O34"}, + {"CM1_IO", NULL, "O35"}, + {"CM1_IO", NULL, "O36"}, + + {"O17", "I14 Switch", "I14"}, + {"O18", "I15 Switch", "I15"}, + {"O19", "I16 Switch", "I16"}, + {"O20", "I17 Switch", "I17"}, + {"O21", "I18 Switch", "I18"}, + {"O22", "I19 Switch", "I19"}, + {"O23", "I20 Switch", "I20"}, + {"O24", "I21 Switch", "I21"}, + {"O25", "I23 Switch", "I23"}, + {"O26", "I24 Switch", "I24"}, + {"O25", "I25 Switch", "I25"}, + {"O26", "I26 Switch", "I26"}, + + {"O17", "I03 Switch", "I03"}, + {"O18", "I04 Switch", "I04"}, + {"O18", "I23 Switch", "I23"}, + {"O18", "I25 Switch", "I25"}, + {"O19", "I04 Switch", "I04"}, + {"O19", "I23 Switch", "I23"}, + {"O19", "I24 Switch", "I24"}, + {"O19", "I25 Switch", "I25"}, + {"O19", "I26 Switch", "I26"}, + {"O20", "I24 Switch", "I24"}, + {"O20", "I26 Switch", "I26"}, + {"O21", "I23 Switch", "I23"}, + {"O21", "I25 Switch", "I25"}, + {"O22", "I24 Switch", "I24"}, + {"O22", "I26 Switch", "I26"}, + + {"O23", "I23 Switch", "I23"}, + {"O23", "I25 Switch", "I25"}, + {"O24", "I24 Switch", "I24"}, + {"O24", "I26 Switch", "I26"}, + {"O24", "I23 Switch", "I23"}, + {"O24", "I25 Switch", "I25"}, + {"O13", "I00 Switch", "I00"}, + {"O14", "I01 Switch", "I01"}, + {"O03", "I10 Switch", "I10"}, + {"O04", "I11 Switch", "I11"}, +}; + +static const struct mtk_base_memif_data memif_data[MT8365_AFE_MEMIF_NUM] = { + { + .name = "DL1", + .id = MT8365_AFE_MEMIF_DL1, + .reg_ofs_base = AFE_DL1_BASE, + .reg_ofs_cur = AFE_DL1_CUR, + .fs_reg = AFE_DAC_CON1, + .fs_shift = 0, + .fs_maskbit = 0xf, + .mono_reg = AFE_DAC_CON1, + .mono_shift = 21, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 16, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 1, + .msb_reg = -1, + .msb_shift = -1, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "DL2", + .id = MT8365_AFE_MEMIF_DL2, + .reg_ofs_base = AFE_DL2_BASE, + .reg_ofs_cur = AFE_DL2_CUR, + .fs_reg = AFE_DAC_CON1, + .fs_shift = 4, + .fs_maskbit = 0xf, + .mono_reg = AFE_DAC_CON1, + .mono_shift = 22, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 18, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 2, + .msb_reg = -1, + .msb_shift = -1, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "TDM OUT", + .id = MT8365_AFE_MEMIF_TDM_OUT, + .reg_ofs_base = AFE_HDMI_OUT_BASE, + .reg_ofs_cur = AFE_HDMI_OUT_CUR, + .fs_reg = -1, + .fs_shift = -1, + .fs_maskbit = -1, + .mono_reg = -1, + .mono_shift = -1, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 28, + .enable_reg = AFE_HDMI_OUT_CON0, + .enable_shift = 0, + .msb_reg = -1, + .msb_shift = -1, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "AWB", + .id = MT8365_AFE_MEMIF_AWB, + .reg_ofs_base = AFE_AWB_BASE, + .reg_ofs_cur = AFE_AWB_CUR, + .fs_reg = AFE_DAC_CON1, + .fs_shift = 12, + .fs_maskbit = 0xf, + .mono_reg = AFE_DAC_CON1, + .mono_shift = 24, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 20, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 6, + .msb_reg = AFE_MEMIF_MSB, + .msb_shift = 17, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "VUL", + .id = MT8365_AFE_MEMIF_VUL, + .reg_ofs_base = AFE_VUL_BASE, + .reg_ofs_cur = AFE_VUL_CUR, + .fs_reg = AFE_DAC_CON1, + .fs_shift = 16, + .fs_maskbit = 0xf, + .mono_reg = AFE_DAC_CON1, + .mono_shift = 27, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 22, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 3, + .msb_reg = AFE_MEMIF_MSB, + .msb_shift = 20, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "VUL2", + .id = MT8365_AFE_MEMIF_VUL2, + .reg_ofs_base = AFE_VUL_D2_BASE, + .reg_ofs_cur = AFE_VUL_D2_CUR, + .fs_reg = AFE_DAC_CON0, + .fs_shift = 20, + .fs_maskbit = 0xf, + .mono_reg = -1, + .mono_shift = -1, + .hd_reg = AFE_MEMIF_PBUF_SIZE, + .hd_shift = 14, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 9, + .msb_reg = AFE_MEMIF_MSB, + .msb_shift = 21, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "VUL3", + .id = MT8365_AFE_MEMIF_VUL3, + .reg_ofs_base = AFE_VUL3_BASE, + .reg_ofs_cur = AFE_VUL3_CUR, + .fs_reg = AFE_DAC_CON1, + .fs_shift = 8, + .fs_maskbit = 0xf, + .mono_reg = AFE_DAC_CON0, + .mono_shift = 13, + .hd_reg = AFE_MEMIF_PBUF2_SIZE, + .hd_shift = 10, + .enable_reg = AFE_DAC_CON0, + .enable_shift = 12, + .msb_reg = AFE_MEMIF_MSB, + .msb_shift = 27, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, { + .name = "TDM IN", + .id = MT8365_AFE_MEMIF_TDM_IN, + .reg_ofs_base = AFE_HDMI_IN_2CH_BASE, + .reg_ofs_cur = AFE_HDMI_IN_2CH_CUR, + .fs_reg = -1, + .fs_shift = -1, + .fs_maskbit = -1, + .mono_reg = AFE_HDMI_IN_2CH_CON0, + .mono_shift = 1, + .hd_reg = AFE_MEMIF_PBUF2_SIZE, + .hd_shift = 8, + .hd_align_mshift = 5, + .enable_reg = AFE_HDMI_IN_2CH_CON0, + .enable_shift = 0, + .msb_reg = AFE_MEMIF_MSB, + .msb_shift = 28, + .agent_disable_reg = -1, + .agent_disable_shift = -1, + }, +}; + +static const struct mtk_base_irq_data irq_data[MT8365_AFE_IRQ_NUM] = { + { + .id = MT8365_AFE_IRQ1, + .irq_cnt_reg = AFE_IRQ_MCU_CNT1, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 0, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 4, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 0, + }, { + .id = MT8365_AFE_IRQ2, + .irq_cnt_reg = AFE_IRQ_MCU_CNT2, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 1, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 8, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 1, + }, { + .id = MT8365_AFE_IRQ3, + .irq_cnt_reg = AFE_IRQ_MCU_CNT3, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 2, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 16, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 2, + }, { + .id = MT8365_AFE_IRQ4, + .irq_cnt_reg = AFE_IRQ_MCU_CNT4, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 3, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 20, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 3, + }, { + .id = MT8365_AFE_IRQ5, + .irq_cnt_reg = AFE_IRQ_MCU_CNT5, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON2, + .irq_en_shift = 3, + .irq_fs_reg = -1, + .irq_fs_shift = 0, + .irq_fs_maskbit = 0x0, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 4, + }, { + .id = MT8365_AFE_IRQ6, + .irq_cnt_reg = -1, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x0, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 13, + .irq_fs_reg = -1, + .irq_fs_shift = 0, + .irq_fs_maskbit = 0x0, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 5, + }, { + .id = MT8365_AFE_IRQ7, + .irq_cnt_reg = AFE_IRQ_MCU_CNT7, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 14, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 24, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 6, + }, { + .id = MT8365_AFE_IRQ8, + .irq_cnt_reg = AFE_IRQ_MCU_CNT8, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON, + .irq_en_shift = 15, + .irq_fs_reg = AFE_IRQ_MCU_CON, + .irq_fs_shift = 28, + .irq_fs_maskbit = 0xf, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 7, + }, { + .id = MT8365_AFE_IRQ9, + .irq_cnt_reg = -1, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x0, + .irq_en_reg = AFE_IRQ_MCU_CON2, + .irq_en_shift = 2, + .irq_fs_reg = -1, + .irq_fs_shift = 0, + .irq_fs_maskbit = 0x0, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 8, + }, { + .id = MT8365_AFE_IRQ10, + .irq_cnt_reg = AFE_IRQ_MCU_CNT10, + .irq_cnt_shift = 0, + .irq_cnt_maskbit = 0x3ffff, + .irq_en_reg = AFE_IRQ_MCU_CON2, + .irq_en_shift = 4, + .irq_fs_reg = -1, + .irq_fs_shift = 0, + .irq_fs_maskbit = 0x0, + .irq_clr_reg = AFE_IRQ_MCU_CLR, + .irq_clr_shift = 9, + }, +}; + +static int memif_specified_irqs[MT8365_AFE_MEMIF_NUM] = { + [MT8365_AFE_MEMIF_DL1] = MT8365_AFE_IRQ1, + [MT8365_AFE_MEMIF_DL2] = MT8365_AFE_IRQ2, + [MT8365_AFE_MEMIF_TDM_OUT] = MT8365_AFE_IRQ5, + [MT8365_AFE_MEMIF_AWB] = MT8365_AFE_IRQ3, + [MT8365_AFE_MEMIF_VUL] = MT8365_AFE_IRQ4, + [MT8365_AFE_MEMIF_VUL2] = MT8365_AFE_IRQ7, + [MT8365_AFE_MEMIF_VUL3] = MT8365_AFE_IRQ8, + [MT8365_AFE_MEMIF_TDM_IN] = MT8365_AFE_IRQ10, +}; + +static const struct regmap_config mt8365_afe_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = MAX_REGISTER, + .cache_type = REGCACHE_NONE, +}; + +static irqreturn_t mt8365_afe_irq_handler(int irq, void *dev_id) +{ + struct mtk_base_afe *afe = dev_id; + unsigned int reg_value; + unsigned int mcu_irq_mask; + int i, ret; + + ret = regmap_read(afe->regmap, AFE_IRQ_MCU_STATUS, ®_value); + if (ret) { + dev_err_ratelimited(afe->dev, "%s irq status err\n", __func__); + reg_value = AFE_IRQ_STATUS_BITS; + goto err_irq; + } + + ret = regmap_read(afe->regmap, AFE_IRQ_MCU_EN, &mcu_irq_mask); + if (ret) { + dev_err_ratelimited(afe->dev, "%s irq mcu_en err\n", __func__); + reg_value = AFE_IRQ_STATUS_BITS; + goto err_irq; + } + + /* only clr cpu irq */ + reg_value &= mcu_irq_mask; + + for (i = 0; i < MT8365_AFE_MEMIF_NUM; i++) { + struct mtk_base_afe_memif *memif = &afe->memif[i]; + struct mtk_base_afe_irq *mcu_irq; + + if (memif->irq_usage < 0) + continue; + + mcu_irq = &afe->irqs[memif->irq_usage]; + + if (!(reg_value & (1 << mcu_irq->irq_data->irq_clr_shift))) + continue; + + snd_pcm_period_elapsed(memif->substream); + } + +err_irq: + /* clear irq */ + regmap_write(afe->regmap, AFE_IRQ_MCU_CLR, + reg_value & AFE_IRQ_STATUS_BITS); + + return IRQ_HANDLED; +} + +static int __maybe_unused mt8365_afe_runtime_suspend(struct device *dev) +{ + return 0; +} + +static int mt8365_afe_runtime_resume(struct device *dev) +{ + return 0; +} + +static int __maybe_unused mt8365_afe_suspend(struct device *dev) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dev); + struct regmap *regmap = afe->regmap; + int i; + + mt8365_afe_enable_main_clk(afe); + + if (!afe->reg_back_up) + afe->reg_back_up = + devm_kcalloc(dev, afe->reg_back_up_list_num, + sizeof(unsigned int), GFP_KERNEL); + + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_read(regmap, afe->reg_back_up_list[i], + &afe->reg_back_up[i]); + + mt8365_afe_disable_main_clk(afe); + + return 0; +} + +static int __maybe_unused mt8365_afe_resume(struct device *dev) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dev); + struct regmap *regmap = afe->regmap; + int i = 0; + + if (!afe->reg_back_up) + return 0; + + mt8365_afe_enable_main_clk(afe); + + for (i = 0; i < afe->reg_back_up_list_num; i++) + regmap_write(regmap, afe->reg_back_up_list[i], + afe->reg_back_up[i]); + + mt8365_afe_disable_main_clk(afe); + + return 0; +} + +static int __maybe_unused mt8365_afe_dev_runtime_suspend(struct device *dev) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dev); + + if (pm_runtime_status_suspended(dev) || afe->suspended) + return 0; + + mt8365_afe_suspend(dev); + afe->suspended = true; + return 0; +} + +static int __maybe_unused mt8365_afe_dev_runtime_resume(struct device *dev) +{ + struct mtk_base_afe *afe = dev_get_drvdata(dev); + + if (pm_runtime_status_suspended(dev) || !afe->suspended) + return 0; + + mt8365_afe_resume(dev); + afe->suspended = false; + return 0; +} + +static int mt8365_afe_init_registers(struct mtk_base_afe *afe) +{ + size_t i; + + static struct { + unsigned int reg; + unsigned int mask; + unsigned int val; + } init_regs[] = { + { AFE_CONN_24BIT, GENMASK(31, 0), GENMASK(31, 0) }, + { AFE_CONN_24BIT_1, GENMASK(21, 0), GENMASK(21, 0) }, + }; + + mt8365_afe_enable_main_clk(afe); + + for (i = 0; i < ARRAY_SIZE(init_regs); i++) + regmap_update_bits(afe->regmap, init_regs[i].reg, + init_regs[i].mask, init_regs[i].val); + + mt8365_afe_disable_main_clk(afe); + + return 0; +} + +static int mt8365_dai_memif_register(struct mtk_base_afe *afe) +{ + struct mtk_base_afe_dai *dai; + + dai = devm_kzalloc(afe->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + + list_add(&dai->list, &afe->sub_dais); + + dai->dai_drivers = mt8365_memif_dai_driver; + dai->num_dai_drivers = ARRAY_SIZE(mt8365_memif_dai_driver); + + dai->dapm_widgets = mt8365_memif_widgets; + dai->num_dapm_widgets = ARRAY_SIZE(mt8365_memif_widgets); + dai->dapm_routes = mt8365_memif_routes; + dai->num_dapm_routes = ARRAY_SIZE(mt8365_memif_routes); + return 0; +} + +typedef int (*dai_register_cb)(struct mtk_base_afe *); +static const dai_register_cb dai_register_cbs[] = { + mt8365_dai_pcm_register, + mt8365_dai_i2s_register, + mt8365_dai_adda_register, + mt8365_dai_dmic_register, + mt8365_dai_memif_register, +}; + +static int mt8365_afe_pcm_dev_probe(struct platform_device *pdev) +{ + struct mtk_base_afe *afe; + struct mt8365_afe_private *afe_priv; + struct device *dev; + int ret, i, sel_irq; + unsigned int irq_id; + struct resource *res; + + afe = devm_kzalloc(&pdev->dev, sizeof(*afe), GFP_KERNEL); + if (!afe) + return -ENOMEM; + platform_set_drvdata(pdev, afe); + + afe->platform_priv = devm_kzalloc(&pdev->dev, sizeof(*afe_priv), + GFP_KERNEL); + if (!afe->platform_priv) + return -ENOMEM; + + afe_priv = afe->platform_priv; + afe->dev = &pdev->dev; + dev = afe->dev; + + spin_lock_init(&afe_priv->afe_ctrl_lock); + mutex_init(&afe_priv->afe_clk_mutex); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + afe->base_addr = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(afe->base_addr)) + return PTR_ERR(afe->base_addr); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (res) { + afe_priv->afe_sram_vir_addr = + devm_ioremap_resource(&pdev->dev, res); + if (!IS_ERR(afe_priv->afe_sram_vir_addr)) { + afe_priv->afe_sram_phy_addr = res->start; + afe_priv->afe_sram_size = resource_size(res); + } + } + + /* initial audio related clock */ + ret = mt8365_afe_init_audio_clk(afe); + if (ret) + return dev_err_probe(afe->dev, ret, "mt8365_afe_init_audio_clk fail\n"); + + afe->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "top_audio_sel", + afe->base_addr, + &mt8365_afe_regmap_config); + if (IS_ERR(afe->regmap)) + return PTR_ERR(afe->regmap); + + /* memif % irq initialize*/ + afe->memif_size = MT8365_AFE_MEMIF_NUM; + afe->memif = devm_kcalloc(afe->dev, afe->memif_size, + sizeof(*afe->memif), GFP_KERNEL); + if (!afe->memif) + return -ENOMEM; + + afe->irqs_size = MT8365_AFE_IRQ_NUM; + afe->irqs = devm_kcalloc(afe->dev, afe->irqs_size, + sizeof(*afe->irqs), GFP_KERNEL); + if (!afe->irqs) + return -ENOMEM; + + for (i = 0; i < afe->irqs_size; i++) + afe->irqs[i].irq_data = &irq_data[i]; + + irq_id = platform_get_irq(pdev, 0); + if (!irq_id) { + dev_err_probe(afe->dev, irq_id, "np %s no irq\n", afe->dev->of_node->name); + return -ENXIO; + } + ret = devm_request_irq(afe->dev, irq_id, mt8365_afe_irq_handler, + 0, "Afe_ISR_Handle", (void *)afe); + if (ret) + return dev_err_probe(afe->dev, ret, "could not request_irq\n"); + + /* init sub_dais */ + INIT_LIST_HEAD(&afe->sub_dais); + + for (i = 0; i < ARRAY_SIZE(dai_register_cbs); i++) { + ret = dai_register_cbs[i](afe); + if (ret) { + dev_warn(afe->dev, "dai register i %d fail, ret %d\n", + i, ret); + return ret; + } + } + + /* init dai_driver and component_driver */ + ret = mtk_afe_combine_sub_dai(afe); + if (ret) { + dev_warn(afe->dev, "mtk_afe_combine_sub_dai fail, ret %d\n", + ret); + return ret; + } + + for (i = 0; i < afe->memif_size; i++) { + afe->memif[i].data = &memif_data[i]; + sel_irq = memif_specified_irqs[i]; + if (sel_irq >= 0) { + afe->memif[i].irq_usage = sel_irq; + afe->memif[i].const_irq = 1; + afe->irqs[sel_irq].irq_occupyed = true; + } else { + afe->memif[i].irq_usage = -1; + } + } + + afe->mtk_afe_hardware = &mt8365_afe_hardware; + afe->memif_fs = mt8365_memif_fs; + afe->irq_fs = mt8365_irq_fs; + + ret = devm_pm_runtime_enable(&pdev->dev); + if (ret) + return ret; + + pm_runtime_get_sync(&pdev->dev); + afe->reg_back_up_list = mt8365_afe_backup_list; + afe->reg_back_up_list_num = ARRAY_SIZE(mt8365_afe_backup_list); + afe->runtime_resume = mt8365_afe_runtime_resume; + afe->runtime_suspend = mt8365_afe_runtime_suspend; + + /* open afe pdn for dapm read/write audio register */ + mt8365_afe_enable_top_cg(afe, MT8365_TOP_CG_AFE); + + /* Set 26m parent clk */ + mt8365_afe_set_clk_parent(afe, + afe_priv->clocks[MT8365_CLK_TOP_AUD_SEL], + afe_priv->clocks[MT8365_CLK_CLK26M]); + + ret = devm_snd_soc_register_component(&pdev->dev, + &mtk_afe_pcm_platform, + afe->dai_drivers, + afe->num_dai_drivers); + if (ret) { + dev_warn(dev, "err_platform\n"); + return ret; + } + + mt8365_afe_init_registers(afe); + + return 0; +} + +static void mt8365_afe_pcm_dev_remove(struct platform_device *pdev) +{ + struct mtk_base_afe *afe = platform_get_drvdata(pdev); + + mt8365_afe_disable_top_cg(afe, MT8365_TOP_CG_AFE); + + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + mt8365_afe_runtime_suspend(&pdev->dev); +} + +static const struct of_device_id mt8365_afe_pcm_dt_match[] = { + { .compatible = "mediatek,mt8365-afe-pcm", }, + { } +}; +MODULE_DEVICE_TABLE(of, mt8365_afe_pcm_dt_match); + +static const struct dev_pm_ops mt8365_afe_pm_ops = { + SET_RUNTIME_PM_OPS(mt8365_afe_dev_runtime_suspend, + mt8365_afe_dev_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(mt8365_afe_suspend, + mt8365_afe_resume) +}; + +static struct platform_driver mt8365_afe_pcm_driver = { + .driver = { + .name = "mt8365-afe-pcm", + .of_match_table = mt8365_afe_pcm_dt_match, + .pm = &mt8365_afe_pm_ops, + }, + .probe = mt8365_afe_pcm_dev_probe, + .remove_new = mt8365_afe_pcm_dev_remove, +}; + +module_platform_driver(mt8365_afe_pcm_driver); + +MODULE_DESCRIPTION("MediaTek ALSA SoC AFE platform driver"); +MODULE_AUTHOR("Jia Zeng "); +MODULE_AUTHOR("Alexandre Mergnat "); +MODULE_LICENSE("GPL"); From fcca6d05ef49d5650514ea1dcfd12e4ae3ff2be6 Mon Sep 17 00:00:00 2001 From: Ma Ke Date: Fri, 30 Aug 2024 22:31:54 +0800 Subject: [PATCH 319/410] ASoC: rt5682: Return devm_of_clk_add_hw_provider to transfer the error Return devm_of_clk_add_hw_provider() in order to transfer the error, if it fails due to resource allocation failure or device tree clock provider registration failure. Cc: stable@vger.kernel.org Fixes: ebbfabc16d23 ("ASoC: rt5682: Add CCF usage for providing I2S clks") Signed-off-by: Ma Ke Link: https://patch.msgid.link/20240830143154.3448004-1-make24@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e3aca9c785a0..aa163ec40862 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2903,8 +2903,10 @@ int rt5682_register_dai_clks(struct rt5682_priv *rt5682) } if (dev->of_node) { - devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + ret = devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + if (ret) + return ret; } else { ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, init.name, From 568dc2fae5d3697033f5e28cf4d6225b3db00461 Mon Sep 17 00:00:00 2001 From: Marek Vasut Date: Sat, 31 Aug 2024 01:09:43 +0200 Subject: [PATCH 320/410] ASoC: tlv320aic32x4: Add multi endpoint support Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. >From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut Link: https://patch.msgid.link/20240830231007.205707-1-marex@denx.de Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06db..54ea4bc58c27 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, From 97688a9c5b1fd2b826c682cdfa36d411a5c99828 Mon Sep 17 00:00:00 2001 From: tangbin Date: Tue, 3 Sep 2024 17:06:20 +0800 Subject: [PATCH 321/410] ASoC: loongson: fix error release In function loongson_card_parse_of(), when get device_node 'codec' failed, the function of_node_put(codec) should not be invoked, thus fix error release. Fixes: d24028606e76 ("ASoC: loongson: Add Loongson ASoC Sound Card Support") Signed-off-by: tangbin Link: https://patch.msgid.link/20240903090620.6276-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index fae5e9312bf0..2c8dbdba27c5 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -127,8 +127,8 @@ static int loongson_card_parse_of(struct loongson_card_data *data) codec = of_get_child_by_name(dev->of_node, "codec"); if (!codec) { dev_err(dev, "audio-codec property missing or invalid\n"); - ret = -EINVAL; - goto err; + of_node_put(cpu); + return -EINVAL; } for (i = 0; i < card->num_links; i++) { From 77e6a5e40aa393492cf3f1d2623d7d1dff7f33de Mon Sep 17 00:00:00 2001 From: Liu Jing Date: Tue, 3 Sep 2024 17:36:23 +0800 Subject: [PATCH 322/410] ASoC: mediatek: mt2701-cs42448: Optimize redundant code in mt2701_cs42448_machine_probe Utilize the defined parameter 'dev' to make the code cleaner. Signed-off-by: Liu Jing Reviewed-by: Matthias Brugger Link: https://patch.msgid.link/20240903093623.7120-1-liujing@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 1262e8a1bc9a..4974b0536b7b 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -329,10 +329,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) int ret; int i; struct device_node *platform_node, *codec_node, *codec_node_bt_mrg; - struct mt2701_cs42448_private *priv = - devm_kzalloc(&pdev->dev, sizeof(struct mt2701_cs42448_private), - GFP_KERNEL); struct device *dev = &pdev->dev; + struct mt2701_cs42448_private *priv = + devm_kzalloc(dev, sizeof(struct mt2701_cs42448_private), + GFP_KERNEL); struct snd_soc_dai_link *dai_link; if (!priv) @@ -341,7 +341,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) platform_node = of_parse_phandle(pdev->dev.of_node, "mediatek,platform", 0); if (!platform_node) { - dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); + dev_err(dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { @@ -355,7 +355,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) codec_node = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0); if (!codec_node) { - dev_err(&pdev->dev, + dev_err(dev, "Property 'audio-codec' missing or invalid\n"); return -EINVAL; } @@ -368,7 +368,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) codec_node_bt_mrg = of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec-bt-mrg", 0); if (!codec_node_bt_mrg) { - dev_err(&pdev->dev, + dev_err(dev, "Property 'audio-codec-bt-mrg' missing or invalid\n"); return -EINVAL; } @@ -377,7 +377,7 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); if (ret) { - dev_err(&pdev->dev, "failed to parse audio-routing: %d\n", ret); + dev_err(dev, "failed to parse audio-routing: %d\n", ret); return ret; } @@ -395,10 +395,10 @@ static int mt2701_cs42448_machine_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); - ret = devm_snd_soc_register_card(&pdev->dev, card); + ret = devm_snd_soc_register_card(dev, card); if (ret) - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + dev_err(dev, "%s snd_soc_register_card fail %d\n", __func__, ret); return ret; } From de849243404e8ce02320a7178d4448e4d0191377 Mon Sep 17 00:00:00 2001 From: Zhang Zekun Date: Tue, 27 Aug 2024 15:06:49 +0800 Subject: [PATCH 323/410] ASoC: audio-graph-card: Use for_each_child_of_node_scoped() to simplify code for_each_child_of_node_scoped() can put the device_node automatically. So, using it to make the code logic more simple and remove the device_node clean up code. Signed-off-by: Zhang Zekun Reviewed-by: Kuninori Morimoto Link: https://patch.msgid.link/20240827070650.11424-2-zhangzekun11@huawei.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card.c | 12 ++---------- 1 file changed, 2 insertions(+), 10 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 3425fbbcbd7e..30152a681aa8 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -363,7 +363,6 @@ static int __graph_for_each_link(struct simple_util_priv *priv, struct device *dev = simple_priv_to_dev(priv); struct device_node *node = dev->of_node; struct device_node *cpu_port; - struct device_node *cpu_ep; struct device_node *codec_ep; struct device_node *codec_port; struct device_node *codec_port_old = NULL; @@ -373,14 +372,9 @@ static int __graph_for_each_link(struct simple_util_priv *priv, /* loop for all listed CPU port */ of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { cpu_port = it.node; - cpu_ep = NULL; /* loop for all CPU endpoint */ - while (1) { - cpu_ep = of_get_next_child(cpu_port, cpu_ep); - if (!cpu_ep) - break; - + for_each_child_of_node_scoped(cpu_port, cpu_ep) { /* get codec */ codec_ep = of_graph_get_remote_endpoint(cpu_ep); codec_port = ep_to_port(codec_ep); @@ -410,10 +404,8 @@ static int __graph_for_each_link(struct simple_util_priv *priv, of_node_put(codec_ep); of_node_put(codec_port); - if (ret < 0) { - of_node_put(cpu_ep); + if (ret < 0) return ret; - } codec_port_old = codec_port; } From 815f1fcf2403454904cbbc5cf370df6bc300f392 Mon Sep 17 00:00:00 2001 From: Zhang Zekun Date: Tue, 27 Aug 2024 15:06:50 +0800 Subject: [PATCH 324/410] ASoC: audio-graph-card2: Use helper function of_get_child_count() of_get_child_count() can help to get the num of child directly and we don't need to manually count the child num. No functional change with this conversion. Signed-off-by: Zhang Zekun Reviewed-by: Kuninori Morimoto Link: https://patch.msgid.link/20240827070650.11424-3-zhangzekun11@huawei.com Signed-off-by: Mark Brown --- sound/soc/generic/audio-graph-card2.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 56f7f946882e..ebbe12692f57 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1141,21 +1141,12 @@ static int graph_counter(struct device_node *lnk) */ if (graph_lnk_is_multi(lnk)) { struct device_node *ports = port_to_ports(lnk); - struct device_node *port = NULL; - int cnt = 0; /* * CPU/Codec = N:M case has many endpoints. * We can't use of_graph_get_endpoint_count() here */ - while(1) { - port = of_get_next_child(ports, port); - if (!port) - break; - cnt++; - } - - return cnt - 1; + return of_get_child_count(ports) - 1; } /* * Single CPU / Codec From 5e2404493f9f6028bf143972b18c88156c9893e6 Mon Sep 17 00:00:00 2001 From: Nicolas Belin Date: Thu, 5 Sep 2024 11:06:58 +0200 Subject: [PATCH 325/410] ASoC: codecs: add MT6357 support Add the support of MT6357 PMIC audio codec. Signed-off-by: Nicolas Belin Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v8-1-e80a57d026ce@baylibre.com Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6357.c | 1855 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6357.h | 660 +++++++++++++ 4 files changed, 2524 insertions(+) create mode 100644 sound/soc/codecs/mt6357.c create mode 100644 sound/soc/codecs/mt6357.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b5e6d0a986c8..7092842480ef 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -157,6 +157,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_MC13783 imply SND_SOC_ML26124 imply SND_SOC_MT6351 + imply SND_SOC_MT6357 imply SND_SOC_MT6358 imply SND_SOC_MT6359 imply SND_SOC_MT6660 @@ -2501,6 +2502,12 @@ config SND_SOC_ML26124 config SND_SOC_MT6351 tristate "MediaTek MT6351 Codec" +config SND_SOC_MT6357 + tristate "MediaTek MT6357 Codec" + help + Enable support for the platform which uses MT6357 as + external codec device. + config SND_SOC_MT6358 tristate "MediaTek MT6358 Codec" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 622e360f0086..54cbc3feae32 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -177,6 +177,7 @@ snd-soc-ml26124-y := ml26124.o snd-soc-msm8916-analog-y := msm8916-wcd-analog.o snd-soc-msm8916-digital-y := msm8916-wcd-digital.o snd-soc-mt6351-y := mt6351.o +snd-soc-mt6357-y := mt6357.o snd-soc-mt6358-y := mt6358.o snd-soc-mt6359-y := mt6359.o snd-soc-mt6359-accdet-y := mt6359-accdet.o @@ -578,6 +579,7 @@ obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o +obj-$(CONFIG_SND_SOC_MT6357) += snd-soc-mt6357.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o obj-$(CONFIG_SND_SOC_MT6359) += snd-soc-mt6359.o obj-$(CONFIG_SND_SOC_MT6359_ACCDET) += mt6359-accdet.o diff --git a/sound/soc/codecs/mt6357.c b/sound/soc/codecs/mt6357.c new file mode 100644 index 000000000000..988728df15e4 --- /dev/null +++ b/sound/soc/codecs/mt6357.c @@ -0,0 +1,1855 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * MT6357 ALSA SoC audio codec driver + * + * Copyright (c) 2024 Baylibre + * Author: Nicolas Belin + */ + +#include +#include +#include +#include +#include + +#include "mt6357.h" + +static void set_playback_gpio(struct mt6357_priv *priv, bool enable) +{ + regmap_write(priv->regmap, MT6357_GPIO_MODE2_CLR, MT6357_GPIO_MODE2_CLEAR_ALL); + if (enable) { + /* set gpio mosi mode */ + regmap_write(priv->regmap, MT6357_GPIO_MODE2_SET, + MT6357_GPIO8_MODE_SET_AUD_CLK_MOSI | + MT6357_GPIO9_MODE_SET_AUD_DAT_MOSI0 | + MT6357_GPIO10_MODE_SET_AUD_DAT_MOSI1 | + MT6357_GPIO11_MODE_SET_AUD_SYNC_MOSI); + } else { + /* pad_aud_*_mosi are GPIO mode after clear and set them to dir input + * reason: + * pad_aud_dat_mosi*, because the pin is used as boot strap + */ + regmap_update_bits(priv->regmap, MT6357_GPIO_DIR0, + MT6357_GPIO8_DIR_MASK | + MT6357_GPIO9_DIR_MASK | + MT6357_GPIO10_DIR_MASK | + MT6357_GPIO11_DIR_MASK, + MT6357_GPIO8_DIR_INPUT | + MT6357_GPIO9_DIR_INPUT | + MT6357_GPIO10_DIR_INPUT | + MT6357_GPIO11_DIR_INPUT); + } +} + +static void set_capture_gpio(struct mt6357_priv *priv, bool enable) +{ + regmap_write(priv->regmap, MT6357_GPIO_MODE3_CLR, MT6357_GPIO_MODE3_CLEAR_ALL); + if (enable) { + /* set gpio miso mode */ + regmap_write(priv->regmap, MT6357_GPIO_MODE3_SET, + MT6357_GPIO12_MODE_SET_AUD_CLK_MISO | + MT6357_GPIO13_MODE_SET_AUD_DAT_MISO0 | + MT6357_GPIO14_MODE_SET_AUD_DAT_MISO1 | + MT6357_GPIO15_MODE_SET_AUD_SYNC_MISO); + } else { + /* pad_aud_*_mosi are GPIO mode after clear and set them to dir input + * reason: + * pad_aud_clk_miso, because when playback only the miso_clk + * will also have 26m, so will have power leak + * pad_aud_dat_miso*, because the pin is used as boot strap + */ + regmap_update_bits(priv->regmap, MT6357_GPIO_DIR0, + MT6357_GPIO12_DIR_MASK | + MT6357_GPIO13_DIR_MASK | + MT6357_GPIO14_DIR_MASK | + MT6357_GPIO15_DIR_MASK, + MT6357_GPIO12_DIR_INPUT | + MT6357_GPIO13_DIR_INPUT | + MT6357_GPIO14_DIR_INPUT | + MT6357_GPIO15_DIR_INPUT); + } +} + +static void hp_main_output_ramp(struct mt6357_priv *priv, bool up) +{ + int i, stage; + + /* Enable/Reduce HPL/R main output stage step by step */ + for (i = 0; i <= MT6357_HPLOUT_STG_CTRL_VAUDP15_MAX; i++) { + stage = up ? i : MT6357_HPLOUT_STG_CTRL_VAUDP15_MAX - i; + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPLOUT_STG_CTRL_VAUDP15_MASK, + stage << MT6357_HPLOUT_STG_CTRL_VAUDP15_SFT); + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_STG_CTRL_VAUDP15_MASK, + stage << MT6357_HPROUT_STG_CTRL_VAUDP15_SFT); + usleep_range(600, 700); + } +} + +static void hp_aux_feedback_loop_gain_ramp(struct mt6357_priv *priv, bool up) +{ + int i, stage; + + /* Reduce HP aux feedback loop gain step by step */ + for (i = 0; i <= MT6357_HP_AUX_LOOP_GAIN_MAX; i++) { + stage = up ? i : MT6357_HP_AUX_LOOP_GAIN_MAX - i; + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HP_AUX_LOOP_GAIN_MASK, + stage << MT6357_HP_AUX_LOOP_GAIN_SFT); + usleep_range(600, 700); + } +} + +static void hp_pull_down(struct mt6357_priv *priv, bool enable) +{ + if (enable) + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON2, + MT6357_HPP_SHORT_2VCM_VAUDP15_MASK, + MT6357_HPP_SHORT_2VCM_VAUDP15_ENABLE); + else + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON2, + MT6357_HPP_SHORT_2VCM_VAUDP15_MASK, + MT6357_HPP_SHORT_2VCM_VAUDP15_DISABLE); +} + +static bool is_valid_hp_pga_idx(int reg_idx) +{ + return (reg_idx >= DL_GAIN_8DB && reg_idx <= DL_GAIN_N_12DB) || reg_idx == DL_GAIN_N_40DB; +} + +static void volume_ramp(struct mt6357_priv *priv, int lfrom, int lto, + int rfrom, int rto, unsigned int reg_addr) +{ + int lcount, rcount, sleep = 0; + + if (!is_valid_hp_pga_idx(lfrom) || !is_valid_hp_pga_idx(lto)) + pr_debug("%s(), invalid left volume index, from %d, to %d\n", + __func__, lfrom, lto); + + if (!is_valid_hp_pga_idx(rfrom) || !is_valid_hp_pga_idx(rto)) + pr_debug("%s(), invalid right volume index, from %d, to %d\n", + __func__, rfrom, rto); + + if (lto > lfrom) + lcount = 1; + else + lcount = -1; + + if (rto > rfrom) + rcount = 1; + else + rcount = -1; + + while ((lto != lfrom) || (rto != rfrom)) { + if (lto != lfrom) { + lfrom += lcount; + if (is_valid_hp_pga_idx(lfrom)) { + regmap_update_bits(priv->regmap, reg_addr, + MT6357_DL_GAIN_REG_LEFT_MASK, + lfrom << MT6357_DL_GAIN_REG_LEFT_SHIFT); + sleep = 1; + } + } + if (rto != rfrom) { + rfrom += rcount; + if (is_valid_hp_pga_idx(rfrom)) { + regmap_update_bits(priv->regmap, reg_addr, + MT6357_DL_GAIN_REG_RIGHT_MASK, + rfrom << MT6357_DL_GAIN_REG_RIGHT_SHIFT); + sleep = 1; + } + } + if (sleep) + usleep_range(200, 300); + } +} + +static void lo_volume_ramp(struct mt6357_priv *priv, int lfrom, int lto, int rfrom, int rto) +{ + volume_ramp(priv, lfrom, lto, rfrom, rto, MT6357_ZCD_CON1); +} + +static void hp_volume_ramp(struct mt6357_priv *priv, int lfrom, int lto, int rfrom, int rto) +{ + volume_ramp(priv, lfrom, lto, rfrom, rto, MT6357_ZCD_CON2); +} + +static void hs_volume_ramp(struct mt6357_priv *priv, int from, int to) +{ + volume_ramp(priv, from, to, 0, 0, MT6357_ZCD_CON3); +} + +/* Volume and channel swap controls */ +static const DECLARE_TLV_DB_SCALE(playback_tlv, -1000, 100, 0); +static const DECLARE_TLV_DB_SCALE(capture_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_degain_tlv, -1200, 1200, 0); + +static const struct snd_kcontrol_new mt6357_controls[] = { + /* dl pga gain */ + SOC_DOUBLE_TLV("Headphone Volume", + MT6357_ZCD_CON2, MT6357_AUD_HPL_GAIN_SFT, + MT6357_AUD_HPR_GAIN_SFT, MT6357_AUD_HP_GAIN_MAX, + 1, playback_tlv), + SOC_SINGLE_TLV("Headphone Vin Volume", + MT6357_AUDDEC_ANA_CON7, MT6357_HP_IVBUF_DEGAIN_SFT, + MT6357_HP_IVBUF_DEGAIN_MAX, 1, hp_degain_tlv), + SOC_DOUBLE_TLV("Lineout Volume", + MT6357_ZCD_CON1, MT6357_AUD_LOL_GAIN_SFT, + MT6357_AUD_LOR_GAIN_SFT, MT6357_AUD_LO_GAIN_MAX, + 1, playback_tlv), + SOC_SINGLE_TLV("Handset Volume", + MT6357_ZCD_CON3, MT6357_AUD_HS_GAIN_SFT, + MT6357_AUD_HS_GAIN_MAX, 1, playback_tlv), + /* ul pga gain */ + SOC_DOUBLE_R_TLV("Mic Volume", + MT6357_AUDENC_ANA_CON0, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPLGAIN_SFT, MT6357_AUDPREAMPLGAIN_MAX, + 0, capture_tlv), +}; + +/* Uplink controls */ + +enum { + MIC_TYPE_MUX_IDLE, + MIC_TYPE_MUX_ACC, + MIC_TYPE_MUX_DMIC, + MIC_TYPE_MUX_DCC, + MIC_TYPE_MUX_DCC_ECM_DIFF, + MIC_TYPE_MUX_DCC_ECM_SINGLE, + MIC_TYPE_MUX_LPBK, + MIC_TYPE_MUX_SGEN, +}; + +#define IS_DCC_BASE(type) ((type) == MIC_TYPE_MUX_DCC || \ + (type) == MIC_TYPE_MUX_DCC_ECM_DIFF || \ + (type) == MIC_TYPE_MUX_DCC_ECM_SINGLE) + +static const char * const mic_type_mux_map[] = { + "Idle", + "ACC", + "DMIC", + "DCC", + "DCC_ECM_DIFF", + "DCC_ECM_SINGLE", + "Loopback", + "Sine Generator", +}; + +static SOC_ENUM_SINGLE_DECL(mic_type_mux_map_enum, SND_SOC_NOPM, + 0, mic_type_mux_map); + +static const struct snd_kcontrol_new mic_type_mux_control = + SOC_DAPM_ENUM("Mic Type Select", mic_type_mux_map_enum); + +static const char * const pga_mux_map[] = { + "None", "AIN0", "AIN1", "AIN2" +}; + +static SOC_ENUM_SINGLE_DECL(pga_left_mux_map_enum, + MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLINPUTSEL_SFT, + pga_mux_map); + +static const struct snd_kcontrol_new pga_left_mux_control = + SOC_DAPM_ENUM("PGA L Select", pga_left_mux_map_enum); + +static SOC_ENUM_SINGLE_DECL(pga_right_mux_map_enum, + MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRINPUTSEL_SFT, + pga_mux_map); + +static const struct snd_kcontrol_new pga_right_mux_control = + SOC_DAPM_ENUM("PGA R Select", pga_right_mux_map_enum); + +/* Downlink controls */ +static const char * const hslo_mux_map[] = { + "Open", "DACR", "Playback", "Test mode" +}; + +static SOC_ENUM_SINGLE_DECL(lo_mux_map_enum, + MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_MUX_INPUT_VAUDP15_SFT, + hslo_mux_map); + +static const struct snd_kcontrol_new lo_mux_control = + SOC_DAPM_ENUM("Line out source", lo_mux_map_enum); + +static SOC_ENUM_SINGLE_DECL(hs_mux_map_enum, + MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_MUX_INPUT_VAUDP15_SFT, + hslo_mux_map); + +static const struct snd_kcontrol_new hs_mux_control = + SOC_DAPM_ENUM("Handset source", hs_mux_map_enum); + +static const char * const hplr_mux_map[] = { + "Open", "Line Out", "DAC", "Handset" +}; + +static SOC_ENUM_SINGLE_DECL(hpr_mux_map_enum, + MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_MUX_INPUT_VAUDP15_SFT, + hplr_mux_map); + +static const struct snd_kcontrol_new hpr_mux_control = + SOC_DAPM_ENUM("Headphone Right source", hpr_mux_map_enum); + +static SOC_ENUM_SINGLE_DECL(hpl_mux_map_enum, + MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPL_MUX_INPUT_VAUDP15_SFT, + hplr_mux_map); + +static const struct snd_kcontrol_new hpl_mux_control = + SOC_DAPM_ENUM("Headphone Left source", hpl_mux_map_enum); + +static const char * const dac_mux_map[] = { + "Normal Path", "Sine Generator" +}; + +static SOC_ENUM_SINGLE_DECL(dac_mux_map_enum, + MT6357_AFE_TOP_CON0, + MT6357_DL_SINE_ON_SFT, + dac_mux_map); + +static const struct snd_kcontrol_new dac_mux_control = + SOC_DAPM_ENUM("DAC Select", dac_mux_map_enum); + +static int mt6357_set_dmic(struct mt6357_priv *priv, bool enable) +{ + if (enable) { + /* DMIC enable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON7, + MT6357_AUDDIGMICBIAS_MASK | MT6357_AUDDIGMICEN_MASK, + MT6357_AUDDIGMICBIAS_DEFAULT_VALUE | MT6357_AUDDIGMICEN_ENABLE); + /* enable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_ENABLE); + /* UL dmic setting: dual mode */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_H, + MT6357_C_TWO_DIGITAL_MIC_CTL_MASK, + MT6357_C_TWO_DIGITAL_MIC_ENABLE); + /* UL turn on SDM 3 level mode */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SDM_3_LEVEL_CTL_MASK, + MT6357_UL_SDM_3_LEVEL_SELECT); + /* UL turn on */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_ENABLE); + /* Wait to avoid any pop noises */ + msleep(100); + } else { + /* UL turn off */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_DISABLE); + /* UL turn on SDM 3 level mode */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SDM_3_LEVEL_CTL_MASK, + MT6357_UL_SDM_3_LEVEL_DESELECT); + /* disable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_DISABLE); + /* UL dmic setting: dual mode */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_H, + MT6357_C_TWO_DIGITAL_MIC_CTL_MASK, + MT6357_C_TWO_DIGITAL_MIC_DISABLE); + /* DMIC disable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON7, + MT6357_AUDDIGMICBIAS_MASK | MT6357_AUDDIGMICEN_MASK, + MT6357_AUDDIGMICBIAS_OFF | MT6357_AUDDIGMICEN_DISABLE); + } + return 0; +} + +static int mt6357_set_amic(struct mt6357_priv *priv, bool enable, unsigned int mic_type) +{ + if (enable) { + if (IS_DCC_BASE(mic_type)) { + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_DIV_MASK, MT6357_DCCLK_DIV_RUN_VALUE); + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_PDN_MASK, MT6357_DCCLK_OUTPUT); + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_GEN_ON_MASK, MT6357_DCCLK_GEN_ON); + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG1, + MT6357_DCCLK_RESYNC_BYPASS_MASK, + MT6357_DCCLK_RESYNC_BYPASS); + + /* mic bias 0: set the correct DC couple*/ + switch (mic_type) { + case MIC_TYPE_MUX_DCC_ECM_DIFF: + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_DC_MASK, + MT6357_AUD_MICBIAS0_DC_ENABLE_ALL); + break; + case MIC_TYPE_MUX_DCC_ECM_SINGLE: + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_DC_MASK, + MT6357_AUD_MICBIAS0_DC_ENABLE_P1); + break; + default: + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_DC_MASK, + MT6357_AUD_MICBIAS0_DC_DISABLE_ALL); + break; + } + + /* mic bias 1: set the correct DC couple */ + if (mic_type == MIC_TYPE_MUX_DCC_ECM_SINGLE) + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON9, + MT6357_AUD_MICBIAS1_DCSW1P_EN_MASK, + MT6357_AUD_MICBIAS1_DCSW1P_ENABLE); + + /* Audio L/R preamplifier DCC precharge */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCPRECHARGE_MASK, + MT6357_AUDPREAMPLDCPRECHARGE_ENABLE); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCPRECHARGE_MASK, + MT6357_AUDPREAMPRDCPRECHARGE_ENABLE); + /* L preamplifier DCCEN */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCCEN_MASK, + MT6357_AUDPREAMPLDCCEN_DC); + /* R preamplifier DCCEN */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCCEN_MASK, + MT6357_AUDPREAMPRDCCEN_DC); + } else { + /* Audio L preamplifier DCC precharge disable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCPRECHARGE_MASK, + MT6357_AUDPREAMPLDCPRECHARGE_DISABLE); + /* L preamplifier ACC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCCEN_MASK, + MT6357_AUDPREAMPLDCCEN_AC); + /* Audio R preamplifier DCC precharge disable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCPRECHARGE_MASK, + MT6357_AUDPREAMPRDCPRECHARGE_DISABLE); + /* R preamplifier ACC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCCEN_MASK, + MT6357_AUDPREAMPRDCCEN_AC); + } + } else { + /* disable any Mic Bias 0 DC couple */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_DC_MASK, + MT6357_AUD_MICBIAS0_DC_DISABLE_ALL); + /* disable any Mic Bias 1 DC couple */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON9, + MT6357_AUD_MICBIAS1_DCSW1P_EN_MASK, + MT6357_AUD_MICBIAS1_DCSW1P_DISABLE); + if (IS_DCC_BASE(mic_type)) { + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_GEN_ON_MASK, MT6357_DCCLK_GEN_OFF); + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_PDN_MASK, MT6357_DCCLK_PDN); + regmap_update_bits(priv->regmap, MT6357_AFE_DCCLK_CFG0, + MT6357_DCCLK_DIV_MASK, MT6357_DCCLK_DIV_STOP_VALUE); + } + } + + return 0; +} + +static int mt6357_set_loopback(struct mt6357_priv *priv, bool enable) +{ + if (enable) { + /* enable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_ENABLE); + /* enable aud_pad lpk TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_LPBK_MASK, + MT6357_AUD_PAD_TX_FIFO_LPBK_ENABLE); + /* Set UL Part: enable new lpbk 2 */ + regmap_update_bits(priv->regmap, MT6357_AFE_ADDA_MTKAIF_CFG0, + MT6357_ADDA_MTKAIF_LPBK_CTL_MASK, + MT6357_ADDA_MTKAIF_LPBK_ENABLE); + /* UL turn on */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_ENABLE); + } else { + /* UL turn off */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_DISABLE); + /* disable new lpbk 2 */ + regmap_update_bits(priv->regmap, MT6357_AFE_ADDA_MTKAIF_CFG0, + MT6357_ADDA_MTKAIF_LPBK_CTL_MASK, + MT6357_ADDA_MTKAIF_LPBK_DISABLE); + /* disable aud_pad lpbk TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_LPBK_MASK, + MT6357_AUD_PAD_TX_FIFO_LPBK_DISABLE); + /* disable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_DISABLE); + } + + return 0; +} + +static int mt6357_set_ul_sine_gen(struct mt6357_priv *priv, bool enable) +{ + if (enable) { + /* enable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_ENABLE); + /* UL turn on */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_ENABLE); + } else { + /* UL turn off */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, + MT6357_UL_SRC_DISABLE); + /* disable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_DISABLE); + } + + return 0; +} + +static int mt_aif_out_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + set_capture_gpio(priv, true); + break; + case SND_SOC_DAPM_POST_PMD: + set_capture_gpio(priv, false); + break; + default: + break; + } + + return 0; +} + +static int mt_adc_supply_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable audio ADC CLKGEN */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON11, + MT6357_RSTB_ENCODER_VA28_MASK, MT6357_RSTB_ENCODER_VA28_ENABLE); + /* Enable LCLDO_ENC 2P8V */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_LCLDO_ENC_EN_VA28_MASK, MT6357_LCLDO_ENC_EN_VA28_ENABLE); + /* LCLDO_ENC remote sense */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_VA28REFGEN_EN_VA28_MASK | + MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_MASK, + MT6357_VA28REFGEN_EN_VA28_ENABLE | + MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_ENABLE); + break; + case SND_SOC_DAPM_POST_PMD: + /* LCLDO_ENC remote sense off */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_VA28REFGEN_EN_VA28_MASK | + MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_MASK, + MT6357_VA28REFGEN_EN_VA28_DISABLE | + MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_DISABLE); + /* disable LCLDO_ENC 2P8V */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_LCLDO_ENC_EN_VA28_MASK, + MT6357_LCLDO_ENC_EN_VA28_DISABLE); + /* disable audio ADC CLKGEN */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON11, + MT6357_RSTB_ENCODER_VA28_MASK, + MT6357_RSTB_ENCODER_VA28_DISABLE); + break; + default: + break; + } + + return 0; +} + +static int mt_mic_type_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + unsigned int mic_type = dapm_kcontrol_get_value(w->kcontrols[0]); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (mic_type) { + case MIC_TYPE_MUX_DMIC: + mt6357_set_dmic(priv, true); + break; + case MIC_TYPE_MUX_LPBK: + mt6357_set_loopback(priv, true); + break; + case MIC_TYPE_MUX_SGEN: + mt6357_set_ul_sine_gen(priv, true); + break; + default: + mt6357_set_amic(priv, true, mic_type); + break; + } + break; + case SND_SOC_DAPM_POST_PMD: + switch (mic_type) { + case MIC_TYPE_MUX_DMIC: + mt6357_set_dmic(priv, false); + break; + case MIC_TYPE_MUX_LPBK: + mt6357_set_loopback(priv, false); + break; + case MIC_TYPE_MUX_SGEN: + mt6357_set_ul_sine_gen(priv, false); + break; + default: + mt6357_set_amic(priv, false, mic_type); + break; + } + break; + default: + break; + } + + return 0; +} + +static int mt_pga_left_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* L preamplifier enable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLON_MASK, + MT6357_AUDPREAMPLON_ENABLE); + /* L ADC input sel : L PGA. Enable audio L ADC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDADCLINPUTSEL_MASK, + MT6357_AUDADCLINPUTSEL_PREAMPLIFIER); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDADCLPWRUP_MASK, + MT6357_AUDADCLPWRUP); + /* Audio L preamplifier DCC precharge off */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCPRECHARGE_MASK, + MT6357_AUDPREAMPLDCPRECHARGE_DISABLE); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Audio L ADC input sel : off, disable audio L ADC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDADCLPWRUP_MASK, + MT6357_AUDADCLPWRDOWN); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDADCLINPUTSEL_MASK, + MT6357_AUDADCLINPUTSEL_IDLE); + /* L preamplifier ACC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCCEN_MASK, + MT6357_AUDPREAMPLDCCEN_AC); + /* L preamplifier disable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLON_MASK, + MT6357_AUDPREAMPLON_DISABLE); + /* disable Audio L preamplifier DCC precharge */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLDCPRECHARGE_MASK, + MT6357_AUDPREAMPLDCPRECHARGE_DISABLE); + break; + default: + break; + } + + return 0; +} + +static int mt_pga_right_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* R preamplifier enable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRON_MASK, MT6357_AUDPREAMPRON_ENABLE); + /* R ADC input sel : R PGA. Enable audio R ADC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDADCRINPUTSEL_MASK, + MT6357_AUDADCRINPUTSEL_PREAMPLIFIER); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDADCRPWRUP_MASK, MT6357_AUDADCRPWRUP); + /* Audio R preamplifier DCC precharge off */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCPRECHARGE_MASK, + MT6357_AUDPREAMPRDCPRECHARGE_DISABLE); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Audio R ADC input sel : off, disable audio R ADC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDADCRPWRUP_MASK, MT6357_AUDADCRPWRDOWN); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDADCRINPUTSEL_MASK, MT6357_AUDADCRINPUTSEL_IDLE); + /* R preamplifier ACC */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCCEN_MASK, MT6357_AUDPREAMPRDCCEN_AC); + /* R preamplifier disable */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRON_MASK, MT6357_AUDPREAMPRON_DISABLE); + /* disable Audio R preamplifier DCC precharge */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRDCPRECHARGE_MASK, + MT6357_AUDPREAMPRDCPRECHARGE_DISABLE); + break; + default: + break; + } + + return 0; +} + +static int adc_enable_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + int lgain, rgain; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_read(priv->regmap, MT6357_AUDENC_ANA_CON0, &lgain); + regmap_read(priv->regmap, MT6357_AUDENC_ANA_CON1, &rgain); + /* L PGA 0 dB gain */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLGAIN_MASK, + UL_GAIN_0DB << MT6357_AUDPREAMPLGAIN_SFT); + /* R PGA 0 dB gain */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRGAIN_MASK, + UL_GAIN_0DB << MT6357_AUDPREAMPRGAIN_SFT); + /* enable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_ENABLE); + /* UL turn on */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, MT6357_UL_SRC_ENABLE); + /* Wait to avoid any pop noises */ + msleep(100); + /* set the mic gains to the stored values */ + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON0, + MT6357_AUDPREAMPLGAIN_MASK, lgain); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON1, + MT6357_AUDPREAMPRGAIN_MASK, rgain); + break; + case SND_SOC_DAPM_POST_PMD: + /* UL turn off */ + regmap_update_bits(priv->regmap, MT6357_AFE_UL_SRC_CON0_L, + MT6357_UL_SRC_ON_TMP_CTL_MASK, MT6357_UL_SRC_DISABLE); + /* disable aud_pad TX fifos */ + regmap_update_bits(priv->regmap, MT6357_AFE_AUD_PAD_TOP, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK, + MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_DISABLE); + break; + default: + break; + } + + return 0; +} + +static void configure_downlinks(struct mt6357_priv *priv, bool enable) +{ + if (enable) { + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ELR_0, + MT6357_AUD_HP_TRIM_EN_VAUDP15_MASK, + MT6357_AUD_HP_TRIM_EN_VAUDP15_ENABLE); + /* Disable headphone short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_SC_VAUDP15_MASK | MT6357_AUD_HPL_SC_VAUDP15_MASK, + MT6357_AUD_HPR_SC_VAUDP15_DISABLE | + MT6357_AUD_HPL_SC_VAUDP15_DISABLE); + /* Disable handset short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_SC_VAUDP15_MASK, + MT6357_AUD_HS_SC_VAUDP15_DISABLE); + /* Disable lineout short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_SC_VAUDP15_MASK, + MT6357_AUD_LOL_SC_VAUDP15_DISABLE); + /* Reduce ESD resistance of AU_REFN */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON2, + MT6357_AUD_REFN_DERES_VAUDP15_MASK, + MT6357_AUD_REFN_DERES_VAUDP15_ENABLE); + /* Turn on DA_600K_NCP_VA18 */ + regmap_write(priv->regmap, MT6357_AUDNCP_CLKDIV_CON1, MT6357_DIVCKS_ON); + /* Set NCP clock as 604kHz // 26MHz/43 = 604KHz */ + regmap_write(priv->regmap, MT6357_AUDNCP_CLKDIV_CON2, 0x002c); + /* Toggle DIVCKS_CHG */ + regmap_write(priv->regmap, MT6357_AUDNCP_CLKDIV_CON0, MT6357_DIVCKS_CHG); + /* Set NCP soft start mode as default mode: 150us */ + regmap_write(priv->regmap, MT6357_AUDNCP_CLKDIV_CON4, + MT6357_DIVCKS_PWD_NCP_ST_150US); + /* Enable NCP */ + regmap_write(priv->regmap, MT6357_AUDNCP_CLKDIV_CON3, + MT6357_DIVCKS_PWD_NCP_ENABLE); + usleep_range(250, 270); + /* Enable cap-less LDOs (1.5V) */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_VA33REFGEN_EN_VA18_MASK | + MT6357_LCLDO_REMOTE_SENSE_VA18_MASK | + MT6357_LCLDO_EN_VA18_MASK | + MT6357_HCLDO_REMOTE_SENSE_VA18_MASK | + MT6357_HCLDO_EN_VA18_MASK, + MT6357_VA33REFGEN_EN_VA18_ENABLE | + MT6357_LCLDO_REMOTE_SENSE_VA18_ENABLE | + MT6357_LCLDO_EN_VA18_ENABLE | + MT6357_HCLDO_REMOTE_SENSE_VA18_ENABLE | + MT6357_HCLDO_EN_VA18_ENABLE); + /* Enable NV regulator (-1.2V) */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON13, + MT6357_NVREG_EN_VAUDP15_MASK, MT6357_NVREG_EN_VAUDP15_ENABLE); + usleep_range(100, 120); + /* Enable IBIST */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON10, + MT6357_AUD_IBIAS_PWRDN_VAUDP15_MASK, + MT6357_AUD_IBIAS_PWRDN_VAUDP15_ENABLE); + /* Enable AUD_CLK */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON11, + MT6357_RSTB_DECODER_VA28_MASK, + MT6357_RSTB_DECODER_VA28_ENABLE); + /* Enable low-noise mode of DAC */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_DAC_LOW_NOISE_MODE_MASK, + MT6357_DAC_LOW_NOISE_MODE_ENABLE); + usleep_range(100, 120); + } else { + /* Disable low-noise mode of DAC */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_DAC_LOW_NOISE_MODE_MASK, + MT6357_DAC_LOW_NOISE_MODE_DISABLE); + /* Disable AUD_CLK */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON11, + MT6357_RSTB_DECODER_VA28_MASK, + MT6357_RSTB_DECODER_VA28_DISABLE); + /* Enable linout short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_SC_VAUDP15_MASK, + MT6357_AUD_LOL_SC_VAUDP15_ENABLE); + /* Enable handset short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_SC_VAUDP15_MASK, + MT6357_AUD_HS_SC_VAUDP15_ENABLE); + /* Enable headphone short-circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_SC_VAUDP15_MASK | + MT6357_AUD_HPL_SC_VAUDP15_MASK, + MT6357_AUD_HPR_SC_VAUDP15_ENABLE | + MT6357_AUD_HPL_SC_VAUDP15_ENABLE); + /* Disable IBIST */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON10, + MT6357_AUD_IBIAS_PWRDN_VAUDP15_MASK, + MT6357_AUD_IBIAS_PWRDN_VAUDP15_DISABLE); + /* Disable NV regulator (-1.2V) */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON13, + MT6357_NVREG_EN_VAUDP15_MASK, + MT6357_NVREG_EN_VAUDP15_DISABLE); + /* Disable cap-less LDOs (1.5V) */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON12, + MT6357_VA33REFGEN_EN_VA18_MASK | + MT6357_LCLDO_REMOTE_SENSE_VA18_MASK | + MT6357_LCLDO_EN_VA18_MASK | + MT6357_HCLDO_REMOTE_SENSE_VA18_MASK | + MT6357_HCLDO_EN_VA18_MASK, + MT6357_VA33REFGEN_EN_VA18_DISABLE | + MT6357_LCLDO_REMOTE_SENSE_VA18_DISABLE | + MT6357_LCLDO_EN_VA18_DISABLE | + MT6357_HCLDO_REMOTE_SENSE_VA18_DISABLE | + MT6357_HCLDO_EN_VA18_DISABLE); + /* Disable NCP */ + regmap_update_bits(priv->regmap, MT6357_AUDNCP_CLKDIV_CON3, + MT6357_DIVCKS_PWD_NCP_MASK, MT6357_DIVCKS_PWD_NCP_DISABLE); + } +} + +static int mt_audio_in_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + set_playback_gpio(priv, true); + + /* Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, true); + + /* Disable HP main CMFB Switch */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HPRL_MAIN_CMFB_LOOP_MASK, + MT6357_HPRL_MAIN_CMFB_LOOP_DISABLE); + /* Audio system digital clock power down release */ + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON2, + MT6357_CCI_AUDIO_FIFO_DISABLE | + MT6357_CCI_ACD_MODE_NORMAL_PATH | + MT6357_CCI_AFIFO_CLK_PWDB_ON | + MT6357_CCI_ACD_FUNC_RSTB_RESET); + /* sdm audio fifo clock power on */ + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON0, + MT6357_CCI_AUD_ANACK_INVERT | + (4 << MT6357_CCI_AUDIO_FIFO_WPTR_SFT) | + MT6357_CCI_SCRAMBLER_CG_ENABLE | + MT6357_CCI_RAND_ENABLE | + MT6357_CCI_SPLT_SCRMB_CLK_ON | + MT6357_CCI_SPLT_SCRMB_ON | + MT6357_CCI_ZERO_PADDING_DISABLE | + MT6357_CCI_SCRAMBLER_ENABLE); + /* scrambler clock on enable */ + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON2, + MT6357_CCI_AUDIO_FIFO_DISABLE | + MT6357_CCI_ACD_MODE_TEST_PATH | + MT6357_CCI_AFIFO_CLK_PWDB_ON | + MT6357_CCI_ACD_FUNC_RSTB_RELEASE); + /* sdm power on */ + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON2, + MT6357_CCI_AUDIO_FIFO_ENABLE | + MT6357_CCI_ACD_MODE_TEST_PATH | + MT6357_CCI_AFIFO_CLK_PWDB_ON | + MT6357_CCI_ACD_FUNC_RSTB_RELEASE); + + configure_downlinks(priv, true); + break; + case SND_SOC_DAPM_POST_PMD: + configure_downlinks(priv, false); + /* DL scrambler disabling sequence */ + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON2, + MT6357_CCI_AUDIO_FIFO_DISABLE | + MT6357_CCI_ACD_MODE_TEST_PATH | + MT6357_CCI_AFIFO_CLK_PWDB_DOWN | + MT6357_CCI_ACD_FUNC_RSTB_RESET); + regmap_write(priv->regmap, MT6357_AFUNC_AUD_CON0, + MT6357_CCI_AUD_ANACK_INVERT | + (4 << MT6357_CCI_AUDIO_FIFO_WPTR_SFT) | + MT6357_CCI_SCRAMBLER_CG_ENABLE | + MT6357_CCI_RAND_ENABLE | + MT6357_CCI_SPLT_SCRMB_CLK_ON | + MT6357_CCI_SPLT_SCRMB_ON | + MT6357_CCI_ZERO_PADDING_DISABLE | + MT6357_CCI_SCRAMBLER_DISABLE); + + set_playback_gpio(priv, false); + + /* disable Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, false); + break; + default: + break; + } + + return 0; +} + +static int mt_delay_250_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(250, 270); + break; + case SND_SOC_DAPM_PRE_PMD: + usleep_range(250, 270); + break; + default: + break; + } + + return 0; +} + +static int lo_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + int lgain, rgain; + + /* Get current gain value */ + regmap_read(priv->regmap, MT6357_ZCD_CON1, &lgain); + rgain = (lgain & MT6357_AUD_LOR_GAIN_MASK) >> MT6357_AUD_LOR_GAIN_SFT; + lgain = lgain & MT6357_AUD_LOL_GAIN_MASK; + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Set -40dB before enable HS to avoid POP noise */ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON1, + MT6357_AUD_LOL_GAIN_MASK | + MT6357_AUD_LOR_GAIN_MASK, + MT6357_DL_GAIN_N_40DB_REG); + /* Set LO STB enhance circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_MASK, + MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_ENABLE); + /* Enable LO driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_MASK, + MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_ENABLE); + /* Enable LO driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_PWRUP_VAUDP15_MASK, + MT6357_AUD_LOL_PWRUP_VAUDP15_ENABLE); + /* Set LOL gain to normal gain step by step */ + lo_volume_ramp(priv, DL_GAIN_N_40DB, lgain, + DL_GAIN_N_40DB, rgain); + break; + case SND_SOC_DAPM_PRE_PMD: + /* decrease LOL gain to minimum gain step by step */ + + lo_volume_ramp(priv, lgain, DL_GAIN_N_40DB, + rgain, DL_GAIN_N_40DB); + /* Disable LO driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_PWRUP_VAUDP15_MASK, + MT6357_AUD_LOL_PWRUP_VAUDP15_DISABLE); + /* Disable LO driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_MASK, + MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_DISABLE); + /* Clear LO STB enhance circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_MASK, + MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_DISABLE); + /* Save the gain value into the register*/ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON1, + MT6357_AUD_LOL_GAIN_MASK | + MT6357_AUD_LOR_GAIN_MASK, + lgain << MT6357_AUD_LOL_GAIN_SFT | + rgain << MT6357_AUD_LOR_GAIN_SFT); + + break; + default: + break; + } + + return 0; +} + +static int hs_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + int gain; /* HS register has only one gain slot */ + + /* Get current gain value */ + regmap_read(priv->regmap, MT6357_ZCD_CON3, &gain); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Set -40dB before enable HS to avoid POP noise */ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON3, + MT6357_AUD_HS_GAIN_MASK, + DL_GAIN_N_40DB); + + /* Set HS STB enhance circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HSOUT_STB_ENH_VAUDP15_MASK, + MT6357_AUD_HSOUT_STB_ENH_VAUDP15_ENABLE); + /* Enable HS driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_MASK, + MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_ENABLE); + /* Enable HS driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_PWRUP_VAUDP15_MASK, + MT6357_AUD_HS_PWRUP_VAUDP15_ENABLE); + /* Set HS gain to normal gain step by step */ + hs_volume_ramp(priv, DL_GAIN_N_40DB, gain); + break; + case SND_SOC_DAPM_PRE_PMD: + /* decrease HS gain to minimum gain step by step */ + hs_volume_ramp(priv, gain, DL_GAIN_N_40DB); + /* Disable HS driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_PWRUP_VAUDP15_MASK, + MT6357_AUD_HS_PWRUP_VAUDP15_DISABLE); + /* Disable HS driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_MASK, + MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_ENABLE); + /* Clear HS STB enhance circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HSOUT_STB_ENH_VAUDP15_MASK, + MT6357_AUD_HSOUT_STB_ENH_VAUDP15_DISABLE); + /* Save the gain value into the register*/ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON3, + MT6357_AUD_HS_GAIN_MASK, gain); + break; + default: + break; + } + + return 0; +} + +static int hp_main_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + int lgain, rgain; + + /* Get current gain value */ + regmap_read(priv->regmap, MT6357_ZCD_CON2, &lgain); + rgain = (lgain & MT6357_AUD_HPR_GAIN_MASK) >> MT6357_AUD_HPR_GAIN_SFT; + lgain = lgain & MT6357_AUD_HPL_GAIN_MASK; + switch (event) { + case SND_SOC_DAPM_POST_PMU: + priv->hp_channel_number++; + if (priv->hp_channel_number > 1) + break; + /* Set -40dB before enable HS to avoid POP noise */ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON2, + MT6357_AUD_HPL_GAIN_MASK | + MT6357_AUD_HPR_GAIN_MASK, + MT6357_DL_GAIN_N_40DB_REG); + /* Set HPP/N STB enhance circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON2, + MT6357_HPROUT_STB_ENH_VAUDP15_MASK | + MT6357_HPLOUT_STB_ENH_VAUDP15_MASK, + MT6357_HPROUT_STB_ENH_VAUDP15_N470_P250 | + MT6357_HPLOUT_STB_ENH_VAUDP15_N470_P250); + /* Enable HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_ENABLE | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_ENABLE); + /* Enable HP aux feedback loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_AUX_FBRSW_VAUDP15_MASK | + MT6357_HPL_AUX_FBRSW_VAUDP15_MASK, + MT6357_HPR_AUX_FBRSW_VAUDP15_ENABLE | + MT6357_HPL_AUX_FBRSW_VAUDP15_ENABLE); + /* Enable HP aux CMFB loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HP_CMFB_RST_MASK | + MT6357_HPL_AUX_CMFB_LOOP_MASK | + MT6357_HPR_AUX_CMFB_LOOP_MASK, + MT6357_HP_CMFB_RST_NORMAL | + MT6357_HPL_AUX_CMFB_LOOP_ENABLE | + MT6357_HPR_AUX_CMFB_LOOP_ENABLE); + /* Enable HP driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_BIAS_VAUDP15_MASK | + MT6357_AUD_HPL_BIAS_VAUDP15_MASK, + MT6357_AUD_HPR_BIAS_VAUDP15_ENABLE | + MT6357_AUD_HPL_BIAS_VAUDP15_ENABLE); + /* Enable HP driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_PWRUP_VAUDP15_MASK | + MT6357_AUD_HPL_PWRUP_VAUDP15_MASK, + MT6357_AUD_HPR_PWRUP_VAUDP15_ENABLE | + MT6357_AUD_HPL_PWRUP_VAUDP15_ENABLE); + /* Short HP main output to HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_MASK | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_MASK, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_ENABLE | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_ENABLE); + /* Enable HP main CMFB loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HPRL_MAIN_CMFB_LOOP_MASK, + MT6357_HPRL_MAIN_CMFB_LOOP_ENABLE); + /* Disable HP aux CMFB loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HPR_AUX_CMFB_LOOP_MASK | + MT6357_HPL_AUX_CMFB_LOOP_MASK, + MT6357_HPR_AUX_CMFB_LOOP_DISABLE | + MT6357_HPL_AUX_CMFB_LOOP_DISABLE); + /* Enable HP main output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_PWRUP_VAUDP15_ENABLE | + MT6357_HPLOUT_PWRUP_VAUDP15_ENABLE); + /* Enable HPR/L main output stage step by step */ + hp_main_output_ramp(priv, true); + usleep_range(1000, 1200); + /* Reduce HP aux feedback loop gain */ + hp_aux_feedback_loop_gain_ramp(priv, true); + /* Disable HP aux feedback loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_AUX_FBRSW_VAUDP15_MASK | + MT6357_HPL_AUX_FBRSW_VAUDP15_MASK, + MT6357_HPR_AUX_FBRSW_VAUDP15_DISABLE | + MT6357_HPL_AUX_FBRSW_VAUDP15_DISABLE); + /* apply volume setting */ + hp_volume_ramp(priv, DL_GAIN_N_40DB, lgain, + DL_GAIN_N_40DB, rgain); + /* Disable HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_DISABLE | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_DISABLE); + /* Unshort HP main output to HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_MASK | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_MASK, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_DISABLE | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_DISABLE); + usleep_range(100, 120); + break; + case SND_SOC_DAPM_PRE_PMD: + priv->hp_channel_number--; + if (priv->hp_channel_number > 0) + break; + /* Short HP main output to HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_MASK | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_MASK, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_ENABLE | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_ENABLE); + /* Enable HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_ENABLE | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_ENABLE); + /* decrease HPL/R gain to normal gain step by step */ + hp_volume_ramp(priv, lgain, DL_GAIN_N_40DB, + rgain, DL_GAIN_N_40DB); + /* Enable HP aux feedback loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_AUX_FBRSW_VAUDP15_MASK | + MT6357_HPL_AUX_FBRSW_VAUDP15_MASK, + MT6357_HPR_AUX_FBRSW_VAUDP15_ENABLE | + MT6357_HPL_AUX_FBRSW_VAUDP15_ENABLE); + /* Reduce HP aux feedback loop gain */ + hp_aux_feedback_loop_gain_ramp(priv, false); + /* decrease HPR/L main output stage step by step */ + hp_main_output_ramp(priv, false); + /* Disable HP main output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_PWRUP_VAUDP15_DISABLE | + MT6357_HPLOUT_PWRUP_VAUDP15_DISABLE); + /* Enable HP aux CMFB loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HP_CMFB_RST_MASK | + MT6357_HPL_AUX_CMFB_LOOP_MASK | + MT6357_HPR_AUX_CMFB_LOOP_MASK, + MT6357_HP_CMFB_RST_RESET | + MT6357_HPL_AUX_CMFB_LOOP_ENABLE | + MT6357_HPR_AUX_CMFB_LOOP_ENABLE); + /* Disable HP main CMFB loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HPRL_MAIN_CMFB_LOOP_MASK, + MT6357_HPRL_MAIN_CMFB_LOOP_DISABLE); + /* Unshort HP main output to HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_MASK | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_MASK, + MT6357_HPR_SHORT2HPR_AUX_VAUDP15_DISABLE | + MT6357_HPL_SHORT2HPR_AUX_VAUDP15_DISABLE); + /* Disable HP driver core circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_PWRUP_VAUDP15_MASK | + MT6357_AUD_HPL_PWRUP_VAUDP15_MASK, + MT6357_AUD_HPR_PWRUP_VAUDP15_DISABLE | + MT6357_AUD_HPL_PWRUP_VAUDP15_DISABLE); + /* Disable HP driver bias circuits */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_BIAS_VAUDP15_MASK | + MT6357_AUD_HPL_BIAS_VAUDP15_MASK, + MT6357_AUD_HPR_BIAS_VAUDP15_DISABLE | + MT6357_AUD_HPL_BIAS_VAUDP15_DISABLE); + /* Disable HP aux CMFB loop, + * Enable HP main CMFB for HP off state + */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON6, + MT6357_HPRL_MAIN_CMFB_LOOP_MASK | + MT6357_HPR_AUX_CMFB_LOOP_MASK | + MT6357_HPL_AUX_CMFB_LOOP_MASK, + MT6357_HPRL_MAIN_CMFB_LOOP_ENABLE | + MT6357_HPR_AUX_CMFB_LOOP_DISABLE | + MT6357_HPL_AUX_CMFB_LOOP_DISABLE); + /* Disable HP aux feedback loop */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPR_AUX_FBRSW_VAUDP15_MASK | + MT6357_HPL_AUX_FBRSW_VAUDP15_MASK, + MT6357_HPR_AUX_FBRSW_VAUDP15_DISABLE | + MT6357_HPL_AUX_FBRSW_VAUDP15_DISABLE); + /* Disable HP aux output stage */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON1, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_MASK | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_MASK, + MT6357_HPROUT_AUX_PWRUP_VAUDP15_DISABLE | + MT6357_HPLOUT_AUX_PWRUP_VAUDP15_DISABLE); + /* Save the gain value into the register*/ + regmap_update_bits(priv->regmap, MT6357_ZCD_CON2, + MT6357_AUD_HPL_GAIN_MASK | + MT6357_AUD_HPR_GAIN_MASK, + lgain << MT6357_AUD_HPL_GAIN_SFT | + rgain << MT6357_AUD_HPR_GAIN_SFT); + break; + default: + break; + } + + return 0; +} + +static int right_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable Audio DAC and control audio bias gen */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_DACR_PWRUP_VA28_MASK | + MT6357_AUD_DACR_PWRUP_VAUDP15_MASK, + MT6357_AUD_DACR_PWRUP_VA28_ENABLE | + MT6357_AUD_DACR_PWRUP_VAUDP15_ENABLE); + break; + case SND_SOC_DAPM_POST_PMU: + /* disable Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, false); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, true); + /* Disable Audio DAC and control audio bias gen */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_DACR_PWRUP_VA28_MASK | + MT6357_AUD_DACR_PWRUP_VAUDP15_MASK, + MT6357_AUD_DACR_PWRUP_VA28_DISABLE | + MT6357_AUD_DACR_PWRUP_VAUDP15_DISABLE); + break; + default: + break; + } + + return 0; +} + +static int left_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_component *cmpnt = snd_soc_dapm_to_component(w->dapm); + struct mt6357_priv *priv = snd_soc_component_get_drvdata(cmpnt); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable Audio DAC and control audio bias gen */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_DACL_PWRUP_VA28_MASK | + MT6357_AUD_DACL_PWRUP_VAUDP15_MASK, + MT6357_AUD_DACL_PWRUP_VA28_ENABLE | + MT6357_AUD_DACL_PWRUP_VAUDP15_ENABLE); + break; + case SND_SOC_DAPM_POST_PMU: + /* disable Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, false); + break; + case SND_SOC_DAPM_PRE_PMD: + /* Pull-down HPL/R to AVSS28_AUD */ + if (priv->pull_down_needed) + hp_pull_down(priv, true); + /* Disable Audio DAC and control audio bias gen */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_DACL_PWRUP_VA28_MASK | + MT6357_AUD_DACL_PWRUP_VAUDP15_MASK, + MT6357_AUD_DACL_PWRUP_VA28_DISABLE | + MT6357_AUD_DACL_PWRUP_VAUDP15_DISABLE); + break; + default: + break; + } + + return 0; +} + +/* Supply widgets subsequence */ +enum { + /* common */ + SUPPLY_SEQ_CLK_BUF, + SUPPLY_SEQ_AUD_GLB, + SUPPLY_SEQ_CLKSQ, + SUPPLY_SEQ_VOW_AUD_LPW, + SUPPLY_SEQ_AUD_VOW, + SUPPLY_SEQ_VOW_CLK, + SUPPLY_SEQ_VOW_LDO, + SUPPLY_SEQ_TOP_CK, + SUPPLY_SEQ_TOP_CK_LAST, + SUPPLY_SEQ_AUD_TOP, + SUPPLY_SEQ_AUD_TOP_LAST, + SUPPLY_SEQ_AFE, + /* capture */ + SUPPLY_SEQ_ADC_SUPPLY, +}; + +/* DAPM Widgets */ +static const struct snd_soc_dapm_widget mt6357_dapm_widgets[] = { + /* Analog Clocks */ + SND_SOC_DAPM_SUPPLY_S("CLK_BUF", SUPPLY_SEQ_CLK_BUF, + MT6357_DCXO_CW14, + MT6357_XO_AUDIO_EN_M_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDGLB", SUPPLY_SEQ_AUD_GLB, + MT6357_AUDDEC_ANA_CON11, + MT6357_AUDGLB_PWRDN_VA28_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("CLKSQ Audio", SUPPLY_SEQ_CLKSQ, + MT6357_AUDENC_ANA_CON6, + MT6357_CLKSQ_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDNCP_CK", SUPPLY_SEQ_TOP_CK, + MT6357_AUD_TOP_CKPDN_CON0, + MT6357_AUDNCP_CK_PDN_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ZCD13M_CK", SUPPLY_SEQ_TOP_CK, + MT6357_AUD_TOP_CKPDN_CON0, + MT6357_ZCD13M_CK_PDN_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUD_CK", SUPPLY_SEQ_TOP_CK_LAST, + MT6357_AUD_TOP_CKPDN_CON0, + MT6357_AUD_CK_PDN_SFT, 1, + mt_delay_250_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY_S("AUDIF_CK", SUPPLY_SEQ_TOP_CK, + MT6357_AUD_TOP_CKPDN_CON0, + MT6357_AUDIF_CK_PDN_SFT, 1, NULL, 0), + + /* Digital Clocks */ + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_AFE_CTL", SUPPLY_SEQ_AUD_TOP_LAST, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_AFE_CTL_SFT, 1, + mt_delay_250_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_DAC_CTL", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_DAC_CTL_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_ADC_CTL", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_ADC_CTL_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_I2S_DL", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_I2S_DL_CTL_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_PWR_CLK", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PWR_CLK_DIS_CTL_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_PDN_AFE_TESTMODEL", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_AFE_TESTMODEL_CTL_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_PDN_RESERVED", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_RESERVED_SFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AUDIO_TOP_LPBK", SUPPLY_SEQ_AUD_TOP, + MT6357_AUDIO_TOP_CON0, + MT6357_PDN_LPBK_CTL_SFT, 1, NULL, 0), + + /* General */ + SND_SOC_DAPM_SUPPLY_S("AFE_ON", SUPPLY_SEQ_AFE, + MT6357_AFE_UL_DL_CON0, + MT6357_AFE_ON_SFT, 0, NULL, 0), + + /* Uplinks */ + SND_SOC_DAPM_AIF_OUT_E("AIF1TX", "MT6357 Capture", 0, + SND_SOC_NOPM, 0, 0, + mt_aif_out_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY_S("ADC Supply", SUPPLY_SEQ_ADC_SUPPLY, + SND_SOC_NOPM, 0, 0, + mt_adc_supply_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC", NULL, SND_SOC_NOPM, 0, 0, adc_enable_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("PGA L Mux", SND_SOC_NOPM, 0, 0, + &pga_left_mux_control, + mt_pga_left_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX_E("PGA R Mux", SND_SOC_NOPM, 0, 0, + &pga_right_mux_control, + mt_pga_right_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("PGA L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX_E("Mic Type Mux", SND_SOC_NOPM, 0, 0, + &mic_type_mux_control, + mt_mic_type_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("MICBIAS0", MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_PWD_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", MT6357_AUDENC_ANA_CON9, + MT6357_AUD_MICBIAS1_PWD_SFT, 0, NULL, 0), + + /* UL inputs */ + SND_SOC_DAPM_INPUT("AIN0"), + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("LPBK"), + SND_SOC_DAPM_INPUT("SGEN UL"), + + /* Downlinks */ + SND_SOC_DAPM_AIF_IN_E("AIF_RX", "MT6357 Playback", 0, + SND_SOC_NOPM, 0, 0, + mt_audio_in_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_INPUT("SGEN DL"), + SND_SOC_DAPM_MUX("DAC Mux", SND_SOC_NOPM, 0, 0, &dac_mux_control), + + SND_SOC_DAPM_DAC_E("DACR", NULL, SND_SOC_NOPM, 0, 0, right_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DACL", NULL, SND_SOC_NOPM, 0, 0, left_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_SUPPLY("DL Digital Supply", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DL Analog Supply", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DL SRC", MT6357_AFE_DL_SRC2_CON0_L, + MT6357_DL_2_SRC_ON_TMP_CTL_PRE_SFT, 0, NULL, 0), + + SND_SOC_DAPM_MUX_E("Line Out Source", SND_SOC_NOPM, 0, 0, &lo_mux_control, + lo_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX_E("Handset Source", SND_SOC_NOPM, 0, 0, &hs_mux_control, + hs_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX_E("Headphone Right Source", SND_SOC_NOPM, 0, 0, &hpr_mux_control, + hp_main_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_MUX_E("Headphone Left Source", SND_SOC_NOPM, 0, 0, &hpl_mux_control, + hp_main_mux_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + /* DL outputs */ + SND_SOC_DAPM_OUTPUT("Headphones"), + SND_SOC_DAPM_OUTPUT("Hansdet"), + SND_SOC_DAPM_OUTPUT("Line out"), + + /* Sine generator */ + SND_SOC_DAPM_SUPPLY("SGEN UL Enable", + MT6357_AFE_TOP_CON0, MT6357_UL_SINE_ON_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SGEN Enable", + MT6357_AFE_SGEN_CFG0, + MT6357_SGEN_DAC_EN_CTL_SFT, 0, mt_audio_in_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SGEN MUTE", + MT6357_AFE_SGEN_CFG0, + MT6357_SGEN_MUTE_SW_CTL_SFT, 1, NULL, 0) +}; + +static const struct snd_soc_dapm_route mt6357_dapm_routes[] = { + /* Capture */ + {"AIF1TX", NULL, "Mic Type Mux"}, + {"AIF1TX", NULL, "CLK_BUF"}, + {"AIF1TX", NULL, "AUDGLB"}, + {"AIF1TX", NULL, "CLKSQ Audio"}, + {"AIF1TX", NULL, "AUD_CK"}, + {"AIF1TX", NULL, "AUDIF_CK"}, + + {"AIF1TX", NULL, "AUDIO_TOP_AFE_CTL"}, + {"AIF1TX", NULL, "AUDIO_TOP_ADC_CTL"}, + {"AIF1TX", NULL, "AUDIO_TOP_PWR_CLK"}, + {"AIF1TX", NULL, "AUDIO_TOP_PDN_RESERVED"}, + {"AIF1TX", NULL, "AUDIO_TOP_I2S_DL"}, + {"AIF1TX", NULL, "AFE_ON"}, + + {"Mic Type Mux", "ACC", "ADC"}, + {"Mic Type Mux", "DCC", "ADC"}, + {"Mic Type Mux", "DCC_ECM_DIFF", "ADC"}, + {"Mic Type Mux", "DCC_ECM_SINGLE", "ADC"}, + {"Mic Type Mux", "DMIC", "AIN0"}, + {"Mic Type Mux", "DMIC", "AIN2"}, + {"Mic Type Mux", "Loopback", "LPBK"}, + {"Mic Type Mux", "Sine Generator", "SGEN UL"}, + + {"SGEN UL", NULL, "AUDIO_TOP_PDN_AFE_TESTMODEL"}, + {"SGEN UL", NULL, "SGEN UL Enable"}, + {"SGEN UL", NULL, "SGEN MUTE"}, + {"SGEN UL", NULL, "SGEN Enable"}, + + {"ADC", NULL, "PGA L Mux"}, + {"ADC", NULL, "PGA R Mux"}, + {"ADC", NULL, "ADC Supply"}, + + {"PGA L Mux", "AIN0", "AIN0"}, + {"PGA L Mux", "AIN1", "AIN1"}, + {"PGA L Mux", "AIN2", "AIN2"}, + + {"PGA R Mux", "AIN0", "AIN0"}, + {"PGA R Mux", "AIN1", "AIN1"}, + {"PGA R Mux", "AIN2", "AIN2"}, + + {"AIN0", NULL, "MICBIAS0"}, + {"AIN1", NULL, "MICBIAS1"}, + {"AIN2", NULL, "MICBIAS0"}, + {"LPBK", NULL, "AUDIO_TOP_LPBK"}, + + /* Playback */ + {"DAC Mux", "Normal Path", "AIF_RX"}, + {"DAC Mux", "Sine Generator", "SGEN DL"}, + + {"AIF_RX", NULL, "DL SRC"}, + + {"SGEN DL", NULL, "DL SRC"}, + {"SGEN DL", NULL, "SGEN MUTE"}, + {"SGEN DL", NULL, "SGEN Enable"}, + {"SGEN DL", NULL, "DL Digital Supply"}, + {"SGEN DL", NULL, "AUDIO_TOP_PDN_AFE_TESTMODEL"}, + + {"DACL", NULL, "DAC Mux"}, + {"DACR", NULL, "DAC Mux"}, + + {"DL Analog Supply", NULL, "CLK_BUF"}, + {"DL Analog Supply", NULL, "AUDGLB"}, + {"DL Analog Supply", NULL, "CLKSQ Audio"}, + {"DL Analog Supply", NULL, "AUDNCP_CK"}, + {"DL Analog Supply", NULL, "ZCD13M_CK"}, + {"DL Analog Supply", NULL, "AUD_CK"}, + {"DL Analog Supply", NULL, "AUDIF_CK"}, + + {"DL Digital Supply", NULL, "AUDIO_TOP_AFE_CTL"}, + {"DL Digital Supply", NULL, "AUDIO_TOP_DAC_CTL"}, + {"DL Digital Supply", NULL, "AUDIO_TOP_PWR_CLK"}, + {"DL Digital Supply", NULL, "AFE_ON"}, + + {"DACR", NULL, "DL Digital Supply"}, + {"DACR", NULL, "DL Analog Supply"}, + {"DACL", NULL, "DL Digital Supply"}, + {"DACL", NULL, "DL Analog Supply"}, + + {"Line Out Source", "DACR", "DACR"}, + {"Line Out Source", "Playback", "DACL"}, + {"Line Out Source", "Test mode", "DACL"}, + + {"Handset Source", "DACR", "DACR"}, + {"Handset Source", "Playback", "DACL"}, + {"Handset Source", "Test mode", "DACL"}, + + {"Headphone Right Source", "DAC", "DACR"}, + {"Headphone Right Source", "Line Out", "Line Out Source"}, + {"Headphone Right Source", "Handset", "Handset Source"}, + + {"Headphone Left Source", "DAC", "DACL"}, + {"Headphone Left Source", "Line Out", "Line Out Source"}, + {"Headphone Left Source", "Handset", "Handset Source"}, + + {"Line out", NULL, "Line Out Source"}, + {"Hansdet", NULL, "Handset Source"}, + + {"Headphones", NULL, "Headphone Right Source"}, + {"Headphones", NULL, "Headphone Left Source"}, +}; + +static struct snd_soc_dai_driver mtk_6357_dai_codecs[] = { + { + .name = "mt6357-snd-codec-aif1", + .playback = { + .stream_name = "MT6357 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = MT6357_SND_SOC_ADV_MT_FMTS, + }, + .capture = { + .stream_name = "MT6357 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MT6357_SOC_HIGH_USE_RATE, + .formats = MT6357_SND_SOC_ADV_MT_FMTS, + }, + }, +}; + +static int mt6357_codec_probe(struct snd_soc_component *codec) +{ + struct mt6357_priv *priv = snd_soc_component_get_drvdata(codec); + + snd_soc_component_init_regmap(codec, priv->regmap); + + /* Enable audio part */ + regmap_update_bits(priv->regmap, MT6357_DCXO_CW14, + MT6357_XO_AUDIO_EN_M_MASK, MT6357_XO_AUDIO_EN_M_ENABLE); + /* Disable HeadphoneL/HeadphoneR short circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON0, + MT6357_AUD_HPR_SC_VAUDP15_MASK | + MT6357_AUD_HPL_SC_VAUDP15_MASK, + MT6357_AUD_HPR_SC_VAUDP15_DISABLE | + MT6357_AUD_HPL_SC_VAUDP15_DISABLE); + /* Disable voice short circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON3, + MT6357_AUD_HS_SC_VAUDP15_MASK, + MT6357_AUD_HS_SC_VAUDP15_DISABLE); + /* disable LO buffer left short circuit protection */ + regmap_update_bits(priv->regmap, MT6357_AUDDEC_ANA_CON4, + MT6357_AUD_LOL_SC_VAUDP15_MASK, + MT6357_AUD_LOL_SC_VAUDP15_DISABLE); + /* set gpio */ + set_playback_gpio(priv, false); + set_capture_gpio(priv, false); + /* Disable audio part */ + regmap_update_bits(priv->regmap, MT6357_DCXO_CW14, + MT6357_XO_AUDIO_EN_M_MASK, + MT6357_XO_AUDIO_EN_M_DISABLE); + + return 0; +} + +static const struct snd_soc_component_driver mt6357_soc_component_driver = { + .probe = mt6357_codec_probe, + .read = snd_soc_component_read, + .write = snd_soc_component_write, + .controls = mt6357_controls, + .num_controls = ARRAY_SIZE(mt6357_controls), + .dapm_widgets = mt6357_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6357_dapm_widgets), + .dapm_routes = mt6357_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6357_dapm_routes), +}; + +static const u32 micbias_values[] = { + 1700000, 1800000, 1900000, 2000000, + 2100000, 2500000, 2600000, 2700000 +}; + +static u32 mt6357_get_micbias_idx(struct device_node *np, const char *micbias) +{ + int err; + u32 idx, val; + + err = of_property_read_u32(np, micbias, &val); + if (err) + return 0; + + for (idx = 0; idx < ARRAY_SIZE(micbias_values); idx++) { + if (val == micbias_values[idx]) + return idx; + } + return 0; +} + +static int mt6357_parse_dt(struct mt6357_priv *priv) +{ + u32 micbias_voltage_index = 0; + struct device_node *np = priv->dev->parent->of_node; + + if (!np) + return -EINVAL; + + priv->pull_down_needed = false; + if (of_property_read_bool(np, "mediatek,hp-pull-down")) + priv->pull_down_needed = true; + + micbias_voltage_index = mt6357_get_micbias_idx(np, "mediatek,micbias0-microvolt"); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON8, + MT6357_AUD_MICBIAS0_VREF_MASK, + micbias_voltage_index << MT6357_AUD_MICBIAS0_VREF_SFT); + + micbias_voltage_index = mt6357_get_micbias_idx(np, "mediatek,micbias1-microvolt"); + regmap_update_bits(priv->regmap, MT6357_AUDENC_ANA_CON9, + MT6357_AUD_MICBIAS1_VREF_MASK, + micbias_voltage_index << MT6357_AUD_MICBIAS1_VREF_SFT); + + return 0; +} + +static int mt6357_platform_driver_probe(struct platform_device *pdev) +{ + struct mt6397_chip *mt6397 = dev_get_drvdata(pdev->dev.parent); + struct mt6357_priv *priv; + int ret; + + ret = devm_regulator_get_enable(&pdev->dev, "vaud28"); + if (ret) + return dev_err_probe(&pdev->dev, ret, "Failed to enable vaud28 regulator\n"); + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->dev = &pdev->dev; + + priv->regmap = mt6397->regmap; + if (IS_ERR(priv->regmap)) + return PTR_ERR(priv->regmap); + + ret = mt6357_parse_dt(priv); + if (ret) + return dev_err_probe(&pdev->dev, ret, "Failed to parse dts\n"); + + pdev->dev.coherent_dma_mask = DMA_BIT_MASK(64); + if (!pdev->dev.dma_mask) + pdev->dev.dma_mask = &pdev->dev.coherent_dma_mask; + + return devm_snd_soc_register_component(&pdev->dev, + &mt6357_soc_component_driver, + mtk_6357_dai_codecs, + ARRAY_SIZE(mtk_6357_dai_codecs)); +} + +static const struct platform_device_id mt6357_platform_ids[] = { + {"mt6357-sound", 0}, + { /* sentinel */ }, +}; +MODULE_DEVICE_TABLE(platform, mt6357_platform_ids); + +static struct platform_driver mt6357_platform_driver = { + .driver = { + .name = "mt6357-sound", + .probe_type = PROBE_PREFER_ASYNCHRONOUS, + }, + .probe = mt6357_platform_driver_probe, + .id_table = mt6357_platform_ids, +}; + +module_platform_driver(mt6357_platform_driver) + +MODULE_DESCRIPTION("MT6357 ALSA SoC codec driver"); +MODULE_AUTHOR("Nicolas Belin "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mt6357.h b/sound/soc/codecs/mt6357.h new file mode 100644 index 000000000000..7f6fccada6a2 --- /dev/null +++ b/sound/soc/codecs/mt6357.h @@ -0,0 +1,660 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * mt6357.h -- mt6357 ALSA SoC audio codec driver + * + * Copyright (c) 2024 Baylibre + * Author: Nicolas Belin + */ + +#ifndef __MT6357_H__ +#define __MT6357_H__ + +#include + +/* Reg bit defines */ +/* MT6357_GPIO_DIR0 */ +#define MT6357_GPIO8_DIR_MASK BIT(8) +#define MT6357_GPIO8_DIR_INPUT 0 +#define MT6357_GPIO8_DIR_OUTPUT BIT(8) +#define MT6357_GPIO9_DIR_MASK BIT(9) +#define MT6357_GPIO9_DIR_INPUT 0 +#define MT6357_GPIO9_DIR_OUTPUT BIT(9) +#define MT6357_GPIO10_DIR_MASK BIT(10) +#define MT6357_GPIO10_DIR_INPUT 0 +#define MT6357_GPIO10_DIR_OUTPUT BIT(10) +#define MT6357_GPIO11_DIR_MASK BIT(11) +#define MT6357_GPIO11_DIR_INPUT 0 +#define MT6357_GPIO11_DIR_OUTPUT BIT(11) +#define MT6357_GPIO12_DIR_MASK BIT(12) +#define MT6357_GPIO12_DIR_INPUT 0 +#define MT6357_GPIO12_DIR_OUTPUT BIT(12) +#define MT6357_GPIO13_DIR_MASK BIT(13) +#define MT6357_GPIO13_DIR_INPUT 0 +#define MT6357_GPIO13_DIR_OUTPUT BIT(13) +#define MT6357_GPIO14_DIR_MASK BIT(14) +#define MT6357_GPIO14_DIR_INPUT 0 +#define MT6357_GPIO14_DIR_OUTPUT BIT(14) +#define MT6357_GPIO15_DIR_MASK BIT(15) +#define MT6357_GPIO15_DIR_INPUT 0 +#define MT6357_GPIO15_DIR_OUTPUT BIT(15) + +/* MT6357_GPIO_MODE2 */ +#define MT6357_GPIO8_MODE_MASK GENMASK(2, 0) +#define MT6357_GPIO8_MODE_AUD_CLK_MOSI BIT(0) +#define MT6357_GPIO8_MODE_GPIO 0 +#define MT6357_GPIO9_MODE_MASK GENMASK(5, 3) +#define MT6357_GPIO9_MODE_AUD_DAT_MOSI0 BIT(3) +#define MT6357_GPIO9_MODE_GPIO 0 +#define MT6357_GPIO10_MODE_MASK GENMASK(8, 6) +#define MT6357_GPIO10_MODE_AUD_DAT_MOSI1 BIT(6) +#define MT6357_GPIO10_MODE_GPIO 0 +#define MT6357_GPIO11_MODE_MASK GENMASK(11, 9) +#define MT6357_GPIO11_MODE_AUD_SYNC_MOSI BIT(9) +#define MT6357_GPIO11_MODE_GPIO 0 + +/* MT6357_GPIO_MODE2_SET */ +#define MT6357_GPIO8_MODE_SET_MASK GENMASK(2, 0) +#define MT6357_GPIO8_MODE_SET_AUD_CLK_MOSI BIT(0) +#define MT6357_GPIO9_MODE_SET_MASK GENMASK(5, 3) +#define MT6357_GPIO9_MODE_SET_AUD_DAT_MOSI0 BIT(3) +#define MT6357_GPIO10_MODE_SET_MASK GENMASK(8, 6) +#define MT6357_GPIO10_MODE_SET_AUD_DAT_MOSI1 BIT(6) +#define MT6357_GPIO11_MODE_SET_MASK GENMASK(11, 9) +#define MT6357_GPIO11_MODE_SET_AUD_SYNC_MOSI BIT(9) + +/* MT6357_GPIO_MODE2_CLR */ +#define MT6357_GPIO_MODE2_CLEAR_ALL GENMASK(15, 0) + +/* MT6357_GPIO_MODE3 */ +#define MT6357_GPIO12_MODE_MASK GENMASK(2, 0) +#define MT6357_GPIO12_MODE_AUD_CLK_MISO BIT(0) +#define MT6357_GPIO12_MODE_GPIO 0 +#define MT6357_GPIO13_MODE_MASK GENMASK(5, 3) +#define MT6357_GPIO13_MODE_AUD_DAT_MISO0 BIT(3) +#define MT6357_GPIO13_MODE_GPIO 0 +#define MT6357_GPIO14_MODE_MASK GENMASK(8, 6) +#define MT6357_GPIO14_MODE_AUD_DAT_MISO1 BIT(6) +#define MT6357_GPIO14_MODE_GPIO 0 +#define MT6357_GPIO15_MODE_MASK GENMASK(11, 9) +#define MT6357_GPIO15_MODE_AUD_SYNC_MISO BIT(9) +#define MT6357_GPIO15_MODE_GPIO 0 + +/* MT6357_GPIO_MODE3_SET */ +#define MT6357_GPIO12_MODE_SET_MASK GENMASK(2, 0) +#define MT6357_GPIO12_MODE_SET_AUD_CLK_MISO BIT(0) +#define MT6357_GPIO13_MODE_SET_MASK GENMASK(5, 3) +#define MT6357_GPIO13_MODE_SET_AUD_DAT_MISO0 BIT(3) +#define MT6357_GPIO14_MODE_SET_MASK GENMASK(8, 6) +#define MT6357_GPIO14_MODE_SET_AUD_DAT_MISO1 BIT(6) +#define MT6357_GPIO15_MODE_SET_MASK GENMASK(11, 9) +#define MT6357_GPIO15_MODE_SET_AUD_SYNC_MISO BIT(9) + +/* MT6357_GPIO_MODE3_CLR */ +#define MT6357_GPIO_MODE3_CLEAR_ALL GENMASK(15, 0) + +/* MT6357_DCXO_CW14 */ +#define MT6357_XO_AUDIO_EN_M_SFT 13 +#define MT6357_XO_AUDIO_EN_M_MASK BIT(13) +#define MT6357_XO_AUDIO_EN_M_ENABLE BIT(13) +#define MT6357_XO_AUDIO_EN_M_DISABLE 0 + +/* MT6357_AUD_TOP_CKPDN_CON0 */ +#define MT6357_AUDNCP_CK_PDN_SFT 6 +#define MT6357_ZCD13M_CK_PDN_SFT 5 +#define MT6357_AUDIF_CK_PDN_SFT 2 +#define MT6357_AUD_CK_PDN_SFT 1 + +/* MT6357_AUDNCP_CLKDIV_CON0 */ +#define MT6357_DIVCKS_CHG BIT(0) + +/* MT6357_AUDNCP_CLKDIV_CON1 */ +#define MT6357_DIVCKS_ON BIT(0) + +/* MT6357_AUDNCP_CLKDIV_CON3 */ +#define MT6357_DIVCKS_PWD_NCP_MASK BIT(0) +#define MT6357_DIVCKS_PWD_NCP_DISABLE BIT(0) +#define MT6357_DIVCKS_PWD_NCP_ENABLE 0 + +/* MT6357_AUDNCP_CLKDIV_CON4 */ +#define MT6357_DIVCKS_PWD_NCP_ST_SEL_MASK GENMASK(1, 0) +#define MT6357_DIVCKS_PWD_NCP_ST_50US 0 +#define MT6357_DIVCKS_PWD_NCP_ST_100US 1 +#define MT6357_DIVCKS_PWD_NCP_ST_150US 2 +#define MT6357_DIVCKS_PWD_NCP_ST_200US 3 + +/* MT6357_AFE_UL_DL_CON0 */ +#define MT6357_AFE_UL_LR_SWAP_SFT 15 +#define MT6357_AFE_ON_SFT 0 + +/* MT6357_AFE_DL_SRC2_CON0_L */ +#define MT6357_DL_2_SRC_ON_TMP_CTL_PRE_SFT 0 + +/* MT6357_AFE_UL_SRC_CON0_H */ +#define MT6357_C_TWO_DIGITAL_MIC_CTL_MASK BIT(7) +#define MT6357_C_TWO_DIGITAL_MIC_ENABLE BIT(7) +#define MT6357_C_TWO_DIGITAL_MIC_DISABLE 0 + +/* MT6357_AFE_UL_SRC_CON0_L */ +#define MT6357_UL_SDM_3_LEVEL_CTL_MASK BIT(1) +#define MT6357_UL_SDM_3_LEVEL_SELECT BIT(1) +#define MT6357_UL_SDM_3_LEVEL_DESELECT 0 +#define MT6357_UL_SRC_ON_TMP_CTL_MASK BIT(0) +#define MT6357_UL_SRC_ENABLE BIT(0) +#define MT6357_UL_SRC_DISABLE 0 + +/* MT6357_AFE_TOP_CON0 */ +#define MT6357_UL_SINE_ON_SFT 1 +#define MT6357_UL_SINE_ON_MASK BIT(1) +#define MT6357_DL_SINE_ON_SFT 0 +#define MT6357_DL_SINE_ON_MASK BIT(0) + +/* MT6357_AUDIO_TOP_CON0 */ +#define MT6357_PDN_LPBK_CTL_SFT 15 +#define MT6357_PDN_AFE_CTL_SFT 7 +#define MT6357_PDN_DAC_CTL_SFT 6 +#define MT6357_PDN_ADC_CTL_SFT 5 +#define MT6357_PDN_I2S_DL_CTL_SFT 3 +#define MT6357_PWR_CLK_DIS_CTL_SFT 2 +#define MT6357_PDN_AFE_TESTMODEL_CTL_SFT 1 +#define MT6357_PDN_RESERVED_SFT 0 + +/* MT6357_AFUNC_AUD_CON0 */ +#define MT6357_CCI_AUD_ANACK_INVERT BIT(15) +#define MT6357_CCI_AUD_ANACK_NORMAL 0 +#define MT6357_CCI_AUDIO_FIFO_WPTR_SFT 12 +#define MT6357_CCI_SCRAMBLER_CG_ENABLE BIT(11) +#define MT6357_CCI_SCRAMBLER_CG_DISABLE 0 +#define MT6357_CCI_LCK_INV_OUT_OF_PHASE BIT(10) +#define MT6357_CCI_LCK_INV_IN_PHASE 0 +#define MT6357_CCI_RAND_ENABLE BIT(9) +#define MT6357_CCI_RAND_DISABLE 0 +#define MT6357_CCI_SPLT_SCRMB_CLK_ON BIT(8) +#define MT6357_CCI_SPLT_SCRMB_CLK_OFF 0 +#define MT6357_CCI_SPLT_SCRMB_ON BIT(7) +#define MT6357_CCI_SPLT_SCRMB_OFF 0 +#define MT6357_CCI_AUD_IDAC_TEST_EN_FROM_TEST_IN BIT(6) +#define MT6357_CCI_AUD_IDAC_TEST_EN_NORMAL_PATH 0 +#define MT6357_CCI_ZERO_PADDING_DISABLE BIT(5) +#define MT6357_CCI_ZERO_PADDING_ENABLE 0 +#define MT6357_CCI_AUD_SPLIT_TEST_EN_FROM_TEST_IN BIT(4) +#define MT6357_CCI_AUD_SPLIT_TEST_EN_NORMAL_PATH 0 +#define MT6357_CCI_AUD_SDM_MUTE_L_REG_CTL BIT(3) +#define MT6357_CCI_AUD_SDM_MUTE_L_NO_CTL 0 +#define MT6357_CCI_AUD_SDM_MUTE_R_REG_CTL BIT(2) +#define MT6357_CCI_AUD_SDM_MUTE_R_NO_CTL 0 +#define MT6357_CCI_AUD_SDM_7BIT_FROM_SPLITTER3 BIT(1) +#define MT6357_CCI_AUD_SDM_7BIT_FROM_SPLITTER1 0 +#define MT6357_CCI_SCRAMBLER_ENABLE BIT(0) +#define MT6357_CCI_SCRAMBLER_DISABLE 0 + +/* MT6357_AFUNC_AUD_CON2 */ +#define MT6357_CCI_AUDIO_FIFO_ENABLE BIT(3) +#define MT6357_CCI_AUDIO_FIFO_DISABLE 0 +#define MT6357_CCI_ACD_MODE_NORMAL_PATH BIT(2) +#define MT6357_CCI_ACD_MODE_TEST_PATH 0 +#define MT6357_CCI_AFIFO_CLK_PWDB_ON BIT(1) +#define MT6357_CCI_AFIFO_CLK_PWDB_DOWN 0 +#define MT6357_CCI_ACD_FUNC_RSTB_RELEASE BIT(0) +#define MT6357_CCI_ACD_FUNC_RSTB_RESET 0 + +/* MT6357_AFE_ADDA_MTKAIF_CFG0 */ +#define MT6357_ADDA_MTKAIF_LPBK_CTL_MASK BIT(1) +#define MT6357_ADDA_MTKAIF_LPBK_ENABLE BIT(1) +#define MT6357_ADDA_MTKAIF_LPBK_DISABLE 0 + +/* MT6357_AFE_SGEN_CFG0 */ +#define MT6357_SGEN_DAC_EN_CTL_SFT 7 +#define MT6357_SGEN_DAC_ENABLE BIT(7) +#define MT6357_SGEN_MUTE_SW_CTL_SFT 6 +#define MT6357_SGEN_MUTE_SW_DISABLE 0 + +/* MT6357_AFE_DCCLK_CFG0 */ +#define MT6357_DCCLK_DIV_MASK GENMASK(15, 5) +#define MT6357_DCCLK_DIV_SFT 5 +#define MT6357_DCCLK_DIV_RUN_VALUE (32 << MT6357_DCCLK_DIV_SFT) +#define MT6357_DCCLK_DIV_STOP_VALUE (259 << MT6357_DCCLK_DIV_SFT) +#define MT6357_DCCLK_PDN_MASK BIT(1) +#define MT6357_DCCLK_PDN BIT(1) +#define MT6357_DCCLK_OUTPUT 0 +#define MT6357_DCCLK_GEN_ON_MASK BIT(0) +#define MT6357_DCCLK_GEN_ON BIT(0) +#define MT6357_DCCLK_GEN_OFF 0 + +/* MT6357_AFE_DCCLK_CFG1 */ +#define MT6357_DCCLK_RESYNC_BYPASS_MASK BIT(8) +#define MT6357_DCCLK_RESYNC_BYPASS BIT(8) + +/* MT6357_AFE_AUD_PAD_TOP */ +#define MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_MASK GENMASK(15, 8) +#define MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_ENABLE (BIT(13) | BIT(12) | BIT(8)) +#define MT6357_AUD_PAD_TX_FIFO_NORMAL_PATH_DISABLE (BIT(13) | BIT(12)) +#define MT6357_AUD_PAD_TX_FIFO_LPBK_MASK GENMASK(7, 0) +#define MT6357_AUD_PAD_TX_FIFO_LPBK_ENABLE (BIT(5) | BIT(4) | BIT(0)) +#define MT6357_AUD_PAD_TX_FIFO_LPBK_DISABLE 0 + +/* MT6357_AUDENC_ANA_CON0 */ +#define MT6357_AUDADCLINPUTSEL_MASK GENMASK(14, 13) +#define MT6357_AUDADCLINPUTSEL_PREAMPLIFIER BIT(14) +#define MT6357_AUDADCLINPUTSEL_IDLE 0 +#define MT6357_AUDADCLPWRUP_SFT 12 +#define MT6357_AUDADCLPWRUP_MASK BIT(12) +#define MT6357_AUDADCLPWRUP BIT(12) +#define MT6357_AUDADCLPWRDOWN 0 +#define MT6357_AUDPREAMPLGAIN_SFT 8 +#define MT6357_AUDPREAMPLGAIN_MASK GENMASK(10, 8) +#define MT6357_AUDPREAMPLGAIN_MAX 4 +#define MT6357_AUDPREAMPLINPUTSEL_SFT 6 +#define MT6357_AUDPREAMPLINPUTSEL_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUDPREAMPLDCPRECHARGE_MASK BIT(2) +#define MT6357_AUDPREAMPLDCPRECHARGE_ENABLE BIT(2) +#define MT6357_AUDPREAMPLDCPRECHARGE_DISABLE 0 +#define MT6357_AUDPREAMPLDCCEN_MASK BIT(1) +#define MT6357_AUDPREAMPLDCCEN_DC BIT(1) +#define MT6357_AUDPREAMPLDCCEN_AC 0 +#define MT6357_AUDPREAMPLON_MASK BIT(0) +#define MT6357_AUDPREAMPLON_ENABLE BIT(0) +#define MT6357_AUDPREAMPLON_DISABLE 0 + +/* MT6357_AUDENC_ANA_CON1 */ +#define MT6357_AUDADCRINPUTSEL_MASK GENMASK(14, 13) +#define MT6357_AUDADCRINPUTSEL_PREAMPLIFIER BIT(14) +#define MT6357_AUDADCRINPUTSEL_IDLE 0 +#define MT6357_AUDADCRPWRUP_SFT 12 +#define MT6357_AUDADCRPWRUP_MASK BIT(12) +#define MT6357_AUDADCRPWRUP BIT(12) +#define MT6357_AUDADCRPWRDOWN 0 +#define MT6357_AUDPREAMPRGAIN_SFT 8 +#define MT6357_AUDPREAMPRGAIN_MASK GENMASK(10, 8) +#define MT6357_AUDPREAMPRGAIN_MAX 4 +#define MT6357_AUDPREAMPRINPUTSEL_SFT 6 +#define MT6357_AUDPREAMPRINPUTSEL_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUDPREAMPRDCPRECHARGE_MASK BIT(2) +#define MT6357_AUDPREAMPRDCPRECHARGE_ENABLE BIT(2) +#define MT6357_AUDPREAMPRDCPRECHARGE_DISABLE 0 +#define MT6357_AUDPREAMPRDCCEN_MASK BIT(1) +#define MT6357_AUDPREAMPRDCCEN_DC BIT(1) +#define MT6357_AUDPREAMPRDCCEN_AC 0 +#define MT6357_AUDPREAMPRON_MASK BIT(0) +#define MT6357_AUDPREAMPRON_ENABLE BIT(0) +#define MT6357_AUDPREAMPRON_DISABLE 0 + +/* MT6357_AUDENC_ANA_CON6 */ +#define MT6357_CLKSQ_EN_SFT 0 + +/* MT6357_AUDENC_ANA_CON7 */ +#define MT6357_AUDDIGMICBIAS_MASK GENMASK(2, 1) +#define MT6357_AUDDIGMICBIAS_DEFAULT_VALUE BIT(2) +#define MT6357_AUDDIGMICBIAS_OFF 0 +#define MT6357_AUDDIGMICEN_MASK BIT(0) +#define MT6357_AUDDIGMICEN_ENABLE BIT(0) +#define MT6357_AUDDIGMICEN_DISABLE 0 + +/* MT6357_AUDENC_ANA_CON8 */ +#define MT6357_AUD_MICBIAS0_DCSW2N_EN_MASK BIT(14) +#define MT6357_AUD_MICBIAS0_DCSW2N_ENABLE BIT(14) +#define MT6357_AUD_MICBIAS0_DCSW2N_DISABLE 0 +#define MT6357_AUD_MICBIAS0_DCSW2P2_EN_MASK BIT(13) +#define MT6357_AUD_MICBIAS0_DCSW2P2_ENABLE BIT(13) +#define MT6357_AUD_MICBIAS0_DCSW2P2_DISABLE 0 +#define MT6357_AUD_MICBIAS0_DCSW2P1_EN_MASK BIT(12) +#define MT6357_AUD_MICBIAS0_DCSW2P1_ENABLE BIT(12) +#define MT6357_AUD_MICBIAS0_DCSW2P1_DISABLE 0 +#define MT6357_AUD_MICBIAS0_DCSW0N_EN_MASK BIT(10) +#define MT6357_AUD_MICBIAS0_DCSW0N_ENABLE BIT(10) +#define MT6357_AUD_MICBIAS0_DCSWN_DISABLE 0 +#define MT6357_AUD_MICBIAS0_DCSW0P2_EN_MASK BIT(9) +#define MT6357_AUD_MICBIAS0_DCSW0P2_ENABLE BIT(9) +#define MT6357_AUD_MICBIAS0_DCSW0P2_DISABLE 0 +#define MT6357_AUD_MICBIAS0_DCSW0P1_EN_MASK BIT(8) +#define MT6357_AUD_MICBIAS0_DCSW0P1_ENABLE BIT(8) +#define MT6357_AUD_MICBIAS0_DCSW0P1_DISABLE 0 +#define MT6357_AUD_MICBIAS0_VREF_MASK GENMASK(6, 4) +#define MT6357_AUD_MICBIAS0_VREF_SFT 4 +#define MT6357_AUD_MICBIAS0_PWD_SFT 0 + +#define MT6357_AUD_MICBIAS0_DC_MASK (MT6357_AUD_MICBIAS0_DCSW2N_EN_MASK | \ + MT6357_AUD_MICBIAS0_DCSW2P2_EN_MASK | \ + MT6357_AUD_MICBIAS0_DCSW2P1_EN_MASK | \ + MT6357_AUD_MICBIAS0_DCSW0N_EN_MASK | \ + MT6357_AUD_MICBIAS0_DCSW0P2_EN_MASK | \ + MT6357_AUD_MICBIAS0_DCSW0P1_EN_MASK) + +#define MT6357_AUD_MICBIAS0_DC_ENABLE_ALL (MT6357_AUD_MICBIAS0_DCSW2N_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW2P2_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW2P1_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW0N_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW0P2_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW0P1_ENABLE) + +#define MT6357_AUD_MICBIAS0_DC_ENABLE_P1 (MT6357_AUD_MICBIAS0_DCSW2P1_ENABLE | \ + MT6357_AUD_MICBIAS0_DCSW0P1_ENABLE) + +#define MT6357_AUD_MICBIAS0_DC_DISABLE_ALL 0 + +/* MT6357_AUDENC_ANA_CON9 */ +#define MT6357_AUD_MICBIAS1_DCSW1P_EN_MASK BIT(8) +#define MT6357_AUD_MICBIAS1_DCSW1P_ENABLE BIT(8) +#define MT6357_AUD_MICBIAS1_DCSW1P_DISABLE 0 +#define MT6357_AUD_MICBIAS1_VREF_MASK GENMASK(6, 4) +#define MT6357_AUD_MICBIAS1_VREF_SFT 4 +#define MT6357_AUD_MICBIAS1_PWD_SFT 0 + +/* MT6357_AUDDEC_ANA_CON0 */ +#define MT6357_AUD_HPR_SC_VAUDP15_MASK BIT(13) +#define MT6357_AUD_HPR_SC_VAUDP15_DISABLE BIT(13) +#define MT6357_AUD_HPR_SC_VAUDP15_ENABLE 0 +#define MT6357_AUD_HPL_SC_VAUDP15_MASK BIT(12) +#define MT6357_AUD_HPL_SC_VAUDP15_DISABLE BIT(12) +#define MT6357_AUD_HPL_SC_VAUDP15_ENABLE 0 +#define MT6357_AUD_HPR_MUX_INPUT_VAUDP15_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUD_HPR_MUX_INPUT_VAUDP15_SFT 10 +#define MT6357_AUD_HPL_MUX_INPUT_VAUDP15_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUD_HPL_MUX_INPUT_VAUDP15_SFT 8 +#define MT6357_AUD_HPR_BIAS_VAUDP15_MASK BIT(7) +#define MT6357_AUD_HPR_BIAS_VAUDP15_ENABLE BIT(7) +#define MT6357_AUD_HPR_BIAS_VAUDP15_DISABLE 0 +#define MT6357_AUD_HPL_BIAS_VAUDP15_MASK BIT(6) +#define MT6357_AUD_HPL_BIAS_VAUDP15_ENABLE BIT(6) +#define MT6357_AUD_HPL_BIAS_VAUDP15_DISABLE 0 +#define MT6357_AUD_HPR_PWRUP_VAUDP15_MASK BIT(5) +#define MT6357_AUD_HPR_PWRUP_VAUDP15_ENABLE BIT(5) +#define MT6357_AUD_HPR_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_AUD_HPL_PWRUP_VAUDP15_MASK BIT(4) +#define MT6357_AUD_HPL_PWRUP_VAUDP15_ENABLE BIT(4) +#define MT6357_AUD_HPL_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_AUD_DACL_PWRUP_VA28_MASK BIT(3) +#define MT6357_AUD_DACL_PWRUP_VA28_ENABLE BIT(3) +#define MT6357_AUD_DACL_PWRUP_VA28_DISABLE 0 +#define MT6357_AUD_DACR_PWRUP_VA28_MASK BIT(2) +#define MT6357_AUD_DACR_PWRUP_VA28_ENABLE BIT(2) +#define MT6357_AUD_DACR_PWRUP_VA28_DISABLE 0 +#define MT6357_AUD_DACR_PWRUP_VAUDP15_MASK BIT(1) +#define MT6357_AUD_DACR_PWRUP_VAUDP15_ENABLE BIT(1) +#define MT6357_AUD_DACR_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_AUD_DACL_PWRUP_VAUDP15_MASK BIT(0) +#define MT6357_AUD_DACL_PWRUP_VAUDP15_ENABLE BIT(0) +#define MT6357_AUD_DACL_PWRUP_VAUDP15_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON1 */ +#define MT6357_HPROUT_STG_CTRL_VAUDP15_MASK GENMASK(14, 12) +#define MT6357_HPROUT_STG_CTRL_VAUDP15_SFT 12 +#define MT6357_HPLOUT_STG_CTRL_VAUDP15_MASK GENMASK(10, 8) +#define MT6357_HPLOUT_STG_CTRL_VAUDP15_SFT 8 +#define MT6357_HPLOUT_STG_CTRL_VAUDP15_MAX 7 +#define MT6357_HPR_SHORT2HPR_AUX_VAUDP15_MASK BIT(7) +#define MT6357_HPR_SHORT2HPR_AUX_VAUDP15_ENABLE BIT(7) +#define MT6357_HPR_SHORT2HPR_AUX_VAUDP15_DISABLE 0 +#define MT6357_HPL_SHORT2HPR_AUX_VAUDP15_MASK BIT(6) +#define MT6357_HPL_SHORT2HPR_AUX_VAUDP15_ENABLE BIT(6) +#define MT6357_HPL_SHORT2HPR_AUX_VAUDP15_DISABLE 0 +#define MT6357_HPR_AUX_FBRSW_VAUDP15_MASK BIT(5) +#define MT6357_HPR_AUX_FBRSW_VAUDP15_ENABLE BIT(5) +#define MT6357_HPR_AUX_FBRSW_VAUDP15_DISABLE 0 +#define MT6357_HPL_AUX_FBRSW_VAUDP15_MASK BIT(4) +#define MT6357_HPL_AUX_FBRSW_VAUDP15_ENABLE BIT(4) +#define MT6357_HPL_AUX_FBRSW_VAUDP15_DISABLE 0 +#define MT6357_HPROUT_AUX_PWRUP_VAUDP15_MASK BIT(3) +#define MT6357_HPROUT_AUX_PWRUP_VAUDP15_ENABLE BIT(3) +#define MT6357_HPROUT_AUX_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_HPLOUT_AUX_PWRUP_VAUDP15_MASK BIT(2) +#define MT6357_HPLOUT_AUX_PWRUP_VAUDP15_ENABLE BIT(2) +#define MT6357_HPLOUT_AUX_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_HPROUT_PWRUP_VAUDP15_MASK BIT(1) +#define MT6357_HPROUT_PWRUP_VAUDP15_ENABLE BIT(1) +#define MT6357_HPROUT_PWRUP_VAUDP15_DISABLE 0 +#define MT6357_HPLOUT_PWRUP_VAUDP15_MASK BIT(0) +#define MT6357_HPLOUT_PWRUP_VAUDP15_ENABLE BIT(0) +#define MT6357_HPLOUT_PWRUP_VAUDP15_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON2 */ +#define MT6357_HPP_SHORT_2VCM_VAUDP15_MASK BIT(10) +#define MT6357_HPP_SHORT_2VCM_VAUDP15_ENABLE BIT(10) +#define MT6357_HPP_SHORT_2VCM_VAUDP15_DISABLE 0 +#define MT6357_AUD_REFN_DERES_VAUDP15_MASK BIT(9) +#define MT6357_AUD_REFN_DERES_VAUDP15_ENABLE BIT(9) +#define MT6357_AUD_REFN_DERES_VAUDP15_DISABLE 0 +#define MT6357_HPROUT_STB_ENH_VAUDP15_MASK GENMASK(6, 4) +#define MT6357_HPROUT_STB_ENH_VAUDP15_OPEN 0 +#define MT6357_HPROUT_STB_ENH_VAUDP15_NOPEN_P250 BIT(4) +#define MT6357_HPROUT_STB_ENH_VAUDP15_N470_POPEN BIT(5) +#define MT6357_HPROUT_STB_ENH_VAUDP15_N470_P250 (BIT(4) | BIT(5)) +#define MT6357_HPROUT_STB_ENH_VAUDP15_NOPEN_P470 (BIT(4) | BIT(6)) +#define MT6357_HPROUT_STB_ENH_VAUDP15_N470_P470 (BIT(4) | BIT(5) | BIT(6)) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_MASK GENMASK(2, 0) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_OPEN 0 +#define MT6357_HPLOUT_STB_ENH_VAUDP15_NOPEN_P250 BIT(0) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_N470_POPEN BIT(1) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_N470_P250 (BIT(0) | BIT(1)) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_NOPEN_P470 (BIT(0) | BIT(2)) +#define MT6357_HPLOUT_STB_ENH_VAUDP15_N470_P470 (BIT(0) | BIT(1) | BIT(2)) + +/* MT6357_AUDDEC_ANA_CON3 */ +#define MT6357_AUD_HSOUT_STB_ENH_VAUDP15_MASK BIT(7) +#define MT6357_AUD_HSOUT_STB_ENH_VAUDP15_ENABLE BIT(7) +#define MT6357_AUD_HSOUT_STB_ENH_VAUDP15_DISABLE 0 +#define MT6357_AUD_HS_SC_VAUDP15_MASK BIT(4) +#define MT6357_AUD_HS_SC_VAUDP15_DISABLE BIT(4) +#define MT6357_AUD_HS_SC_VAUDP15_ENABLE 0 +#define MT6357_AUD_HS_MUX_INPUT_VAUDP15_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUD_HS_MUX_INPUT_VAUDP15_SFT 2 +#define MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_MASK BIT(1) +#define MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_ENABLE BIT(1) +#define MT6357_AUD_HS_PWRUP_BIAS_VAUDP15_DISABLE 0 +#define MT6357_AUD_HS_PWRUP_VAUDP15_MASK BIT(0) +#define MT6357_AUD_HS_PWRUP_VAUDP15_ENABLE BIT(0) +#define MT6357_AUD_HS_PWRUP_VAUDP15_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON4 */ +#define MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_MASK BIT(8) +#define MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_ENABLE BIT(8) +#define MT6357_AUD_LOLOUT_STB_ENH_VAUDP15_DISABLE 0 +#define MT6357_AUD_LOL_SC_VAUDP15_MASK BIT(4) +#define MT6357_AUD_LOL_SC_VAUDP15_DISABLE BIT(4) +#define MT6357_AUD_LOL_SC_VAUDP15_ENABLE 0 +#define MT6357_AUD_LOL_MUX_INPUT_VAUDP15_MASK_NOSFT GENMASK(1, 0) +#define MT6357_AUD_LOL_MUX_INPUT_VAUDP15_SFT 2 +#define MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_MASK BIT(1) +#define MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_ENABLE BIT(1) +#define MT6357_AUD_LOL_PWRUP_BIAS_VAUDP15_DISABLE 0 +#define MT6357_AUD_LOL_PWRUP_VAUDP15_MASK BIT(0) +#define MT6357_AUD_LOL_PWRUP_VAUDP15_ENABLE BIT(0) +#define MT6357_AUD_LOL_PWRUP_VAUDP15_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON6 */ +#define MT6357_HP_AUX_LOOP_GAIN_MASK GENMASK(15, 12) +#define MT6357_HP_AUX_LOOP_GAIN_SFT 12 +#define MT6357_HP_AUX_LOOP_GAIN_MAX 0x0f +#define MT6357_HPR_AUX_CMFB_LOOP_MASK BIT(11) +#define MT6357_HPR_AUX_CMFB_LOOP_ENABLE BIT(11) +#define MT6357_HPR_AUX_CMFB_LOOP_DISABLE 0 +#define MT6357_HPL_AUX_CMFB_LOOP_MASK BIT(10) +#define MT6357_HPL_AUX_CMFB_LOOP_ENABLE BIT(10) +#define MT6357_HPL_AUX_CMFB_LOOP_DISABLE 0 +#define MT6357_HPRL_MAIN_CMFB_LOOP_MASK BIT(9) +#define MT6357_HPRL_MAIN_CMFB_LOOP_ENABLE BIT(9) +#define MT6357_HPRL_MAIN_CMFB_LOOP_DISABLE 0 +#define MT6357_HP_CMFB_RST_MASK BIT(7) +#define MT6357_HP_CMFB_RST_NORMAL BIT(7) +#define MT6357_HP_CMFB_RST_RESET 0 +#define MT6357_DAC_LOW_NOISE_MODE_MASK BIT(0) +#define MT6357_DAC_LOW_NOISE_MODE_ENABLE BIT(0) +#define MT6357_DAC_LOW_NOISE_MODE_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON7 */ +#define MT6357_HP_IVBUF_DEGAIN_SFT 2 +#define MT6357_HP_IVBUF_DEGAIN_MAX 1 + +/* MT6357_AUDDEC_ANA_CON10 */ +#define MT6357_AUD_IBIAS_PWRDN_VAUDP15_MASK BIT(8) +#define MT6357_AUD_IBIAS_PWRDN_VAUDP15_DISABLE BIT(8) +#define MT6357_AUD_IBIAS_PWRDN_VAUDP15_ENABLE 0 + +/* MT6357_AUDDEC_ANA_CON11 */ +#define MT6357_RSTB_ENCODER_VA28_MASK BIT(5) +#define MT6357_RSTB_ENCODER_VA28_ENABLE BIT(5) +#define MT6357_RSTB_ENCODER_VA28_DISABLE 0 +#define MT6357_AUDGLB_PWRDN_VA28_SFT 4 +#define MT6357_RSTB_DECODER_VA28_MASK BIT(0) +#define MT6357_RSTB_DECODER_VA28_ENABLE BIT(0) +#define MT6357_RSTB_DECODER_VA28_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON12 */ +#define MT6357_VA28REFGEN_EN_VA28_MASK BIT(13) +#define MT6357_VA28REFGEN_EN_VA28_ENABLE BIT(13) +#define MT6357_VA28REFGEN_EN_VA28_DISABLE 0 +#define MT6357_VA33REFGEN_EN_VA18_MASK BIT(12) +#define MT6357_VA33REFGEN_EN_VA18_ENABLE BIT(12) +#define MT6357_VA33REFGEN_EN_VA18_DISABLE 0 +#define MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_MASK BIT(10) +#define MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_ENABLE BIT(10) +#define MT6357_LCLDO_ENC_REMOTE_SENSE_VA28_DISABLE 0 +#define MT6357_LCLDO_ENC_EN_VA28_MASK BIT(8) +#define MT6357_LCLDO_ENC_EN_VA28_ENABLE BIT(8) +#define MT6357_LCLDO_ENC_EN_VA28_DISABLE 0 +#define MT6357_LCLDO_REMOTE_SENSE_VA18_MASK BIT(6) +#define MT6357_LCLDO_REMOTE_SENSE_VA18_ENABLE BIT(6) +#define MT6357_LCLDO_REMOTE_SENSE_VA18_DISABLE 0 +#define MT6357_LCLDO_EN_VA18_MASK BIT(4) +#define MT6357_LCLDO_EN_VA18_ENABLE BIT(4) +#define MT6357_LCLDO_EN_VA18_DISABLE 0 +#define MT6357_HCLDO_REMOTE_SENSE_VA18_MASK BIT(2) +#define MT6357_HCLDO_REMOTE_SENSE_VA18_ENABLE BIT(2) +#define MT6357_HCLDO_REMOTE_SENSE_VA18_DISABLE 0 +#define MT6357_HCLDO_EN_VA18_MASK BIT(0) +#define MT6357_HCLDO_EN_VA18_ENABLE BIT(0) +#define MT6357_HCLDO_EN_VA18_DISABLE 0 + +/* MT6357_AUDDEC_ANA_CON13 */ +#define MT6357_NVREG_EN_VAUDP15_MASK BIT(0) +#define MT6357_NVREG_EN_VAUDP15_ENABLE BIT(0) +#define MT6357_NVREG_EN_VAUDP15_DISABLE 0 + +/* MT6357_AUDDEC_ELR_0 */ +#define MT6357_AUD_HP_TRIM_EN_VAUDP15_MASK BIT(12) +#define MT6357_AUD_HP_TRIM_EN_VAUDP15_ENABLE BIT(12) +#define MT6357_AUD_HP_TRIM_EN_VAUDP15_DISABLE 0 + +/* MT6357_ZCD_CON1 */ +#define MT6357_AUD_LOL_GAIN_MASK GENMASK(4, 0) +#define MT6357_AUD_LOL_GAIN_SFT 0 +#define MT6357_AUD_LOR_GAIN_MASK GENMASK(11, 7) +#define MT6357_AUD_LOR_GAIN_SFT 7 +#define MT6357_AUD_LO_GAIN_MAX 0x12 + +/* MT6357_ZCD_CON2 */ +#define MT6357_AUD_HPL_GAIN_MASK GENMASK(4, 0) +#define MT6357_AUD_HPL_GAIN_SFT 0 +#define MT6357_AUD_HPR_GAIN_MASK GENMASK(11, 7) +#define MT6357_AUD_HPR_GAIN_SFT 7 +#define MT6357_AUD_HP_GAIN_MAX 0x12 + +/* MT6357_ZCD_CON3 */ +#define MT6357_AUD_HS_GAIN_MASK GENMASK(4, 0) +#define MT6357_AUD_HS_GAIN_SFT 0 +#define MT6357_AUD_HS_GAIN_MAX 0x12 + +/* Registers list */ +/* gpio direction */ +#define MT6357_GPIO_DIR0 0x0088 +/* mosi */ +#define MT6357_GPIO_MODE2 0x00B6 +#define MT6357_GPIO_MODE2_SET 0x00B8 +#define MT6357_GPIO_MODE2_CLR 0x00BA +/* miso */ +#define MT6357_GPIO_MODE3 0x00BC +#define MT6357_GPIO_MODE3_SET 0x00BE +#define MT6357_GPIO_MODE3_CLR 0x00C0 + +#define MT6357_DCXO_CW14 0x07AC + +#define MT6357_AUD_TOP_CKPDN_CON0 0x208C +#define MT6357_AUDNCP_CLKDIV_CON0 0x20B4 +#define MT6357_AUDNCP_CLKDIV_CON1 0x20B6 +#define MT6357_AUDNCP_CLKDIV_CON2 0x20B8 +#define MT6357_AUDNCP_CLKDIV_CON3 0x20BA +#define MT6357_AUDNCP_CLKDIV_CON4 0x20BC +#define MT6357_AFE_UL_DL_CON0 0x2108 +#define MT6357_AFE_DL_SRC2_CON0_L 0x210A +#define MT6357_AFE_UL_SRC_CON0_H 0x210C +#define MT6357_AFE_UL_SRC_CON0_L 0x210E +#define MT6357_AFE_TOP_CON0 0x2110 +#define MT6357_AUDIO_TOP_CON0 0x2112 +#define MT6357_AFUNC_AUD_CON0 0x2116 +#define MT6357_AFUNC_AUD_CON2 0x211A +#define MT6357_AFE_ADDA_MTKAIF_CFG0 0x2134 +#define MT6357_AFE_SGEN_CFG0 0x2140 +#define MT6357_AFE_DCCLK_CFG0 0x2146 +#define MT6357_AFE_DCCLK_CFG1 0x2148 +#define MT6357_AFE_AUD_PAD_TOP 0x214C +#define MT6357_AUDENC_ANA_CON0 0x2188 +#define MT6357_AUDENC_ANA_CON1 0x218A +#define MT6357_AUDENC_ANA_CON6 0x2194 +#define MT6357_AUDENC_ANA_CON7 0x2196 +#define MT6357_AUDENC_ANA_CON8 0x2198 +#define MT6357_AUDENC_ANA_CON9 0x219A +#define MT6357_AUDDEC_ANA_CON0 0x2208 +#define MT6357_AUDDEC_ANA_CON1 0x220A +#define MT6357_AUDDEC_ANA_CON2 0x220C +#define MT6357_AUDDEC_ANA_CON3 0x220E +#define MT6357_AUDDEC_ANA_CON4 0x2210 +#define MT6357_AUDDEC_ANA_CON6 0x2214 +#define MT6357_AUDDEC_ANA_CON7 0x2216 +#define MT6357_AUDDEC_ANA_CON10 0x221C +#define MT6357_AUDDEC_ANA_CON11 0x221E +#define MT6357_AUDDEC_ANA_CON12 0x2220 +#define MT6357_AUDDEC_ANA_CON13 0x2222 +#define MT6357_AUDDEC_ELR_0 0x2226 +#define MT6357_ZCD_CON1 0x228A +#define MT6357_ZCD_CON2 0x228C +#define MT6357_ZCD_CON3 0x228E + +enum { + DL_GAIN_8DB = 0, + DL_GAIN_0DB = 8, + DL_GAIN_N_1DB = 9, + DL_GAIN_N_10DB = 18, + DL_GAIN_N_12DB = 20, + DL_GAIN_N_40DB = 0x1f, +}; + +enum { + UL_GAIN_0DB = 0, + UL_GAIN_6DB, + UL_GAIN_12DB, + UL_GAIN_18DB, + UL_GAIN_24DB, +}; + +#define MT6357_DL_GAIN_N_40DB_REG (DL_GAIN_N_40DB << 7 | DL_GAIN_N_40DB) +#define MT6357_DL_GAIN_REG_LEFT_MASK 0x001f +#define MT6357_DL_GAIN_REG_LEFT_SHIFT 0 +#define MT6357_DL_GAIN_REG_RIGHT_MASK 0x0f80 +#define MT6357_DL_GAIN_REG_RIGHT_SHIFT 7 +#define MT6357_DL_GAIN_REG_MASK 0x0f9f + +#define MT6357_SND_SOC_ADV_MT_FMTS (\ + SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S16_BE |\ + SNDRV_PCM_FMTBIT_U16_LE |\ + SNDRV_PCM_FMTBIT_U16_BE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S24_BE |\ + SNDRV_PCM_FMTBIT_U24_LE |\ + SNDRV_PCM_FMTBIT_U24_BE |\ + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_S32_BE |\ + SNDRV_PCM_FMTBIT_U32_LE |\ + SNDRV_PCM_FMTBIT_U32_BE) + +#define MT6357_SOC_HIGH_USE_RATE (\ + SNDRV_PCM_RATE_CONTINUOUS |\ + SNDRV_PCM_RATE_8000_192000) + +/* codec private structure */ +struct mt6357_priv { + struct device *dev; + struct regmap *regmap; + bool pull_down_needed; + int hp_channel_number; +}; +#endif From 5bbfdad8cf8d4b2c3189fee8e6300b38f65f8d72 Mon Sep 17 00:00:00 2001 From: Alexandre Mergnat Date: Thu, 5 Sep 2024 11:06:59 +0200 Subject: [PATCH 326/410] ASoC: mediatek: Add MT8365 support - Add specific config to enable: - MT8365 sound support - MT6357 audio codec support - Add the mt8365 directory and all drivers under it. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Alexandre Mergnat Link: https://patch.msgid.link/20240226-audio-i350-v8-2-e80a57d026ce@baylibre.com Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 20 ++++++++++++++++++++ sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8365/Makefile | 15 +++++++++++++++ 3 files changed, 36 insertions(+) create mode 100644 sound/soc/mediatek/mt8365/Makefile diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 5a8476e1ecca..e6f7a5a49794 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -298,3 +298,23 @@ config SND_SOC_MT8195_MT6359 boards with the MT6359 and other I2S audio codecs. Select Y if you have such device. If unsure select "N". + +config SND_SOC_MT8365 + tristate "ASoC support for MediaTek MT8365 chip" + depends on ARCH_MEDIATEK + select SND_SOC_MEDIATEK + help + This adds ASoC platform driver support for MediaTek MT8365 chip + that can be used with other codecs. + Select Y if you have such device. + If unsure select "N". + +config SND_SOC_MT8365_MT6357 + tristate "ASoC Audio driver for MT8365 with MT6357 codec" + depends on SND_SOC_MT8365 && MTK_PMIC_WRAP + select SND_SOC_MT6357 + help + This adds support for ASoC machine driver for MediaTek MT8365 + boards with the MT6357 PMIC codec. + Select Y if you have such device. + If unsure select "N". diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile index 3938e7f75c2e..4b55434f2168 100644 --- a/sound/soc/mediatek/Makefile +++ b/sound/soc/mediatek/Makefile @@ -9,3 +9,4 @@ obj-$(CONFIG_SND_SOC_MT8186) += mt8186/ obj-$(CONFIG_SND_SOC_MT8188) += mt8188/ obj-$(CONFIG_SND_SOC_MT8192) += mt8192/ obj-$(CONFIG_SND_SOC_MT8195) += mt8195/ +obj-$(CONFIG_SND_SOC_MT8365) += mt8365/ diff --git a/sound/soc/mediatek/mt8365/Makefile b/sound/soc/mediatek/mt8365/Makefile new file mode 100644 index 000000000000..52ba45a8498a --- /dev/null +++ b/sound/soc/mediatek/mt8365/Makefile @@ -0,0 +1,15 @@ +# SPDX-License-Identifier: GPL-2.0 + +# MTK Platform driver +snd-soc-mt8365-pcm-objs := \ + mt8365-afe-clk.o \ + mt8365-afe-pcm.o \ + mt8365-dai-adda.o \ + mt8365-dai-dmic.o \ + mt8365-dai-i2s.o \ + mt8365-dai-pcm.o + +obj-$(CONFIG_SND_SOC_MT8365) += snd-soc-mt8365-pcm.o + +# Machine driver +obj-$(CONFIG_SND_SOC_MT8365_MT6357) += mt8365-mt6357.o From 03667e3d4fbcaf6228fd642464467366f0b693de Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Thu, 5 Sep 2024 12:56:33 +0300 Subject: [PATCH 327/410] ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance The period size represents the size of the DMA descriptor. To ensure all DMA descriptors start from a well-aligned address, the period size must be divided by (sample size * maxburst), not just by maxburst. This adjustment allows for computing a higher maxburst value, thereby increasing the performance of the DMA transfer. Previously, snd_pcm_lib_period_bytes() returned 0 because the runtime HW parameters are computed after the hw_params() callbacks are used. To address this, we now use params_*() functions to compute the period size accurately. This change optimizes the DMA transfer performance by ensuring proper alignment and efficient use of maxburst values. [andrei.simion@microchip.com: Reword commit message and commit title. Add macros with values for maximum DMA chunk size allowed. Add DMA_BURST_ALIGNED preprocessor function to check the alignment of the DMA burst] Signed-off-by: Codrin Ciubotariu Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20240905095633.113784-1-andrei.simion@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-i2s-mcc.c | 38 +++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 193dd7acceb0..35a56b266b8d 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -221,6 +221,15 @@ #define MCHP_I2SMCC_MAX_CHANNELS 8 #define MCHP_I2MCC_TDM_SLOT_WIDTH 32 +/* + * ---- DMA chunk size allowed ---- + */ +#define MCHP_I2SMCC_DMA_8_WORD_CHUNK 8 +#define MCHP_I2SMCC_DMA_4_WORD_CHUNK 4 +#define MCHP_I2SMCC_DMA_2_WORD_CHUNK 2 +#define MCHP_I2SMCC_DMA_1_WORD_CHUNK 1 +#define DMA_BURST_ALIGNED(_p, _s, _w) !(_p % (_s * _w)) + static const struct regmap_config mchp_i2s_mcc_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -504,12 +513,30 @@ static int mchp_i2s_mcc_is_running(struct mchp_i2s_mcc_dev *dev) return !!(sr & (MCHP_I2SMCC_SR_TXEN | MCHP_I2SMCC_SR_RXEN)); } +static inline int mchp_i2s_mcc_period_to_maxburst(int period_size, int sample_size) +{ + int p_size = period_size; + int s_size = sample_size; + + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_8_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_8_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_4_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_4_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_2_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_2_WORD_CHUNK; + return MCHP_I2SMCC_DMA_1_WORD_CHUNK; +} + static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned long rate = 0; struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + int sample_bytes = params_physical_width(params) / 8; + int period_bytes = params_period_size(params) * + params_channels(params) * sample_bytes; + int maxburst; u32 mra = 0; u32 mrb = 0; unsigned int channels = params_channels(params); @@ -519,9 +546,9 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, int ret; bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u period_bytes=%d\n", __func__, params_rate(params), params_format(params), - params_width(params), params_channels(params)); + params_width(params), params_channels(params), period_bytes); switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -630,11 +657,12 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, * We must have the same burst size configured * in the DMA transfer and in out IP */ - mrb |= MCHP_I2SMCC_MRB_DMACHUNK(channels); + maxburst = mchp_i2s_mcc_period_to_maxburst(period_bytes, sample_bytes); + mrb |= MCHP_I2SMCC_MRB_DMACHUNK(maxburst); if (is_playback) - dev->playback.maxburst = 1 << (fls(channels) - 1); + dev->playback.maxburst = maxburst; else - dev->capture.maxburst = 1 << (fls(channels) - 1); + dev->capture.maxburst = maxburst; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: From 54694840eff5e95d1b0ec0c916db2d889529c5d7 Mon Sep 17 00:00:00 2001 From: Chen Ni Date: Thu, 5 Sep 2024 11:21:48 +0800 Subject: [PATCH 328/410] ASoC: topology-test: Convert comma to semicolon Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni Link: https://patch.msgid.link/20240905032148.1929393-1-nichen@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/soc-topology-test.c | 132 +++++++++++++++++----------------- 1 file changed, 66 insertions(+), 66 deletions(-) diff --git a/sound/soc/soc-topology-test.c b/sound/soc/soc-topology-test.c index d62a02ec5896..a2b08568f4e8 100644 --- a/sound/soc/soc-topology-test.c +++ b/sound/soc/soc-topology-test.c @@ -244,12 +244,12 @@ static void snd_soc_tplg_test_load_with_null_comp(struct kunit *test) kunit_comp->kunit = test; kunit_comp->expect = -EINVAL; /* expect failure */ - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -286,12 +286,12 @@ static void snd_soc_tplg_test_load_with_null_ops(struct kunit *test) kunit_comp->kunit = test; kunit_comp->expect = 0; /* expect success */ - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -348,12 +348,12 @@ static void snd_soc_tplg_test_load_with_null_fw(struct kunit *test) kunit_comp->kunit = test; kunit_comp->expect = -EINVAL; /* expect failure */ - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -396,12 +396,12 @@ static void snd_soc_tplg_test_load_empty_tplg(struct kunit *test) kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -451,12 +451,12 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_magic(struct kunit *test) kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -506,12 +506,12 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_abi(struct kunit *test) kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -561,12 +561,12 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_size(struct kunit *test) kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -617,12 +617,12 @@ static void snd_soc_tplg_test_load_empty_tplg_bad_payload_size(struct kunit *tes kunit_comp->fw.data = (u8 *)data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -665,12 +665,12 @@ static void snd_soc_tplg_test_load_pcm_tplg(struct kunit *test) kunit_comp->fw.data = data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -715,12 +715,12 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_comp(struct kunit *test) kunit_comp->fw.data = data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_register_card(&kunit_comp->card); @@ -767,12 +767,12 @@ static void snd_soc_tplg_test_load_pcm_tplg_reload_card(struct kunit *test) kunit_comp->fw.data = data; kunit_comp->fw.size = size; - kunit_comp->card.dev = test_dev, - kunit_comp->card.name = "kunit-card", - kunit_comp->card.owner = THIS_MODULE, - kunit_comp->card.dai_link = kunit_dai_links, - kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links), - kunit_comp->card.fully_routed = true, + kunit_comp->card.dev = test_dev; + kunit_comp->card.name = "kunit-card"; + kunit_comp->card.owner = THIS_MODULE; + kunit_comp->card.dai_link = kunit_dai_links; + kunit_comp->card.num_links = ARRAY_SIZE(kunit_dai_links); + kunit_comp->card.fully_routed = true; /* run test */ ret = snd_soc_component_initialize(&kunit_comp->comp, &test_component, test_dev); From 813751eaec93bfeb6236aaed99607a44c01b3110 Mon Sep 17 00:00:00 2001 From: Chen Ni Date: Thu, 5 Sep 2024 10:20:17 +0800 Subject: [PATCH 329/410] ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni Link: https://patch.msgid.link/20240905022017.1642550-1-nichen@iscas.ac.cn Signed-off-by: Mark Brown --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 225867bb3310..8e104874d58c 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -217,14 +217,14 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; card = &ctx->card; - card->name = "hda-dsp", - card->owner = THIS_MODULE, - card->dai_link = skl_hda_be_dai_links, - card->dapm_widgets = skl_hda_widgets, - card->dapm_routes = skl_hda_map, - card->add_dai_link = skl_hda_add_dai_link, - card->fully_routed = true, - card->late_probe = skl_hda_card_late_probe, + card->name = "hda-dsp"; + card->owner = THIS_MODULE; + card->dai_link = skl_hda_be_dai_links; + card->dapm_widgets = skl_hda_widgets; + card->dapm_routes = skl_hda_map; + card->add_dai_link = skl_hda_add_dai_link; + card->fully_routed = true; + card->late_probe = skl_hda_card_late_probe; snd_soc_card_set_drvdata(card, ctx); From 090624b7dc8389a516df3adb447a25dc0723adfe Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:52 +0200 Subject: [PATCH 330/410] ALSA: pcm: add more sample rate definitions This adds a sample rate definition for 12kHz, 24kHz and 128kHz. Admittedly, just a few drivers are currently using these sample rates but there is enough of a recurrence to justify adding a definition for them and remove some custom rate constraint code while at it. The new definitions are not added to the interval definitions, such as SNDRV_PCM_RATE_8000_44100, because it would silently add new supported rates to drivers that may or may not support them. For sure the drivers have not been tested for these new rates so it is better to leave them out of interval definitions. That being said, the added rates are multiples of well know rates families, it is very likely that a lot of devices out there actually supports them. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-1-8371948d3921@baylibre.com --- include/sound/pcm.h | 31 +++++++++++++++++-------------- sound/core/pcm_native.c | 6 +++--- 2 files changed, 20 insertions(+), 17 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 732121b934fd..c993350975a9 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -109,20 +109,23 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_5512 (1U<<0) /* 5512Hz */ #define SNDRV_PCM_RATE_8000 (1U<<1) /* 8000Hz */ #define SNDRV_PCM_RATE_11025 (1U<<2) /* 11025Hz */ -#define SNDRV_PCM_RATE_16000 (1U<<3) /* 16000Hz */ -#define SNDRV_PCM_RATE_22050 (1U<<4) /* 22050Hz */ -#define SNDRV_PCM_RATE_32000 (1U<<5) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1U<<6) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1U<<7) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1U<<8) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1U<<9) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1U<<10) /* 96000Hz */ -#define SNDRV_PCM_RATE_176400 (1U<<11) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1U<<12) /* 192000Hz */ -#define SNDRV_PCM_RATE_352800 (1U<<13) /* 352800Hz */ -#define SNDRV_PCM_RATE_384000 (1U<<14) /* 384000Hz */ -#define SNDRV_PCM_RATE_705600 (1U<<15) /* 705600Hz */ -#define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */ +#define SNDRV_PCM_RATE_12000 (1U<<3) /* 12000Hz */ +#define SNDRV_PCM_RATE_16000 (1U<<4) /* 16000Hz */ +#define SNDRV_PCM_RATE_22050 (1U<<5) /* 22050Hz */ +#define SNDRV_PCM_RATE_24000 (1U<<6) /* 24000Hz */ +#define SNDRV_PCM_RATE_32000 (1U<<7) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1U<<8) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1U<<9) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1U<<10) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1U<<11) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1U<<12) /* 96000Hz */ +#define SNDRV_PCM_RATE_128000 (1U<<13) /* 128000Hz */ +#define SNDRV_PCM_RATE_176400 (1U<<14) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1U<<15) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1U<<16) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1U<<17) /* 384000Hz */ +#define SNDRV_PCM_RATE_705600 (1U<<18) /* 705600Hz */ +#define SNDRV_PCM_RATE_768000 (1U<<19) /* 768000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuous rates */ diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 44381514f695..7461a727615c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2418,13 +2418,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_768000 != 1 << 19 #error "Change this table" #endif static const unsigned int rates[] = { - 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000, 705600, 768000 + 5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, + 88200, 96000, 128000, 176400, 192000, 352800, 384000, 705600, 768000, }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { From 91dd20d855d64e5e18a821514189d68058c5bcf1 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:53 +0200 Subject: [PATCH 331/410] ALSA: cmipci: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 128kHz. This rate is now available through SNDRV_PCM_RATE_128000. Use it and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-2-8371948d3921@baylibre.com --- sound/pci/cmipci.c | 32 +++++++++----------------------- 1 file changed, 9 insertions(+), 23 deletions(-) diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 36014501f7ed..e3cac73517d6 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -1570,14 +1570,6 @@ static const struct snd_pcm_hardware snd_cmipci_capture_spdif = .fifo_size = 0, }; -static const unsigned int rate_constraints[] = { 5512, 8000, 11025, 16000, 22050, - 32000, 44100, 48000, 88200, 96000, 128000 }; -static const struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rate_constraints), - .list = rate_constraints, - .mask = 0, -}; - /* * check device open/close */ @@ -1649,11 +1641,9 @@ static int snd_cmipci_playback_open(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - if (err < 0) - return err; - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_128000; runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); @@ -1675,11 +1665,9 @@ static int snd_cmipci_capture_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = 41000; runtime->hw.rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - if (err < 0) - return err; - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_128000; runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); @@ -1715,11 +1703,9 @@ static int snd_cmipci_playback2_open(struct snd_pcm_substream *substream) SNDRV_PCM_RATE_96000; runtime->hw.rate_max = 96000; } else if (cm->chip_version == 55) { - err = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - if (err < 0) - return err; - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; + runtime->hw.rates |= SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_128000; runtime->hw.rate_max = 128000; } snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 0, 0x10000); From 1f40410623b76141f0d80f8fb946ba3ff6910d4d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:54 +0200 Subject: [PATCH 332/410] ALSA: emu10k1: drop SNDRV_PCM_RATE_KNOT The custom rate constraint lists were necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rules. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-3-8371948d3921@baylibre.com --- sound/pci/emu10k1/emupcm.c | 31 +++++-------------------------- 1 file changed, 5 insertions(+), 26 deletions(-) diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index 7f4c1b38d6ec..1bf6e3d652f8 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -147,16 +147,6 @@ static const struct snd_pcm_hw_constraint_list hw_constraints_capture_buffer_siz .mask = 0 }; -static const unsigned int capture_rates[8] = { - 8000, 11025, 16000, 22050, 24000, 32000, 44100, 48000 -}; - -static const struct snd_pcm_hw_constraint_list hw_constraints_capture_rates = { - .count = 8, - .list = capture_rates, - .mask = 0 -}; - static unsigned int snd_emu10k1_capture_rate_reg(unsigned int rate) { switch (rate) { @@ -174,16 +164,6 @@ static unsigned int snd_emu10k1_capture_rate_reg(unsigned int rate) } } -static const unsigned int audigy_capture_rates[9] = { - 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 -}; - -static const struct snd_pcm_hw_constraint_list hw_constraints_audigy_capture_rates = { - .count = 9, - .list = audigy_capture_rates, - .mask = 0 -}; - static unsigned int snd_emu10k1_audigy_capture_rate_reg(unsigned int rate) { switch (rate) { @@ -207,17 +187,16 @@ static void snd_emu10k1_constrain_capture_rates(struct snd_emu10k1 *emu, { if (emu->card_capabilities->emu_model && emu->emu1010.word_clock == 44100) { - // This also sets the rate constraint by deleting SNDRV_PCM_RATE_KNOT runtime->hw.rates = SNDRV_PCM_RATE_11025 | \ SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100; runtime->hw.rate_min = 11025; runtime->hw.rate_max = 44100; - return; + } else if (emu->audigy) { + runtime->hw.rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_12000 | + SNDRV_PCM_RATE_24000; } - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - emu->audigy ? &hw_constraints_audigy_capture_rates : - &hw_constraints_capture_rates); } static void snd_emu1010_constrain_efx_rate(struct snd_emu10k1 *emu, @@ -1053,7 +1032,7 @@ static const struct snd_pcm_hardware snd_emu10k1_capture = SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_24000, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, From 3cc1e94dbc1e65b77a22a8e4e5ab79319ab5ed98 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:55 +0200 Subject: [PATCH 333/410] ALSA: hdsp: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 128kHz. This rate is now available through SNDRV_PCM_RATE_128000. Use it and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-4-8371948d3921@baylibre.com --- sound/pci/rme9652/hdsp.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 713ca262a0e9..1c504a591948 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4301,14 +4301,6 @@ static const struct snd_pcm_hw_constraint_list hdsp_hw_constraints_period_sizes .mask = 0 }; -static const unsigned int hdsp_9632_sample_rates[] = { 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000 }; - -static const struct snd_pcm_hw_constraint_list hdsp_hw_constraints_9632_sample_rates = { - .count = ARRAY_SIZE(hdsp_9632_sample_rates), - .list = hdsp_9632_sample_rates, - .mask = 0 -}; - static int snd_hdsp_hw_rule_in_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -4499,8 +4491,9 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = runtime->hw.rate_max = hdsp->system_sample_rate; } else if (hdsp->io_type == H9632) { runtime->hw.rate_max = 192000; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); + runtime->hw.rates |= (SNDRV_PCM_RATE_128000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000); } if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; @@ -4575,8 +4568,9 @@ static int snd_hdsp_capture_open(struct snd_pcm_substream *substream) runtime->hw.channels_min = hdsp->qs_in_channels; runtime->hw.channels_max = hdsp->ss_in_channels; runtime->hw.rate_max = 192000; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hdsp_hw_constraints_9632_sample_rates); + runtime->hw.rates |= (SNDRV_PCM_RATE_128000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000); } snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_in_channels, hdsp, From 151d82f914e85cc938d76315818d9d250a01ee90 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:56 +0200 Subject: [PATCH 334/410] ALSA: hdspm: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 128kHz. This rate is now available through SNDRV_PCM_RATE_128000. Use it and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-5-8371948d3921@baylibre.com --- sound/pci/rme9652/hdspm.c | 16 +--------------- 1 file changed, 1 insertion(+), 15 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index c3f930a8f78d..dad974378e00 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6032,18 +6032,6 @@ static int snd_hdspm_hw_rule_out_channels(struct snd_pcm_hw_params *params, return snd_interval_list(c, 3, list, 0); } - -static const unsigned int hdspm_aes32_sample_rates[] = { - 32000, 44100, 48000, 64000, 88200, 96000, 128000, 176400, 192000 -}; - -static const struct snd_pcm_hw_constraint_list -hdspm_hw_constraints_aes32_sample_rates = { - .count = ARRAY_SIZE(hdspm_aes32_sample_rates), - .list = hdspm_aes32_sample_rates, - .mask = 0 -}; - static int snd_hdspm_open(struct snd_pcm_substream *substream) { struct hdspm *hdspm = snd_pcm_substream_chip(substream); @@ -6096,9 +6084,7 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream) } if (AES32 == hdspm->io_type) { - runtime->hw.rates |= SNDRV_PCM_RATE_KNOT; - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - &hdspm_hw_constraints_aes32_sample_rates); + runtime->hw.rates |= SNDRV_PCM_RATE_128000; } else { snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, (playback ? From 7067e8142c4b37b4b78c6ba8eb23bf9ce4a2c5ec Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:57 +0200 Subject: [PATCH 335/410] ASoC: cs35l36: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-6-8371948d3921@baylibre.com --- sound/soc/codecs/cs35l36.c | 34 ++++++++++++---------------------- 1 file changed, 12 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index cbea79bd8980..b49c6905e872 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -949,32 +949,22 @@ static const struct cs35l36_pll_config *cs35l36_get_clk_config( return NULL; } -static const unsigned int cs35l36_src_rates[] = { - 8000, 12000, 11025, 16000, 22050, 24000, 32000, - 44100, 48000, 88200, 96000, 176400, 192000, 384000 -}; - -static const struct snd_pcm_hw_constraint_list cs35l36_constraints = { - .count = ARRAY_SIZE(cs35l36_src_rates), - .list = cs35l36_src_rates, -}; - -static int cs35l36_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &cs35l36_constraints); - - return 0; -} - static const struct snd_soc_dai_ops cs35l36_ops = { - .startup = cs35l36_pcm_startup, .set_fmt = cs35l36_set_dai_fmt, .hw_params = cs35l36_pcm_hw_params, .set_sysclk = cs35l36_dai_set_sysclk, }; +#define CS35L36_RATES ( \ + SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000 | \ + SNDRV_PCM_RATE_384000) + static struct snd_soc_dai_driver cs35l36_dai[] = { { .name = "cs35l36-pcm", @@ -983,14 +973,14 @@ static struct snd_soc_dai_driver cs35l36_dai[] = { .stream_name = "AMP Playback", .channels_min = 1, .channels_max = 8, - .rates = SNDRV_PCM_RATE_KNOT, + .rates = CS35L36_RATES, .formats = CS35L36_RX_FORMATS, }, .capture = { .stream_name = "AMP Capture", .channels_min = 1, .channels_max = 8, - .rates = SNDRV_PCM_RATE_KNOT, + .rates = CS35L36_RATES, .formats = CS35L36_TX_FORMATS, }, .ops = &cs35l36_ops, From 79acb4c046ce2ef63b77ea055e1dc559d813aa30 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:58 +0200 Subject: [PATCH 336/410] ASoC: cs35l41: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-7-8371948d3921@baylibre.com --- sound/soc/codecs/cs35l41.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index 1688c2c688f0..07a5cab35fe1 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -808,26 +808,6 @@ static int cs35l41_get_clk_config(int freq) return -EINVAL; } -static const unsigned int cs35l41_src_rates[] = { - 8000, 12000, 11025, 16000, 22050, 24000, 32000, - 44100, 48000, 88200, 96000, 176400, 192000 -}; - -static const struct snd_pcm_hw_constraint_list cs35l41_constraints = { - .count = ARRAY_SIZE(cs35l41_src_rates), - .list = cs35l41_src_rates, -}; - -static int cs35l41_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - if (substream->runtime) - return snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &cs35l41_constraints); - return 0; -} - static int cs35l41_component_set_sysclk(struct snd_soc_component *component, int clk_id, int source, unsigned int freq, int dir) @@ -974,13 +954,21 @@ static void cs35l41_component_remove(struct snd_soc_component *component) } static const struct snd_soc_dai_ops cs35l41_ops = { - .startup = cs35l41_pcm_startup, .set_fmt = cs35l41_set_dai_fmt, .hw_params = cs35l41_pcm_hw_params, .set_sysclk = cs35l41_dai_set_sysclk, .set_channel_map = cs35l41_set_channel_map, }; +#define CS35L41_RATES ( \ + SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver cs35l41_dai[] = { { .name = "cs35l41-pcm", @@ -989,14 +977,14 @@ static struct snd_soc_dai_driver cs35l41_dai[] = { .stream_name = "AMP Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_KNOT, + .rates = CS35L41_RATES, .formats = CS35L41_RX_FORMATS, }, .capture = { .stream_name = "AMP Capture", .channels_min = 1, .channels_max = 4, - .rates = SNDRV_PCM_RATE_KNOT, + .rates = CS35L41_RATES, .formats = CS35L41_TX_FORMATS, }, .ops = &cs35l41_ops, From eab3464be764e1f42e5dedabef633c6e3615c7e5 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:12:59 +0200 Subject: [PATCH 337/410] ASoC: cs53l30: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-8-8371948d3921@baylibre.com --- sound/soc/codecs/cs53l30.c | 24 +++--------------------- 1 file changed, 3 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index bcbaf28a0b2d..28f4be37dec1 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -739,24 +739,6 @@ static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) CS53L30_ASP_3ST_MASK, val); } -static unsigned int const cs53l30_src_rates[] = { - 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 -}; - -static const struct snd_pcm_hw_constraint_list src_constraints = { - .count = ARRAY_SIZE(cs53l30_src_rates), - .list = cs53l30_src_rates, -}; - -static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &src_constraints); - - return 0; -} - /* * Note: CS53L30 counts the slot number per byte while ASoC counts the slot * number per slot_width. So there is a difference between the slots of ASoC @@ -843,14 +825,14 @@ static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream) return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000) #define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops cs53l30_ops = { - .startup = cs53l30_pcm_startup, .hw_params = cs53l30_pcm_hw_params, .set_fmt = cs53l30_set_dai_fmt, .set_sysclk = cs53l30_set_sysclk, From 9469cf57cd95a1f0b7ad9afd104ac120cf0aa7af Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:13:00 +0200 Subject: [PATCH 338/410] ASoC: Intel: avs: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 12kHz, 24kHz and 128kHz. These rates are now available through SNDRV_PCM_RATE_12000, SNDRV_PCM_RATE_24000 and SNDRV_PCM_RATE_128000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-9-8371948d3921@baylibre.com --- sound/soc/intel/avs/pcm.c | 22 ++++++---------------- 1 file changed, 6 insertions(+), 16 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index c76b86254a8b..afc0fc74cf94 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -471,16 +471,6 @@ static int hw_rule_param_size(struct snd_pcm_hw_params *params, struct snd_pcm_h static int avs_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - static const unsigned int rates[] = { - 8000, 11025, 12000, 16000, - 22050, 24000, 32000, 44100, - 48000, 64000, 88200, 96000, - 128000, 176400, 192000, - }; - static const struct snd_pcm_hw_constraint_list rate_list = { - .count = ARRAY_SIZE(rates), - .list = rates, - }; int ret; ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -492,10 +482,6 @@ static int avs_pcm_hw_constraints_init(struct snd_pcm_substream *substream) if (ret < 0) return ret; - ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &rate_list); - if (ret < 0) - return ret; - /* Adjust buffer and period size based on the audio format. */ snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, hw_rule_param_size, NULL, SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_CHANNELS, @@ -1332,7 +1318,9 @@ static const struct snd_soc_dai_driver i2s_dai_template = { .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_KNOT, + SNDRV_PCM_RATE_12000 | + SNDRV_PCM_RATE_24000 | + SNDRV_PCM_RATE_128000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | @@ -1343,7 +1331,9 @@ static const struct snd_soc_dai_driver i2s_dai_template = { .channels_min = 1, .channels_max = 8, .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_KNOT, + SNDRV_PCM_RATE_12000 | + SNDRV_PCM_RATE_24000 | + SNDRV_PCM_RATE_128000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .subformats = SNDRV_PCM_SUBFMTBIT_MSBITS_20 | From c061d1e4b2af8c8c76754ac9fd2cbde6db472e1d Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:13:01 +0200 Subject: [PATCH 339/410] ASoC: qcom: q6asm-dai: drop SNDRV_PCM_RATE_KNOT The custom rate constraint list was necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-10-8371948d3921@baylibre.com --- sound/soc/qcom/qdsp6/q6asm-dai.c | 31 ++++++++++--------------------- 1 file changed, 10 insertions(+), 21 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index 3913706ccdc5..045100c94352 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -128,8 +128,13 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_playback = { #define Q6ASM_FEDAI_DRIVER(num) { \ .playback = { \ .stream_name = "MultiMedia"#num" Playback", \ - .rates = (SNDRV_PCM_RATE_8000_192000| \ - SNDRV_PCM_RATE_KNOT), \ + .rates = (SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000), \ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE), \ .channels_min = 1, \ @@ -139,8 +144,9 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_playback = { }, \ .capture = { \ .stream_name = "MultiMedia"#num" Capture", \ - .rates = (SNDRV_PCM_RATE_8000_48000| \ - SNDRV_PCM_RATE_KNOT), \ + .rates = (SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000), \ .formats = (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE), \ .channels_min = 1, \ @@ -152,18 +158,6 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_playback = { .id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \ } -/* Conventional and unconventional sample rate supported */ -static unsigned int supported_sample_rates[] = { - 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, - 88200, 96000, 176400, 192000 -}; - -static struct snd_pcm_hw_constraint_list constraints_sample_rates = { - .count = ARRAY_SIZE(supported_sample_rates), - .list = supported_sample_rates, - .mask = 0, -}; - static const struct snd_compr_codec_caps q6asm_compr_caps = { .num_descriptors = 1, .descriptor[0].max_ch = 2, @@ -390,11 +384,6 @@ static int q6asm_dai_open(struct snd_soc_component *component, else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) runtime->hw = q6asm_dai_hardware_capture; - ret = snd_pcm_hw_constraint_list(runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &constraints_sample_rates); - if (ret < 0) - dev_info(dev, "snd_pcm_hw_constraint_list failed\n"); /* Ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); From 9dc03a1250d4d645badc5cdd5a3bc47237ffcab9 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:13:02 +0200 Subject: [PATCH 340/410] ASoC: sunxi: sun4i-codec: drop SNDRV_PCM_RATE_KNOT The custom rate constraint lists was necessary to support 12kHz and 24kHz. These rates are now available through SNDRV_PCM_RATE_12000 and SNDRV_PCM_RATE_24000. Use them and drop the custom rate constraint rule. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-11-8371948d3921@baylibre.com --- sound/soc/sunxi/sun4i-codec.c | 28 +++++++++------------------- 1 file changed, 9 insertions(+), 19 deletions(-) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index a2618ed650b0..25af47b63bdd 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -577,28 +577,12 @@ static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, hwrate); } - -static unsigned int sun4i_codec_src_rates[] = { - 8000, 11025, 12000, 16000, 22050, 24000, 32000, - 44100, 48000, 96000, 192000 -}; - - -static struct snd_pcm_hw_constraint_list sun4i_codec_constraints = { - .count = ARRAY_SIZE(sun4i_codec_src_rates), - .list = sun4i_codec_src_rates, -}; - - static int sun4i_codec_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &sun4i_codec_constraints); - /* * Stop issuing DRQ when we have room for less than 16 samples * in our TX FIFO @@ -626,6 +610,13 @@ static const struct snd_soc_dai_ops sun4i_codec_dai_ops = { .prepare = sun4i_codec_prepare, }; +#define SUN4I_CODEC_RATES ( \ + SNDRV_PCM_RATE_8000_48000 | \ + SNDRV_PCM_RATE_12000 | \ + SNDRV_PCM_RATE_24000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver sun4i_codec_dai = { .name = "Codec", .ops = &sun4i_codec_dai_ops, @@ -635,7 +626,7 @@ static struct snd_soc_dai_driver sun4i_codec_dai = { .channels_max = 2, .rate_min = 8000, .rate_max = 192000, - .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rates = SUN4I_CODEC_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .sig_bits = 24, @@ -646,7 +637,7 @@ static struct snd_soc_dai_driver sun4i_codec_dai = { .channels_max = 2, .rate_min = 8000, .rate_max = 48000, - .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rates = SUN4I_CODEC_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, .sig_bits = 24, @@ -1233,7 +1224,6 @@ static const struct snd_soc_component_driver sun4i_codec_component = { #endif }; -#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_CONTINUOUS #define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S32_LE) From 7bc09f7eb5e1f09b91b8edcf492e50bfda54080e Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:13:03 +0200 Subject: [PATCH 341/410] ASoC: cs35l34: drop useless rate contraint The cs35l34 adds a useless rate constraint on startup. It does not set SNDRV_PCM_RATE_KNOT and the rates set are already a subset of the ones provided in the constraint list, so it is a no-op. >From the rest of the code, it is likely HW supports more than the 32, 44.1 and 48kHz listed in CS35L34_RATES but there is no way to know for sure without proper documentation. Keep the driver as it is for now and just drop the useless constraint. Signed-off-by: Jerome Brunet Reviewed-by: David Rhodes Acked-by: Mark Brown Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-12-8371948d3921@baylibre.com --- sound/soc/codecs/cs35l34.c | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index e63a518e3b8e..287b27476a10 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -562,26 +562,6 @@ static int cs35l34_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } -static const unsigned int cs35l34_src_rates[] = { - 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 -}; - - -static const struct snd_pcm_hw_constraint_list cs35l34_constraints = { - .count = ARRAY_SIZE(cs35l34_src_rates), - .list = cs35l34_src_rates, -}; - -static int cs35l34_pcm_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, &cs35l34_constraints); - return 0; -} - - static int cs35l34_set_tristate(struct snd_soc_dai *dai, int tristate) { @@ -639,7 +619,6 @@ static int cs35l34_dai_set_sysclk(struct snd_soc_dai *dai, } static const struct snd_soc_dai_ops cs35l34_ops = { - .startup = cs35l34_pcm_startup, .set_tristate = cs35l34_set_tristate, .set_fmt = cs35l34_set_dai_fmt, .hw_params = cs35l34_pcm_hw_params, From 8055c0cd6ba520c1673d369b0216ab9da98141ea Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Thu, 5 Sep 2024 16:13:04 +0200 Subject: [PATCH 342/410] ASoC: spdif: extend supported rates to 768kHz IEC958-3 defines sampling rate up to 768 kHz. Such rates maybe used with high bandwidth IEC958 links, such as eARC. Signed-off-by: Jerome Brunet Acked-by: Mark Brown Reviewed-by: David Rhodes Reviewed-by: Jaroslav Kysela Signed-off-by: Takashi Iwai Link: https://patch.msgid.link/20240905-alsa-12-24-128-v1-13-8371948d3921@baylibre.com --- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c index 862e0b654a1c..310123d2bb5f 100644 --- a/sound/soc/codecs/spdif_receiver.c +++ b/sound/soc/codecs/spdif_receiver.c @@ -28,7 +28,8 @@ static const struct snd_soc_dapm_route dir_routes[] = { { "Capture", NULL, "spdif-in" }, }; -#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_RATES (SNDRV_PCM_RATE_8000_768000 | \ + SNDRV_PCM_RATE_128000) #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ diff --git a/sound/soc/codecs/spdif_transmitter.c b/sound/soc/codecs/spdif_transmitter.c index 736518921555..db51a46e689d 100644 --- a/sound/soc/codecs/spdif_transmitter.c +++ b/sound/soc/codecs/spdif_transmitter.c @@ -21,7 +21,8 @@ #define DRV_NAME "spdif-dit" -#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_RATES (SNDRV_PCM_RATE_8000_768000 | \ + SNDRV_PCM_RATE_128000) #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ From 1798acef8f50af59abaa0ff4775e7a79a68db671 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Fri, 6 Sep 2024 18:05:23 +0800 Subject: [PATCH 343/410] ASoC: loongson: remove redundant variable assignments In the function loongson_asoc_card_probe, it is simpler to return the value of function devm_snd_soc_register_card directly. Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240906100523.2142-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 2c8dbdba27c5..4d511b7589d0 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -192,9 +192,7 @@ static int loongson_asoc_card_probe(struct platform_device *pdev) if (ret < 0) return ret; - ret = devm_snd_soc_register_card(&pdev->dev, card); - - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } static const struct of_device_id loongson_asoc_dt_ids[] = { From 3c5a18a10a8c6fab27739e6e7bd0e4c0aa854d92 Mon Sep 17 00:00:00 2001 From: Muhammad Usama Anjum Date: Fri, 6 Sep 2024 15:37:24 +0500 Subject: [PATCH 344/410] ASoC: amd: acp: Return in-case of error Return when error occurs instead of proceeding to for loop which will use val uninitialized. Fixes: f6f7d25b1103 ("ASoC: amd: acp: Add pte configuration for ACP7.0 platform") Signed-off-by: Muhammad Usama Anjum Link: https://patch.msgid.link/20240906103727.222749-1-usama.anjum@collabora.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-platform.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index ae63b2e693ab..3a7a467b7063 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -231,7 +231,7 @@ void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int s break; default: dev_err(adata->dev, "Invalid dai id %x\n", stream->dai_id); - break; + return; } break; default: From 4849b2f78020cf0e3ba67777aadd07c620c91811 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 5 Sep 2024 05:14:29 +0000 Subject: [PATCH 345/410] ASoC: makes rtd->initialized bit field MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit rtd->initialized is used to know whether soc_init_pcm_runtime() was correctly fined, and used to call snd_soc_link_exit(). We don't need to have it as bool, let's make it bit-field same as other flags. Signed-off-by: Kuninori Morimoto Cc: Amadeusz Sławiński Cc: Cezary Rojewski Link: https://patch.msgid.link/87o752k7gq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81ef380b2e06..e6e359c1a2ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1207,8 +1207,7 @@ struct snd_soc_pcm_runtime { /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ - - bool initialized; + unsigned int initialized:1; /* CPU/Codec/Platform */ int num_components; From 241c044e743f9c55886828763c99b51b0392c21d Mon Sep 17 00:00:00 2001 From: ying zuxin <11154159@vivo.com> Date: Fri, 6 Sep 2024 16:48:31 +0800 Subject: [PATCH 346/410] ASoC: codecs: Use devm_clk_get_enabled() helpers The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: ying zuxin <11154159@vivo.com> Link: https://patch.msgid.link/20240906084841.19248-1-yingzuxin@vivo.com Signed-off-by: Mark Brown --- sound/soc/codecs/peb2466.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..b67cfad4fc32 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1975,12 +1975,9 @@ static int peb2466_spi_probe(struct spi_device *spi) if (IS_ERR(peb2466->reset_gpio)) return PTR_ERR(peb2466->reset_gpio); - peb2466->mclk = devm_clk_get(&peb2466->spi->dev, "mclk"); + peb2466->mclk = devm_clk_get_enabled(&peb2466->spi->dev, "mclk"); if (IS_ERR(peb2466->mclk)) return PTR_ERR(peb2466->mclk); - ret = clk_prepare_enable(peb2466->mclk); - if (ret) - return ret; if (peb2466->reset_gpio) { gpiod_set_value_cansleep(peb2466->reset_gpio, 1); @@ -2031,17 +2028,9 @@ static int peb2466_spi_probe(struct spi_device *spi) return 0; failed: - clk_disable_unprepare(peb2466->mclk); return ret; } -static void peb2466_spi_remove(struct spi_device *spi) -{ - struct peb2466 *peb2466 = spi_get_drvdata(spi); - - clk_disable_unprepare(peb2466->mclk); -} - static const struct of_device_id peb2466_of_match[] = { { .compatible = "infineon,peb2466", }, { } @@ -2061,7 +2050,6 @@ static struct spi_driver peb2466_spi_driver = { }, .id_table = peb2466_id_table, .probe = peb2466_spi_probe, - .remove = peb2466_spi_remove, }; module_spi_driver(peb2466_spi_driver); From 63e38d0787971d3026bb5e31c86663e0f1e62586 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 7 Sep 2024 10:40:47 +0200 Subject: [PATCH 347/410] ALSA: hda: Allow the default preallocation for x86 again Since there are a few corner cases where the S/G buffer allocation isn't performed (e.g. depending on IOMMU implementations), it'd be better to allow the default buffer preallocation size for x86, too. The default for x86 is still kept to 0, as it should work in most cases. Link: https://patch.msgid.link/20240907084129.28802-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/hda/Kconfig | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index e2ac247fc1d4..eb488a522572 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -21,7 +21,7 @@ config SND_HDA_EXT_CORE select SND_HDA_CORE config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" if !SND_DMA_SGBUF + int "Pre-allocated buffer size for HD-audio driver" range 0 32768 default 0 if SND_DMA_SGBUF default 64 if !SND_DMA_SGBUF @@ -30,7 +30,8 @@ config SND_HDA_PREALLOC_SIZE HD-audio driver. A larger buffer (e.g. 2048) is preferred for systems using PulseAudio. The default 64 is chosen just for compatibility reasons. - On x86 systems, the default is zero as we need no preallocation. + On x86 systems, the default is zero as S/G allocation works + and no preallocation is needed in most cases. Note that the pre-allocation size can be changed dynamically via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. From b12891c7b6d2c29a07b2da5589ee469d7bf37254 Mon Sep 17 00:00:00 2001 From: Jerome Brunet Date: Fri, 6 Sep 2024 11:34:16 +0200 Subject: [PATCH 348/410] ALSA: IEC958 definition for consumer status channel update Add 128kHz, 352.4kHz, 384kHz and 705.6kHz. These definitions have been found working on eARC using a Murideo Seven Generator. Signed-off-by: Jerome Brunet Link: https://patch.msgid.link/20240906093422.2976550-1-jbrunet@baylibre.com Signed-off-by: Takashi Iwai --- include/sound/asoundef.h | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h index 9fdeac19dadb..09b2c3dffb30 100644 --- a/include/sound/asoundef.h +++ b/include/sound/asoundef.h @@ -110,18 +110,22 @@ #define IEC958_AES2_CON_SOURCE_UNSPEC (0<<0) /* unspecified */ #define IEC958_AES2_CON_CHANNEL (15<<4) /* mask - channel number */ #define IEC958_AES2_CON_CHANNEL_UNSPEC (0<<4) /* unspecified */ -#define IEC958_AES3_CON_FS (15<<0) /* mask - sample frequency */ +#define IEC958_AES3_CON_FS ((1<<7) | (15<<0)) /* mask - sample frequency */ #define IEC958_AES3_CON_FS_44100 (0<<0) /* 44.1kHz */ #define IEC958_AES3_CON_FS_NOTID (1<<0) /* non indicated */ #define IEC958_AES3_CON_FS_48000 (2<<0) /* 48kHz */ #define IEC958_AES3_CON_FS_32000 (3<<0) /* 32kHz */ #define IEC958_AES3_CON_FS_22050 (4<<0) /* 22.05kHz */ +#define IEC958_AES3_CON_FS_384000 (5<<0) /* 384kHz */ #define IEC958_AES3_CON_FS_24000 (6<<0) /* 24kHz */ #define IEC958_AES3_CON_FS_88200 (8<<0) /* 88.2kHz */ #define IEC958_AES3_CON_FS_768000 (9<<0) /* 768kHz */ #define IEC958_AES3_CON_FS_96000 (10<<0) /* 96kHz */ #define IEC958_AES3_CON_FS_176400 (12<<0) /* 176.4kHz */ +#define IEC958_AES3_CON_FS_352400 (13<<0) /* 352.4kHz */ #define IEC958_AES3_CON_FS_192000 (14<<0) /* 192kHz */ +#define IEC958_AES3_CON_FS_128000 ((1<<7) | (11<<0)) /* 128kHz */ +#define IEC958_AES3_CON_FS_705600 ((1<<7) | (13<<0)) /* 705.6kHz */ #define IEC958_AES3_CON_CLOCK (3<<4) /* mask - clock accuracy */ #define IEC958_AES3_CON_CLOCK_1000PPM (0<<4) /* 1000 ppm */ #define IEC958_AES3_CON_CLOCK_50PPM (1<<4) /* 50 ppm */ From 0a131eb8551da832e3c04b385305f7d79e5edb89 Mon Sep 17 00:00:00 2001 From: He Lugang Date: Sat, 7 Sep 2024 22:28:54 +0800 Subject: [PATCH 349/410] ALSA: rme9652: remove unused parameter in macro Parameter xindex is not used in macro HDSPM_TCO_LTC_FRAMES and HDSPM_TCO_VIDEO_INPUT_FORMAT,so just remove it to simplify the code. Signed-off-by: He Lugang Link: https://patch.msgid.link/F53E9F10DA24705D+20240907142854.17627-1-helugang@uniontech.com Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index dad974378e00..d7290463d654 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3083,7 +3083,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol, -#define HDSPM_TCO_VIDEO_INPUT_FORMAT(xname, xindex) \ +#define HDSPM_TCO_VIDEO_INPUT_FORMAT(xname) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_READ |\ @@ -3129,7 +3129,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol, -#define HDSPM_TCO_LTC_FRAMES(xname, xindex) \ +#define HDSPM_TCO_LTC_FRAMES(xname) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_READ |\ @@ -4630,8 +4630,8 @@ static const struct snd_kcontrol_new snd_hdspm_controls_tco[] = { HDSPM_TCO_WORD_TERM("TCO Word Term", 0), HDSPM_TCO_LOCK_CHECK("TCO Input Check", 11), HDSPM_TCO_LOCK_CHECK("TCO LTC Valid", 12), - HDSPM_TCO_LTC_FRAMES("TCO Detected Frame Rate", 0), - HDSPM_TCO_VIDEO_INPUT_FORMAT("Video Input Format", 0) + HDSPM_TCO_LTC_FRAMES("TCO Detected Frame Rate"), + HDSPM_TCO_VIDEO_INPUT_FORMAT("Video Input Format") }; From 77b696f489d2fd83bbcffeed363baac8f2f6ed4b Mon Sep 17 00:00:00 2001 From: Romain Gantois Date: Fri, 6 Sep 2024 14:20:58 +0200 Subject: [PATCH 350/410] ASoC: tlv320aic31xx: Add support for loading filter coefficients The TLV320DAC3100 Audio DAC has 25 built-in digital audio processing blocks (PRBs). Each of these PRBs has a static filter structure with programmable coefficients. Once a PRB is selected for use by the DAC, its filter coefficients can be configured via a dedicated set of registers. Define a new optional firmware which can be loaded by the TLV320DAC driver. This firmware describes a full set of filter coefficients for all blocks used by the various PRBs. The firmware's binary format is heavily inspired by the one used in the peb2466 driver. It includes a version marker to allow for potential evolutions of the format. Note that adaptive filtering is not supported i.e. filter coefficients are loaded once before power-on and then cannot be changed while the DAC is powered. This is why only page A coefficients are modified. Page B coefficients are only used for adaptive filtering. Signed-off-by: Romain Gantois Link: https://patch.msgid.link/20240906-tlv320-filter-v1-1-6955f53ff435@bootlin.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 100 +++++++++++++++++++++++++++++++ 1 file changed, 100 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2f94cfda0e33..7e624c4b77b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -12,6 +12,7 @@ * and mono/stereo Class-D speaker driver. */ +#include #include #include #include @@ -22,6 +23,7 @@ #include #include #include +#include #include #include #include @@ -1638,6 +1640,98 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +static int tlv320dac3100_fw_load(struct aic31xx_priv *aic31xx, + const u8 *data, size_t size) +{ + int ret, reg; + u16 val16; + + /* + * Coefficients firmware binary structure. Multi-byte values are big-endian. + * + * @0, 16bits: Magic (0xB30C) + * @2, 16bits: Version (0x0100 for version 1.0) + * @4, 8bits: DAC Processing Block Selection + * @5, 62 16-bit values: Page 8 buffer A DAC programmable filter coefficients + * @129, 12 16-bit values: Page 9 Buffer A DAC programmable filter coefficients + * + * Filter coefficients are interpreted as two's complement values + * ranging from -32 768 to 32 767. For more details on filter coefficients, + * please refer to the TLV320DAC3100 datasheet, tables 6-120 and 6-123. + */ + + if (size != 153) { + dev_err(aic31xx->dev, "firmware size is %zu, expected 153 bytes\n", size); + return -EINVAL; + } + + /* Check magic */ + val16 = get_unaligned_be16(data); + if (val16 != 0xb30c) { + dev_err(aic31xx->dev, "fw magic is 0x%04x expected 0xb30c\n", val16); + return -EINVAL; + } + data += 2; + + /* Check version */ + val16 = get_unaligned_be16(data); + if (val16 != 0x0100) { + dev_err(aic31xx->dev, "invalid firmware version 0x%04x! expected 1", val16); + return -EINVAL; + } + data += 2; + + ret = regmap_write(aic31xx->regmap, AIC31XX_DACPRB, *data); + if (ret) { + dev_err(aic31xx->dev, "failed to write PRB index: err %d\n", ret); + return ret; + } + data += 1; + + /* Page 8 Buffer A coefficients */ + for (reg = 2; reg < 126; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(8, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 8 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + /* Page 9 Buffer A coefficients */ + for (reg = 2; reg < 26; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(9, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 9 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + dev_info(aic31xx->dev, "done loading DAC filter coefficients\n"); + + return ret; +} + +static int tlv320dac3100_load_coeffs(struct aic31xx_priv *aic31xx, + const char *fw_name) +{ + const struct firmware *fw; + int ret; + + ret = request_firmware(&fw, fw_name, aic31xx->dev); + if (ret) + return ret; + + ret = tlv320dac3100_fw_load(aic31xx, fw->data, fw->size); + + release_firmware(fw); + + return ret; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c) { struct aic31xx_priv *aic31xx; @@ -1727,6 +1821,12 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c) } } + if (aic31xx->codec_type == DAC3100) { + ret = tlv320dac3100_load_coeffs(aic31xx, "tlv320dac3100-coeffs.bin"); + if (ret) + dev_warn(aic31xx->dev, "Did not load any filter coefficients\n"); + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, From 6b31d6a4ca3b4d8abfb39127130889f9a9a38aa1 Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sun, 8 Sep 2024 22:17:34 +0000 Subject: [PATCH 351/410] ASoC: mediatek: mt8365: include linux/bitfield.h On x86, the header is not already included implicitly, breaking compile-testing: In file included from sound/soc/mediatek/mt8365/mt8365-afe-common.h:19, from sound/soc/mediatek/mt8365/mt8365-afe-pcm.c:18: sound/soc/mediatek/mt8365/mt8365-afe-pcm.c: In function 'mt8365_afe_cm2_mux_conn': sound/soc/mediatek/mt8365/mt8365-reg.h:952:41: error: implicit declaration of function 'FIELD_PREP' [-Wimplicit-function-declaration] 952 | #define CM2_AFE_CM2_CONN_CFG1(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG1_MASK, (x)) | ^~~~~~~~~~ Included it ahead of the field definitions. Fixes: 38c7c9ddc740 ("ASoC: mediatek: mt8365: Add common header") Signed-off-by: Arnd Bergmann Reviewed-by: Matthias Brugger Link: https://patch.msgid.link/20240908221754.2210857-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-reg.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/mediatek/mt8365/mt8365-reg.h b/sound/soc/mediatek/mt8365/mt8365-reg.h index b7334c2e64ed..1051718c29f3 100644 --- a/sound/soc/mediatek/mt8365/mt8365-reg.h +++ b/sound/soc/mediatek/mt8365/mt8365-reg.h @@ -10,6 +10,8 @@ #ifndef _MT8365_REG_H_ #define _MT8365_REG_H_ +#include + #define AUDIO_TOP_CON0 (0x0000) #define AUDIO_TOP_CON1 (0x0004) #define AUDIO_TOP_CON2 (0x0008) From 876dec03fdfb7eeb592d01a95ba292c9e53b324b Mon Sep 17 00:00:00 2001 From: Arnd Bergmann Date: Sat, 7 Sep 2024 20:00:38 +0000 Subject: [PATCH 352/410] ASoC: mediatek: mt8365: remove unused mt8365_i2s_hd_str The mt8365_i2s_enum and mt8365_i2s_hd_str variables are not used anywhere, but cause a warning when building with C=1 or when enabling -Wunused-const-variable: sound/soc/mediatek/mt8365/mt8365-dai-i2s.c:781:27: error: 'mt8365_i2s_hd_str' defined but not used [-Werror=unused-const-variable=] 781 | static const char * const mt8365_i2s_hd_str[] = { | ^~~~~~~~~~~~~~~~~ Remove these for the moment, they can be added back if a user comes up. Fixes: 402bbb13a195 ("ASoC: mediatek: mt8365: Add I2S DAI support") Signed-off-by: Arnd Bergmann Reviewed-by: Matthias Brugger Link: https://patch.msgid.link/20240907200053.3027553-1-arnd@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c index 5003fe5e5ccf..b4d4d6705c86 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c @@ -777,13 +777,6 @@ static struct snd_soc_dai_driver mtk_dai_i2s_driver[] = { } }; -/* low jitter control */ -static const char * const mt8365_i2s_hd_str[] = { - "Normal", "Low_Jitter" -}; - -static SOC_ENUM_SINGLE_EXT_DECL(mt8365_i2s_enum, mt8365_i2s_hd_str); - static const char * const fmi2sin_text[] = { "OPEN", "FM_2ND_I2S_IN" }; From 28fbfaf6bd1df94cccf1a5c009fe26b9bdccc2b6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 9 Sep 2024 15:47:43 +0200 Subject: [PATCH 353/410] ALSA: hda: Use non-SG allocation for the communication buffers The azx_bus->dma_type is referred only for allocating the communication buffers like CORB/RIRB, and the allocation size is small. Hence it doesn't have to be S/G buffer allocation, which is an obvious overkill. Use the standard SNDRV_DMA_TYPE_DEV_WC instead. This was changed to SNDRV_DMA_TYPE_DEV_WC_SG in the commit 37137ec26c2c ("ALSA: hda: Once again fix regression of page allocations with IOMMU") as a workaround for IOMMU-backed allocations. But this is no longer needed since the allocation with SNDRV_DMA_TYPE_DEV_WC itself was fixed in the commit 9c27301342a5 ("ALSA: memalloc: Use DMA API for x86 WC page allocations, too"). So this patch reverts the previous workaround in this piece of code. Link: https://patch.msgid.link/20240909134744.25426-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b79020adce63..edeaf3ee273c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1821,7 +1821,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, /* use the non-cached pages in non-snoop mode */ if (!azx_snoop(chip)) - azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC_SG; + azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_WC; if (chip->driver_type == AZX_DRIVER_NVIDIA) { dev_dbg(chip->card->dev, "Enable delay in RIRB handling\n"); From f2bd6f5b3777d800fb3cad17b9509e1e2128df62 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:05 +0800 Subject: [PATCH 354/410] ASoC: loongson: Use BIT() macro Where applicable, use BIT() macro instead of shift operation to improve readability. Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/ccca555c96f18c0ecf5f1544c82945ba651d105f.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_dma.c | 10 +++++----- sound/soc/loongson/loongson_i2s.h | 24 ++++++++++++------------ 2 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/loongson/loongson_dma.c b/sound/soc/loongson/loongson_dma.c index 0238f88bc674..20e4a0641340 100644 --- a/sound/soc/loongson/loongson_dma.c +++ b/sound/soc/loongson/loongson_dma.c @@ -17,11 +17,11 @@ #include "loongson_i2s.h" /* DMA dma_order Register */ -#define DMA_ORDER_STOP (1 << 4) /* DMA stop */ -#define DMA_ORDER_START (1 << 3) /* DMA start */ -#define DMA_ORDER_ASK_VALID (1 << 2) /* DMA ask valid flag */ -#define DMA_ORDER_AXI_UNCO (1 << 1) /* Uncache access */ -#define DMA_ORDER_ADDR_64 (1 << 0) /* 64bits address support */ +#define DMA_ORDER_STOP BIT(4) /* DMA stop */ +#define DMA_ORDER_START BIT(3) /* DMA start */ +#define DMA_ORDER_ASK_VALID BIT(2) /* DMA ask valid flag */ +#define DMA_ORDER_AXI_UNCO BIT(1) /* Uncache access */ +#define DMA_ORDER_ADDR_64 BIT(0) /* 64bits address support */ #define DMA_ORDER_ASK_MASK (~0x1fUL) /* Ask addr mask */ #define DMA_ORDER_CTRL_MASK (0x0fUL) /* Control mask */ diff --git a/sound/soc/loongson/loongson_i2s.h b/sound/soc/loongson/loongson_i2s.h index 89492eebf834..c8052a762c1b 100644 --- a/sound/soc/loongson/loongson_i2s.h +++ b/sound/soc/loongson/loongson_i2s.h @@ -27,18 +27,18 @@ #define LS_I2S_RX_ORDER 0x110 /* RX DMA Order */ /* Loongson I2S Control Register */ -#define I2S_CTRL_MCLK_READY (1 << 16) /* MCLK ready */ -#define I2S_CTRL_MASTER (1 << 15) /* Master mode */ -#define I2S_CTRL_MSB (1 << 14) /* MSB bit order */ -#define I2S_CTRL_RX_EN (1 << 13) /* RX enable */ -#define I2S_CTRL_TX_EN (1 << 12) /* TX enable */ -#define I2S_CTRL_RX_DMA_EN (1 << 11) /* DMA RX enable */ -#define I2S_CTRL_CLK_READY (1 << 8) /* BCLK ready */ -#define I2S_CTRL_TX_DMA_EN (1 << 7) /* DMA TX enable */ -#define I2S_CTRL_RESET (1 << 4) /* Controller soft reset */ -#define I2S_CTRL_MCLK_EN (1 << 3) /* Enable MCLK */ -#define I2S_CTRL_RX_INT_EN (1 << 1) /* RX interrupt enable */ -#define I2S_CTRL_TX_INT_EN (1 << 0) /* TX interrupt enable */ +#define I2S_CTRL_MCLK_READY BIT(16) /* MCLK ready */ +#define I2S_CTRL_MASTER BIT(15) /* Master mode */ +#define I2S_CTRL_MSB BIT(14) /* MSB bit order */ +#define I2S_CTRL_RX_EN BIT(13) /* RX enable */ +#define I2S_CTRL_TX_EN BIT(12) /* TX enable */ +#define I2S_CTRL_RX_DMA_EN BIT(11) /* DMA RX enable */ +#define I2S_CTRL_CLK_READY BIT(8) /* BCLK ready */ +#define I2S_CTRL_TX_DMA_EN BIT(7) /* DMA TX enable */ +#define I2S_CTRL_RESET BIT(4) /* Controller soft reset */ +#define I2S_CTRL_MCLK_EN BIT(3) /* Enable MCLK */ +#define I2S_CTRL_RX_INT_EN BIT(1) /* RX interrupt enable */ +#define I2S_CTRL_TX_INT_EN BIT(0) /* TX interrupt enable */ #define LS_I2S_DRVNAME "loongson-i2s" From ce3997ab8b4ae2312b8cdddd7db9b15eb11004a3 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:06 +0800 Subject: [PATCH 355/410] ASoC: loongson: Simplify probe() with local dev variable Simplify the probe() function by using local 'dev' instead of &pdev->dev. Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/1984a20930da515e2a478b02159f83c02498f6be.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 17 ++++++++------- sound/soc/loongson/loongson_i2s_pci.c | 30 +++++++++++++-------------- 2 files changed, 24 insertions(+), 23 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 4d511b7589d0..d3cd23ddd027 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -159,40 +159,41 @@ err: static int loongson_asoc_card_probe(struct platform_device *pdev) { struct loongson_card_data *ls_priv; + struct device *dev = &pdev->dev; struct snd_soc_card *card; int ret; - ls_priv = devm_kzalloc(&pdev->dev, sizeof(*ls_priv), GFP_KERNEL); + ls_priv = devm_kzalloc(dev, sizeof(*ls_priv), GFP_KERNEL); if (!ls_priv) return -ENOMEM; card = &ls_priv->snd_card; - card->dev = &pdev->dev; + card->dev = dev; card->owner = THIS_MODULE; card->dai_link = loongson_dai_links; card->num_links = ARRAY_SIZE(loongson_dai_links); snd_soc_card_set_drvdata(card, ls_priv); - ret = device_property_read_string(&pdev->dev, "model", &card->name); + ret = device_property_read_string(dev, "model", &card->name); if (ret) { - dev_err(&pdev->dev, "Error parsing card name: %d\n", ret); + dev_err(dev, "Error parsing card name: %d\n", ret); return ret; } - ret = device_property_read_u32(&pdev->dev, "mclk-fs", &ls_priv->mclk_fs); + ret = device_property_read_u32(dev, "mclk-fs", &ls_priv->mclk_fs); if (ret) { - dev_err(&pdev->dev, "Error parsing mclk-fs: %d\n", ret); + dev_err(dev, "Error parsing mclk-fs: %d\n", ret); return ret; } - if (has_acpi_companion(&pdev->dev)) + if (has_acpi_companion(dev)) ret = loongson_card_parse_acpi(ls_priv); else ret = loongson_card_parse_of(ls_priv); if (ret < 0) return ret; - return devm_snd_soc_register_card(&pdev->dev, card); + return devm_snd_soc_register_card(dev, card); } static const struct of_device_id loongson_asoc_dt_ids[] = { diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index ec18b122cd79..e8ea28bc5a5f 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -75,32 +75,33 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, { const struct fwnode_handle *fwnode = pdev->dev.fwnode; struct loongson_dma_data *tx_data, *rx_data; + struct device *dev = &pdev->dev; struct loongson_i2s *i2s; int ret; if (pcim_enable_device(pdev)) { - dev_err(&pdev->dev, "pci_enable_device failed\n"); + dev_err(dev, "pci_enable_device failed\n"); return -ENODEV; } - i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); + i2s = devm_kzalloc(dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) return -ENOMEM; i2s->rev_id = pdev->revision; - i2s->dev = &pdev->dev; + i2s->dev = dev; pci_set_drvdata(pdev, i2s); - ret = pcim_iomap_regions(pdev, 1 << 0, dev_name(&pdev->dev)); + ret = pcim_iomap_regions(pdev, 1 << 0, dev_name(dev)); if (ret < 0) { - dev_err(&pdev->dev, "iomap_regions failed\n"); + dev_err(dev, "iomap_regions failed\n"); return ret; } i2s->reg_base = pcim_iomap_table(pdev)[0]; - i2s->regmap = devm_regmap_init_mmio(&pdev->dev, i2s->reg_base, + i2s->regmap = devm_regmap_init_mmio(dev, i2s->reg_base, &loongson_i2s_regmap_config); if (IS_ERR(i2s->regmap)) { - dev_err(&pdev->dev, "regmap_init_mmio failed\n"); + dev_err(dev, "regmap_init_mmio failed\n"); return PTR_ERR(i2s->regmap); } @@ -115,34 +116,33 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); if (tx_data->irq < 0) { - dev_err(&pdev->dev, "dma tx irq invalid\n"); + dev_err(dev, "dma tx irq invalid\n"); return tx_data->irq; } rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); if (rx_data->irq < 0) { - dev_err(&pdev->dev, "dma rx irq invalid\n"); + dev_err(dev, "dma rx irq invalid\n"); return rx_data->irq; } - device_property_read_u32(&pdev->dev, "clock-frequency", &i2s->clk_rate); + device_property_read_u32(dev, "clock-frequency", &i2s->clk_rate); if (!i2s->clk_rate) { - dev_err(&pdev->dev, "clock-frequency property invalid\n"); + dev_err(dev, "clock-frequency property invalid\n"); return -EINVAL; } - dma_set_mask_and_coherent(&pdev->dev, DMA_BIT_MASK(64)); + dma_set_mask_and_coherent(dev, DMA_BIT_MASK(64)); if (i2s->rev_id == 1) { regmap_write(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_RESET); udelay(200); } - ret = devm_snd_soc_register_component(&pdev->dev, - &loongson_i2s_component, + ret = devm_snd_soc_register_component(dev, &loongson_i2s_component, &loongson_i2s_dai, 1); if (ret) { - dev_err(&pdev->dev, "register DAI failed %d\n", ret); + dev_err(dev, "register DAI failed %d\n", ret); return ret; } From 3d2528d6c021cba141a7079ae1a5937190700899 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:07 +0800 Subject: [PATCH 356/410] ASoC: loongson: Simplify with dev_err_probe() Error handling in probe() can be a bit simpler with dev_err_probe(). Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/07855aa6c290ec826d63e68b898e7f4afac5e30d.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 25 ++++++++------------ sound/soc/loongson/loongson_i2s_pci.c | 33 ++++++++++----------------- 2 files changed, 22 insertions(+), 36 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index d3cd23ddd027..f0ed4508f38a 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -176,22 +176,17 @@ static int loongson_asoc_card_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, ls_priv); ret = device_property_read_string(dev, "model", &card->name); - if (ret) { - dev_err(dev, "Error parsing card name: %d\n", ret); - return ret; - } - ret = device_property_read_u32(dev, "mclk-fs", &ls_priv->mclk_fs); - if (ret) { - dev_err(dev, "Error parsing mclk-fs: %d\n", ret); - return ret; - } + if (ret) + dev_err_probe(dev, ret, "Error parsing card name\n"); - if (has_acpi_companion(dev)) - ret = loongson_card_parse_acpi(ls_priv); - else - ret = loongson_card_parse_of(ls_priv); - if (ret < 0) - return ret; + ret = device_property_read_u32(dev, "mclk-fs", &ls_priv->mclk_fs); + if (ret) + dev_err_probe(dev, ret, "Error parsing mclk-fs\n"); + + ret = has_acpi_companion(dev) ? loongson_card_parse_acpi(ls_priv) + : loongson_card_parse_of(ls_priv); + if (ret) + dev_err_probe(dev, ret, "Error parsing acpi/of properties\n"); return devm_snd_soc_register_card(dev, card); } diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index e8ea28bc5a5f..3872b1d8fce0 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -97,13 +97,12 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, dev_err(dev, "iomap_regions failed\n"); return ret; } + i2s->reg_base = pcim_iomap_table(pdev)[0]; i2s->regmap = devm_regmap_init_mmio(dev, i2s->reg_base, &loongson_i2s_regmap_config); - if (IS_ERR(i2s->regmap)) { - dev_err(dev, "regmap_init_mmio failed\n"); - return PTR_ERR(i2s->regmap); - } + if (IS_ERR(i2s->regmap)) + dev_err_probe(dev, PTR_ERR(i2s->regmap), "regmap_init_mmio failed\n"); tx_data = &i2s->tx_dma_data; rx_data = &i2s->rx_dma_data; @@ -115,22 +114,16 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); - if (tx_data->irq < 0) { - dev_err(dev, "dma tx irq invalid\n"); - return tx_data->irq; - } + if (tx_data->irq < 0) + dev_err_probe(dev, tx_data->irq, "dma tx irq invalid\n"); rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); - if (rx_data->irq < 0) { - dev_err(dev, "dma rx irq invalid\n"); - return rx_data->irq; - } + if (rx_data->irq < 0) + dev_err_probe(dev, rx_data->irq, "dma rx irq invalid\n"); - device_property_read_u32(dev, "clock-frequency", &i2s->clk_rate); - if (!i2s->clk_rate) { - dev_err(dev, "clock-frequency property invalid\n"); - return -EINVAL; - } + ret = device_property_read_u32(dev, "clock-frequency", &i2s->clk_rate); + if (ret) + dev_err_probe(dev, ret, "clock-frequency property invalid\n"); dma_set_mask_and_coherent(dev, DMA_BIT_MASK(64)); @@ -141,10 +134,8 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, ret = devm_snd_soc_register_component(dev, &loongson_i2s_component, &loongson_i2s_dai, 1); - if (ret) { - dev_err(dev, "register DAI failed %d\n", ret); - return ret; - } + if (ret) + dev_err_probe(dev, ret, "register DAI failed\n"); return 0; } From e28ee1b8a92e6797fc652fa64e536178dd627e89 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:34 +0800 Subject: [PATCH 357/410] ASoC: loongson: Simplify if statment in loongson_card_hw_params() Deal with illegal conditions first and put the normal process code outside the if condition to improve code readability. Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/98b71f9643970f11bc500c01599c7aeb77ff2a58.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 32 +++++++++++++++--------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index f0ed4508f38a..3dd82caaae3b 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -24,27 +24,27 @@ static int loongson_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); - struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); struct loongson_card_data *ls_card = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); + struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); int ret, mclk; - if (ls_card->mclk_fs) { - mclk = ls_card->mclk_fs * params_rate(params); - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret < 0) { - dev_err(codec_dai->dev, "cpu_dai clock not set\n"); - return ret; - } + if (!ls_card->mclk_fs) + return 0; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(codec_dai->dev, "codec_dai clock not set\n"); - return ret; - } + mclk = ls_card->mclk_fs * params_rate(params); + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "cpu_dai clock not set\n"); + return ret; } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "codec_dai clock not set\n"); + return ret; + } + return 0; } From c7b626a8930d7029c096f0634ae6169af59d92b6 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:36 +0800 Subject: [PATCH 358/410] ASoC: loongson: Replace if with ternary operator Replace an if statement with a ternary operator, making the code a tiny bit shorter. Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/94ec2ac178610f50af4815ef5b719695915bba31.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_i2s.c | 24 ++++++++++-------------- 1 file changed, 10 insertions(+), 14 deletions(-) diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c index 3b9786076501..8bb38e418333 100644 --- a/sound/soc/loongson/loongson_i2s.c +++ b/sound/soc/loongson/loongson_i2s.c @@ -21,34 +21,30 @@ SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) +#define LOONGSON_I2S_TX_ENABLE (I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN) +#define LOONGSON_I2S_RX_ENABLE (I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN) + static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask; int ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, - I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN); - else - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, - I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN); + mask = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + LOONGSON_I2S_TX_ENABLE : LOONGSON_I2S_RX_ENABLE; + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, mask, mask); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN, 0); - else - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN, 0); + mask = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + LOONGSON_I2S_TX_ENABLE : LOONGSON_I2S_RX_ENABLE; + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, mask, 0); break; default: ret = -EINVAL; From ddb538a3004b10a04a14a0d275c5f52a8d161e80 Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:37 +0800 Subject: [PATCH 359/410] ASoC: loongson: Factor out loongson_card_acpi_find_device() function The operations for reading the cpu and codec nodes in loongson_card_parse_acpi() are similar, so we convert them into a simple helper function called loongson_card_acpi_find_device(). Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/3b7da05e5fd4326e7944aa749bf06dd44e964f6c.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 55 +++++++++++++++++------------- 1 file changed, 31 insertions(+), 24 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 3dd82caaae3b..6078cdf09c22 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -68,46 +68,53 @@ static struct snd_soc_dai_link loongson_dai_links[] = { }, }; +static struct acpi_device *loongson_card_acpi_find_device(struct snd_soc_card *card, + const char *name) +{ + struct fwnode_handle *fwnode = card->dev->fwnode; + struct fwnode_reference_args args; + int status; + + memset(&args, 0, sizeof(args)); + status = acpi_node_get_property_reference(fwnode, name, 0, &args); + if (status || !is_acpi_device_node(args.fwnode)) { + dev_err(card->dev, "No matching phy in ACPI table\n"); + return NULL; + } + + return to_acpi_device_node(args.fwnode); +} + static int loongson_card_parse_acpi(struct loongson_card_data *data) { struct snd_soc_card *card = &data->snd_card; - struct fwnode_handle *fwnode = card->dev->fwnode; - struct fwnode_reference_args args; const char *codec_dai_name; struct acpi_device *adev; struct device *phy_dev; - int ret, i; + int i; /* fixup platform name based on reference node */ - memset(&args, 0, sizeof(args)); - ret = acpi_node_get_property_reference(fwnode, "cpu", 0, &args); - if (ret || !is_acpi_device_node(args.fwnode)) { - dev_err(card->dev, "No matching phy in ACPI table\n"); - return ret ?: -ENOENT; - } - adev = to_acpi_device_node(args.fwnode); + adev = loongson_card_acpi_find_device(card, "cpu"); + if (!adev) + return -ENOENT; + phy_dev = acpi_get_first_physical_node(adev); if (!phy_dev) return -EPROBE_DEFER; - for (i = 0; i < card->num_links; i++) - loongson_dai_links[i].platforms->name = dev_name(phy_dev); /* fixup codec name based on reference node */ - memset(&args, 0, sizeof(args)); - ret = acpi_node_get_property_reference(fwnode, "codec", 0, &args); - if (ret || !is_acpi_device_node(args.fwnode)) { - dev_err(card->dev, "No matching phy in ACPI table\n"); - return ret ?: -ENOENT; - } - adev = to_acpi_device_node(args.fwnode); + adev = loongson_card_acpi_find_device(card, "codec"); + if (!adev) + return -ENOENT; snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); - for (i = 0; i < card->num_links; i++) - loongson_dai_links[i].codecs->name = codec_name; - device_property_read_string(card->dev, "codec-dai-name", - &codec_dai_name); - for (i = 0; i < card->num_links; i++) + device_property_read_string(card->dev, "codec-dai-name", &codec_dai_name); + + for (i = 0; i < card->num_links; i++) { + loongson_dai_links[i].platforms->name = dev_name(phy_dev); + loongson_dai_links[i].codecs->name = codec_name; loongson_dai_links[i].codecs->dai_name = codec_dai_name; + } return 0; } From 4c22b04e116e114f18d9cef15f913e1649ccb7ed Mon Sep 17 00:00:00 2001 From: Binbin Zhou Date: Mon, 9 Sep 2024 15:19:55 +0800 Subject: [PATCH 360/410] ASoC: loongson: Factor out loongson i2s enable clock functions There are a few i2s clock enable operations in loongson_i2s_set_fmt(), convert them to simple helper functions called loongson_i2s_enable_mclk() and loongson_i2s_enable_bclk(). Signed-off-by: Binbin Zhou Link: https://patch.msgid.link/d6f6c818b0ecee87277f704b6a801cbbf5e712ce.1725844530.git.zhoubinbin@loongson.cn Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_i2s.c | 86 +++++++++++++++++-------------- 1 file changed, 47 insertions(+), 39 deletions(-) diff --git a/sound/soc/loongson/loongson_i2s.c b/sound/soc/loongson/loongson_i2s.c index 8bb38e418333..40bbf3205391 100644 --- a/sound/soc/loongson/loongson_i2s.c +++ b/sound/soc/loongson/loongson_i2s.c @@ -24,6 +24,9 @@ #define LOONGSON_I2S_TX_ENABLE (I2S_CTRL_TX_EN | I2S_CTRL_TX_DMA_EN) #define LOONGSON_I2S_RX_ENABLE (I2S_CTRL_RX_EN | I2S_CTRL_RX_DMA_EN) +#define LOONGSON_I2S_DEF_DELAY 10 +#define LOONGSON_I2S_DEF_TIMEOUT 500000 + static int loongson_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { @@ -119,10 +122,40 @@ static int loongson_i2s_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, return 0; } +static int loongson_i2s_enable_mclk(struct loongson_i2s *i2s) +{ + u32 val; + + if (i2s->rev_id == 0) + return 0; + + regmap_update_bits(i2s->regmap, LS_I2S_CTRL, + I2S_CTRL_MCLK_EN, I2S_CTRL_MCLK_EN); + + return regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_MCLK_READY, + LOONGSON_I2S_DEF_DELAY, + LOONGSON_I2S_DEF_TIMEOUT); +} + +static int loongson_i2s_enable_bclk(struct loongson_i2s *i2s) +{ + u32 val; + + if (i2s->rev_id == 0) + return 0; + + return regmap_read_poll_timeout_atomic(i2s->regmap, + LS_I2S_CTRL, val, + val & I2S_CTRL_CLK_READY, + LOONGSON_I2S_DEF_DELAY, + LOONGSON_I2S_DEF_TIMEOUT); +} + static int loongson_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct loongson_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 val; int ret; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -144,54 +177,29 @@ static int loongson_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* Enable master mode */ regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, I2S_CTRL_MASTER); - if (i2s->rev_id == 1) { - ret = regmap_read_poll_timeout_atomic(i2s->regmap, - LS_I2S_CTRL, val, - val & I2S_CTRL_CLK_READY, - 10, 500000); - if (ret < 0) - dev_warn(dai->dev, "wait BCLK ready timeout\n"); - } + ret = loongson_i2s_enable_bclk(i2s); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); break; case SND_SOC_DAIFMT_BC_FP: /* Enable MCLK */ - if (i2s->rev_id == 1) { - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_MCLK_EN, - I2S_CTRL_MCLK_EN); - ret = regmap_read_poll_timeout_atomic(i2s->regmap, - LS_I2S_CTRL, val, - val & I2S_CTRL_MCLK_READY, - 10, 500000); - if (ret < 0) - dev_warn(dai->dev, "wait MCLK ready timeout\n"); - } + ret = loongson_i2s_enable_mclk(i2s); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); break; case SND_SOC_DAIFMT_BP_FP: /* Enable MCLK */ - if (i2s->rev_id == 1) { - regmap_update_bits(i2s->regmap, LS_I2S_CTRL, - I2S_CTRL_MCLK_EN, - I2S_CTRL_MCLK_EN); - ret = regmap_read_poll_timeout_atomic(i2s->regmap, - LS_I2S_CTRL, val, - val & I2S_CTRL_MCLK_READY, - 10, 500000); - if (ret < 0) - dev_warn(dai->dev, "wait MCLK ready timeout\n"); - } + ret = loongson_i2s_enable_mclk(i2s); + if (ret < 0) + dev_warn(dai->dev, "wait MCLK ready timeout\n"); /* Enable master mode */ regmap_update_bits(i2s->regmap, LS_I2S_CTRL, I2S_CTRL_MASTER, I2S_CTRL_MASTER); - if (i2s->rev_id == 1) { - ret = regmap_read_poll_timeout_atomic(i2s->regmap, - LS_I2S_CTRL, val, - val & I2S_CTRL_CLK_READY, - 10, 500000); - if (ret < 0) - dev_warn(dai->dev, "wait BCLK ready timeout\n"); - } + + ret = loongson_i2s_enable_bclk(i2s); + if (ret < 0) + dev_warn(dai->dev, "wait BCLK ready timeout\n"); break; default: return -EINVAL; From d01c6a398750aae265c17859b57d7409a6d9181d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:26 +0100 Subject: [PATCH 361/410] ASoC: mt8365: Open code BIT() to avoid spurious warnings The mt8365 driver uses bits.h to define bitfields but BIT() uses unsigned long constants so does not play well with being bitwise negated and converted to an unsigned int, the compiler complains about width reduction on a number of architectures. Just open code the shifting to avoid the issue. Generated with s/BIT(/(1U << / Reported-by: Nathan Chancellor Reviewed-by: Alexandre Mergnat Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-1-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-reg.h | 214 ++++++++++++------------- 1 file changed, 107 insertions(+), 107 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-reg.h b/sound/soc/mediatek/mt8365/mt8365-reg.h index b7334c2e64ed..b763cddc93db 100644 --- a/sound/soc/mediatek/mt8365/mt8365-reg.h +++ b/sound/soc/mediatek/mt8365/mt8365-reg.h @@ -734,57 +734,57 @@ #define AFE_IRQ_STATUS_BITS 0x3ff /* AUDIO_TOP_CON0 (0x0000) */ -#define AUD_TCON0_PDN_TML BIT(27) -#define AUD_TCON0_PDN_DAC_PREDIS BIT(26) -#define AUD_TCON0_PDN_DAC BIT(25) -#define AUD_TCON0_PDN_ADC BIT(24) -#define AUD_TCON0_PDN_TDM_IN BIT(23) -#define AUD_TCON0_PDN_TDM_OUT BIT(22) -#define AUD_TCON0_PDN_SPDIF BIT(21) -#define AUD_TCON0_PDN_APLL_TUNER BIT(19) -#define AUD_TCON0_PDN_APLL2_TUNER BIT(18) -#define AUD_TCON0_PDN_INTDIR BIT(15) -#define AUD_TCON0_PDN_24M BIT(9) -#define AUD_TCON0_PDN_22M BIT(8) -#define AUD_TCON0_PDN_I2S_IN BIT(6) -#define AUD_TCON0_PDN_AFE BIT(2) +#define AUD_TCON0_PDN_TML (1U << 27) +#define AUD_TCON0_PDN_DAC_PREDIS (1U << 26) +#define AUD_TCON0_PDN_DAC (1U << 25) +#define AUD_TCON0_PDN_ADC (1U << 24) +#define AUD_TCON0_PDN_TDM_IN (1U << 23) +#define AUD_TCON0_PDN_TDM_OUT (1U << 22) +#define AUD_TCON0_PDN_SPDIF (1U << 21) +#define AUD_TCON0_PDN_APLL_TUNER (1U << 19) +#define AUD_TCON0_PDN_APLL2_TUNER (1U << 18) +#define AUD_TCON0_PDN_INTDIR (1U << 15) +#define AUD_TCON0_PDN_24M (1U << 9) +#define AUD_TCON0_PDN_22M (1U << 8) +#define AUD_TCON0_PDN_I2S_IN (1U << 6) +#define AUD_TCON0_PDN_AFE (1U << 2) /* AUDIO_TOP_CON1 (0x0004) */ -#define AUD_TCON1_PDN_TDM_ASRC BIT(15) -#define AUD_TCON1_PDN_GENERAL2_ASRC BIT(14) -#define AUD_TCON1_PDN_GENERAL1_ASRC BIT(13) -#define AUD_TCON1_PDN_CONNSYS_I2S_ASRC BIT(12) -#define AUD_TCON1_PDN_DMIC3_ADC BIT(11) -#define AUD_TCON1_PDN_DMIC2_ADC BIT(10) -#define AUD_TCON1_PDN_DMIC1_ADC BIT(9) -#define AUD_TCON1_PDN_DMIC0_ADC BIT(8) -#define AUD_TCON1_PDN_I2S4_BCLK BIT(7) -#define AUD_TCON1_PDN_I2S3_BCLK BIT(6) -#define AUD_TCON1_PDN_I2S2_BCLK BIT(5) -#define AUD_TCON1_PDN_I2S1_BCLK BIT(4) +#define AUD_TCON1_PDN_TDM_ASRC (1U << 15) +#define AUD_TCON1_PDN_GENERAL2_ASRC (1U << 14) +#define AUD_TCON1_PDN_GENERAL1_ASRC (1U << 13) +#define AUD_TCON1_PDN_CONNSYS_I2S_ASRC (1U << 12) +#define AUD_TCON1_PDN_DMIC3_ADC (1U << 11) +#define AUD_TCON1_PDN_DMIC2_ADC (1U << 10) +#define AUD_TCON1_PDN_DMIC1_ADC (1U << 9) +#define AUD_TCON1_PDN_DMIC0_ADC (1U << 8) +#define AUD_TCON1_PDN_I2S4_BCLK (1U << 7) +#define AUD_TCON1_PDN_I2S3_BCLK (1U << 6) +#define AUD_TCON1_PDN_I2S2_BCLK (1U << 5) +#define AUD_TCON1_PDN_I2S1_BCLK (1U << 4) /* AUDIO_TOP_CON3 (0x000C) */ -#define AUD_TCON3_HDMI_BCK_INV BIT(3) +#define AUD_TCON3_HDMI_BCK_INV (1U << 3) /* AFE_I2S_CON (0x0018) */ -#define AFE_I2S_CON_PHASE_SHIFT_FIX BIT(31) -#define AFE_I2S_CON_FROM_IO_MUX BIT(28) -#define AFE_I2S_CON_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON_PHASE_SHIFT_FIX (1U << 31) +#define AFE_I2S_CON_FROM_IO_MUX (1U << 28) +#define AFE_I2S_CON_LOW_JITTER_CLK (1U << 12) #define AFE_I2S_CON_RATE_MASK GENMASK(11, 8) -#define AFE_I2S_CON_FORMAT_I2S BIT(3) -#define AFE_I2S_CON_SRC_SLAVE BIT(2) +#define AFE_I2S_CON_FORMAT_I2S (1U << 3) +#define AFE_I2S_CON_SRC_SLAVE (1U << 2) /* AFE_ASRC_2CH_CON0 */ -#define ONE_HEART BIT(31) -#define CHSET_STR_CLR BIT(4) -#define COEFF_SRAM_CTRL BIT(1) -#define ASM_ON BIT(0) +#define ONE_HEART (1U << 31) +#define CHSET_STR_CLR (1U << 4) +#define COEFF_SRAM_CTRL (1U << 1) +#define ASM_ON (1U << 0) /* CON2 */ -#define O16BIT BIT(19) -#define CLR_IIR_HISTORY BIT(17) -#define IS_MONO BIT(16) -#define IIR_EN BIT(11) +#define O16BIT (1U << 19) +#define CLR_IIR_HISTORY (1U << 17) +#define IS_MONO (1U << 16) +#define IIR_EN (1U << 11) #define IIR_STAGE_MASK GENMASK(10, 8) /* CON5 */ @@ -793,80 +793,80 @@ #define CALI_96_CYCLE FIELD_PREP(CALI_CYCLE_MASK, 0x5F) #define CALI_441_CYCLE FIELD_PREP(CALI_CYCLE_MASK, 0x1B8) -#define CALI_AUTORST BIT(15) -#define AUTO_TUNE_FREQ5 BIT(12) -#define COMP_FREQ_RES BIT(11) +#define CALI_AUTORST (1U << 15) +#define AUTO_TUNE_FREQ5 (1U << 12) +#define COMP_FREQ_RES (1U << 11) #define CALI_SEL_MASK GENMASK(9, 8) #define CALI_SEL_00 FIELD_PREP(CALI_SEL_MASK, 0) #define CALI_SEL_01 FIELD_PREP(CALI_SEL_MASK, 1) -#define CALI_BP_DGL BIT(7) /* Bypass the deglitch circuit */ -#define AUTO_TUNE_FREQ4 BIT(3) -#define CALI_AUTO_RESTART BIT(2) -#define CALI_USE_FREQ_OUT BIT(1) -#define CALI_ON BIT(0) +#define CALI_BP_DGL (1U << 7) /* Bypass the deglitch circuit */ +#define AUTO_TUNE_FREQ4 (1U << 3) +#define CALI_AUTO_RESTART (1U << 2) +#define CALI_USE_FREQ_OUT (1U << 1) +#define CALI_ON (1U << 0) -#define AFE_I2S_CON_WLEN_32BIT BIT(1) -#define AFE_I2S_CON_EN BIT(0) +#define AFE_I2S_CON_WLEN_32BIT (1U << 1) +#define AFE_I2S_CON_EN (1U << 0) -#define AFE_CONN3_I03_O03_S BIT(3) -#define AFE_CONN4_I04_O04_S BIT(4) -#define AFE_CONN4_I03_O04_S BIT(3) +#define AFE_CONN3_I03_O03_S (1U << 3) +#define AFE_CONN4_I04_O04_S (1U << 4) +#define AFE_CONN4_I03_O04_S (1U << 3) /* AFE_I2S_CON1 (0x0034) */ -#define AFE_I2S_CON1_I2S2_TO_PAD BIT(18) +#define AFE_I2S_CON1_I2S2_TO_PAD (1U << 18) #define AFE_I2S_CON1_TDMOUT_TO_PAD (0 << 18) #define AFE_I2S_CON1_RATE GENMASK(11, 8) -#define AFE_I2S_CON1_FORMAT_I2S BIT(3) -#define AFE_I2S_CON1_WLEN_32BIT BIT(1) -#define AFE_I2S_CON1_EN BIT(0) +#define AFE_I2S_CON1_FORMAT_I2S (1U << 3) +#define AFE_I2S_CON1_WLEN_32BIT (1U << 1) +#define AFE_I2S_CON1_EN (1U << 0) /* AFE_I2S_CON2 (0x0038) */ -#define AFE_I2S_CON2_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON2_LOW_JITTER_CLK (1U << 12) #define AFE_I2S_CON2_RATE GENMASK(11, 8) -#define AFE_I2S_CON2_FORMAT_I2S BIT(3) -#define AFE_I2S_CON2_WLEN_32BIT BIT(1) -#define AFE_I2S_CON2_EN BIT(0) +#define AFE_I2S_CON2_FORMAT_I2S (1U << 3) +#define AFE_I2S_CON2_WLEN_32BIT (1U << 1) +#define AFE_I2S_CON2_EN (1U << 0) /* AFE_I2S_CON3 (0x004C) */ -#define AFE_I2S_CON3_LOW_JITTER_CLK BIT(12) +#define AFE_I2S_CON3_LOW_JITTER_CLK (1U << 12) #define AFE_I2S_CON3_RATE GENMASK(11, 8) -#define AFE_I2S_CON3_FORMAT_I2S BIT(3) -#define AFE_I2S_CON3_WLEN_32BIT BIT(1) -#define AFE_I2S_CON3_EN BIT(0) +#define AFE_I2S_CON3_FORMAT_I2S (1U << 3) +#define AFE_I2S_CON3_WLEN_32BIT (1U << 1) +#define AFE_I2S_CON3_EN (1U << 0) /* AFE_ADDA_DL_SRC2_CON0 (0x0108) */ #define AFE_ADDA_DL_SAMPLING_RATE GENMASK(31, 28) #define AFE_ADDA_DL_8X_UPSAMPLE GENMASK(25, 24) -#define AFE_ADDA_DL_MUTE_OFF_CH1 BIT(12) -#define AFE_ADDA_DL_MUTE_OFF_CH2 BIT(11) -#define AFE_ADDA_DL_VOICE_DATA BIT(5) -#define AFE_ADDA_DL_DEGRADE_GAIN BIT(1) +#define AFE_ADDA_DL_MUTE_OFF_CH1 (1U << 12) +#define AFE_ADDA_DL_MUTE_OFF_CH2 (1U << 11) +#define AFE_ADDA_DL_VOICE_DATA (1U << 5) +#define AFE_ADDA_DL_DEGRADE_GAIN (1U << 1) /* AFE_ADDA_UL_SRC_CON0 (0x0114) */ #define AFE_ADDA_UL_SAMPLING_RATE GENMASK(19, 17) /* AFE_ADDA_UL_DL_CON0 */ -#define AFE_ADDA_UL_DL_ADDA_AFE_ON BIT(0) -#define AFE_ADDA_UL_DL_DMIC_CLKDIV_ON BIT(1) +#define AFE_ADDA_UL_DL_ADDA_AFE_ON (1U << 0) +#define AFE_ADDA_UL_DL_DMIC_CLKDIV_ON (1U << 1) /* AFE_APLL_TUNER_CFG (0x03f0) */ #define AFE_APLL_TUNER_CFG_MASK GENMASK(15, 1) -#define AFE_APLL_TUNER_CFG_EN_MASK BIT(0) +#define AFE_APLL_TUNER_CFG_EN_MASK (1U << 0) /* AFE_APLL_TUNER_CFG1 (0x03f4) */ #define AFE_APLL_TUNER_CFG1_MASK GENMASK(15, 1) -#define AFE_APLL_TUNER_CFG1_EN_MASK BIT(0) +#define AFE_APLL_TUNER_CFG1_EN_MASK (1U << 0) /* PCM_INTF_CON1 (0x0550) */ -#define PCM_INTF_CON1_EXT_MODEM BIT(17) +#define PCM_INTF_CON1_EXT_MODEM (1U << 17) #define PCM_INTF_CON1_16BIT (0 << 16) -#define PCM_INTF_CON1_24BIT BIT(16) +#define PCM_INTF_CON1_24BIT (1U << 16) #define PCM_INTF_CON1_32BCK (0 << 14) -#define PCM_INTF_CON1_64BCK BIT(14) +#define PCM_INTF_CON1_64BCK (1U << 14) #define PCM_INTF_CON1_MASTER_MODE (0 << 5) -#define PCM_INTF_CON1_SLAVE_MODE BIT(5) +#define PCM_INTF_CON1_SLAVE_MODE (1U << 5) #define PCM_INTF_CON1_FS_MASK GENMASK(4, 3) #define PCM_INTF_CON1_FS_8K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 0) #define PCM_INTF_CON1_FS_16K FIELD_PREP(PCM_INTF_CON1_FS_MASK, 1) @@ -875,12 +875,12 @@ #define PCM_INTF_CON1_SYNC_LEN_MASK GENMASK(13, 9) #define PCM_INTF_CON1_SYNC_LEN(x) FIELD_PREP(PCM_INTF_CON1_SYNC_LEN_MASK, ((x) - 1)) #define PCM_INTF_CON1_FORMAT_MASK GENMASK(2, 1) -#define PCM_INTF_CON1_SYNC_OUT_INV BIT(23) -#define PCM_INTF_CON1_BCLK_OUT_INV BIT(22) -#define PCM_INTF_CON1_SYNC_IN_INV BIT(21) -#define PCM_INTF_CON1_BCLK_IN_INV BIT(20) -#define PCM_INTF_CON1_BYPASS_ASRC BIT(6) -#define PCM_INTF_CON1_EN BIT(0) +#define PCM_INTF_CON1_SYNC_OUT_INV (1U << 23) +#define PCM_INTF_CON1_BCLK_OUT_INV (1U << 22) +#define PCM_INTF_CON1_SYNC_IN_INV (1U << 21) +#define PCM_INTF_CON1_BCLK_IN_INV (1U << 20) +#define PCM_INTF_CON1_BYPASS_ASRC (1U << 6) +#define PCM_INTF_CON1_EN (1U << 0) #define PCM_INTF_CON1_CONFIG_MASK (0xf3fffe) /* AFE_DMIC0_UL_SRC_CON0 (0x05b4) @@ -890,9 +890,9 @@ */ #define DMIC_TOP_CON_CK_PHASE_SEL_CH1 GENMASK(29, 27) #define DMIC_TOP_CON_CK_PHASE_SEL_CH2 GENMASK(26, 24) -#define DMIC_TOP_CON_TWO_WIRE_MODE BIT(23) -#define DMIC_TOP_CON_CH2_ON BIT(22) -#define DMIC_TOP_CON_CH1_ON BIT(21) +#define DMIC_TOP_CON_TWO_WIRE_MODE (1U << 23) +#define DMIC_TOP_CON_CH2_ON (1U << 22) +#define DMIC_TOP_CON_CH1_ON (1U << 21) #define DMIC_TOP_CON_VOICE_MODE_MASK GENMASK(19, 17) #define DMIC_TOP_CON_VOICE_MODE_8K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 0) #define DMIC_TOP_CON_VOICE_MODE_16K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 1) @@ -900,28 +900,28 @@ #define DMIC_TOP_CON_VOICE_MODE_48K FIELD_PREP(DMIC_TOP_CON_VOICE_MODE_MASK, 3) #define DMIC_TOP_CON_LOW_POWER_MODE_MASK GENMASK(15, 14) #define DMIC_TOP_CON_LOW_POWER_MODE(x) FIELD_PREP(DMIC_TOP_CON_LOW_POWER_MODE_MASK, (x)) -#define DMIC_TOP_CON_IIR_ON BIT(10) +#define DMIC_TOP_CON_IIR_ON (1U << 10) #define DMIC_TOP_CON_IIR_MODE GENMASK(9, 7) -#define DMIC_TOP_CON_INPUT_MODE BIT(5) -#define DMIC_TOP_CON_SDM3_LEVEL_MODE BIT(1) -#define DMIC_TOP_CON_SRC_ON BIT(0) +#define DMIC_TOP_CON_INPUT_MODE (1U << 5) +#define DMIC_TOP_CON_SDM3_LEVEL_MODE (1U << 1) +#define DMIC_TOP_CON_SRC_ON (1U << 0) #define DMIC_TOP_CON_SDM3_DE_SELECT (0 << 1) #define DMIC_TOP_CON_CONFIG_MASK (0x3f8ed7a6) /* AFE_CONN_24BIT (0x0AA4) */ -#define AFE_CONN_24BIT_O10 BIT(10) -#define AFE_CONN_24BIT_O09 BIT(9) -#define AFE_CONN_24BIT_O06 BIT(6) -#define AFE_CONN_24BIT_O05 BIT(5) -#define AFE_CONN_24BIT_O04 BIT(4) -#define AFE_CONN_24BIT_O03 BIT(3) -#define AFE_CONN_24BIT_O02 BIT(2) -#define AFE_CONN_24BIT_O01 BIT(1) -#define AFE_CONN_24BIT_O00 BIT(0) +#define AFE_CONN_24BIT_O10 (1U << 10) +#define AFE_CONN_24BIT_O09 (1U << 9) +#define AFE_CONN_24BIT_O06 (1U << 6) +#define AFE_CONN_24BIT_O05 (1U << 5) +#define AFE_CONN_24BIT_O04 (1U << 4) +#define AFE_CONN_24BIT_O03 (1U << 3) +#define AFE_CONN_24BIT_O02 (1U << 2) +#define AFE_CONN_24BIT_O01 (1U << 1) +#define AFE_CONN_24BIT_O00 (1U << 0) /* AFE_HD_ENGEN_ENABLE */ -#define AFE_22M_PLL_EN BIT(0) -#define AFE_24M_PLL_EN BIT(1) +#define AFE_22M_PLL_EN (1U << 0) +#define AFE_24M_PLL_EN (1U << 1) /* AFE_GAIN1_CON0 (0x0410) */ #define AFE_GAIN1_CON0_EN_MASK GENMASK(0, 0) @@ -938,15 +938,15 @@ /* AFE_CM2_CON0 (0x0e60) */ #define CM_AFE_CM_CH_NUM_MASK GENMASK(3, 0) #define CM_AFE_CM_CH_NUM(x) FIELD_PREP(CM_AFE_CM_CH_NUM_MASK, ((x) - 1)) -#define CM_AFE_CM_ON BIT(4) +#define CM_AFE_CM_ON (1U << 4) #define CM_AFE_CM_START_DATA_MASK GENMASK(11, 8) -#define CM_AFE_CM1_VUL_SEL BIT(12) +#define CM_AFE_CM1_VUL_SEL (1U << 12) #define CM_AFE_CM1_IN_MODE_MASK GENMASK(19, 16) -#define CM_AFE_CM2_TDM_SEL BIT(12) -#define CM_AFE_CM2_CLK_SEL BIT(13) -#define CM_AFE_CM2_GASRC1_OUT_SEL BIT(17) -#define CM_AFE_CM2_GASRC2_OUT_SEL BIT(16) +#define CM_AFE_CM2_TDM_SEL (1U << 12) +#define CM_AFE_CM2_CLK_SEL (1U << 13) +#define CM_AFE_CM2_GASRC1_OUT_SEL (1U << 17) +#define CM_AFE_CM2_GASRC2_OUT_SEL (1U << 16) /* AFE_CM2_CONN* */ #define CM2_AFE_CM2_CONN_CFG1(x) FIELD_PREP(CM2_AFE_CM2_CONN_CFG1_MASK, (x)) From 1b084d8e3b98ca460b815cff14617719ebe605ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:27 +0100 Subject: [PATCH 362/410] ASoC: mt8365: Remove spurious unsigned long casts The regmap APIs take unsigned ints not unsigned longs so casting their arguments to unsigned longs is not a good choice, the constants being cast here are all unsigned ints anyway. Reviewed-by: Alexandre Mergnat Reviewed-by: AngeloGioacchino Del Regno Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-2-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c index 5003fe5e5ccf..6b4d8b7e24ca 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c @@ -385,7 +385,7 @@ static int mt8365_afe_set_2nd_i2s_asrc(struct mtk_base_afe *afe, /* disable IIR coeff SRAM access */ regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, COEFF_SRAM_CTRL, - (unsigned long)~COEFF_SRAM_CTRL); + ~COEFF_SRAM_CTRL); regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, CLR_IIR_HISTORY | IIR_EN | IIR_STAGE_MASK, CLR_IIR_HISTORY | IIR_EN | @@ -393,7 +393,7 @@ static int mt8365_afe_set_2nd_i2s_asrc(struct mtk_base_afe *afe, } else { /* disable IIR */ regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON2, - IIR_EN, (unsigned long)~IIR_EN); + IIR_EN, ~IIR_EN); } /* CON3 setting (RX OFS) */ @@ -456,7 +456,7 @@ static int mt8365_afe_set_2nd_i2s_asrc_enable(struct mtk_base_afe *afe, ASM_ON, ASM_ON); else regmap_update_bits(afe->regmap, AFE_ASRC_2CH_CON0, - ASM_ON, (unsigned long)~ASM_ON); + ASM_ON, ~ASM_ON); return 0; } From 3e61df7d2ff67875c770a7f548038054d05a0f15 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:28 +0100 Subject: [PATCH 363/410] ASoC: mt8365: Remove unused prototype for mt8365_afe_clk_group_48k() The function is not used outside of the file it is defined and the equivalent function for 44.1kHz is not prototyped so remove the prototype for this function. Reviewed-by: AngeloGioacchino Del Regno Reviewed-by: Alexandre Mergnat Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-3-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-common.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-common.h b/sound/soc/mediatek/mt8365/mt8365-afe-common.h index 1fa87e54a57f..731406e15ac7 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-common.h +++ b/sound/soc/mediatek/mt8365/mt8365-afe-common.h @@ -421,7 +421,6 @@ static inline u32 AutoRstThLo(unsigned int fs) } } -bool mt8365_afe_clk_group_48k(int sample_rate); bool mt8365_afe_rate_supported(unsigned int rate, unsigned int id); bool mt8365_afe_channel_supported(unsigned int channel, unsigned int id); From 63157d994025639075b3faa372976a96186322c1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:29 +0100 Subject: [PATCH 364/410] ASoC: mt8365: Make non-exported functions static The compilers warn if functions without a prototype are not static so add appropriate static declarations. Reported-by: Nathan Chancellor Reviewed-by: Alexandre Mergnat Reviewed-by: AngeloGioacchino Del Regno Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-4-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-clk.c | 4 ++-- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 12 ++++++------ 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-clk.c b/sound/soc/mediatek/mt8365/mt8365-afe-clk.c index 300d1f0ae660..8a0af2ea8546 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-clk.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-clk.c @@ -295,7 +295,7 @@ int mt8365_afe_disable_afe_on(struct mtk_base_afe *afe) return 0; } -int mt8365_afe_hd_engen_enable(struct mtk_base_afe *afe, bool apll1) +static int mt8365_afe_hd_engen_enable(struct mtk_base_afe *afe, bool apll1) { if (apll1) regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, @@ -307,7 +307,7 @@ int mt8365_afe_hd_engen_enable(struct mtk_base_afe *afe, bool apll1) return 0; } -int mt8365_afe_hd_engen_disable(struct mtk_base_afe *afe, bool apll1) +static int mt8365_afe_hd_engen_disable(struct mtk_base_afe *afe, bool apll1) { if (apll1) regmap_update_bits(afe->regmap, AFE_HD_ENGEN_ENABLE, diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index df6dd8c5bbe5..54d2112d2e92 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -170,7 +170,7 @@ bool mt8365_afe_channel_supported(unsigned int channel, unsigned int id) return false; } -bool mt8365_afe_clk_group_44k(int sample_rate) +static bool mt8365_afe_clk_group_44k(int sample_rate) { if (sample_rate == 11025 || sample_rate == 22050 || @@ -182,7 +182,7 @@ bool mt8365_afe_clk_group_44k(int sample_rate) return false; } -bool mt8365_afe_clk_group_48k(int sample_rate) +static bool mt8365_afe_clk_group_48k(int sample_rate) { return (!mt8365_afe_clk_group_44k(sample_rate)); } @@ -496,8 +496,8 @@ static int mt8365_afe_configure_cm(struct mtk_base_afe *afe, return 0; } -int mt8365_afe_fe_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) +static int mt8365_afe_fe_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); @@ -714,8 +714,8 @@ static int mt8365_afe_fe_prepare(struct snd_pcm_substream *substream, return 0; } -int mt8365_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) +static int mt8365_afe_fe_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); From 067d832806225cfaabeec8f5f683b7e2bc508a6d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:30 +0100 Subject: [PATCH 365/410] ASoC: mt8365: Remove unused variables Silence compiler warnings by removing unused variables. Reported-by: Nathan Chancellor Reviewed-by: Alexandre Mergnat Reviewed-by: AngeloGioacchino Del Regno Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-5-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 1 - sound/soc/mediatek/mt8365/mt8365-mt6357.c | 3 +-- 2 files changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index 54d2112d2e92..21b1319a6c28 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -651,7 +651,6 @@ static int mt8365_afe_fe_hw_free(struct snd_pcm_substream *substream, struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); struct mt8365_afe_private *afe_priv = afe->platform_priv; int dai_id = snd_soc_rtd_to_cpu(rtd, 0)->id; - struct mtk_base_afe_memif *memif = &afe->memif[dai_id]; struct mt8365_fe_dai_data *fe_data = &afe_priv->fe_data[dai_id]; int ret = 0; diff --git a/sound/soc/mediatek/mt8365/mt8365-mt6357.c b/sound/soc/mediatek/mt8365/mt8365-mt6357.c index fef76118f801..1b8d1656101b 100644 --- a/sound/soc/mediatek/mt8365/mt8365-mt6357.c +++ b/sound/soc/mediatek/mt8365/mt8365-mt6357.c @@ -290,9 +290,8 @@ static int mt8365_mt6357_dev_probe(struct mtk_soc_card_data *soc_card_data, bool struct mtk_platform_card_data *card_data = soc_card_data->card_data; struct snd_soc_card *card = card_data->card; struct device *dev = card->dev; - struct device_node *platform_node; struct mt8365_mt6357_priv *mach_priv; - int i, ret; + int ret; card->dev = dev; ret = parse_dai_link_info(card); From d70ce6d3105a6bd02b1708c399105631643a550a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:31 +0100 Subject: [PATCH 366/410] ASoC: mt8365: Remove unused DMIC IIR coefficient configuration Nothing ever calls mt8365_dai_load_dmic_iirc_coeff_table() so the compiler warns about an unused static function. While it seems likely that something should be calling the function I don't know what and this is breaking -Werror builds like allmodconfig so let's just remove it. It can be added again along with the user. Reported-by: Nathan Chancellor Reviewed-by: AngeloGioacchino Del Regno Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-6-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 30 --------------------- 1 file changed, 30 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c index a3bf54751420..f9945c2a2cd1 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-dmic.c @@ -108,36 +108,6 @@ static void mt8365_dai_disable_dmic(struct mtk_base_afe *afe, regmap_update_bits(afe->regmap, reg, mask, 0); } -static const struct reg_sequence mt8365_dmic_iir_coeff[] = { - { AFE_DMIC0_IIR_COEF_02_01, 0x00000000 }, - { AFE_DMIC0_IIR_COEF_04_03, 0x00003FB8 }, - { AFE_DMIC0_IIR_COEF_06_05, 0x3FB80000 }, - { AFE_DMIC0_IIR_COEF_08_07, 0x3FB80000 }, - { AFE_DMIC0_IIR_COEF_10_09, 0x0000C048 }, - { AFE_DMIC1_IIR_COEF_02_01, 0x00000000 }, - { AFE_DMIC1_IIR_COEF_04_03, 0x00003FB8 }, - { AFE_DMIC1_IIR_COEF_06_05, 0x3FB80000 }, - { AFE_DMIC1_IIR_COEF_08_07, 0x3FB80000 }, - { AFE_DMIC1_IIR_COEF_10_09, 0x0000C048 }, - { AFE_DMIC2_IIR_COEF_02_01, 0x00000000 }, - { AFE_DMIC2_IIR_COEF_04_03, 0x00003FB8 }, - { AFE_DMIC2_IIR_COEF_06_05, 0x3FB80000 }, - { AFE_DMIC2_IIR_COEF_08_07, 0x3FB80000 }, - { AFE_DMIC2_IIR_COEF_10_09, 0x0000C048 }, - { AFE_DMIC3_IIR_COEF_02_01, 0x00000000 }, - { AFE_DMIC3_IIR_COEF_04_03, 0x00003FB8 }, - { AFE_DMIC3_IIR_COEF_06_05, 0x3FB80000 }, - { AFE_DMIC3_IIR_COEF_08_07, 0x3FB80000 }, - { AFE_DMIC3_IIR_COEF_10_09, 0x0000C048 }, -}; - -static int mt8365_dai_load_dmic_iir_coeff_table(struct mtk_base_afe *afe) -{ - return regmap_multi_reg_write(afe->regmap, - mt8365_dmic_iir_coeff, - ARRAY_SIZE(mt8365_dmic_iir_coeff)); -} - static int mt8365_dai_configure_dmic(struct mtk_base_afe *afe, struct snd_pcm_substream *substream, struct snd_soc_dai *dai) From 36fa259b214c37bbae3e0b7a47e7fb49cb0ab462 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 7 Sep 2024 01:53:32 +0100 Subject: [PATCH 367/410] ASoC: mt8365: Allow build coverage There is no build time dependency on anything specific to ARCH_MEDIATEK so enable COMPILE_TEST builds. Reviewed-by: AngeloGioacchino Del Regno Reviewed-by: Alexandre Mergnat Tested-by: Nathan Chancellor # build Signed-off-by: Mark Brown Link: https://patch.msgid.link/20240907-asoc-fix-mt8365-build-v1-7-7ad0bac20161@kernel.org Signed-off-by: Mark Brown --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index e6f7a5a49794..3033e2d3fe16 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -301,7 +301,7 @@ config SND_SOC_MT8195_MT6359 config SND_SOC_MT8365 tristate "ASoC support for MediaTek MT8365 chip" - depends on ARCH_MEDIATEK + depends on ARCH_MEDIATEK || COMPILE_TEST select SND_SOC_MEDIATEK help This adds ASoC platform driver support for MediaTek MT8365 chip From 130eb72d3cb38a629205a2a192800b4b1b9bc5c9 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Sun, 8 Sep 2024 21:46:04 +0800 Subject: [PATCH 368/410] ASoC: codecs: fix the right check and simplify code In the file drivers/base/platform.c, the return description of platform_get_irq is 'non-zero IRQ number on success, negative error number on failure.', so the check is wrong, fix it. And when get irq failed, the function platform_get_irq logs an error message. Fixes: 5e2404493f9f ("ASoC: codecs: add MT6357 support") Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240908134604.3652-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index df6dd8c5bbe5..4ec86b71dd88 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -2155,11 +2155,11 @@ static int mt8365_afe_pcm_dev_probe(struct platform_device *pdev) for (i = 0; i < afe->irqs_size; i++) afe->irqs[i].irq_data = &irq_data[i]; - irq_id = platform_get_irq(pdev, 0); - if (!irq_id) { - dev_err_probe(afe->dev, irq_id, "np %s no irq\n", afe->dev->of_node->name); - return -ENXIO; - } + ret = platform_get_irq(pdev, 0); + if (ret < 0) + return ret; + + irq_id = ret; ret = devm_request_irq(afe->dev, irq_id, mt8365_afe_irq_handler, 0, "Afe_ISR_Handle", (void *)afe); if (ret) From b09c71f3e8413ac0a9749f9d0d06f6f0d0b2cc65 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Mon, 9 Sep 2024 11:35:29 +0300 Subject: [PATCH 369/410] ASoC: atmel: mchp-i2s-mcc: Remove interface name from stream_name Remove the interface name from the stream_name. The interface name (and the index of the interface) can be set in DT using the sound-name-prefix string property. [andrei.simion@microchip: Adjust the commit title] Signed-off-by: Codrin Ciubotariu Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20240909083530.14695-2-andrei.simion@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-i2s-mcc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 35a56b266b8d..f40508668ac5 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -936,14 +936,14 @@ static const struct snd_soc_dai_ops mchp_i2s_mcc_dai_ops = { static struct snd_soc_dai_driver mchp_i2s_mcc_dai = { .playback = { - .stream_name = "I2SMCC-Playback", + .stream_name = "Playback", .channels_min = 1, .channels_max = 8, .rates = MCHP_I2SMCC_RATES, .formats = MCHP_I2SMCC_FORMATS, }, .capture = { - .stream_name = "I2SMCC-Capture", + .stream_name = "Capture", .channels_min = 1, .channels_max = 8, .rates = MCHP_I2SMCC_RATES, From 130af75b5c05eef4ecd8593371f3e924bcd41241 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Mon, 9 Sep 2024 17:12:30 +0200 Subject: [PATCH 370/410] ASoC: Switch back to struct platform_driver::remove() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit After commit 0edb555a65d1 ("platform: Make platform_driver::remove() return void") .remove() is (again) the right callback to implement for platform drivers. Convert all drivers below sound/soc to use .remove(), with the eventual goal to drop struct platform_driver::remove_new(). As .remove() and .remove_new() have the same prototypes, conversion is done by just changing the structure member name in the driver initializer. Signed-off-by: Uwe Kleine-König Link: https://patch.msgid.link/20240909151230.909818-2-u.kleine-koenig@baylibre.com Signed-off-by: Mark Brown --- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 2 +- sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- sound/soc/apple/mca.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l85.c | 2 +- sound/soc/codecs/cs47l90.c | 2 +- sound/soc/codecs/cs47l92.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rk817_codec.c | 2 +- sound/soc/codecs/wcd937x.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_audmix.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_rpmsg.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2-custom-sample.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/test-component.c | 2 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/img/pistachio-internal-dac.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2 +- sound/soc/meson/aiu.c | 2 +- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/gx-card.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/arndale.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/imx/imx8ulp.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sprd/sprd-mcdt.c | 2 +- sound/soc/starfive/jh7110_pwmdac.c | 2 +- sound/soc/starfive/jh7110_tdm.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra186_asrc.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/tegra/tegra210_amx.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra210_mixer.c | 2 +- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_ope.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/soc/xilinx/xlnx_spdif.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 184 files changed, 184 insertions(+), 184 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 4107eddc9ca8..41f89384f8fd 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -293,7 +293,7 @@ static struct platform_driver axi_i2s_driver = { .of_match_table = axi_i2s_of_match, }, .probe = axi_i2s_probe, - .remove_new = axi_i2s_dev_remove, + .remove = axi_i2s_dev_remove, }; module_platform_driver(axi_i2s_driver); diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 10545bd99704..5581134201a3 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -258,7 +258,7 @@ static struct platform_driver axi_spdif_driver = { .of_match_table = axi_spdif_of_match, }, .probe = axi_spdif_probe, - .remove_new = axi_spdif_dev_remove, + .remove = axi_spdif_dev_remove, }; module_platform_driver(axi_spdif_driver); diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b857e2676fe8..897dde630022 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1426,7 +1426,7 @@ static const struct dev_pm_ops acp_pm_ops = { static struct platform_driver acp_dma_driver = { .probe = acp_audio_probe, - .remove_new = acp_audio_remove, + .remove = acp_audio_remove, .driver = { .name = DRV_NAME, .pm = &acp_pm_ops, diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index e19981c7d65a..396434a45eea 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -295,7 +295,7 @@ static const struct dev_pm_ops rmb_dma_pm_ops = { static struct platform_driver rembrandt_driver = { .probe = rembrandt_audio_probe, - .remove_new = rembrandt_audio_remove, + .remove = rembrandt_audio_remove, .driver = { .name = "acp_asoc_rembrandt", .pm = &rmb_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index db835ed7c208..5e3f730aa6bf 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -244,7 +244,7 @@ static const struct dev_pm_ops rn_dma_pm_ops = { static struct platform_driver renoir_driver = { .probe = renoir_audio_probe, - .remove_new = renoir_audio_remove, + .remove = renoir_audio_remove, .driver = { .name = "acp_asoc_renoir", .pm = &rn_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 08f368b3bbc8..b1cd173f607d 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -729,7 +729,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/amd/acp/acp63.c b/sound/soc/amd/acp/acp63.c index f340920b3289..f325c374f228 100644 --- a/sound/soc/amd/acp/acp63.c +++ b/sound/soc/amd/acp/acp63.c @@ -304,7 +304,7 @@ static const struct dev_pm_ops acp63_dma_pm_ops = { static struct platform_driver acp63_driver = { .probe = acp63_audio_probe, - .remove_new = acp63_audio_remove, + .remove = acp63_audio_remove, .driver = { .name = "acp_asoc_acp63", .pm = &acp63_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 1b0b59a22924..68d2590e1a4e 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -278,7 +278,7 @@ static const struct dev_pm_ops acp70_dma_pm_ops = { static struct platform_driver acp70_driver = { .probe = acp_acp70_audio_probe, - .remove_new = acp_acp70_audio_remove, + .remove = acp_acp70_audio_remove, .driver = { .name = "acp_asoc_acp70", .pm = &acp70_dma_pm_ops, diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 7bbacbab1095..318fc260f293 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -448,7 +448,7 @@ static const struct dev_pm_ops acp63_pdm_pm_ops = { static struct platform_driver acp63_pdm_dma_driver = { .probe = acp63_pdm_audio_probe, - .remove_new = acp63_pdm_audio_remove, + .remove = acp63_pdm_audio_remove, .driver = { .name = "acp_ps_pdm_dma", .pm = &acp63_pdm_pm_ops, diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 2f630753278d..3b4b9c6b3171 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -551,7 +551,7 @@ static const struct dev_pm_ops acp63_pm_ops = { static struct platform_driver acp63_sdw_dma_driver = { .probe = acp63_sdw_platform_probe, - .remove_new = acp63_sdw_platform_remove, + .remove = acp63_sdw_platform_remove, .driver = { .name = "amd_ps_sdw_dma", .pm = &acp63_pm_ops, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3a50558f6751..bb9ed52d744d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -509,7 +509,7 @@ static const struct dev_pm_ops acp3x_pm_ops = { static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, - .remove_new = acp3x_audio_remove, + .remove = acp3x_audio_remove, .driver = { .name = "acp3x_rv_i2s_dma", .pm = &acp3x_pm_ops, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index c3b47e9bd239..95ac8c680037 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -489,7 +489,7 @@ static const struct dev_pm_ops acp_pdm_pm_ops = { static struct platform_driver acp_pdm_dma_driver = { .probe = acp_pdm_audio_probe, - .remove_new = acp_pdm_audio_remove, + .remove = acp_pdm_audio_remove, .driver = { .name = "acp_rn_pdm_dma", .pm = &acp_pdm_pm_ops, diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 491b16e52a72..d5965f2b09bc 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -499,7 +499,7 @@ static const struct dev_pm_ops acp5x_pm_ops = { static struct platform_driver acp5x_dma_driver = { .probe = acp5x_audio_probe, - .remove_new = acp5x_audio_remove, + .remove = acp5x_audio_remove, .driver = { .name = "acp5x_i2s_dma", .pm = &acp5x_pm_ops, diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index 72c4591e451b..3eb3e82efb10 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -440,7 +440,7 @@ static const struct dev_pm_ops acp6x_pdm_pm_ops = { static struct platform_driver acp6x_pdm_dma_driver = { .probe = acp6x_pdm_audio_probe, - .remove_new = acp6x_pdm_audio_remove, + .remove = acp6x_pdm_audio_remove, .driver = { .name = "acp_yc_pdm_dma", .pm = &acp6x_pdm_pm_ops, diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 3780aca71076..c9e7d40c47cc 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -1179,7 +1179,7 @@ static struct platform_driver apple_mca_driver = { .of_match_table = apple_mca_of_match, }, .probe = apple_mca_probe, - .remove_new = apple_mca_remove, + .remove = apple_mca_remove, }; module_platform_driver(apple_mca_driver); diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6c20c643f321..762199faf872 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -733,7 +733,7 @@ static struct platform_driver atmel_i2s_driver = { .of_match_table = atmel_i2s_dt_ids, }, .probe = atmel_i2s_probe, - .remove_new = atmel_i2s_remove, + .remove = atmel_i2s_remove, }; module_platform_driver(atmel_i2s_driver); diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index b7f16ea0cdfc..0f4021c6c588 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -187,7 +187,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, - .remove_new = atmel_asoc_wm8904_remove, + .remove = atmel_asoc_wm8904_remove, }; module_platform_driver(atmel_asoc_wm8904_driver); diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index f40508668ac5..17d138bb9064 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1129,7 +1129,7 @@ static struct platform_driver mchp_i2s_mcc_driver = { .of_match_table = mchp_i2s_mcc_dt_ids, }, .probe = mchp_i2s_mcc_probe, - .remove_new = mchp_i2s_mcc_remove, + .remove = mchp_i2s_mcc_remove, }; module_platform_driver(mchp_i2s_mcc_driver); diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index dcc4e14b3dde..260074018da9 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -1153,7 +1153,7 @@ static struct platform_driver mchp_pdmc_driver = { .pm = pm_ptr(&mchp_pdmc_pm_ops), }, .probe = mchp_pdmc_probe, - .remove_new = mchp_pdmc_remove, + .remove = mchp_pdmc_remove, }; module_platform_driver(mchp_pdmc_driver); diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 33ce5e54482b..b2507a1491b7 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -1194,7 +1194,7 @@ static void mchp_spdifrx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdifrx_driver = { .probe = mchp_spdifrx_probe, - .remove_new = mchp_spdifrx_remove, + .remove = mchp_spdifrx_remove, .driver = { .name = "mchp_spdifrx", .of_match_table = mchp_spdifrx_dt_ids, diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index a201a96fa690..4c60ea652896 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -888,7 +888,7 @@ static void mchp_spdiftx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdiftx_driver = { .probe = mchp_spdiftx_probe, - .remove_new = mchp_spdiftx_remove, + .remove = mchp_spdiftx_remove, .driver = { .name = "mchp_spdiftx", .of_match_table = mchp_spdiftx_dt_ids, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index d3ec9826d505..335e216ea7b4 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static struct platform_driver at91sam9g20ek_audio_driver = { .of_match_table = of_match_ptr(at91sam9g20ek_wm8731_dt_ids), }, .probe = at91sam9g20ek_audio_probe, - .remove_new = at91sam9g20ek_audio_remove, + .remove = at91sam9g20ek_audio_remove, }; module_platform_driver(at91sam9g20ek_audio_driver); diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index d1c1f370a9cd..1b5ef4e9d2b8 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -196,7 +196,7 @@ static struct platform_driver sam9x5_wm8731_driver = { .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), }, .probe = sam9x5_wm8731_driver_probe, - .remove_new = sam9x5_wm8731_driver_remove, + .remove = sam9x5_wm8731_driver_remove, }; module_platform_driver(sam9x5_wm8731_driver); diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 5d208e0b4b90..0a9efd5f2861 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -431,7 +431,7 @@ static struct platform_driver tse850_driver = { .of_match_table = tse850_dt_ids, }, .probe = tse850_probe, - .remove_new = tse850_remove, + .remove = tse850_remove, }; module_platform_driver(tse850_driver); diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index b0e1a1253e10..f8ab936250dc 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -336,7 +336,7 @@ static struct platform_driver au1xac97c_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xac97c_drvprobe, - .remove_new = au1xac97c_drvremove, + .remove = au1xac97c_drvremove, }; module_platform_driver(au1xac97c_driver); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 064406080d72..7d296f29dade 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -313,7 +313,7 @@ static struct platform_driver au1xi2s_driver = { .pm = AU1XI2SC_PMOPS, }, .probe = au1xi2s_drvprobe, - .remove_new = au1xi2s_drvremove, + .remove = au1xi2s_drvremove, }; module_platform_driver(au1xi2s_driver); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 1727eeb12b64..8a59a50978b9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -486,7 +486,7 @@ static struct platform_driver au1xpsc_ac97_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xpsc_ac97_drvprobe, - .remove_new = au1xpsc_ac97_drvremove, + .remove = au1xpsc_ac97_drvremove, }; module_platform_driver(au1xpsc_ac97_driver); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 52734dec8247..bee013555e7a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -404,7 +404,7 @@ static struct platform_driver au1xpsc_i2s_driver = { .pm = AU1XPSCI2S_PMOPS, }, .probe = au1xpsc_i2s_drvprobe, - .remove_new = au1xpsc_i2s_drvremove, + .remove = au1xpsc_i2s_drvremove, }; module_platform_driver(au1xpsc_i2s_driver); diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index c64609718738..c47ed1e6ea2b 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -293,7 +293,7 @@ static struct platform_driver bcm63xx_i2s_driver = { .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), }, .probe = bcm63xx_i2s_dev_probe, - .remove_new = bcm63xx_i2s_dev_remove, + .remove = bcm63xx_i2s_dev_remove, }; module_platform_driver(bcm63xx_i2s_driver); diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 90088516fed0..e0ce0232eb1e 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1390,7 +1390,7 @@ MODULE_DEVICE_TABLE(of, cygnus_ssp_of_match); static struct platform_driver cygnus_ssp_driver = { .probe = cygnus_ssp_probe, - .remove_new = cygnus_ssp_remove, + .remove = cygnus_ssp_remove, .driver = { .name = "cygnus-ssp", .of_match_table = cygnus_ssp_of_match, diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 8bb67d7d2b4b..8dac754ddb0d 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -105,7 +105,7 @@ static struct platform_driver edb93xx_driver = { .name = "edb93xx-audio", }, .probe = edb93xx_probe, - .remove_new = edb93xx_remove, + .remove = edb93xx_remove, }; module_platform_driver(edb93xx_driver); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 522de4b80293..d45862ceb0c9 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -523,7 +523,7 @@ MODULE_DEVICE_TABLE(of, ep93xx_i2s_of_ids); static struct platform_driver ep93xx_i2s_driver = { .probe = ep93xx_i2s_probe, - .remove_new = ep93xx_i2s_remove, + .remove = ep93xx_i2s_remove, .driver = { .name = "ep93xx-i2s", .of_match_table = ep93xx_i2s_of_ids, diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 5183b4586424..d0098b4558b5 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2461,7 +2461,7 @@ static struct platform_driver cs42l43_codec_driver = { }, .probe = cs42l43_codec_probe, - .remove_new = cs42l43_codec_remove, + .remove = cs42l43_codec_remove, .id_table = cs42l43_codec_id_table, }; module_platform_driver(cs42l43_codec_driver); diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index ab6e7cd99733..29a2bcfb3048 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1493,7 +1493,7 @@ static struct platform_driver cs47l15_codec_driver = { .name = "cs47l15-codec", }, .probe = &cs47l15_probe, - .remove_new = cs47l15_remove, + .remove = cs47l15_remove, }; module_platform_driver(cs47l15_codec_driver); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index ec405ef66a8e..e2a839fae4fc 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1344,7 +1344,7 @@ static struct platform_driver cs47l24_codec_driver = { .name = "cs47l24-codec", }, .probe = cs47l24_probe, - .remove_new = cs47l24_remove, + .remove = cs47l24_remove, }; module_platform_driver(cs47l24_codec_driver); diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 0d7ee7ea6257..85555c7a2e4b 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1769,7 +1769,7 @@ static struct platform_driver cs47l35_codec_driver = { .name = "cs47l35-codec", }, .probe = &cs47l35_probe, - .remove_new = cs47l35_remove, + .remove = cs47l35_remove, }; module_platform_driver(cs47l35_codec_driver); diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index 2dfb867e6edd..d34f4e8c26d3 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2720,7 +2720,7 @@ static struct platform_driver cs47l85_codec_driver = { .name = "cs47l85-codec", }, .probe = &cs47l85_probe, - .remove_new = cs47l85_remove, + .remove = cs47l85_remove, }; module_platform_driver(cs47l85_codec_driver); diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 2549cb1fc121..a9e703981f37 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2644,7 +2644,7 @@ static struct platform_driver cs47l90_codec_driver = { .name = "cs47l90-codec", }, .probe = &cs47l90_probe, - .remove_new = cs47l90_remove, + .remove = cs47l90_remove, }; module_platform_driver(cs47l90_codec_driver); diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 0c05ae0b09fb..2c355c61acd8 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -2092,7 +2092,7 @@ static struct platform_driver cs47l92_codec_driver = { .name = "cs47l92-codec", }, .probe = &cs47l92_probe, - .remove_new = cs47l92_remove, + .remove = cs47l92_remove, }; module_platform_driver(cs47l92_codec_driver); diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 11320423c69c..fdd19f8e8864 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -476,7 +476,7 @@ static struct platform_driver rk3036_codec_platform_driver = { .of_match_table = of_match_ptr(rk3036_codec_of_match), }, .probe = rk3036_codec_platform_probe, - .remove_new = rk3036_codec_platform_remove, + .remove = rk3036_codec_platform_remove, }; module_platform_driver(rk3036_codec_platform_driver); diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ce42749660c8..71e0d3bffd3f 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -4024,7 +4024,7 @@ static struct platform_driver rx_macro_driver = { .pm = &rx_macro_pm_ops, }, .probe = rx_macro_probe, - .remove_new = rx_macro_remove, + .remove = rx_macro_remove, }; module_platform_driver(rx_macro_driver); diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 209c12ec16dd..a134584acf90 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -2534,7 +2534,7 @@ static struct platform_driver tx_macro_driver = { .pm = &tx_macro_pm_ops, }, .probe = tx_macro_probe, - .remove_new = tx_macro_remove, + .remove = tx_macro_remove, }; module_platform_driver(tx_macro_driver); diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..24689cad0094 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1729,7 +1729,7 @@ static struct platform_driver va_macro_driver = { .pm = &va_macro_pm_ops, }, .probe = va_macro_probe, - .remove_new = va_macro_remove, + .remove = va_macro_remove, }; module_platform_driver(va_macro_driver); diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 76900274acf3..c989d82d1d3c 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2980,7 +2980,7 @@ static struct platform_driver wsa_macro_driver = { .pm = &wsa_macro_pm_ops, }, .probe = wsa_macro_probe, - .remove_new = wsa_macro_remove, + .remove = wsa_macro_remove, }; module_platform_driver(wsa_macro_driver); diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 978c4d056e81..ebb6f2e84818 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1241,7 +1241,7 @@ static struct platform_driver msm8916_wcd_digital_driver = { .of_match_table = msm8916_wcd_digital_match_table, }, .probe = msm8916_wcd_digital_probe, - .remove_new = msm8916_wcd_digital_remove, + .remove = msm8916_wcd_digital_remove, }; module_platform_driver(msm8916_wcd_digital_driver); diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 5fea600bc3a4..3c5b66357661 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -529,7 +529,7 @@ static struct platform_driver rk817_codec_driver = { .name = "rk817-codec", }, .probe = rk817_platform_probe, - .remove_new = rk817_platform_remove, + .remove = rk817_platform_remove, }; module_platform_driver(rk817_codec_driver); diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index af296b77a723..45f32d281908 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2957,7 +2957,7 @@ MODULE_DEVICE_TABLE(of, wcd937x_of_match); static struct platform_driver wcd937x_codec_driver = { .probe = wcd937x_probe, - .remove_new = wcd937x_remove, + .remove = wcd937x_remove, .driver = { .name = "wcd937x_codec", .of_match_table = of_match_ptr(wcd937x_of_match), diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index aa92f1bfc921..f2a4f3262bdb 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3596,7 +3596,7 @@ MODULE_DEVICE_TABLE(of, wcd938x_dt_match); static struct platform_driver wcd938x_codec_driver = { .probe = wcd938x_probe, - .remove_new = wcd938x_remove, + .remove = wcd938x_remove, .driver = { .name = "wcd938x_codec", .of_match_table = of_match_ptr(wcd938x_dt_match), diff --git a/sound/soc/codecs/wcd939x.c b/sound/soc/codecs/wcd939x.c index 68fc591670dc..4a417a92514d 100644 --- a/sound/soc/codecs/wcd939x.c +++ b/sound/soc/codecs/wcd939x.c @@ -3683,7 +3683,7 @@ MODULE_DEVICE_TABLE(of, wcd939x_dt_match); static struct platform_driver wcd939x_codec_driver = { .probe = wcd939x_probe, - .remove_new = wcd939x_remove, + .remove = wcd939x_remove, .driver = { .name = "wcd939x_codec", .of_match_table = of_match_ptr(wcd939x_dt_match), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4ecf07c7448c..651f1319204d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2174,7 +2174,7 @@ static struct platform_driver wm5102_codec_driver = { .name = "wm5102-codec", }, .probe = wm5102_probe, - .remove_new = wm5102_remove, + .remove = wm5102_remove, }; module_platform_driver(wm5102_codec_driver); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0f299cd07b2e..502196253d42 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2534,7 +2534,7 @@ static struct platform_driver wm5110_codec_driver = { .name = "wm5110-codec", }, .probe = wm5110_probe, - .remove_new = wm5110_remove, + .remove = wm5110_remove, }; module_platform_driver(wm5110_codec_driver); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a99908582a50..a4abe6e53bfc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4699,7 +4699,7 @@ static struct platform_driver wm8994_codec_driver = { .pm = &wm8994_pm_ops, }, .probe = wm8994_probe, - .remove_new = wm8994_remove, + .remove = wm8994_remove, }; module_platform_driver(wm8994_codec_driver); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 87442840f0af..5389c363b14e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1210,7 +1210,7 @@ static struct platform_driver wm8997_codec_driver = { .name = "wm8997-codec", }, .probe = wm8997_probe, - .remove_new = wm8997_remove, + .remove = wm8997_remove, }; module_platform_driver(wm8997_codec_driver); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 3c2c4d12c08e..b72b8a64be8f 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1426,7 +1426,7 @@ static struct platform_driver wm8998_codec_driver = { .name = "wm8998-codec", }, .probe = wm8998_probe, - .remove_new = wm8998_remove, + .remove = wm8998_remove, }; module_platform_driver(wm8998_codec_driver); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e6a5eebe5bc9..57b789d7fbed 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -1089,7 +1089,7 @@ static const struct dev_pm_ops dwc_pm_ops = { static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, - .remove_new = dw_i2s_remove, + .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .of_match_table = of_match_ptr(dw_i2s_of_match), diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b793263291dc..bd5c46d763c0 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1392,7 +1392,7 @@ MODULE_DEVICE_TABLE(of, fsl_asrc_ids); static struct platform_driver fsl_asrc_driver = { .probe = fsl_asrc_probe, - .remove_new = fsl_asrc_remove, + .remove = fsl_asrc_remove, .driver = { .name = "fsl-asrc", .of_match_table = fsl_asrc_ids, diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..021d73e409aa 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -296,7 +296,7 @@ static const struct dev_pm_ops fsl_aud2htx_pm_ops = { static struct platform_driver fsl_aud2htx_driver = { .probe = fsl_aud2htx_probe, - .remove_new = fsl_aud2htx_remove, + .remove = fsl_aud2htx_remove, .driver = { .name = "fsl-aud2htx", .pm = pm_ptr(&fsl_aud2htx_pm_ops), diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index f3a24758aedb..3cd9a66b70a1 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -548,7 +548,7 @@ static const struct dev_pm_ops fsl_audmix_pm = { static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, - .remove_new = fsl_audmix_remove, + .remove = fsl_audmix_remove, .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c4bc9395dff7..aca066b5a43c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -911,7 +911,7 @@ static struct platform_driver fsl_soc_dma_driver = { .of_match_table = fsl_soc_dma_ids, }, .probe = fsl_soc_dma_probe, - .remove_new = fsl_soc_dma_remove, + .remove = fsl_soc_dma_remove, }; module_platform_driver(fsl_soc_dma_driver); diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..82359edd6a8b 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -2093,7 +2093,7 @@ static const struct dev_pm_ops fsl_easrc_pm_ops = { static struct platform_driver fsl_easrc_driver = { .probe = fsl_easrc_probe, - .remove_new = fsl_easrc_remove, + .remove = fsl_easrc_remove, .driver = { .name = "fsl-easrc", .pm = pm_ptr(&fsl_easrc_pm_ops), diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0d8a01da9bd..a65f5b9935a2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1198,7 +1198,7 @@ static const struct dev_pm_ops fsl_esai_pm_ops = { static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, - .remove_new = fsl_esai_remove, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..193be098fa5e 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1317,7 +1317,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = { static struct platform_driver fsl_micfil_driver = { .probe = fsl_micfil_probe, - .remove_new = fsl_micfil_remove, + .remove = fsl_micfil_remove, .driver = { .name = "fsl-micfil-dai", .pm = &fsl_micfil_pm_ops, diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index df160834c81b..145f9ca15e43 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -381,7 +381,7 @@ MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); static struct platform_driver fsl_mqs_driver = { .probe = fsl_mqs_probe, - .remove_new = fsl_mqs_remove, + .remove = fsl_mqs_remove, .driver = { .name = "fsl-mqs", .of_match_table = fsl_mqs_dt_ids, diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index be46fbfd487a..0a551be3053b 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -328,7 +328,7 @@ static const struct dev_pm_ops fsl_rpmsg_pm_ops = { static struct platform_driver fsl_rpmsg_driver = { .probe = fsl_rpmsg_probe, - .remove_new = fsl_rpmsg_remove, + .remove = fsl_rpmsg_remove, .driver = { .name = "fsl_rpmsg", .pm = pm_ptr(&fsl_rpmsg_pm_ops), diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad..ab58a4461073 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1817,7 +1817,7 @@ static const struct dev_pm_ops fsl_sai_pm_ops = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove_new = fsl_sai_remove, + .remove = fsl_sai_remove, .driver = { .name = "fsl-sai", .pm = &fsl_sai_pm_ops, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index eace399cb064..b6ff04f7138a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1763,7 +1763,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = pm_ptr(&fsl_spdif_pm), }, .probe = fsl_spdif_probe, - .remove_new = fsl_spdif_remove, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..320108bebf30 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1734,7 +1734,7 @@ static struct platform_driver fsl_ssi_driver = { .pm = pm_sleep_ptr(&fsl_ssi_pm), }, .probe = fsl_ssi_probe, - .remove_new = fsl_ssi_remove, + .remove = fsl_ssi_remove, }; module_platform_driver(fsl_ssi_driver); diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index bf9a4e90978e..e9efe51d8fd0 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1540,7 +1540,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = pm_ptr(&fsl_xcvr_pm_ops), .of_match_table = fsl_xcvr_dt_ids, }, - .remove_new = fsl_xcvr_remove, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index f97ae14dc452..43e14f2eca8d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -354,7 +354,7 @@ static const struct dev_pm_ops imx_audmux_pm = { static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove_new = imx_audmux_remove, + .remove = imx_audmux_remove, .driver = { .name = DRIVER_NAME, .pm = pm_sleep_ptr(&imx_audmux_pm), diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 22156f99dcee..1daf0be3d100 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -822,7 +822,7 @@ MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table); static struct platform_driver imx_pcm_rpmsg_driver = { .probe = imx_rpmsg_pcm_probe, - .remove_new = imx_rpmsg_pcm_remove, + .remove = imx_rpmsg_pcm_remove, .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3c1e69092d2f..8bcf54ef709e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -214,7 +214,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove_new = imx_sgtl5000_remove, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 0423cf43c7a0..8554fb690772 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -327,7 +327,7 @@ MODULE_DEVICE_TABLE(of, psc_ac97_match); static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, - .remove_new = psc_ac97_of_remove, + .remove = psc_ac97_of_remove, .driver = { .name = "mpc5200-psc-ac97", .of_match_table = psc_ac97_match, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 931b430fa983..9ad44eeed6ad 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -225,7 +225,7 @@ MODULE_DEVICE_TABLE(of, psc_i2s_match); static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, - .remove_new = psc_i2s_of_remove, + .remove = psc_i2s_of_remove, .driver = { .name = "mpc5200-psc-i2s", .of_match_table = psc_i2s_match, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 6f5eecf6d88c..66db05970d82 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -408,7 +408,7 @@ static void p1022_ds_remove(struct platform_device *pdev) static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, - .remove_new = p1022_ds_remove, + .remove = p1022_ds_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index a42149311c8b..d4568346714f 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -370,7 +370,7 @@ static void p1022_rdk_remove(struct platform_device *pdev) static struct platform_driver p1022_rdk_driver = { .probe = p1022_rdk_probe, - .remove_new = p1022_rdk_remove, + .remove = p1022_rdk_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 2bab0fc1de59..5542c4ee6d12 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -124,7 +124,7 @@ MODULE_DEVICE_TABLE(of, pcm030_audio_match); static struct platform_driver pcm030_fabric_driver = { .probe = pcm030_fabric_probe, - .remove_new = pcm030_fabric_remove, + .remove = pcm030_fabric_remove, .driver = { .name = DRV_NAME, .of_match_table = pcm030_audio_match, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 03add80e75f3..4b384475b972 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -664,7 +664,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.c b/sound/soc/generic/audio-graph-card2-custom-sample.c index 8e5a51098490..7151d426bee9 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.c +++ b/sound/soc/generic/audio-graph-card2-custom-sample.c @@ -177,7 +177,7 @@ static struct platform_driver custom_card = { .of_match_table = custom_of_match, }, .probe = custom_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(custom_card); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index bf4c1ab4ec58..4ad3d1b0714f 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1438,7 +1438,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 42c60b92cca5..76a1d05e2ebe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -838,7 +838,7 @@ static struct platform_driver simple_card = { .of_match_table = simple_of_match, }, .probe = simple_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(simple_card); diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index df2487b700cc..407288055741 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -637,7 +637,7 @@ static struct platform_driver test_driver = { .of_match_table = test_of_match, }, .probe = test_driver_probe, - .remove_new = test_driver_remove, + .remove = test_driver_remove, }; module_platform_driver(test_driver); diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index b69a364d619e..6a988976fb0d 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_in_driver = { .pm = &img_i2s_in_pm_ops }, .probe = img_i2s_in_probe, - .remove_new = img_i2s_in_dev_remove + .remove = img_i2s_in_dev_remove }; module_platform_driver(img_i2s_in_driver); diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 6f9831c6d6e0..1211e6184d97 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_out_driver = { .pm = &img_i2s_out_pm_ops }, .probe = img_i2s_out_probe, - .remove_new = img_i2s_out_dev_remove + .remove = img_i2s_out_dev_remove }; module_platform_driver(img_i2s_out_driver); diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 815e68a7048c..4ec63119d67c 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -311,7 +311,7 @@ static struct platform_driver img_prl_out_driver = { .pm = &img_prl_out_pm_ops }, .probe = img_prl_out_probe, - .remove_new = img_prl_out_dev_remove + .remove = img_prl_out_dev_remove }; module_platform_driver(img_prl_out_driver); diff --git a/sound/soc/img/img-spdif-in.c b/sound/soc/img/img-spdif-in.c index 9646e9d3f0bc..3c513f5b8c54 100644 --- a/sound/soc/img/img-spdif-in.c +++ b/sound/soc/img/img-spdif-in.c @@ -878,7 +878,7 @@ static struct platform_driver img_spdif_in_driver = { .pm = &img_spdif_in_pm_ops }, .probe = img_spdif_in_probe, - .remove_new = img_spdif_in_dev_remove + .remove = img_spdif_in_dev_remove }; module_platform_driver(img_spdif_in_driver); diff --git a/sound/soc/img/img-spdif-out.c b/sound/soc/img/img-spdif-out.c index dfa72afa946e..402695b5fc41 100644 --- a/sound/soc/img/img-spdif-out.c +++ b/sound/soc/img/img-spdif-out.c @@ -468,7 +468,7 @@ static struct platform_driver img_spdif_out_driver = { .pm = &img_spdif_out_pm_ops }, .probe = img_spdif_out_probe, - .remove_new = img_spdif_out_dev_remove + .remove = img_spdif_out_dev_remove }; module_platform_driver(img_spdif_out_driver); diff --git a/sound/soc/img/pistachio-internal-dac.c b/sound/soc/img/pistachio-internal-dac.c index da6251680e41..fdeceb271e7f 100644 --- a/sound/soc/img/pistachio-internal-dac.c +++ b/sound/soc/img/pistachio-internal-dac.c @@ -271,7 +271,7 @@ static struct platform_driver pistachio_internal_dac_plat_driver = { .pm = &pistachio_internal_dac_pm_ops }, .probe = pistachio_internal_dac_probe, - .remove_new = pistachio_internal_dac_remove + .remove = pistachio_internal_dac_remove }; module_platform_driver(pistachio_internal_dac_plat_driver); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 8652b4a20020..373d68b4cf88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -812,7 +812,7 @@ static struct platform_driver sst_platform_driver = { .pm = &sst_platform_pm, }, .probe = sst_platform_probe, - .remove_new = sst_platform_remove, + .remove = sst_platform_remove, }; module_platform_driver(sst_platform_driver); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 29d44c989e5f..9956dc63db74 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -358,7 +358,7 @@ static struct platform_driver sst_acpi_driver = { .pm = &intel_sst_pm, }, .probe = sst_acpi_probe, - .remove_new = sst_acpi_remove, + .remove = sst_acpi_remove, }; module_platform_driver(sst_acpi_driver); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 3539c9ff0fd2..d3327bc237b5 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -709,7 +709,7 @@ static struct platform_driver snd_byt_cht_es8316_mc_driver = { .name = "bytcht_es8316", }, .probe = snd_byt_cht_es8316_mc_probe, - .remove_new = snd_byt_cht_es8316_mc_remove, + .remove = snd_byt_cht_es8316_mc_remove, }; module_platform_driver(snd_byt_cht_es8316_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4479825c08b5..2ed49acb4e36 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1918,7 +1918,7 @@ static struct platform_driver snd_byt_rt5640_mc_driver = { .name = "bytcr_rt5640", }, .probe = snd_byt_rt5640_mc_probe, - .remove_new = snd_byt_rt5640_mc_remove, + .remove = snd_byt_rt5640_mc_remove, }; module_platform_driver(snd_byt_rt5640_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1f54da98aacf..8e4b821efe92 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -1142,7 +1142,7 @@ static struct platform_driver snd_byt_rt5651_mc_driver = { .name = "bytcr_rt5651", }, .probe = snd_byt_rt5651_mc_probe, - .remove_new = snd_byt_rt5651_mc_remove, + .remove = snd_byt_rt5651_mc_remove, }; module_platform_driver(snd_byt_rt5651_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index e5a7cc606aa9..0b10d89cb189 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -663,7 +663,7 @@ static struct platform_driver snd_byt_wm5102_mc_driver = { .name = "bytcr_wm5102", }, .probe = snd_byt_wm5102_mc_probe, - .remove_new = snd_byt_wm5102_mc_remove, + .remove = snd_byt_wm5102_mc_remove, }; module_platform_driver(snd_byt_wm5102_mc_driver); diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index f43bc20d6aae..d7c114858833 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -637,7 +637,7 @@ static struct platform_driver snd_cht_mc_driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, - .remove_new = snd_cht_mc_remove, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 578271b4230a..fc998fe4b196 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -838,7 +838,7 @@ static struct platform_driver sof_es8336_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_es8336_probe, - .remove_new = sof_es8336_remove, + .remove = sof_es8336_remove, .id_table = board_ids, }; module_platform_driver(sof_es8336_driver); diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index b01cb2329542..5de883b9563e 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -430,7 +430,7 @@ static void sof_pcm512x_remove(struct platform_device *pdev) static struct platform_driver sof_audio = { .probe = sof_audio_probe, - .remove_new = sof_pcm512x_remove, + .remove = sof_pcm512x_remove, .driver = { .name = "sof_pcm512x", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 855109d22131..f10ed95770ea 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1368,7 +1368,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index 0a5ce34d7f7b..facc6c32cbfe 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -294,7 +294,7 @@ static struct platform_driver sof_wm8804_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_wm8804_probe, - .remove_new = sof_wm8804_remove, + .remove = sof_wm8804_remove, }; module_platform_driver(sof_wm8804_driver); diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 2e1fa79a04d4..2aa637124bec 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -374,7 +374,7 @@ MODULE_DEVICE_TABLE(acpi, catpt_ids); static struct platform_driver catpt_acpi_driver = { .probe = catpt_acpi_probe, - .remove_new = catpt_acpi_remove, + .remove = catpt_acpi_remove, .driver = { .name = "intel_catpt", .acpi_match_table = catpt_ids, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d1eb90310afa..99bd066c7309 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -759,7 +759,7 @@ MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove_new = kirkwood_i2s_dev_remove, + .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(mvebu_audio_of_match), diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c12d170fa1de..d07f288f9752 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1400,7 +1400,7 @@ static struct platform_driver mtk_btcvsd_snd_driver = { .of_match_table = mtk_btcvsd_snd_dt_match, }, .probe = mtk_btcvsd_snd_probe, - .remove_new = mtk_btcvsd_snd_remove, + .remove = mtk_btcvsd_snd_remove, }; module_platform_driver(mtk_btcvsd_snd_driver); diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 6a17deb874df..5f11bc5438bd 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1473,7 +1473,7 @@ static struct platform_driver mt2701_afe_pcm_driver = { .pm = &mt2701_afe_pm_ops, }, .probe = mt2701_afe_pcm_dev_probe, - .remove_new = mt2701_afe_pcm_dev_remove, + .remove = mt2701_afe_pcm_dev_remove, }; module_platform_driver(mt2701_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index c1dee119e93a..9159b42adf6a 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -890,7 +890,7 @@ static struct platform_driver mt6797_afe_pcm_driver = { .pm = &mt6797_afe_pm_ops, }, .probe = mt6797_afe_pcm_dev_probe, - .remove_new = mt6797_afe_pcm_dev_remove, + .remove = mt6797_afe_pcm_dev_remove, }; module_platform_driver(mt6797_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index 572ded279b53..d400bea0ef9c 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -601,7 +601,7 @@ static struct platform_driver mt7986_afe_pcm_driver = { .pm = &mt7986_afe_pm_ops, }, .probe = mt7986_afe_pcm_dev_probe, - .remove_new = mt7986_afe_pcm_dev_remove, + .remove = mt7986_afe_pcm_dev_remove, }; module_platform_driver(mt7986_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index b6291b7c811e..03250273ea9c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1223,7 +1223,7 @@ static struct platform_driver mt8173_afe_pcm_driver = { .pm = &mt8173_afe_pm_ops, }, .probe = mt8173_afe_pcm_dev_probe, - .remove_new = mt8173_afe_pcm_dev_remove, + .remove = mt8173_afe_pcm_dev_remove, }; module_platform_driver(mt8173_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 25348fdf75fa..3f377ba4ad53 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1268,7 +1268,7 @@ static struct platform_driver mt8183_afe_pcm_driver = { .pm = &mt8183_afe_pm_ops, }, .probe = mt8183_afe_pcm_dev_probe, - .remove_new = mt8183_afe_pcm_dev_remove, + .remove = mt8183_afe_pcm_dev_remove, }; module_platform_driver(mt8183_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 424c5c68f78a..9b502f4cd6ea 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2325,7 +2325,7 @@ static struct platform_driver mt8192_afe_pcm_driver = { .pm = &mt8192_afe_pm_ops, }, .probe = mt8192_afe_pcm_dev_probe, - .remove_new = mt8192_afe_pcm_dev_remove, + .remove = mt8192_afe_pcm_dev_remove, }; module_platform_driver(mt8192_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 38891d1bd18a..8016bfb35015 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3199,7 +3199,7 @@ static struct platform_driver mt8195_afe_pcm_driver = { .pm = &mt8195_afe_pm_ops, }, .probe = mt8195_afe_pcm_dev_probe, - .remove_new = mt8195_afe_pcm_dev_remove, + .remove = mt8195_afe_pcm_dev_remove, }; module_platform_driver(mt8195_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index 4ec86b71dd88..6ca10e2046c8 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -2264,7 +2264,7 @@ static struct platform_driver mt8365_afe_pcm_driver = { .pm = &mt8365_afe_pm_ops, }, .probe = mt8365_afe_pcm_dev_probe, - .remove_new = mt8365_afe_pcm_dev_remove, + .remove = mt8365_afe_pcm_dev_remove, }; module_platform_driver(mt8365_afe_pcm_driver); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 5d1419ed7a62..f2890111c1d2 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -345,7 +345,7 @@ MODULE_DEVICE_TABLE(of, aiu_of_match); static struct platform_driver aiu_pdrv = { .probe = aiu_probe, - .remove_new = aiu_remove, + .remove = aiu_remove, .driver = { .name = "meson-aiu", .of_match_table = aiu_of_match, diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 646ab87afac2..59deb332bd35 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -360,7 +360,7 @@ MODULE_DEVICE_TABLE(of, axg_card_of_match); static struct platform_driver axg_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "axg-sound-card", .of_match_table = axg_card_of_match, diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 7edca3e49c8f..455f6bfc9f8f 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -129,7 +129,7 @@ MODULE_DEVICE_TABLE(of, gx_card_of_match); static struct platform_driver gx_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "gx-sound-card", .of_match_table = gx_card_of_match, diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 310e3ac77424..a41a13ae38a5 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -185,7 +185,7 @@ static struct platform_driver mxs_sgtl5000_audio_driver = { .of_match_table = mxs_sgtl5000_dt_ids, }, .probe = mxs_sgtl5000_probe, - .remove_new = mxs_sgtl5000_remove, + .remove = mxs_sgtl5000_remove, }; module_platform_driver(mxs_sgtl5000_audio_driver); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index abfaf3cdf5bb..73f36c9dd35c 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -574,7 +574,7 @@ static struct platform_driver asoc_mmp_sspa_driver = { .of_match_table = of_match_ptr(mmp_sspa_of_match), }, .probe = asoc_mmp_sspa_probe, - .remove_new = asoc_mmp_sspa_remove, + .remove = asoc_mmp_sspa_remove, }; module_platform_driver(asoc_mmp_sspa_driver); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 80e0ea0ec9fb..78f50032afc5 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -286,7 +286,7 @@ static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, - .remove_new = pxa2xx_ac97_dev_remove, + .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", .pm = &pxa2xx_ac97_pm_ops, diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 9005c85f8c54..b8f23414eb77 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -300,7 +300,7 @@ static struct platform_driver apq8016_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(apq8016_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(apq8016_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 5c874139f39d..e57d29ea4ce7 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -172,7 +172,7 @@ static struct platform_driver ipq806x_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(ipq806x_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(ipq806x_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index e6bcdf6ed796..fbead6af3d95 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -315,7 +315,7 @@ static struct platform_driver sc7180_lpass_cpu_platform_driver = { .pm = &sc7180_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 47c622327a8d..7cd3e291382a 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -445,7 +445,7 @@ static struct platform_driver sc7280_lpass_cpu_platform_driver = { .pm = &sc7280_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 81fde0681f95..90228699ba7d 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -1161,7 +1161,7 @@ static struct platform_driver q6pcm_routing_platform_driver = { .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, - .remove_new = q6pcm_routing_remove, + .remove = q6pcm_routing_remove, }; module_platform_driver(q6pcm_routing_platform_driver); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b378f870b3ad..4315da4a47c1 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -866,7 +866,7 @@ static const struct dev_pm_ops rockchip_i2s_pm_ops = { static struct platform_driver rockchip_i2s_driver = { .probe = rockchip_i2s_probe, - .remove_new = rockchip_i2s_remove, + .remove = rockchip_i2s_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(rockchip_i2s_match), diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index ee517d7b5b7b..d1f28699652f 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1423,7 +1423,7 @@ static const struct dev_pm_ops rockchip_i2s_tdm_pm_ops = { static struct platform_driver rockchip_i2s_tdm_driver = { .probe = rockchip_i2s_tdm_probe, - .remove_new = rockchip_i2s_tdm_remove, + .remove = rockchip_i2s_tdm_remove, .driver = { .name = DRV_NAME, .of_match_table = rockchip_i2s_tdm_match, diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index d16a4a67a6a2..cae91108f7a8 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -703,7 +703,7 @@ static const struct dev_pm_ops rockchip_pdm_pm_ops = { static struct platform_driver rockchip_pdm_driver = { .probe = rockchip_pdm_probe, - .remove_new = rockchip_pdm_remove, + .remove = rockchip_pdm_remove, .driver = { .name = "rockchip-pdm", .of_match_table = of_match_ptr(rockchip_pdm_match), diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 449f62820045..b085d80ea2e4 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -233,7 +233,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, - .remove_new = snd_rk_mc_remove, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index eb9d5dee196e..d87c0e4f6f91 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -380,7 +380,7 @@ static const struct dev_pm_ops rk_spdif_pm_ops = { static struct platform_driver rk_spdif_driver = { .probe = rk_spdif_probe, - .remove_new = rk_spdif_remove, + .remove = rk_spdif_remove, .driver = { .name = "rockchip-spdif", .of_match_table = of_match_ptr(rk_spdif_match), diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index f02873b6ce7f..9619f550608c 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -207,7 +207,7 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = arndale_audio_of_match, }, .probe = arndale_audio_probe, - .remove_new = arndale_audio_remove, + .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1bcabb114e29..8f6deb06e234 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1741,7 +1741,7 @@ static const struct dev_pm_ops samsung_i2s_pm = { static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove_new = samsung_i2s_remove, + .remove = samsung_i2s_remove, .id_table = samsung_i2s_driver_ids, .driver = { .name = "samsung-i2s", diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 110ae14dd7ea..ed865cc07e2e 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -341,7 +341,7 @@ static struct platform_driver odroid_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = odroid_audio_probe, - .remove_new = odroid_audio_remove, + .remove = odroid_audio_remove, }; module_platform_driver(odroid_audio_driver); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 573b2dee7f07..a03ba9374c2e 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -590,7 +590,7 @@ static void s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove_new = s3c_pcm_dev_remove, + .remove = s3c_pcm_dev_remove, .driver = { .name = "samsung-pcm", }, diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index aad0f9b4d4fc..4bbe7bcdb845 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -245,7 +245,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, - .remove_new = snow_remove, + .remove = snow_remove, }; module_platform_driver(snow_driver); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index f44e3180e8d3..235d0063d1b3 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -476,7 +476,7 @@ static void spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove_new = spdif_remove, + .remove = spdif_remove, .driver = { .name = "samsung-spdif", }, diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 087e379aa3bc..221ce91f1950 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2107,7 +2107,7 @@ static struct platform_driver fsi_driver = { .of_match_table = fsi_of_match, }, .probe = fsi_probe, - .remove_new = fsi_remove, + .remove = fsi_remove, .id_table = fsi_id_table, }; diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index cc200f45826c..db618c09d1e0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -334,7 +334,7 @@ static struct platform_driver hac_pcm_driver = { }, .probe = hac_soc_platform_probe, - .remove_new = hac_soc_platform_remove, + .remove = hac_soc_platform_remove, }; module_platform_driver(hac_pcm_driver); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 15cb5e7008f9..9784718a2b6f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -2087,7 +2087,7 @@ static struct platform_driver rsnd_driver = { .of_match_table = rsnd_of_match, }, .probe = rsnd_probe, - .remove_new = rsnd_remove, + .remove = rsnd_remove, }; module_platform_driver(rsnd_driver); diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index d0bf0487bf1b..040ce0431fd2 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1204,7 +1204,7 @@ static struct platform_driver rz_ssi_driver = { .of_match_table = rz_ssi_of_match, }, .probe = rz_ssi_probe, - .remove_new = rz_ssi_remove, + .remove = rz_ssi_remove, }; module_platform_driver(rz_ssi_driver); diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index d0b5c543fd2f..7e771a164a80 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -788,7 +788,7 @@ static struct platform_driver siu_driver = { .name = "siu-pcm-audio", }, .probe = siu_probe, - .remove_new = siu_remove, + .remove = siu_remove, }; module_platform_driver(siu_driver); diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 9f24e3c283dd..f09ee0dea2cc 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -658,7 +658,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cdd1e79ef9f6..01d3ad3314f3 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -505,7 +505,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8m_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8m_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8m", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 2585b1beef23..e5eee1c9f6da 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -507,7 +507,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8ulp_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8ulp_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8ulp", .pm = &sof_of_pm, diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 7f18080e4e19..322ff118f0f6 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -684,7 +684,7 @@ static int sof_broadwell_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_bdw_driver = { .probe = sof_broadwell_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-bdw", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 7a57e162fb1c..f531710cf94e 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -465,7 +465,7 @@ static int sof_baytrail_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_byt_driver = { .probe = sof_baytrail_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-byt", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 74522400207e..9466f7d2e535 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -656,7 +656,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8186_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8186_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8186", diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 82d221f53a46..5b4423ed8023 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -612,7 +612,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8195_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8195_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8195", diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 688419c6b092..814a1cde1d35 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -993,7 +993,7 @@ MODULE_DEVICE_TABLE(of, sprd_mcdt_of_match); static struct platform_driver sprd_mcdt_driver = { .probe = sprd_mcdt_probe, - .remove_new = sprd_mcdt_remove, + .remove = sprd_mcdt_remove, .driver = { .name = "sprd-mcdt", .of_match_table = sprd_mcdt_of_match, diff --git a/sound/soc/starfive/jh7110_pwmdac.c b/sound/soc/starfive/jh7110_pwmdac.c index 4a4dd431b82b..a603dd17931c 100644 --- a/sound/soc/starfive/jh7110_pwmdac.c +++ b/sound/soc/starfive/jh7110_pwmdac.c @@ -516,7 +516,7 @@ static struct platform_driver jh7110_pwmdac_driver = { .pm = pm_ptr(&jh7110_pwmdac_pm_ops), }, .probe = jh7110_pwmdac_probe, - .remove_new = jh7110_pwmdac_remove, + .remove = jh7110_pwmdac_remove, }; module_platform_driver(jh7110_pwmdac_driver); diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 1e0ff6720747..d38090e68df5 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -660,7 +660,7 @@ static struct platform_driver jh7110_tdm_driver = { .pm = pm_ptr(&jh7110_tdm_pm_ops), }, .probe = jh7110_tdm_probe, - .remove_new = jh7110_tdm_dev_remove, + .remove = jh7110_tdm_dev_remove, }; module_platform_driver(jh7110_tdm_driver); diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index fb5dd9a68bea..9351727dce1a 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -398,7 +398,7 @@ static struct platform_driver stm32_adfsdm_driver = { .of_match_table = stm32_adfsdm_of_match, }, .probe = stm32_adfsdm_probe, - .remove_new = stm32_adfsdm_remove, + .remove = stm32_adfsdm_remove, }; module_platform_driver(stm32_adfsdm_driver); diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index a96aa308681a..faa00103ee7f 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -1216,7 +1216,7 @@ static struct platform_driver stm32_i2s_driver = { .pm = &stm32_i2s_pm_ops, }, .probe = stm32_i2s_probe, - .remove_new = stm32_i2s_remove, + .remove = stm32_i2s_remove, }; module_platform_driver(stm32_i2s_driver); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ad2492efb1cd..7bc4a96b7503 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1619,7 +1619,7 @@ static struct platform_driver stm32_sai_sub_driver = { .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, - .remove_new = stm32_sai_sub_remove, + .remove = stm32_sai_sub_remove, }; module_platform_driver(stm32_sai_sub_driver); diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 9eed3c57e3f1..d1b32ba1e1a2 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1072,7 +1072,7 @@ static struct platform_driver stm32_spdifrx_driver = { .pm = &stm32_spdifrx_pm_ops, }, .probe = stm32_spdifrx_probe, - .remove_new = stm32_spdifrx_remove, + .remove = stm32_spdifrx_remove, }; module_platform_driver(stm32_spdifrx_driver); diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index a2618ed650b0..596a1ade4658 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1838,7 +1838,7 @@ static struct platform_driver sun4i_codec_driver = { .of_match_table = sun4i_codec_of_match, }, .probe = sun4i_codec_probe, - .remove_new = sun4i_codec_remove, + .remove = sun4i_codec_remove, }; module_platform_driver(sun4i_codec_driver); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 5f8d979585b6..909b5f2b65f8 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1681,7 +1681,7 @@ static const struct dev_pm_ops sun4i_i2s_pm_ops = { static struct platform_driver sun4i_i2s_driver = { .probe = sun4i_i2s_probe, - .remove_new = sun4i_i2s_remove, + .remove = sun4i_i2s_remove, .driver = { .name = "sun4i-i2s", .of_match_table = sun4i_i2s_match, diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index f41c30955857..0aa416423246 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -726,7 +726,7 @@ static struct platform_driver sun4i_spdif_driver = { .pm = &sun4i_spdif_pm, }, .probe = sun4i_spdif_probe, - .remove_new = sun4i_spdif_remove, + .remove = sun4i_spdif_remove, }; module_platform_driver(sun4i_spdif_driver); diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 884394ddaf86..3e751b5694fe 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -426,7 +426,7 @@ static struct platform_driver sun50i_dmic_driver = { .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, - .remove_new = sun50i_dmic_remove, + .remove = sun50i_dmic_remove, }; module_platform_driver(sun50i_dmic_driver); diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index b5dafb749c3f..8c645e04d571 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -1713,7 +1713,7 @@ static struct platform_driver sun8i_codec_driver = { .pm = &sun8i_codec_pm_ops, }, .probe = sun8i_codec_probe, - .remove_new = sun8i_codec_remove, + .remove = sun8i_codec_remove, }; module_platform_driver(sun8i_codec_driver); diff --git a/sound/soc/tegra/tegra186_asrc.c b/sound/soc/tegra/tegra186_asrc.c index 22af5135d77a..d914dba56013 100644 --- a/sound/soc/tegra/tegra186_asrc.c +++ b/sound/soc/tegra/tegra186_asrc.c @@ -1034,7 +1034,7 @@ static struct platform_driver tegra186_asrc_driver = { .pm = &tegra186_asrc_pm_ops, }, .probe = tegra186_asrc_platform_probe, - .remove_new = tegra186_asrc_platform_remove, + .remove = tegra186_asrc_platform_remove, }; module_platform_driver(tegra186_asrc_driver) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 21cd41fec7a9..508128b7783e 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -542,7 +542,7 @@ static struct platform_driver tegra186_dspk_driver = { .pm = &tegra186_dspk_pm_ops, }, .probe = tegra186_dspk_platform_probe, - .remove_new = tegra186_dspk_platform_remove, + .remove = tegra186_dspk_platform_remove, }; module_platform_driver(tegra186_dspk_driver); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 8011afe93c96..08c58e8f3c22 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -448,7 +448,7 @@ static struct platform_driver tegra20_ac97_driver = { .of_match_table = tegra20_ac97_of_match, }, .probe = tegra20_ac97_platform_probe, - .remove_new = tegra20_ac97_platform_remove, + .remove = tegra20_ac97_platform_remove, }; module_platform_driver(tegra20_ac97_driver); diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index f11618e8f13e..3b9823d1a87a 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -500,7 +500,7 @@ static struct platform_driver tegra20_i2s_driver = { .pm = &tegra20_i2s_pm_ops, }, .probe = tegra20_i2s_platform_probe, - .remove_new = tegra20_i2s_platform_remove, + .remove = tegra20_i2s_platform_remove, }; module_platform_driver(tegra20_i2s_driver); diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 9f9334e48049..a866aeb2719d 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -856,7 +856,7 @@ static const struct dev_pm_ops tegra_admaif_pm_ops = { static struct platform_driver tegra_admaif_driver = { .probe = tegra_admaif_probe, - .remove_new = tegra_admaif_remove, + .remove = tegra_admaif_remove, .driver = { .name = "tegra210-admaif", .of_match_table = tegra_admaif_of_match, diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index d2530443a221..109f763fe211 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -532,7 +532,7 @@ static struct platform_driver tegra210_adx_driver = { .pm = &tegra210_adx_pm_ops, }, .probe = tegra210_adx_platform_probe, - .remove_new = tegra210_adx_platform_remove, + .remove = tegra210_adx_platform_remove, }; module_platform_driver(tegra210_adx_driver); diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index 3f114a2adfce..2e25b6403599 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -1414,7 +1414,7 @@ static const struct dev_pm_ops tegra_ahub_pm_ops = { static struct platform_driver tegra_ahub_driver = { .probe = tegra_ahub_probe, - .remove_new = tegra_ahub_remove, + .remove = tegra_ahub_remove, .driver = { .name = "tegra210-ahub", .of_match_table = tegra_ahub_of_match, diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 91e405909e0f..38a2d6ec033b 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -589,7 +589,7 @@ static struct platform_driver tegra210_amx_driver = { .pm = &tegra210_amx_pm_ops, }, .probe = tegra210_amx_platform_probe, - .remove_new = tegra210_amx_platform_remove, + .remove = tegra210_amx_platform_remove, }; module_platform_driver(tegra210_amx_driver); diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index e53c0278ae9a..d9b577f146dc 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -559,7 +559,7 @@ static struct platform_driver tegra210_dmic_driver = { .pm = &tegra210_dmic_pm_ops, }, .probe = tegra210_dmic_probe, - .remove_new = tegra210_dmic_remove, + .remove = tegra210_dmic_remove, }; module_platform_driver(tegra210_dmic_driver) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index e93ceb7afb4c..a3908b15dfdc 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -1019,7 +1019,7 @@ static struct platform_driver tegra210_i2s_driver = { .pm = &tegra210_i2s_pm_ops, }, .probe = tegra210_i2s_probe, - .remove_new = tegra210_i2s_remove, + .remove = tegra210_i2s_remove, }; module_platform_driver(tegra210_i2s_driver) diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 024614f6ec0b..e07e2f1d2f70 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -674,7 +674,7 @@ static struct platform_driver tegra210_mixer_driver = { .pm = &tegra210_mixer_pm_ops, }, .probe = tegra210_mixer_platform_probe, - .remove_new = tegra210_mixer_platform_remove, + .remove = tegra210_mixer_platform_remove, }; module_platform_driver(tegra210_mixer_driver); diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index b89f5edafa03..4ead52564ab6 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -766,7 +766,7 @@ static struct platform_driver tegra210_mvc_driver = { .pm = &tegra210_mvc_pm_ops, }, .probe = tegra210_mvc_platform_probe, - .remove_new = tegra210_mvc_platform_remove, + .remove = tegra210_mvc_platform_remove, }; module_platform_driver(tegra210_mvc_driver) diff --git a/sound/soc/tegra/tegra210_ope.c b/sound/soc/tegra/tegra210_ope.c index 136ed17f3650..e2bc604e8b79 100644 --- a/sound/soc/tegra/tegra210_ope.c +++ b/sound/soc/tegra/tegra210_ope.c @@ -407,7 +407,7 @@ static struct platform_driver tegra210_ope_driver = { .pm = &tegra210_ope_pm_ops, }, .probe = tegra210_ope_probe, - .remove_new = tegra210_ope_remove, + .remove = tegra210_ope_remove, }; module_platform_driver(tegra210_ope_driver) diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 028747c44f37..e16bbb44cc77 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3631,7 +3631,7 @@ static struct platform_driver tegra210_sfc_driver = { .pm = &tegra210_sfc_pm_ops, }, .probe = tegra210_sfc_platform_probe, - .remove_new = tegra210_sfc_platform_remove, + .remove = tegra210_sfc_platform_remove, }; module_platform_driver(tegra210_sfc_driver) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d2e8078e444a..c9b52f54cea8 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -608,7 +608,7 @@ static const struct dev_pm_ops tegra30_ahub_pm_ops = { static struct platform_driver tegra30_ahub_driver = { .probe = tegra30_ahub_probe, - .remove_new = tegra30_ahub_remove, + .remove = tegra30_ahub_remove, .driver = { .name = DRV_NAME, .of_match_table = tegra30_ahub_of_match, diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a8ff51d12edb..0d3badfbe143 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -560,7 +560,7 @@ static struct platform_driver tegra30_i2s_driver = { .pm = &tegra30_i2s_pm_ops, }, .probe = tegra30_i2s_platform_probe, - .remove_new = tegra30_i2s_platform_remove, + .remove = tegra30_i2s_platform_remove, }; module_platform_driver(tegra30_i2s_driver); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index feba9d42bbc5..8b48813c2c59 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -248,7 +248,7 @@ static struct platform_driver tegra_audio_graph_card = { .of_match_table = graph_of_tegra_match, }, .probe = tegra_audio_graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(tegra_audio_graph_card); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 76bda188e992..94645f275495 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -595,7 +595,7 @@ static struct platform_driver ams_delta_driver = { .name = DRV_NAME, }, .probe = ams_delta_probe, - .remove_new = ams_delta_remove, + .remove = ams_delta_remove, }; module_platform_driver(ams_delta_driver); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 0f15a743c798..d682b98d6848 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -920,7 +920,7 @@ MODULE_DEVICE_TABLE(of, davinci_i2s_match); static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, - .remove_new = davinci_i2s_remove, + .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", .of_match_table = of_match_ptr(davinci_i2s_match), diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2b1ed91a736c..a0b8cca31cba 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2535,7 +2535,7 @@ static const struct dev_pm_ops davinci_mcasp_pm_ops = { static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, - .remove_new = davinci_mcasp_remove, + .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .pm = &davinci_mcasp_pm_ops, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 2110ffe5281c..411970399271 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1430,7 +1430,7 @@ static struct platform_driver asoc_mcbsp_driver = { }, .probe = asoc_mcbsp_probe, - .remove_new = asoc_mcbsp_remove, + .remove = asoc_mcbsp_remove, }; module_platform_driver(asoc_mcbsp_driver); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index 01cc3b961999..a041ce8e23a4 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -347,7 +347,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/aio-pxs2.c b/sound/soc/uniphier/aio-pxs2.c index fba13a212bdb..889f64b2c01f 100644 --- a/sound/soc/uniphier/aio-pxs2.c +++ b/sound/soc/uniphier/aio-pxs2.c @@ -256,7 +256,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index d90b3e4b0104..662e45882c90 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -560,7 +560,7 @@ static struct platform_driver evea_codec_driver = { .of_match_table = of_match_ptr(evea_of_match), }, .probe = evea_probe, - .remove_new = evea_remove, + .remove = evea_remove, }; module_platform_driver(evea_codec_driver); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index e0ab4534fe3e..ae6d326167d1 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -157,7 +157,7 @@ static struct platform_driver snd_soc_mop500_driver = { .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, - .remove_new = mop500_remove, + .remove = mop500_remove, }; module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 3fd13e8dd110..a2dd739fdf2d 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -816,7 +816,7 @@ static struct platform_driver msp_i2s_driver = { .of_match_table = ux500_msp_i2s_match, }, .probe = ux500_msp_drv_probe, - .remove_new = ux500_msp_drv_remove, + .remove = ux500_msp_drv_remove, }; module_platform_driver(msp_i2s_driver); diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 158fc21a86c1..17ef05309469 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -713,7 +713,7 @@ MODULE_DEVICE_TABLE(of, xlnx_formatter_pcm_of_match); static struct platform_driver xlnx_formatter_pcm_driver = { .probe = xlnx_formatter_pcm_probe, - .remove_new = xlnx_formatter_pcm_remove, + .remove = xlnx_formatter_pcm_remove, .driver = { .name = DRV_NAME, .of_match_table = xlnx_formatter_pcm_of_match, diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c index d52d5fc7b5b8..7febb3830dc2 100644 --- a/sound/soc/xilinx/xlnx_spdif.c +++ b/sound/soc/xilinx/xlnx_spdif.c @@ -325,7 +325,7 @@ static struct platform_driver xlnx_spdif_driver = { .of_match_table = xlnx_spdif_of_match, }, .probe = xlnx_spdif_probe, - .remove_new = xlnx_spdif_remove, + .remove = xlnx_spdif_remove, }; module_platform_driver(xlnx_spdif_driver); diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 6e2b72d7a65d..4eaa9011405f 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -635,7 +635,7 @@ static const struct dev_pm_ops xtfpga_i2s_pm_ops = { static struct platform_driver xtfpga_i2s_driver = { .probe = xtfpga_i2s_probe, - .remove_new = xtfpga_i2s_remove, + .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", .of_match_table = of_match_ptr(xtfpga_i2s_of_match), From fc09ea51ddc01119d7cbb3ff56d51af2de6f71d8 Mon Sep 17 00:00:00 2001 From: Edson Juliano Drosdeck Date: Mon, 9 Sep 2024 13:27:51 -0300 Subject: [PATCH 371/410] ALSA: hda/realtek: Enable mic on Vaio VJFH52 Vaio VJFH52 is equipped with ACL256, and needs a fix to make the internal mic and headphone mic to work. Also must to limits the internal microphone boost. Signed-off-by: Edson Juliano Drosdeck Link: https://patch.msgid.link/20240909162751.4790-1-edson.drosdeck@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2e90c713b4a2..037556bf46e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7608,6 +7608,7 @@ enum { ALC287_FIXUP_LENOVO_14ARP8_LEGION_IAH7, ALC287_FIXUP_LENOVO_SSID_17AA3820, ALC245_FIXUP_CLEVO_NOISY_MIC, + ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9935,6 +9936,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, }, + [ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a1113c }, /* use as headset mic, without its own jack detect */ + { 0x1b, 0x20a11040 }, /* dock mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10571,6 +10582,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), From 7e4d4b32ab9532bd1babcd5d0763d727ebb04be0 Mon Sep 17 00:00:00 2001 From: Joshua Grisham Date: Mon, 9 Sep 2024 21:30:00 +0200 Subject: [PATCH 372/410] ALSA: hda/realtek: Refactor and simplify Samsung Galaxy Book init I have done a lot of analysis for these type of devices and collaborated quite a bit with Nick Weihs (author of the first patch submitted for this including adding samsung_helper.c). More information can be found in the issue on Github [1] including additional rationale and testing. The existing implementation includes a large number of equalizer coef values that are not necessary to actually init and enable the speaker amps, as well as create a somewhat worse sound profile. Users have reported "muffled" or "muddy" sound; more information about this including my analysis of the differences can be found in the linked Github issue. This patch refactors the "v2" version of ALC298_FIXUP_SAMSUNG_AMP to a much simpler implementation which removes the new samsung_helper.c, reuses more of the existing patch_realtek.c, and sends significantly fewer unnecessary coef values (including removing all of these EQ-specific coef values). A pcm_playback_hook is used to dynamically enable and disable the speaker amps only when there will be audio playback; this is to match the behavior of how the driver for these devices is working in Windows, and is suspected but not yet tested or confirmed to help with power consumption. Support for models with 2 speaker amps vs 4 speaker amps is controlled by a specific quirk name for both types. A new int num_speaker_amps has been added to alc_spec so that the hooks can know how many speaker amps to enable or disable. This design was chosen to limit the number of places that subsystem ids will need to be maintained: like this, they can be maintained only once in the quirk table and there will not be another separate list of subsystem ids to maintain elsewhere in the code. Also updated the quirk name from ALC298_FIXUP_SAMSUNG_AMP2 to ALC298_FIXUP_SAMSUNG_AMP_V2_.. as this is not a quirk for "Amp #2" on ALC298 but is instead a different version of how to handle it. More devices have been added (see Github issue for testing confirmation), as well as a small cleanup to existing names. [1]: https://github.com/thesofproject/linux/issues/4055#issuecomment-2323411911 Signed-off-by: Joshua Grisham Link: https://patch.msgid.link/20240909193000.838815-1-josh@joshuagrisham.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 151 +++++++++++++++- sound/pci/hda/samsung_helper.c | 310 --------------------------------- 2 files changed, 144 insertions(+), 317 deletions(-) delete mode 100644 sound/pci/hda/samsung_helper.c diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 037556bf46e9..ef6d9ac28a61 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -125,6 +125,7 @@ struct alc_spec { unsigned int has_hs_key:1; unsigned int no_internal_mic_pin:1; unsigned int en_3kpull_low:1; + int num_speaker_amps; /* for PLL fix */ hda_nid_t pll_nid; @@ -4813,7 +4814,133 @@ static void alc298_fixup_samsung_amp(struct hda_codec *codec, } } -#include "samsung_helper.c" +struct alc298_samsung_v2_amp_desc { + unsigned short nid; + int init_seq_size; + unsigned short init_seq[18][2]; +}; + +static const struct alc298_samsung_v2_amp_desc +alc298_samsung_v2_amp_desc_tbl[] = { + { 0x38, 18, { + { 0x23e1, 0x0000 }, { 0x2012, 0x006f }, { 0x2014, 0x0000 }, + { 0x201b, 0x0001 }, { 0x201d, 0x0001 }, { 0x201f, 0x00fe }, + { 0x2021, 0x0000 }, { 0x2022, 0x0010 }, { 0x203d, 0x0005 }, + { 0x203f, 0x0003 }, { 0x2050, 0x002c }, { 0x2076, 0x000e }, + { 0x207c, 0x004a }, { 0x2081, 0x0003 }, { 0x2399, 0x0003 }, + { 0x23a4, 0x00b5 }, { 0x23a5, 0x0001 }, { 0x23ba, 0x0094 } + }}, + { 0x39, 18, { + { 0x23e1, 0x0000 }, { 0x2012, 0x006f }, { 0x2014, 0x0000 }, + { 0x201b, 0x0002 }, { 0x201d, 0x0002 }, { 0x201f, 0x00fd }, + { 0x2021, 0x0001 }, { 0x2022, 0x0010 }, { 0x203d, 0x0005 }, + { 0x203f, 0x0003 }, { 0x2050, 0x002c }, { 0x2076, 0x000e }, + { 0x207c, 0x004a }, { 0x2081, 0x0003 }, { 0x2399, 0x0003 }, + { 0x23a4, 0x00b5 }, { 0x23a5, 0x0001 }, { 0x23ba, 0x0094 } + }}, + { 0x3c, 15, { + { 0x23e1, 0x0000 }, { 0x2012, 0x006f }, { 0x2014, 0x0000 }, + { 0x201b, 0x0001 }, { 0x201d, 0x0001 }, { 0x201f, 0x00fe }, + { 0x2021, 0x0000 }, { 0x2022, 0x0010 }, { 0x203d, 0x0005 }, + { 0x203f, 0x0003 }, { 0x2050, 0x002c }, { 0x2076, 0x000e }, + { 0x207c, 0x004a }, { 0x2081, 0x0003 }, { 0x23ba, 0x008d } + }}, + { 0x3d, 15, { + { 0x23e1, 0x0000 }, { 0x2012, 0x006f }, { 0x2014, 0x0000 }, + { 0x201b, 0x0002 }, { 0x201d, 0x0002 }, { 0x201f, 0x00fd }, + { 0x2021, 0x0001 }, { 0x2022, 0x0010 }, { 0x203d, 0x0005 }, + { 0x203f, 0x0003 }, { 0x2050, 0x002c }, { 0x2076, 0x000e }, + { 0x207c, 0x004a }, { 0x2081, 0x0003 }, { 0x23ba, 0x008d } + }} +}; + +static void alc298_samsung_v2_enable_amps(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + static const unsigned short enable_seq[][2] = { + { 0x203a, 0x0081 }, { 0x23ff, 0x0001 }, + }; + int i, j; + + for (i = 0; i < spec->num_speaker_amps; i++) { + alc_write_coef_idx(codec, 0x22, alc298_samsung_v2_amp_desc_tbl[i].nid); + for (j = 0; j < ARRAY_SIZE(enable_seq); j++) + alc298_samsung_write_coef_pack(codec, enable_seq[j]); + codec_dbg(codec, "alc298_samsung_v2: Enabled speaker amp 0x%02x\n", + alc298_samsung_v2_amp_desc_tbl[i].nid); + } +} + +static void alc298_samsung_v2_disable_amps(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + static const unsigned short disable_seq[][2] = { + { 0x23ff, 0x0000 }, { 0x203a, 0x0080 }, + }; + int i, j; + + for (i = 0; i < spec->num_speaker_amps; i++) { + alc_write_coef_idx(codec, 0x22, alc298_samsung_v2_amp_desc_tbl[i].nid); + for (j = 0; j < ARRAY_SIZE(disable_seq); j++) + alc298_samsung_write_coef_pack(codec, disable_seq[j]); + codec_dbg(codec, "alc298_samsung_v2: Disabled speaker amp 0x%02x\n", + alc298_samsung_v2_amp_desc_tbl[i].nid); + } +} + +static void alc298_samsung_v2_playback_hook(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream, + int action) +{ + /* Dynamically enable/disable speaker amps before and after playback */ + if (action == HDA_GEN_PCM_ACT_OPEN) + alc298_samsung_v2_enable_amps(codec); + if (action == HDA_GEN_PCM_ACT_CLOSE) + alc298_samsung_v2_disable_amps(codec); +} + +static void alc298_samsung_v2_init_amps(struct hda_codec *codec, + int num_speaker_amps) +{ + struct alc_spec *spec = codec->spec; + int i, j; + + /* Set spec's num_speaker_amps before doing anything else */ + spec->num_speaker_amps = num_speaker_amps; + + /* Disable speaker amps before init to prevent any physical damage */ + alc298_samsung_v2_disable_amps(codec); + + /* Initialize the speaker amps */ + for (i = 0; i < spec->num_speaker_amps; i++) { + alc_write_coef_idx(codec, 0x22, alc298_samsung_v2_amp_desc_tbl[i].nid); + for (j = 0; j < alc298_samsung_v2_amp_desc_tbl[i].init_seq_size; j++) { + alc298_samsung_write_coef_pack(codec, + alc298_samsung_v2_amp_desc_tbl[i].init_seq[j]); + } + alc_write_coef_idx(codec, 0x89, 0x0); + codec_dbg(codec, "alc298_samsung_v2: Initialized speaker amp 0x%02x\n", + alc298_samsung_v2_amp_desc_tbl[i].nid); + } + + /* register hook to enable speaker amps only when they are needed */ + spec->gen.pcm_playback_hook = alc298_samsung_v2_playback_hook; +} + +static void alc298_fixup_samsung_amp_v2_2_amps(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PROBE) + alc298_samsung_v2_init_amps(codec, 2); +} + +static void alc298_fixup_samsung_amp_v2_4_amps(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PROBE) + alc298_samsung_v2_init_amps(codec, 4); +} #if IS_REACHABLE(CONFIG_INPUT) static void gpio2_mic_hotkey_event(struct hda_codec *codec, @@ -7514,7 +7641,8 @@ enum { ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, ALC236_FIXUP_LENOVO_INV_DMIC, ALC298_FIXUP_SAMSUNG_AMP, - ALC298_FIXUP_SAMSUNG_AMP2, + ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS, + ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -9151,9 +9279,13 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET }, - [ALC298_FIXUP_SAMSUNG_AMP2] = { + [ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS] = { .type = HDA_FIXUP_FUNC, - .v.func = alc298_fixup_samsung_amp2 + .v.func = alc298_fixup_samsung_amp_v2_2_amps + }, + [ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc298_fixup_samsung_amp_v2_4_amps }, [ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET] = { .type = HDA_FIXUP_VERBS, @@ -10506,8 +10638,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), - SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG-KB1US)", ALC298_FIXUP_SAMSUNG_AMP2), - SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH-XD92G))", ALC298_FIXUP_SAMSUNG_AMP2), + SND_PCI_QUIRK(0x144d, 0xc870, "Samsung Galaxy Book2 Pro (NP950XED)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), + SND_PCI_QUIRK(0x144d, 0xc886, "Samsung Galaxy Book3 Pro (NP964XFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), + SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), + SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -10739,6 +10873,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), + SND_PCI_QUIRK(0x1854, 0x0488, "LG gram 16 (16Z90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), + SND_PCI_QUIRK(0x1854, 0x048a, "LG gram 17 (17ZD90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), @@ -10949,7 +11085,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_AMP, .name = "alc298-samsung-amp"}, - {.id = ALC298_FIXUP_SAMSUNG_AMP2, .name = "alc298-samsung-amp2"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS, .name = "alc298-samsung-amp-v2-2-amps"}, + {.id = ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS, .name = "alc298-samsung-amp-v2-4-amps"}, {.id = ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc256-samsung-headphone"}, {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {.id = ALC274_FIXUP_HP_MIC, .name = "alc274-hp-mic-detect"}, diff --git a/sound/pci/hda/samsung_helper.c b/sound/pci/hda/samsung_helper.c deleted file mode 100644 index a40175b69015..000000000000 --- a/sound/pci/hda/samsung_helper.c +++ /dev/null @@ -1,310 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-or-later -/* Helper functions for Samsung Galaxy Book3 audio initialization */ - -struct alc298_samsung_coeff_fixup_desc { - unsigned char coeff_idx; - unsigned short coeff_value; -}; - -struct alc298_samsung_coeff_seq_desc { - unsigned short coeff_0x23; - unsigned short coeff_0x24; - unsigned short coeff_0x25; - unsigned short coeff_0x26; -}; - - -static inline void alc298_samsung_write_coef_pack2(struct hda_codec *codec, - const struct alc298_samsung_coeff_seq_desc *seq) -{ - int i; - - for (i = 0; i < 100; i++) { - if ((alc_read_coef_idx(codec, 0x26) & 0x0010) == 0) - break; - - usleep_range(500, 1000); - } - - alc_write_coef_idx(codec, 0x23, seq->coeff_0x23); - alc_write_coef_idx(codec, 0x24, seq->coeff_0x24); - alc_write_coef_idx(codec, 0x25, seq->coeff_0x25); - alc_write_coef_idx(codec, 0x26, seq->coeff_0x26); -} - -static inline void alc298_samsung_write_coef_pack_seq( - struct hda_codec *codec, - unsigned char target, - const struct alc298_samsung_coeff_seq_desc seq[], - int count) -{ - alc_write_coef_idx(codec, 0x22, target); - for (int i = 0; i < count; i++) - alc298_samsung_write_coef_pack2(codec, &seq[i]); -} - -static void alc298_fixup_samsung_amp2(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - int i; - static const struct alc298_samsung_coeff_fixup_desc fixups1[] = { - { 0x99, 0x8000 }, { 0x82, 0x4408 }, { 0x32, 0x3f00 }, { 0x0e, 0x6f80 }, - { 0x10, 0x0e21 }, { 0x55, 0x8000 }, { 0x08, 0x2fcf }, { 0x08, 0x2fcf }, - { 0x2d, 0xc020 }, { 0x19, 0x0017 }, { 0x50, 0x1000 }, { 0x0e, 0x6f80 }, - { 0x08, 0x2fcf }, { 0x80, 0x0011 }, { 0x2b, 0x0c10 }, { 0x2d, 0xc020 }, - { 0x03, 0x0042 }, { 0x0f, 0x0062 }, { 0x08, 0x2fcf }, - }; - - static const struct alc298_samsung_coeff_seq_desc amp_0x38[] = { - { 0x2000, 0x0000, 0x0001, 0xb011 }, { 0x23ff, 0x0000, 0x0000, 0xb011 }, - { 0x203a, 0x0000, 0x0080, 0xb011 }, { 0x23e1, 0x0000, 0x0000, 0xb011 }, - { 0x2012, 0x0000, 0x006f, 0xb011 }, { 0x2014, 0x0000, 0x0000, 0xb011 }, - { 0x201b, 0x0000, 0x0001, 0xb011 }, { 0x201d, 0x0000, 0x0001, 0xb011 }, - { 0x201f, 0x0000, 0x00fe, 0xb011 }, { 0x2021, 0x0000, 0x0000, 0xb011 }, - { 0x2022, 0x0000, 0x0010, 0xb011 }, { 0x203d, 0x0000, 0x0005, 0xb011 }, - { 0x203f, 0x0000, 0x0003, 0xb011 }, { 0x2050, 0x0000, 0x002c, 0xb011 }, - { 0x2076, 0x0000, 0x000e, 0xb011 }, { 0x207c, 0x0000, 0x004a, 0xb011 }, - { 0x2081, 0x0000, 0x0003, 0xb011 }, { 0x2399, 0x0000, 0x0003, 0xb011 }, - { 0x23a4, 0x0000, 0x00b5, 0xb011 }, { 0x23a5, 0x0000, 0x0001, 0xb011 }, - { 0x23ba, 0x0000, 0x0094, 0xb011 }, { 0x2100, 0x00d0, 0x950e, 0xb017 }, - { 0x2104, 0x0061, 0xd4e2, 0xb017 }, { 0x2108, 0x00d0, 0x950e, 0xb017 }, - { 0x210c, 0x0075, 0xf4e2, 0xb017 }, { 0x2110, 0x00b4, 0x4b0d, 0xb017 }, - { 0x2114, 0x000a, 0x1000, 0xb017 }, { 0x2118, 0x0015, 0x2000, 0xb017 }, - { 0x211c, 0x000a, 0x1000, 0xb017 }, { 0x2120, 0x0075, 0xf4e2, 0xb017 }, - { 0x2124, 0x00b4, 0x4b0d, 0xb017 }, { 0x2128, 0x0000, 0x0010, 0xb017 }, - { 0x212c, 0x0000, 0x0000, 0xb017 }, { 0x2130, 0x0000, 0x0000, 0xb017 }, - { 0x2134, 0x0000, 0x0000, 0xb017 }, { 0x2138, 0x0000, 0x0000, 0xb017 }, - { 0x213c, 0x0000, 0x0010, 0xb017 }, { 0x2140, 0x0000, 0x0000, 0xb017 }, - { 0x2144, 0x0000, 0x0000, 0xb017 }, { 0x2148, 0x0000, 0x0000, 0xb017 }, - { 0x214c, 0x0000, 0x0000, 0xb017 }, { 0x2150, 0x0000, 0x0010, 0xb017 }, - { 0x2154, 0x0000, 0x0000, 0xb017 }, { 0x2158, 0x0000, 0x0000, 0xb017 }, - { 0x215c, 0x0000, 0x0000, 0xb017 }, { 0x2160, 0x0000, 0x0000, 0xb017 }, - { 0x2164, 0x0000, 0x0010, 0xb017 }, { 0x2168, 0x0000, 0x0000, 0xb017 }, - { 0x216c, 0x0000, 0x0000, 0xb017 }, { 0x2170, 0x0000, 0x0000, 0xb017 }, - { 0x2174, 0x0000, 0x0000, 0xb017 }, { 0x2178, 0x0000, 0x0010, 0xb017 }, - { 0x217c, 0x0000, 0x0000, 0xb017 }, { 0x2180, 0x0000, 0x0000, 0xb017 }, - { 0x2184, 0x0000, 0x0000, 0xb017 }, { 0x2188, 0x0000, 0x0000, 0xb017 }, - { 0x218c, 0x0064, 0x5800, 0xb017 }, { 0x2190, 0x00c8, 0xb000, 0xb017 }, - { 0x2194, 0x0064, 0x5800, 0xb017 }, { 0x2198, 0x003d, 0x5be7, 0xb017 }, - { 0x219c, 0x0054, 0x060a, 0xb017 }, { 0x21a0, 0x00c8, 0xa310, 0xb017 }, - { 0x21a4, 0x0029, 0x4de5, 0xb017 }, { 0x21a8, 0x0032, 0x420c, 0xb017 }, - { 0x21ac, 0x0029, 0x4de5, 0xb017 }, { 0x21b0, 0x00fa, 0xe50c, 0xb017 }, - { 0x21b4, 0x0000, 0x0010, 0xb017 }, { 0x21b8, 0x0000, 0x0000, 0xb017 }, - { 0x21bc, 0x0000, 0x0000, 0xb017 }, { 0x21c0, 0x0000, 0x0000, 0xb017 }, - { 0x21c4, 0x0000, 0x0000, 0xb017 }, { 0x21c8, 0x0056, 0xc50f, 0xb017 }, - { 0x21cc, 0x007b, 0xd7e1, 0xb017 }, { 0x21d0, 0x0077, 0xa70e, 0xb017 }, - { 0x21d4, 0x00e0, 0xbde1, 0xb017 }, { 0x21d8, 0x0032, 0x530e, 0xb017 }, - { 0x2204, 0x00fb, 0x7e0f, 0xb017 }, { 0x2208, 0x000b, 0x02e1, 0xb017 }, - { 0x220c, 0x00fb, 0x7e0f, 0xb017 }, { 0x2210, 0x00d5, 0x17e1, 0xb017 }, - { 0x2214, 0x00c0, 0x130f, 0xb017 }, { 0x2218, 0x00e5, 0x0a00, 0xb017 }, - { 0x221c, 0x00cb, 0x1500, 0xb017 }, { 0x2220, 0x00e5, 0x0a00, 0xb017 }, - { 0x2224, 0x00d5, 0x17e1, 0xb017 }, { 0x2228, 0x00c0, 0x130f, 0xb017 }, - { 0x222c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2230, 0x0017, 0x48e2, 0xb017 }, - { 0x2234, 0x00f5, 0xdb0e, 0xb017 }, { 0x2238, 0x00ef, 0x5ce2, 0xb017 }, - { 0x223c, 0x00c1, 0xcc0d, 0xb017 }, { 0x2240, 0x00f5, 0xdb0e, 0xb017 }, - { 0x2244, 0x0017, 0x48e2, 0xb017 }, { 0x2248, 0x00f5, 0xdb0e, 0xb017 }, - { 0x224c, 0x00ef, 0x5ce2, 0xb017 }, { 0x2250, 0x00c1, 0xcc0d, 0xb017 }, - { 0x2254, 0x00f5, 0xdb0e, 0xb017 }, { 0x2258, 0x0017, 0x48e2, 0xb017 }, - { 0x225c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2260, 0x00ef, 0x5ce2, 0xb017 }, - { 0x2264, 0x00c1, 0xcc0d, 0xb017 }, { 0x2268, 0x00f5, 0xdb0e, 0xb017 }, - { 0x226c, 0x0017, 0x48e2, 0xb017 }, { 0x2270, 0x00f5, 0xdb0e, 0xb017 }, - { 0x2274, 0x00ef, 0x5ce2, 0xb017 }, { 0x2278, 0x00c1, 0xcc0d, 0xb017 }, - { 0x227c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2280, 0x0017, 0x48e2, 0xb017 }, - { 0x2284, 0x00f5, 0xdb0e, 0xb017 }, { 0x2288, 0x00ef, 0x5ce2, 0xb017 }, - { 0x228c, 0x00c1, 0xcc0d, 0xb017 }, { 0x22cc, 0x00e8, 0x8d00, 0xb017 }, - { 0x22d0, 0x0000, 0x0000, 0xb017 }, { 0x22d4, 0x0018, 0x72ff, 0xb017 }, - { 0x22d8, 0x00ce, 0x25e1, 0xb017 }, { 0x22dc, 0x002f, 0xe40e, 0xb017 }, - { 0x238e, 0x0000, 0x0099, 0xb011 }, { 0x238f, 0x0000, 0x0011, 0xb011 }, - { 0x2390, 0x0000, 0x0056, 0xb011 }, { 0x2391, 0x0000, 0x0004, 0xb011 }, - { 0x2392, 0x0000, 0x00bb, 0xb011 }, { 0x2393, 0x0000, 0x006d, 0xb011 }, - { 0x2394, 0x0000, 0x0010, 0xb011 }, { 0x2395, 0x0000, 0x0064, 0xb011 }, - { 0x2396, 0x0000, 0x00b6, 0xb011 }, { 0x2397, 0x0000, 0x0028, 0xb011 }, - { 0x2398, 0x0000, 0x000b, 0xb011 }, { 0x239a, 0x0000, 0x0099, 0xb011 }, - { 0x239b, 0x0000, 0x000d, 0xb011 }, { 0x23a6, 0x0000, 0x0064, 0xb011 }, - { 0x23a7, 0x0000, 0x0078, 0xb011 }, { 0x23b9, 0x0000, 0x0000, 0xb011 }, - { 0x23e0, 0x0000, 0x0021, 0xb011 }, { 0x23e1, 0x0000, 0x0001, 0xb011 }, - }; - - static const struct alc298_samsung_coeff_seq_desc amp_0x39[] = { - { 0x2000, 0x0000, 0x0001, 0xb011 }, { 0x23ff, 0x0000, 0x0000, 0xb011 }, - { 0x203a, 0x0000, 0x0080, 0xb011 }, { 0x23e1, 0x0000, 0x0000, 0xb011 }, - { 0x2012, 0x0000, 0x006f, 0xb011 }, { 0x2014, 0x0000, 0x0000, 0xb011 }, - { 0x201b, 0x0000, 0x0002, 0xb011 }, { 0x201d, 0x0000, 0x0002, 0xb011 }, - { 0x201f, 0x0000, 0x00fd, 0xb011 }, { 0x2021, 0x0000, 0x0001, 0xb011 }, - { 0x2022, 0x0000, 0x0010, 0xb011 }, { 0x203d, 0x0000, 0x0005, 0xb011 }, - { 0x203f, 0x0000, 0x0003, 0xb011 }, { 0x2050, 0x0000, 0x002c, 0xb011 }, - { 0x2076, 0x0000, 0x000e, 0xb011 }, { 0x207c, 0x0000, 0x004a, 0xb011 }, - { 0x2081, 0x0000, 0x0003, 0xb011 }, { 0x2399, 0x0000, 0x0003, 0xb011 }, - { 0x23a4, 0x0000, 0x00b5, 0xb011 }, { 0x23a5, 0x0000, 0x0001, 0xb011 }, - { 0x23ba, 0x0000, 0x0094, 0xb011 }, { 0x2100, 0x00d0, 0x950e, 0xb017 }, - { 0x2104, 0x0061, 0xd4e2, 0xb017 }, { 0x2108, 0x00d0, 0x950e, 0xb017 }, - { 0x210c, 0x0075, 0xf4e2, 0xb017 }, { 0x2110, 0x00b4, 0x4b0d, 0xb017 }, - { 0x2114, 0x000a, 0x1000, 0xb017 }, { 0x2118, 0x0015, 0x2000, 0xb017 }, - { 0x211c, 0x000a, 0x1000, 0xb017 }, { 0x2120, 0x0075, 0xf4e2, 0xb017 }, - { 0x2124, 0x00b4, 0x4b0d, 0xb017 }, { 0x2128, 0x0000, 0x0010, 0xb017 }, - { 0x212c, 0x0000, 0x0000, 0xb017 }, { 0x2130, 0x0000, 0x0000, 0xb017 }, - { 0x2134, 0x0000, 0x0000, 0xb017 }, { 0x2138, 0x0000, 0x0000, 0xb017 }, - { 0x213c, 0x0000, 0x0010, 0xb017 }, { 0x2140, 0x0000, 0x0000, 0xb017 }, - { 0x2144, 0x0000, 0x0000, 0xb017 }, { 0x2148, 0x0000, 0x0000, 0xb017 }, - { 0x214c, 0x0000, 0x0000, 0xb017 }, { 0x2150, 0x0000, 0x0010, 0xb017 }, - { 0x2154, 0x0000, 0x0000, 0xb017 }, { 0x2158, 0x0000, 0x0000, 0xb017 }, - { 0x215c, 0x0000, 0x0000, 0xb017 }, { 0x2160, 0x0000, 0x0000, 0xb017 }, - { 0x2164, 0x0000, 0x0010, 0xb017 }, { 0x2168, 0x0000, 0x0000, 0xb017 }, - { 0x216c, 0x0000, 0x0000, 0xb017 }, { 0x2170, 0x0000, 0x0000, 0xb017 }, - { 0x2174, 0x0000, 0x0000, 0xb017 }, { 0x2178, 0x0000, 0x0010, 0xb017 }, - { 0x217c, 0x0000, 0x0000, 0xb017 }, { 0x2180, 0x0000, 0x0000, 0xb017 }, - { 0x2184, 0x0000, 0x0000, 0xb017 }, { 0x2188, 0x0000, 0x0000, 0xb017 }, - { 0x218c, 0x0064, 0x5800, 0xb017 }, { 0x2190, 0x00c8, 0xb000, 0xb017 }, - { 0x2194, 0x0064, 0x5800, 0xb017 }, { 0x2198, 0x003d, 0x5be7, 0xb017 }, - { 0x219c, 0x0054, 0x060a, 0xb017 }, { 0x21a0, 0x00c8, 0xa310, 0xb017 }, - { 0x21a4, 0x0029, 0x4de5, 0xb017 }, { 0x21a8, 0x0032, 0x420c, 0xb017 }, - { 0x21ac, 0x0029, 0x4de5, 0xb017 }, { 0x21b0, 0x00fa, 0xe50c, 0xb017 }, - { 0x21b4, 0x0000, 0x0010, 0xb017 }, { 0x21b8, 0x0000, 0x0000, 0xb017 }, - { 0x21bc, 0x0000, 0x0000, 0xb017 }, { 0x21c0, 0x0000, 0x0000, 0xb017 }, - { 0x21c4, 0x0000, 0x0000, 0xb017 }, { 0x21c8, 0x0056, 0xc50f, 0xb017 }, - { 0x21cc, 0x007b, 0xd7e1, 0xb017 }, { 0x21d0, 0x0077, 0xa70e, 0xb017 }, - { 0x21d4, 0x00e0, 0xbde1, 0xb017 }, { 0x21d8, 0x0032, 0x530e, 0xb017 }, - { 0x2204, 0x00fb, 0x7e0f, 0xb017 }, { 0x2208, 0x000b, 0x02e1, 0xb017 }, - { 0x220c, 0x00fb, 0x7e0f, 0xb017 }, { 0x2210, 0x00d5, 0x17e1, 0xb017 }, - { 0x2214, 0x00c0, 0x130f, 0xb017 }, { 0x2218, 0x00e5, 0x0a00, 0xb017 }, - { 0x221c, 0x00cb, 0x1500, 0xb017 }, { 0x2220, 0x00e5, 0x0a00, 0xb017 }, - { 0x2224, 0x00d5, 0x17e1, 0xb017 }, { 0x2228, 0x00c0, 0x130f, 0xb017 }, - { 0x222c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2230, 0x0017, 0x48e2, 0xb017 }, - { 0x2234, 0x00f5, 0xdb0e, 0xb017 }, { 0x2238, 0x00ef, 0x5ce2, 0xb017 }, - { 0x223c, 0x00c1, 0xcc0d, 0xb017 }, { 0x2240, 0x00f5, 0xdb0e, 0xb017 }, - { 0x2244, 0x0017, 0x48e2, 0xb017 }, { 0x2248, 0x00f5, 0xdb0e, 0xb017 }, - { 0x224c, 0x00ef, 0x5ce2, 0xb017 }, { 0x2250, 0x00c1, 0xcc0d, 0xb017 }, - { 0x2254, 0x00f5, 0xdb0e, 0xb017 }, { 0x2258, 0x0017, 0x48e2, 0xb017 }, - { 0x225c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2260, 0x00ef, 0x5ce2, 0xb017 }, - { 0x2264, 0x00c1, 0xcc0d, 0xb017 }, { 0x2268, 0x00f5, 0xdb0e, 0xb017 }, - { 0x226c, 0x0017, 0x48e2, 0xb017 }, { 0x2270, 0x00f5, 0xdb0e, 0xb017 }, - { 0x2274, 0x00ef, 0x5ce2, 0xb017 }, { 0x2278, 0x00c1, 0xcc0d, 0xb017 }, - { 0x227c, 0x00f5, 0xdb0e, 0xb017 }, { 0x2280, 0x0017, 0x48e2, 0xb017 }, - { 0x2284, 0x00f5, 0xdb0e, 0xb017 }, { 0x2288, 0x00ef, 0x5ce2, 0xb017 }, - { 0x228c, 0x00c1, 0xcc0d, 0xb017 }, { 0x22cc, 0x00e8, 0x8d00, 0xb017 }, - { 0x22d0, 0x0000, 0x0000, 0xb017 }, { 0x22d4, 0x0018, 0x72ff, 0xb017 }, - { 0x22d8, 0x00ce, 0x25e1, 0xb017 }, { 0x22dc, 0x002f, 0xe40e, 0xb017 }, - { 0x238e, 0x0000, 0x0099, 0xb011 }, { 0x238f, 0x0000, 0x0011, 0xb011 }, - { 0x2390, 0x0000, 0x0056, 0xb011 }, { 0x2391, 0x0000, 0x0004, 0xb011 }, - { 0x2392, 0x0000, 0x00bb, 0xb011 }, { 0x2393, 0x0000, 0x006d, 0xb011 }, - { 0x2394, 0x0000, 0x0010, 0xb011 }, { 0x2395, 0x0000, 0x0064, 0xb011 }, - { 0x2396, 0x0000, 0x00b6, 0xb011 }, { 0x2397, 0x0000, 0x0028, 0xb011 }, - { 0x2398, 0x0000, 0x000b, 0xb011 }, { 0x239a, 0x0000, 0x0099, 0xb011 }, - { 0x239b, 0x0000, 0x000d, 0xb011 }, { 0x23a6, 0x0000, 0x0064, 0xb011 }, - { 0x23a7, 0x0000, 0x0078, 0xb011 }, { 0x23b9, 0x0000, 0x0000, 0xb011 }, - { 0x23e0, 0x0000, 0x0021, 0xb011 }, { 0x23e1, 0x0000, 0x0001, 0xb011 }, - }; - - static const struct alc298_samsung_coeff_seq_desc amp_0x3c[] = { - { 0x2000, 0x0000, 0x0001, 0xb011 }, { 0x23ff, 0x0000, 0x0000, 0xb011 }, - { 0x203a, 0x0000, 0x0080, 0xb011 }, { 0x23e1, 0x0000, 0x0000, 0xb011 }, - { 0x2012, 0x0000, 0x006f, 0xb011 }, { 0x2014, 0x0000, 0x0000, 0xb011 }, - { 0x201b, 0x0000, 0x0001, 0xb011 }, { 0x201d, 0x0000, 0x0001, 0xb011 }, - { 0x201f, 0x0000, 0x00fe, 0xb011 }, { 0x2021, 0x0000, 0x0000, 0xb011 }, - { 0x2022, 0x0000, 0x0010, 0xb011 }, { 0x203d, 0x0000, 0x0005, 0xb011 }, - { 0x203f, 0x0000, 0x0003, 0xb011 }, { 0x2050, 0x0000, 0x002c, 0xb011 }, - { 0x2076, 0x0000, 0x000e, 0xb011 }, { 0x207c, 0x0000, 0x004a, 0xb011 }, - { 0x2081, 0x0000, 0x0003, 0xb011 }, { 0x23ba, 0x0000, 0x008d, 0xb011 }, - { 0x2128, 0x0005, 0x460d, 0xb017 }, { 0x212c, 0x00f6, 0x73e5, 0xb017 }, - { 0x2130, 0x0005, 0x460d, 0xb017 }, { 0x2134, 0x00c0, 0xe9e5, 0xb017 }, - { 0x2138, 0x00d5, 0x010b, 0xb017 }, { 0x213c, 0x009d, 0x7809, 0xb017 }, - { 0x2140, 0x00c5, 0x0eed, 0xb017 }, { 0x2144, 0x009d, 0x7809, 0xb017 }, - { 0x2148, 0x00c4, 0x4ef0, 0xb017 }, { 0x214c, 0x003a, 0x3106, 0xb017 }, - { 0x2150, 0x00af, 0x750e, 0xb017 }, { 0x2154, 0x008c, 0x1ff1, 0xb017 }, - { 0x2158, 0x009e, 0x360c, 0xb017 }, { 0x215c, 0x008c, 0x1ff1, 0xb017 }, - { 0x2160, 0x004d, 0xac0a, 0xb017 }, { 0x2164, 0x007d, 0xa00f, 0xb017 }, - { 0x2168, 0x00e1, 0x9ce3, 0xb017 }, { 0x216c, 0x00e8, 0x590e, 0xb017 }, - { 0x2170, 0x00e1, 0x9ce3, 0xb017 }, { 0x2174, 0x0066, 0xfa0d, 0xb017 }, - { 0x2178, 0x0000, 0x0010, 0xb017 }, { 0x217c, 0x0000, 0x0000, 0xb017 }, - { 0x2180, 0x0000, 0x0000, 0xb017 }, { 0x2184, 0x0000, 0x0000, 0xb017 }, - { 0x2188, 0x0000, 0x0000, 0xb017 }, { 0x218c, 0x0000, 0x0010, 0xb017 }, - { 0x2190, 0x0000, 0x0000, 0xb017 }, { 0x2194, 0x0000, 0x0000, 0xb017 }, - { 0x2198, 0x0000, 0x0000, 0xb017 }, { 0x219c, 0x0000, 0x0000, 0xb017 }, - { 0x21a0, 0x0000, 0x0010, 0xb017 }, { 0x21a4, 0x0000, 0x0000, 0xb017 }, - { 0x21a8, 0x0000, 0x0000, 0xb017 }, { 0x21ac, 0x0000, 0x0000, 0xb017 }, - { 0x21b0, 0x0000, 0x0000, 0xb017 }, { 0x21b4, 0x0000, 0x0010, 0xb017 }, - { 0x21b8, 0x0000, 0x0000, 0xb017 }, { 0x21bc, 0x0000, 0x0000, 0xb017 }, - { 0x21c0, 0x0000, 0x0000, 0xb017 }, { 0x21c4, 0x0000, 0x0000, 0xb017 }, - { 0x23b9, 0x0000, 0x0000, 0xb011 }, { 0x23e0, 0x0000, 0x0020, 0xb011 }, - { 0x23e1, 0x0000, 0x0001, 0xb011 }, - }; - - static const struct alc298_samsung_coeff_seq_desc amp_0x3d[] = { - { 0x2000, 0x0000, 0x0001, 0xb011 }, { 0x23ff, 0x0000, 0x0000, 0xb011 }, - { 0x203a, 0x0000, 0x0080, 0xb011 }, { 0x23e1, 0x0000, 0x0000, 0xb011 }, - { 0x2012, 0x0000, 0x006f, 0xb011 }, { 0x2014, 0x0000, 0x0000, 0xb011 }, - { 0x201b, 0x0000, 0x0002, 0xb011 }, { 0x201d, 0x0000, 0x0002, 0xb011 }, - { 0x201f, 0x0000, 0x00fd, 0xb011 }, { 0x2021, 0x0000, 0x0001, 0xb011 }, - { 0x2022, 0x0000, 0x0010, 0xb011 }, { 0x203d, 0x0000, 0x0005, 0xb011 }, - { 0x203f, 0x0000, 0x0003, 0xb011 }, { 0x2050, 0x0000, 0x002c, 0xb011 }, - { 0x2076, 0x0000, 0x000e, 0xb011 }, { 0x207c, 0x0000, 0x004a, 0xb011 }, - { 0x2081, 0x0000, 0x0003, 0xb011 }, { 0x23ba, 0x0000, 0x008d, 0xb011 }, - { 0x2128, 0x0005, 0x460d, 0xb017 }, { 0x212c, 0x00f6, 0x73e5, 0xb017 }, - { 0x2130, 0x0005, 0x460d, 0xb017 }, { 0x2134, 0x00c0, 0xe9e5, 0xb017 }, - { 0x2138, 0x00d5, 0x010b, 0xb017 }, { 0x213c, 0x009d, 0x7809, 0xb017 }, - { 0x2140, 0x00c5, 0x0eed, 0xb017 }, { 0x2144, 0x009d, 0x7809, 0xb017 }, - { 0x2148, 0x00c4, 0x4ef0, 0xb017 }, { 0x214c, 0x003a, 0x3106, 0xb017 }, - { 0x2150, 0x00af, 0x750e, 0xb017 }, { 0x2154, 0x008c, 0x1ff1, 0xb017 }, - { 0x2158, 0x009e, 0x360c, 0xb017 }, { 0x215c, 0x008c, 0x1ff1, 0xb017 }, - { 0x2160, 0x004d, 0xac0a, 0xb017 }, { 0x2164, 0x007d, 0xa00f, 0xb017 }, - { 0x2168, 0x00e1, 0x9ce3, 0xb017 }, { 0x216c, 0x00e8, 0x590e, 0xb017 }, - { 0x2170, 0x00e1, 0x9ce3, 0xb017 }, { 0x2174, 0x0066, 0xfa0d, 0xb017 }, - { 0x2178, 0x0000, 0x0010, 0xb017 }, { 0x217c, 0x0000, 0x0000, 0xb017 }, - { 0x2180, 0x0000, 0x0000, 0xb017 }, { 0x2184, 0x0000, 0x0000, 0xb017 }, - { 0x2188, 0x0000, 0x0000, 0xb017 }, { 0x218c, 0x0000, 0x0010, 0xb017 }, - { 0x2190, 0x0000, 0x0000, 0xb017 }, { 0x2194, 0x0000, 0x0000, 0xb017 }, - { 0x2198, 0x0000, 0x0000, 0xb017 }, { 0x219c, 0x0000, 0x0000, 0xb017 }, - { 0x21a0, 0x0000, 0x0010, 0xb017 }, { 0x21a4, 0x0000, 0x0000, 0xb017 }, - { 0x21a8, 0x0000, 0x0000, 0xb017 }, { 0x21ac, 0x0000, 0x0000, 0xb017 }, - { 0x21b0, 0x0000, 0x0000, 0xb017 }, { 0x21b4, 0x0000, 0x0010, 0xb017 }, - { 0x21b8, 0x0000, 0x0000, 0xb017 }, { 0x21bc, 0x0000, 0x0000, 0xb017 }, - { 0x21c0, 0x0000, 0x0000, 0xb017 }, { 0x21c4, 0x0000, 0x0000, 0xb017 }, - { 0x23b9, 0x0000, 0x0000, 0xb011 }, { 0x23e0, 0x0000, 0x0020, 0xb011 }, - { 0x23e1, 0x0000, 0x0001, 0xb011 }, - }; - - static const struct alc298_samsung_coeff_seq_desc amp_seq1[] = { - { 0x23ff, 0x0000, 0x0000, 0xb011 }, { 0x203a, 0x0000, 0x0080, 0xb011 }, - }; - - static const struct alc298_samsung_coeff_fixup_desc fixups2[] = { - { 0x4f, 0xb029 }, { 0x05, 0x2be0 }, { 0x30, 0x2421 }, - }; - - - static const struct alc298_samsung_coeff_seq_desc amp_seq2[] = { - { 0x203a, 0x0000, 0x0081, 0xb011 }, { 0x23ff, 0x0000, 0x0001, 0xb011 }, - }; - - if (action != HDA_FIXUP_ACT_INIT) - return; - - // First set of fixups - for (i = 0; i < ARRAY_SIZE(fixups1); i++) - alc_write_coef_idx(codec, fixups1[i].coeff_idx, fixups1[i].coeff_value); - - // First set of writes - alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_0x38, ARRAY_SIZE(amp_0x38)); - alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_0x39, ARRAY_SIZE(amp_0x39)); - alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_0x3c, ARRAY_SIZE(amp_0x3c)); - alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_0x3d, ARRAY_SIZE(amp_0x3d)); - - // Second set of writes - alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq1, ARRAY_SIZE(amp_seq1)); - alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq1, ARRAY_SIZE(amp_seq1)); - alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq1, ARRAY_SIZE(amp_seq1)); - alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq1, ARRAY_SIZE(amp_seq1)); - - // Second set of fixups - for (i = 0; i < ARRAY_SIZE(fixups2); i++) - alc_write_coef_idx(codec, fixups2[i].coeff_idx, fixups2[i].coeff_value); - - // Third set of writes - alc298_samsung_write_coef_pack_seq(codec, 0x38, amp_seq2, ARRAY_SIZE(amp_seq2)); - alc298_samsung_write_coef_pack_seq(codec, 0x39, amp_seq2, ARRAY_SIZE(amp_seq2)); - alc298_samsung_write_coef_pack_seq(codec, 0x3c, amp_seq2, ARRAY_SIZE(amp_seq2)); - alc298_samsung_write_coef_pack_seq(codec, 0x3d, amp_seq2, ARRAY_SIZE(amp_seq2)); - - // Final fixup - alc_write_coef_idx(codec, 0x10, 0x0F21); -} From 5ced8b914ed46edff0d265fb6f397bc851f5fd85 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Sep 2024 13:31:41 +0200 Subject: [PATCH 373/410] ALSA: memalloc: Move snd_malloc_ops definition into memalloc.c again The definition of struct snd_malloc_ops was moved out to memalloc_local.h since there was another code for S/G buffer allocation referring to the struct. But since the code change to use non-contiguous allocators, it's solely referred in memalloc.c, hence it makes little sense to have a separate header file. Let's move it back to memalloc.c. Link: https://patch.msgid.link/20240910113141.32618-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 12 +++++++++++- sound/core/memalloc_local.h | 16 ---------------- 2 files changed, 11 insertions(+), 17 deletions(-) delete mode 100644 sound/core/memalloc_local.h diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f3ad9f85adf1..7237d77713be 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -17,7 +17,17 @@ #include #endif #include -#include "memalloc_local.h" + +struct snd_malloc_ops { + void *(*alloc)(struct snd_dma_buffer *dmab, size_t size); + void (*free)(struct snd_dma_buffer *dmab); + dma_addr_t (*get_addr)(struct snd_dma_buffer *dmab, size_t offset); + struct page *(*get_page)(struct snd_dma_buffer *dmab, size_t offset); + unsigned int (*get_chunk_size)(struct snd_dma_buffer *dmab, + unsigned int ofs, unsigned int size); + int (*mmap)(struct snd_dma_buffer *dmab, struct vm_area_struct *area); + void (*sync)(struct snd_dma_buffer *dmab, enum snd_dma_sync_mode mode); +}; #define DEFAULT_GFP \ (GFP_KERNEL | \ diff --git a/sound/core/memalloc_local.h b/sound/core/memalloc_local.h deleted file mode 100644 index 8b19f3a68a4b..000000000000 --- a/sound/core/memalloc_local.h +++ /dev/null @@ -1,16 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -#ifndef __MEMALLOC_LOCAL_H -#define __MEMALLOC_LOCAL_H - -struct snd_malloc_ops { - void *(*alloc)(struct snd_dma_buffer *dmab, size_t size); - void (*free)(struct snd_dma_buffer *dmab); - dma_addr_t (*get_addr)(struct snd_dma_buffer *dmab, size_t offset); - struct page *(*get_page)(struct snd_dma_buffer *dmab, size_t offset); - unsigned int (*get_chunk_size)(struct snd_dma_buffer *dmab, - unsigned int ofs, unsigned int size); - int (*mmap)(struct snd_dma_buffer *dmab, struct vm_area_struct *area); - void (*sync)(struct snd_dma_buffer *dmab, enum snd_dma_sync_mode mode); -}; - -#endif /* __MEMALLOC_LOCAL_H */ From 86a7f453e99c3c202ac1623557e4f57bd73fc88c Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Tue, 10 Sep 2024 09:33:03 +0800 Subject: [PATCH 374/410] ASoC: soc-ac97: Fix the incorrect description In the function snd_soc_alloc_ac97_component & snd_soc_new_ac97_component, the error return is ERR_PTR, so fix the incorrect description. Fixes: 47e039413cac ("ASoC: Add support for allocating AC'97 device before registering it") Fixes: 7361fbeaeaab ("ASoC: ac97: Add support for resetting device before registration") Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240910013303.2044-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/soc-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 4e4fe29ade50..079e4ff5a14e 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -168,7 +168,7 @@ static void snd_soc_ac97_free_gpio(struct snd_ac97 *ac97) * it. The caller is responsible to either call device_add(&ac97->dev) to * register the device, or to call put_device(&ac97->dev) to free the device. * - * Returns: A snd_ac97 device or a PTR_ERR in case of an error. + * Returns: A snd_ac97 device or an ERR_PTR in case of an error. */ struct snd_ac97 *snd_soc_alloc_ac97_component(struct snd_soc_component *component) { @@ -207,7 +207,7 @@ EXPORT_SYMBOL(snd_soc_alloc_ac97_component); * the device and check if it matches the expected ID. If it doesn't match an * error will be returned and device will not be registered. * - * Returns: A PTR_ERR() on failure or a valid snd_ac97 struct on success. + * Returns: An ERR_PTR on failure or a valid snd_ac97 struct on success. */ struct snd_ac97 *snd_soc_new_ac97_component(struct snd_soc_component *component, unsigned int id, unsigned int id_mask) From 5e6f78cb5f53c52a11090657e917d2d7202aea23 Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Tue, 10 Sep 2024 10:11:04 +0800 Subject: [PATCH 375/410] ASoC: loongson: Add the correct judgement return Use the function dev_err_probe can simplify code, but the error return should not be deleted, that is unreasonable, thus fix it. Fixes: 3d2528d6c021 ("ASoC: loongson: Simplify with dev_err_probe()") Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240910021104.3400-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/loongson/loongson_card.c | 6 +++--- sound/soc/loongson/loongson_i2s_pci.c | 10 +++++----- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/loongson/loongson_card.c b/sound/soc/loongson/loongson_card.c index 6078cdf09c22..7379f24d385c 100644 --- a/sound/soc/loongson/loongson_card.c +++ b/sound/soc/loongson/loongson_card.c @@ -184,16 +184,16 @@ static int loongson_asoc_card_probe(struct platform_device *pdev) ret = device_property_read_string(dev, "model", &card->name); if (ret) - dev_err_probe(dev, ret, "Error parsing card name\n"); + return dev_err_probe(dev, ret, "Error parsing card name\n"); ret = device_property_read_u32(dev, "mclk-fs", &ls_priv->mclk_fs); if (ret) - dev_err_probe(dev, ret, "Error parsing mclk-fs\n"); + return dev_err_probe(dev, ret, "Error parsing mclk-fs\n"); ret = has_acpi_companion(dev) ? loongson_card_parse_acpi(ls_priv) : loongson_card_parse_of(ls_priv); if (ret) - dev_err_probe(dev, ret, "Error parsing acpi/of properties\n"); + return dev_err_probe(dev, ret, "Error parsing acpi/of properties\n"); return devm_snd_soc_register_card(dev, card); } diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index 3872b1d8fce0..d2d0e5d8cac9 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -102,7 +102,7 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, i2s->regmap = devm_regmap_init_mmio(dev, i2s->reg_base, &loongson_i2s_regmap_config); if (IS_ERR(i2s->regmap)) - dev_err_probe(dev, PTR_ERR(i2s->regmap), "regmap_init_mmio failed\n"); + return dev_err_probe(dev, PTR_ERR(i2s->regmap), "regmap_init_mmio failed\n"); tx_data = &i2s->tx_dma_data; rx_data = &i2s->rx_dma_data; @@ -115,15 +115,15 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); if (tx_data->irq < 0) - dev_err_probe(dev, tx_data->irq, "dma tx irq invalid\n"); + return dev_err_probe(dev, tx_data->irq, "dma tx irq invalid\n"); rx_data->irq = fwnode_irq_get_byname(fwnode, "rx"); if (rx_data->irq < 0) - dev_err_probe(dev, rx_data->irq, "dma rx irq invalid\n"); + return dev_err_probe(dev, rx_data->irq, "dma rx irq invalid\n"); ret = device_property_read_u32(dev, "clock-frequency", &i2s->clk_rate); if (ret) - dev_err_probe(dev, ret, "clock-frequency property invalid\n"); + return dev_err_probe(dev, ret, "clock-frequency property invalid\n"); dma_set_mask_and_coherent(dev, DMA_BIT_MASK(64)); @@ -135,7 +135,7 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, ret = devm_snd_soc_register_component(dev, &loongson_i2s_component, &loongson_i2s_dai, 1); if (ret) - dev_err_probe(dev, ret, "register DAI failed\n"); + return dev_err_probe(dev, ret, "register DAI failed\n"); return 0; } From 851e3a2a4490b03bb8dd0cda1b8b2a78f6a92805 Mon Sep 17 00:00:00 2001 From: Jens Reidel Date: Mon, 26 Aug 2024 15:49:20 +0200 Subject: [PATCH 376/410] ASoC: qcom: sm8250: enable primary mi2s When using primary mi2s on sm8250-compatible SoCs, the correct clock needs to get enabled to be able to use the mi2s interface. Signed-off-by: Jens Reidel Tested-by: Danila Tikhonov # sm7325-nothing-spacewar Link: https://patch.msgid.link/20240826134920.55148-2-adrian@travitia.xyz Reviewed-by: Dmitry Baryshkov Signed-off-by: Mark Brown --- sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index a15dafb99b33..274bab28209a 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -55,6 +55,14 @@ static int sm8250_snd_startup(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = snd_soc_rtd_to_codec(rtd, 0); switch (cpu_dai->id) { + case PRIMARY_MI2S_RX: + codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; + snd_soc_dai_set_sysclk(cpu_dai, + Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, + MI2S_BCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_fmt(cpu_dai, fmt); + snd_soc_dai_set_fmt(codec_dai, codec_dai_fmt); + break; case TERTIARY_MI2S_RX: codec_dai_fmt |= SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_I2S; snd_soc_dai_set_sysclk(cpu_dai, From afe671ac3e9309e55d556d066250bd37d9c47d1a Mon Sep 17 00:00:00 2001 From: Zhang Zekun Date: Tue, 10 Sep 2024 20:23:30 +0800 Subject: [PATCH 377/410] ASoC: meson: Remove unused declartion in header file The declaration of aiu_fifo_hw_free() has been removed since commit e05cde84eabc ("ASoC: meson: Use managed DMA buffer allocation"). Let's remove the unused declaration. Fixes: e05cde84eabc ("ASoC: meson: Use managed DMA buffer allocation") Signed-off-by: Zhang Zekun Reviewed-by: Jerome Brunet Link: https://patch.msgid.link/20240910122330.70684-1-zhangzekun11@huawei.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-fifo.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/meson/aiu-fifo.h b/sound/soc/meson/aiu-fifo.h index 84ab4577815a..b02cfcc4de7f 100644 --- a/sound/soc/meson/aiu-fifo.h +++ b/sound/soc/meson/aiu-fifo.h @@ -38,8 +38,6 @@ int aiu_fifo_prepare(struct snd_pcm_substream *substream, int aiu_fifo_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai); -int aiu_fifo_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai); int aiu_fifo_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); void aiu_fifo_shutdown(struct snd_pcm_substream *substream, From 69f3014248f0f10e24f07a66ae650061ecaf732b Mon Sep 17 00:00:00 2001 From: Andrew Kreimer Date: Wed, 11 Sep 2024 00:12:41 +0300 Subject: [PATCH 378/410] ASoC: tlv320aic31xx: Fix typos Fix typos in comments. Reported-by: Matthew Wilcox Signed-off-by: Andrew Kreimer Link: https://patch.msgid.link/20240910211302.8909-1-algonell@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7e624c4b77b6..187d68e8688c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -895,7 +895,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component, dev_err(component->dev, "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); - /* See bellow for details how fix this. */ + /* See below for details on how to fix this. */ return -EINVAL; } if (bclk_score != 0) { From 0ccbc99e05ec80a7a2348e12132f8341f5b890aa Mon Sep 17 00:00:00 2001 From: Leo Tsai Date: Tue, 10 Sep 2024 14:55:42 +0800 Subject: [PATCH 379/410] ALSA: hda: Add a new CM9825 standard driver The CM9825 is a High Definition Audio Codec. There are 2 independent stereo outputs, one of the stereo outputs is cap-less with HP AMP, and the other is line out to connect the active speaker. The inputs can be Line-in and MIC-in. Signed-off-by: Leo Tsai Link: https://patch.msgid.link/20240910065542.6534-1-antivirus621@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cmedia.c | 269 +++++++++++++++++++++++++++++++++++ 1 file changed, 269 insertions(+) diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index 2ddd33f8dd6c..fe946d407830 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -17,10 +17,231 @@ #include "hda_jack.h" #include "hda_generic.h" +/* CM9825 Offset Definitions */ + +#define CM9825_VERB_SET_HPF_1 0x781 +#define CM9825_VERB_SET_HPF_2 0x785 +#define CM9825_VERB_SET_PLL 0x7a0 +#define CM9825_VERB_SET_NEG 0x7a1 +#define CM9825_VERB_SET_ADCL 0x7a2 +#define CM9825_VERB_SET_DACL 0x7a3 +#define CM9825_VERB_SET_MBIAS 0x7a4 +#define CM9825_VERB_SET_VNEG 0x7a8 +#define CM9825_VERB_SET_D2S 0x7a9 +#define CM9825_VERB_SET_DACTRL 0x7aa +#define CM9825_VERB_SET_PDNEG 0x7ac +#define CM9825_VERB_SET_VDO 0x7ad +#define CM9825_VERB_SET_CDALR 0x7b0 +#define CM9825_VERB_SET_MTCBA 0x7b1 +#define CM9825_VERB_SET_OTP 0x7b2 +#define CM9825_VERB_SET_OCP 0x7b3 +#define CM9825_VERB_SET_GAD 0x7b4 +#define CM9825_VERB_SET_TMOD 0x7b5 +#define CM9825_VERB_SET_SNR 0x7b6 + struct cmi_spec { struct hda_gen_spec gen; + const struct hda_verb *chip_d0_verbs; + const struct hda_verb *chip_d3_verbs; + const struct hda_verb *chip_hp_present_verbs; + const struct hda_verb *chip_hp_remove_verbs; + struct hda_codec *codec; + struct delayed_work unsol_hp_work; + int quirk; }; +static const struct hda_verb cm9825_std_d3_verbs[] = { + /* chip sleep verbs */ + {0x43, CM9825_VERB_SET_D2S, 0x62}, /* depop */ + {0x43, CM9825_VERB_SET_PLL, 0x01}, /* PLL set */ + {0x43, CM9825_VERB_SET_NEG, 0xc2}, /* NEG set */ + {0x43, CM9825_VERB_SET_ADCL, 0x00}, /* ADC */ + {0x43, CM9825_VERB_SET_DACL, 0x02}, /* DACL */ + {0x43, CM9825_VERB_SET_VNEG, 0x50}, /* VOL NEG */ + {0x43, CM9825_VERB_SET_MBIAS, 0x00}, /* MBIAS */ + {0x43, CM9825_VERB_SET_PDNEG, 0x04}, /* SEL OSC */ + {0x43, CM9825_VERB_SET_CDALR, 0xf6}, /* Class D */ + {0x43, CM9825_VERB_SET_OTP, 0xcd}, /* OTP set */ + {} +}; + +static const struct hda_verb cm9825_std_d0_verbs[] = { + /* chip init verbs */ + {0x34, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, /* EAPD set */ + {0x43, CM9825_VERB_SET_SNR, 0x30}, /* SNR set */ + {0x43, CM9825_VERB_SET_PLL, 0x00}, /* PLL set */ + {0x43, CM9825_VERB_SET_ADCL, 0x00}, /* ADC */ + {0x43, CM9825_VERB_SET_DACL, 0x02}, /* DACL */ + {0x43, CM9825_VERB_SET_MBIAS, 0x00}, /* MBIAS */ + {0x43, CM9825_VERB_SET_VNEG, 0x56}, /* VOL NEG */ + {0x43, CM9825_VERB_SET_D2S, 0x62}, /* depop */ + {0x43, CM9825_VERB_SET_DACTRL, 0x00}, /* DACTRL set */ + {0x43, CM9825_VERB_SET_PDNEG, 0x0c}, /* SEL OSC */ + {0x43, CM9825_VERB_SET_VDO, 0x80}, /* VDO set */ + {0x43, CM9825_VERB_SET_CDALR, 0xf4}, /* Class D */ + {0x43, CM9825_VERB_SET_OTP, 0xcd}, /* OTP set */ + {0x43, CM9825_VERB_SET_MTCBA, 0x61}, /* SR set */ + {0x43, CM9825_VERB_SET_OCP, 0x33}, /* OTP set */ + {0x43, CM9825_VERB_SET_GAD, 0x07}, /* ADC -3db */ + {0x43, CM9825_VERB_SET_TMOD, 0x26}, /* Class D clk */ + {0x3C, AC_VERB_SET_AMP_GAIN_MUTE | + AC_AMP_SET_OUTPUT | AC_AMP_SET_RIGHT, 0x2d}, /* Gain set */ + {0x3C, AC_VERB_SET_AMP_GAIN_MUTE | + AC_AMP_SET_OUTPUT | AC_AMP_SET_LEFT, 0x2d}, /* Gain set */ + {0x43, CM9825_VERB_SET_HPF_1, 0x40}, /* HPF set */ + {0x43, CM9825_VERB_SET_HPF_2, 0x40}, /* HPF set */ + {} +}; + +static const struct hda_verb cm9825_hp_present_verbs[] = { + {0x42, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, /* PIN off */ + {0x43, CM9825_VERB_SET_ADCL, 0x88}, /* ADC */ + {0x43, CM9825_VERB_SET_DACL, 0xaa}, /* DACL */ + {0x43, CM9825_VERB_SET_MBIAS, 0x10}, /* MBIAS */ + {0x43, CM9825_VERB_SET_D2S, 0xf2}, /* depop */ + {0x43, CM9825_VERB_SET_DACTRL, 0x00}, /* DACTRL set */ + {0x43, CM9825_VERB_SET_VDO, 0xc4}, /* VDO set */ + {} +}; + +static const struct hda_verb cm9825_hp_remove_verbs[] = { + {0x43, CM9825_VERB_SET_ADCL, 0x00}, /* ADC */ + {0x43, CM9825_VERB_SET_DACL, 0x56}, /* DACL */ + {0x43, CM9825_VERB_SET_MBIAS, 0x00}, /* MBIAS */ + {0x43, CM9825_VERB_SET_D2S, 0x62}, /* depop */ + {0x43, CM9825_VERB_SET_DACTRL, 0xe0}, /* DACTRL set */ + {0x43, CM9825_VERB_SET_VDO, 0x80}, /* VDO set */ + {0x42, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, /* PIN on */ + {} +}; + +static void cm9825_unsol_hp_delayed(struct work_struct *work) +{ + struct cmi_spec *spec = + container_of(to_delayed_work(work), struct cmi_spec, unsol_hp_work); + struct hda_jack_tbl *jack; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_jack_plugin = false; + int err = 0; + + hp_jack_plugin = snd_hda_jack_detect(spec->codec, hp_pin); + + codec_dbg(spec->codec, "hp_jack_plugin %d, hp_pin 0x%X\n", + (int)hp_jack_plugin, hp_pin); + + if (!hp_jack_plugin) { + err = + snd_hda_codec_write(spec->codec, 0x42, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40); + if (err) + codec_dbg(spec->codec, "codec_write err %d\n", err); + + snd_hda_sequence_write(spec->codec, spec->chip_hp_remove_verbs); + } else { + snd_hda_sequence_write(spec->codec, + spec->chip_hp_present_verbs); + } + + jack = snd_hda_jack_tbl_get(spec->codec, hp_pin); + if (jack) { + jack->block_report = 0; + snd_hda_jack_report_sync(spec->codec); + } +} + +static void hp_callback(struct hda_codec *codec, struct hda_jack_callback *cb) +{ + struct cmi_spec *spec = codec->spec; + struct hda_jack_tbl *tbl; + + /* Delay enabling the HP amp, to let the mic-detection + * state machine run. + */ + + codec_dbg(spec->codec, "cb->nid 0x%X\n", cb->nid); + + tbl = snd_hda_jack_tbl_get(codec, cb->nid); + if (tbl) + tbl->block_report = 1; + schedule_delayed_work(&spec->unsol_hp_work, msecs_to_jiffies(200)); +} + +static void cm9825_setup_unsol(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + + snd_hda_jack_detect_enable_callback(codec, hp_pin, hp_callback); +} + +static int cm9825_init(struct hda_codec *codec) +{ + snd_hda_gen_init(codec); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + + return 0; +} + +static void cm9825_free(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->unsol_hp_work); + snd_hda_gen_free(codec); +} + +static int cm9825_suspend(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + + cancel_delayed_work_sync(&spec->unsol_hp_work); + + snd_hda_sequence_write(codec, spec->chip_d3_verbs); + + return 0; +} + +static int cm9825_resume(struct hda_codec *codec) +{ + struct cmi_spec *spec = codec->spec; + hda_nid_t hp_pin = 0; + bool hp_jack_plugin = false; + int err; + + err = + snd_hda_codec_write(spec->codec, 0x42, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00); + if (err) + codec_dbg(codec, "codec_write err %d\n", err); + + msleep(150); /* for depop noise */ + + codec->patch_ops.init(codec); + + hp_pin = spec->gen.autocfg.hp_pins[0]; + hp_jack_plugin = snd_hda_jack_detect(spec->codec, hp_pin); + + codec_dbg(spec->codec, "hp_jack_plugin %d, hp_pin 0x%X\n", + (int)hp_jack_plugin, hp_pin); + + if (!hp_jack_plugin) { + err = + snd_hda_codec_write(spec->codec, 0x42, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40); + + if (err) + codec_dbg(codec, "codec_write err %d\n", err); + + snd_hda_sequence_write(codec, cm9825_hp_remove_verbs); + } + + snd_hda_regmap_sync(codec); + hda_call_check_power_status(codec, 0x01); + + return 0; +} + /* * stuff for auto-parser */ @@ -32,6 +253,53 @@ static const struct hda_codec_ops cmi_auto_patch_ops = { .unsol_event = snd_hda_jack_unsol_event, }; +static int patch_cm9825(struct hda_codec *codec) +{ + struct cmi_spec *spec; + struct auto_pin_cfg *cfg; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + INIT_DELAYED_WORK(&spec->unsol_hp_work, cm9825_unsol_hp_delayed); + codec->spec = spec; + spec->codec = codec; + codec->patch_ops = cmi_auto_patch_ops; + codec->patch_ops.init = cm9825_init; + codec->patch_ops.suspend = cm9825_suspend; + codec->patch_ops.resume = cm9825_resume; + codec->patch_ops.free = cm9825_free; + codec->patch_ops.check_power_status = snd_hda_gen_check_power_status; + cfg = &spec->gen.autocfg; + snd_hda_gen_spec_init(&spec->gen); + spec->chip_d0_verbs = cm9825_std_d0_verbs; + spec->chip_d3_verbs = cm9825_std_d3_verbs; + spec->chip_hp_present_verbs = cm9825_hp_present_verbs; + spec->chip_hp_remove_verbs = cm9825_hp_remove_verbs; + + snd_hda_sequence_write(codec, spec->chip_d0_verbs); + + err = snd_hda_parse_pin_defcfg(codec, cfg, NULL, 0); + if (err < 0) + goto error; + err = snd_hda_gen_parse_auto_config(codec, cfg); + if (err < 0) + goto error; + + cm9825_setup_unsol(codec); + + return 0; + + error: + cm9825_free(codec); + + codec_info(codec, "Enter err %d\n", err); + + return err; +} + static int patch_cmi9880(struct hda_codec *codec) { struct cmi_spec *spec; @@ -113,6 +381,7 @@ static const struct hda_device_id snd_hda_id_cmedia[] = { HDA_CODEC_ENTRY(0x13f68888, "CMI8888", patch_cmi8888), HDA_CODEC_ENTRY(0x13f69880, "CMI9880", patch_cmi9880), HDA_CODEC_ENTRY(0x434d4980, "CMI9880", patch_cmi9880), + HDA_CODEC_ENTRY(0x13f69825, "CM9825", patch_cm9825), {} /* terminator */ }; MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cmedia); From 9408ace468c317d60159c011b863b2982dae5e05 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 Sep 2024 13:30:59 +0200 Subject: [PATCH 380/410] ALSA: memalloc: Drop Xen PV workaround again Since recently in the commit e469e2045f1b ("ALSA: memalloc: Let IOMMU handle S/G primarily"), the SG buffer allocation code was modified to use the standard DMA code primarily and the fallback is applied only limitedly. This made the Xen PV specific workarounds we took in the commit 53466ebdec61 ("ALSA: memalloc: Workaround for Xen PV") rather superfluous. It was a hackish workaround for the regression at that time, and it seems that it's causing another issues (reportedly memory corruptions). So it's better to clean it up, after all. Link: https://lore.kernel.org/20240906184209.25423-1-ariadne@ariadne.space Cc: Ariadne Conill Link: https://patch.msgid.link/20240910113100.32542-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 19 +++---------------- 1 file changed, 3 insertions(+), 16 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 7237d77713be..b21dba4b374a 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -667,7 +667,6 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { #ifdef CONFIG_SND_DMA_SGBUF /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { - bool use_dma_alloc_coherent; size_t count; struct page **pages; /* DMA address array; the first page contains #pages in ~PAGE_MASK */ @@ -687,13 +686,8 @@ static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, size = sgbuf->addrs[i] & ~PAGE_MASK; if (WARN_ON(!size)) break; - if (sgbuf->use_dma_alloc_coherent) - dma_free_coherent(dmab->dev.dev, size << PAGE_SHIFT, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i] & PAGE_MASK); - else - do_free_pages(page_address(sgbuf->pages[i]), - size << PAGE_SHIFT, false); + do_free_pages(page_address(sgbuf->pages[i]), + size << PAGE_SHIFT, false); i += size; } } @@ -715,7 +709,6 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) return NULL; - sgbuf->use_dma_alloc_coherent = cpu_feature_enabled(X86_FEATURE_XENPV); size = PAGE_ALIGN(size); sgbuf->count = size >> PAGE_SHIFT; sgbuf->pages = kvcalloc(sgbuf->count, sizeof(*sgbuf->pages), GFP_KERNEL); @@ -728,10 +721,7 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) chunk = (PAGE_SIZE - 1) << PAGE_SHIFT; /* to fit in low bits in addrs */ while (size > 0) { chunk = min(size, chunk); - if (sgbuf->use_dma_alloc_coherent) - p = dma_alloc_coherent(dmab->dev.dev, chunk, &addr, DEFAULT_GFP); - else - p = do_alloc_pages(dmab->dev.dev, chunk, &addr, false); + p = do_alloc_pages(dmab->dev.dev, chunk, &addr, false); if (!p) { if (chunk <= PAGE_SIZE) goto error; @@ -803,9 +793,6 @@ static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) int type = dmab->dev.type; void *p; - if (cpu_feature_enabled(X86_FEATURE_XENPV)) - return snd_dma_sg_fallback_alloc(dmab, size); - /* try the standard DMA API allocation at first */ if (type == SNDRV_DMA_TYPE_DEV_WC_SG) dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC; From bacae49eccb9a3acaf74fc275893abc26c0420b5 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 11 Sep 2024 15:05:53 +0530 Subject: [PATCH 381/410] ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver As module device table added for AMD legacy machine driver, MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD legacy machine driver. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240911093554.2076872-1-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-legacy-mach.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 0d529e32e552..d104f7e8fdcd 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -242,11 +242,4 @@ module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); MODULE_DESCRIPTION("ACP chrome audio support"); -MODULE_ALIAS("platform:acp3xalc56821019"); -MODULE_ALIAS("platform:acp3xalc5682sm98360"); -MODULE_ALIAS("platform:acp3xalc5682s1019"); -MODULE_ALIAS("platform:acp3x-es83xx"); -MODULE_ALIAS("platform:rmb-nau8825-max"); -MODULE_ALIAS("platform:rmb-rt5682s-rt1019"); -MODULE_ALIAS("platform:acp-pdm-mach"); MODULE_LICENSE("GPL v2"); From 0b0aa67baa8904e3c1e13be48a2ca125f59ead3d Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 11 Sep 2024 15:05:54 +0530 Subject: [PATCH 382/410] ASoC: amd: acp: remove MODULE_ALIAS for sof based generic machine driver As module device table added for AMD sof based generic machine driver, MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD sof based generic machine driver. Signed-off-by: Vijendar Mukunda Link: https://patch.msgid.link/20240911093554.2076872-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sof-mach.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index b3a702dcd991..f36750167fa2 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -173,11 +173,4 @@ module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); MODULE_DESCRIPTION("ACP SOF Machine Driver"); -MODULE_ALIAS("platform:rt5682-rt1019"); -MODULE_ALIAS("platform:rt5682-max"); -MODULE_ALIAS("platform:rt5682s-max"); -MODULE_ALIAS("platform:rt5682s-rt1019"); -MODULE_ALIAS("platform:nau8825-max"); -MODULE_ALIAS("platform:rt5682s-hs-rt1019"); -MODULE_ALIAS("platform:nau8821-max"); MODULE_LICENSE("GPL v2"); From 5c4e15e63216e7268bb2f1132ee8fad0ec46bbb7 Mon Sep 17 00:00:00 2001 From: Muhammad Usama Anjum Date: Wed, 11 Sep 2024 16:13:07 +0500 Subject: [PATCH 383/410] ASoC: mediatek: mt8365: check validity before usage of i2s_data There may be a case where i2s_data may not get initialized by the for loop which will cause the kernel crash. Initialize the i2s_data to NULL and abort execute if it isn't found. Fixes: 402bbb13a195 ("ASoC: mediatek: mt8365: Add I2S DAI support") Signed-off-by: Muhammad Usama Anjum Reviewed-by: AngeloGioacchino Del Regno Link: https://patch.msgid.link/20240911111317.4072349-1-usama.anjum@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c index 3482d8f8b4e7..11b9a5bc7163 100644 --- a/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c +++ b/sound/soc/mediatek/mt8365/mt8365-dai-i2s.c @@ -465,13 +465,16 @@ void mt8365_afe_set_i2s_out_enable(struct mtk_base_afe *afe, bool enable) int i; unsigned long flags; struct mt8365_afe_private *afe_priv = afe->platform_priv; - struct mtk_afe_i2s_priv *i2s_data; + struct mtk_afe_i2s_priv *i2s_data = NULL; for (i = 0; i < DAI_I2S_NUM; i++) { if (mt8365_i2s_priv[i].adda_link) i2s_data = afe_priv->dai_priv[mt8365_i2s_priv[i].id]; } + if (!i2s_data) + return; + spin_lock_irqsave(&afe_priv->afe_ctrl_lock, flags); if (enable) { From 9a26234423b87e111351eac0e95775e6ba14d752 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 11 Sep 2024 15:57:52 +0200 Subject: [PATCH 384/410] ALSA: pcm: Fix breakage of PCM rates used for topology It turned out that the topology ABI takes the standard PCM rate bits as is, and it means that the recent change of the PCM rate bits would lead to the inconsistent rate values used for topology. This patch reverts the original PCM rate bit definitions while adding the new rates to the extended bits instead. This needed the change of snd_pcm_known_rates, too. And this also required to fix the handling in snd_pcm_hw_limit_rates() that blindly assumed that the list is sorted while it became unsorted now. Fixes: 090624b7dc83 ("ALSA: pcm: add more sample rate definitions") Reported-by: Pierre-Louis Bossart Closes: https://lore.kernel.org/1ab3efaa-863c-4dd0-8f81-b50fd9775fad@linux.intel.com Reviewed-by: Jaroslav Kysela Tested-by: Jerome Brunet Tested-by: Bard Liao Link: https://patch.msgid.link/20240911135756.24434-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 35 ++++++++++++++++++----------------- sound/core/pcm_misc.c | 18 ++++++++++-------- sound/core/pcm_native.c | 10 +++++++--- 3 files changed, 35 insertions(+), 28 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c993350975a9..0bf7d25434d7 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -109,23 +109,24 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_5512 (1U<<0) /* 5512Hz */ #define SNDRV_PCM_RATE_8000 (1U<<1) /* 8000Hz */ #define SNDRV_PCM_RATE_11025 (1U<<2) /* 11025Hz */ -#define SNDRV_PCM_RATE_12000 (1U<<3) /* 12000Hz */ -#define SNDRV_PCM_RATE_16000 (1U<<4) /* 16000Hz */ -#define SNDRV_PCM_RATE_22050 (1U<<5) /* 22050Hz */ -#define SNDRV_PCM_RATE_24000 (1U<<6) /* 24000Hz */ -#define SNDRV_PCM_RATE_32000 (1U<<7) /* 32000Hz */ -#define SNDRV_PCM_RATE_44100 (1U<<8) /* 44100Hz */ -#define SNDRV_PCM_RATE_48000 (1U<<9) /* 48000Hz */ -#define SNDRV_PCM_RATE_64000 (1U<<10) /* 64000Hz */ -#define SNDRV_PCM_RATE_88200 (1U<<11) /* 88200Hz */ -#define SNDRV_PCM_RATE_96000 (1U<<12) /* 96000Hz */ -#define SNDRV_PCM_RATE_128000 (1U<<13) /* 128000Hz */ -#define SNDRV_PCM_RATE_176400 (1U<<14) /* 176400Hz */ -#define SNDRV_PCM_RATE_192000 (1U<<15) /* 192000Hz */ -#define SNDRV_PCM_RATE_352800 (1U<<16) /* 352800Hz */ -#define SNDRV_PCM_RATE_384000 (1U<<17) /* 384000Hz */ -#define SNDRV_PCM_RATE_705600 (1U<<18) /* 705600Hz */ -#define SNDRV_PCM_RATE_768000 (1U<<19) /* 768000Hz */ +#define SNDRV_PCM_RATE_16000 (1U<<3) /* 16000Hz */ +#define SNDRV_PCM_RATE_22050 (1U<<4) /* 22050Hz */ +#define SNDRV_PCM_RATE_32000 (1U<<5) /* 32000Hz */ +#define SNDRV_PCM_RATE_44100 (1U<<6) /* 44100Hz */ +#define SNDRV_PCM_RATE_48000 (1U<<7) /* 48000Hz */ +#define SNDRV_PCM_RATE_64000 (1U<<8) /* 64000Hz */ +#define SNDRV_PCM_RATE_88200 (1U<<9) /* 88200Hz */ +#define SNDRV_PCM_RATE_96000 (1U<<10) /* 96000Hz */ +#define SNDRV_PCM_RATE_176400 (1U<<11) /* 176400Hz */ +#define SNDRV_PCM_RATE_192000 (1U<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1U<<13) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1U<<14) /* 384000Hz */ +#define SNDRV_PCM_RATE_705600 (1U<<15) /* 705600Hz */ +#define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */ +/* extended rates since 6.12 */ +#define SNDRV_PCM_RATE_12000 (1U<<17) /* 12000Hz */ +#define SNDRV_PCM_RATE_24000 (1U<<18) /* 24000Hz */ +#define SNDRV_PCM_RATE_128000 (1U<<19) /* 128000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuous rates */ diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index 5588b6a1ee8b..4f556211bb56 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -494,18 +494,20 @@ EXPORT_SYMBOL(snd_pcm_format_set_silence); int snd_pcm_hw_limit_rates(struct snd_pcm_hardware *hw) { int i; + unsigned int rmin, rmax; + + rmin = UINT_MAX; + rmax = 0; for (i = 0; i < (int)snd_pcm_known_rates.count; i++) { if (hw->rates & (1 << i)) { - hw->rate_min = snd_pcm_known_rates.list[i]; - break; - } - } - for (i = (int)snd_pcm_known_rates.count - 1; i >= 0; i--) { - if (hw->rates & (1 << i)) { - hw->rate_max = snd_pcm_known_rates.list[i]; - break; + rmin = min(rmin, snd_pcm_known_rates.list[i]); + rmax = max(rmax, snd_pcm_known_rates.list[i]); } } + if (rmin > rmax) + return -EINVAL; + hw->rate_min = rmin; + hw->rate_max = rmax; return 0; } EXPORT_SYMBOL(snd_pcm_hw_limit_rates); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7461a727615c..5e1e6006707b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2418,13 +2418,17 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_768000 != 1 << 19 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 ||\ + SNDRV_PCM_RATE_128000 != 1 << 19 #error "Change this table" #endif +/* NOTE: the list is unsorted! */ static const unsigned int rates[] = { - 5512, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, - 88200, 96000, 128000, 176400, 192000, 352800, 384000, 705600, 768000, + 5512, 8000, 11025, 16000, 22050, 32000, 44100, + 48000, 64000, 88200, 96000, 176400, 192000, 352800, 384000, 705600, 768000, + /* extended */ + 12000, 24000, 128000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { From 8f0280c84607afe122788e508a171ba163d71be6 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Wed, 11 Sep 2024 15:29:07 +0300 Subject: [PATCH 385/410] ASoC: atmel: mchp-pdmc: Improve maxburst calculation for better performance Improve the DMA descriptor calculation by dividing the period size by the product of sample size and DMA chunk size, rather than just DMA chunk size. Ensure that all DMA descriptors start from a well-aligned address to improve the reliability and efficiency of DMA operations and avoid potential issues related to misaligned descriptors. [andrei.simion@microchip.com: Adjust the commit title. Reword the commit message. Add MACROS for each DMA size chunk supported by mchp-pdmc. Add DMA_BURST_ALIGNED preprocesor function to check the alignment of the DMA burst.] Signed-off-by: Codrin Ciubotariu Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20240911122909.133399-2-andrei.simion@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-pdmc.c | 39 ++++++++++++++++++++++++++----------- 1 file changed, 28 insertions(+), 11 deletions(-) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index 260074018da9..7a5585839c1d 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -90,6 +90,15 @@ #define MCHP_PDMC_DS_NO 2 #define MCHP_PDMC_EDGE_NO 2 +/* + * ---- DMA chunk size allowed ---- + */ +#define MCHP_PDMC_DMA_8_WORD_CHUNK 8 +#define MCHP_PDMC_DMA_4_WORD_CHUNK 4 +#define MCHP_PDMC_DMA_2_WORD_CHUNK 2 +#define MCHP_PDMC_DMA_1_WORD_CHUNK 1 +#define DMA_BURST_ALIGNED(_p, _s, _w) !(_p % (_s * _w)) + struct mic_map { int ds_pos; int clk_edge; @@ -511,15 +520,18 @@ static u32 mchp_pdmc_mr_set_osr(int audio_filter_en, unsigned int osr) return 0; } -static inline int mchp_pdmc_period_to_maxburst(int period_size) +static inline int mchp_pdmc_period_to_maxburst(int period_size, int sample_size) { - if (!(period_size % 8)) - return 8; - if (!(period_size % 4)) - return 4; - if (!(period_size % 2)) - return 2; - return 1; + int p_size = period_size; + int s_size = sample_size; + + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_PDMC_DMA_8_WORD_CHUNK)) + return MCHP_PDMC_DMA_8_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_PDMC_DMA_4_WORD_CHUNK)) + return MCHP_PDMC_DMA_4_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_PDMC_DMA_2_WORD_CHUNK)) + return MCHP_PDMC_DMA_2_WORD_CHUNK; + return MCHP_PDMC_DMA_1_WORD_CHUNK; } static struct snd_pcm_chmap_elem mchp_pdmc_std_chmaps[] = { @@ -547,14 +559,18 @@ static int mchp_pdmc_hw_params(struct snd_pcm_substream *substream, unsigned int channels = params_channels(params); unsigned int osr = 0, osr_start; unsigned int fs = params_rate(params); + int sample_bytes = params_physical_width(params) / 8; + int period_bytes = params_period_size(params) * + params_channels(params) * sample_bytes; + int maxburst; u32 mr_val = 0; u32 cfgr_val = 0; int i; int ret; - dev_dbg(comp->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + dev_dbg(comp->dev, "%s() rate=%u format=%#x width=%u channels=%u period_bytes=%d\n", __func__, params_rate(params), params_format(params), - params_width(params), params_channels(params)); + params_width(params), params_channels(params), period_bytes); if (channels > dd->mic_no) { dev_err(comp->dev, "more channels %u than microphones %d\n", @@ -608,7 +624,8 @@ static int mchp_pdmc_hw_params(struct snd_pcm_substream *substream, mr_val |= FIELD_PREP(MCHP_PDMC_MR_SINCORDER_MASK, dd->sinc_order); - dd->addr.maxburst = mchp_pdmc_period_to_maxburst(snd_pcm_lib_period_bytes(substream)); + maxburst = mchp_pdmc_period_to_maxburst(period_bytes, sample_bytes); + dd->addr.maxburst = maxburst; mr_val |= FIELD_PREP(MCHP_PDMC_MR_CHUNK_MASK, dd->addr.maxburst); dev_dbg(comp->dev, "maxburst set to %d\n", dd->addr.maxburst); From e6b95bdc1e333e14e4fdf71fd4e8962429d9b6cd Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Wed, 11 Sep 2024 15:29:08 +0300 Subject: [PATCH 386/410] ASoC: atmel: mchp-pdmc: Add snd_soc_dai_driver name Set snd_soc_dai_driver name to improve controller's display of the DAI name. Signed-off-by: Codrin Ciubotariu Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20240911122909.133399-3-andrei.simion@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-pdmc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index 7a5585839c1d..d97d153ee375 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -777,6 +777,7 @@ static const struct snd_soc_dai_ops mchp_pdmc_dai_ops = { }; static struct snd_soc_dai_driver mchp_pdmc_dai = { + .name = "mchp-pdmc", .capture = { .stream_name = "Capture", .channels_min = 1, From a2187d0dadfc308551bbb1b8d6caee69e2ad4744 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 9 Sep 2024 23:13:47 +0000 Subject: [PATCH 387/410] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property At least if rsnd is using DPCM connection on Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto Reviewed-by: Rob Herring (Arm) Link: https://patch.msgid.link/87wmjkifob.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9de..3bc93c59535e 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,12 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: At least if rsnd is using DPCM connection on Audio-Graph-Card2, + fw_devlink might doesn't have enough information to break the cycle. rsnd + driver will not be probed in such case. Same problem might occur with + Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base From 448aa89af07b83be84a58155c60001743342fca0 Mon Sep 17 00:00:00 2001 From: Andrei Simion Date: Tue, 10 Sep 2024 11:22:03 +0300 Subject: [PATCH 388/410] ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference Update the spdifrx yaml file to reference the dai-common.yaml schema, enabling the use of the 'sound-name-prefix' property Signed-off-by: Andrei Simion Acked-by: Rob Herring (Arm) Link: https://patch.msgid.link/20240910082202.45972-1-andrei.simion@microchip.com Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml index 2f43c684ab88..7fbab5871be4 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -13,6 +13,9 @@ description: The Microchip Sony/Philips Digital Interface Receiver is a serial port compliant with the IEC-60958 standard. +allOf: + - $ref: dai-common.yaml# + properties: "#sound-dai-cells": const: 0 @@ -53,7 +56,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | From 12647a7cfbaa865cb291bd36a4c5d8496e28d61b Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 11 Sep 2024 22:50:39 +0300 Subject: [PATCH 389/410] ALSA: ump: Use %*ph to print small buffer Use %*ph format to print small buffer as hex string. Signed-off-by: Andy Shevchenko Link: https://patch.msgid.link/20240911195039.2885979-1-andriy.shevchenko@linux.intel.com Signed-off-by: Takashi Iwai --- sound/core/ump.c | 13 +++---------- 1 file changed, 3 insertions(+), 10 deletions(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index 243ecdbb2a6e..cf22a17e38dd 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -489,11 +489,7 @@ static void snd_ump_proc_read(struct snd_info_entry *entry, ump->info.manufacturer_id); snd_iprintf(buffer, "Family ID: 0x%04x\n", ump->info.family_id); snd_iprintf(buffer, "Model ID: 0x%04x\n", ump->info.model_id); - snd_iprintf(buffer, "SW Revision: 0x%02x%02x%02x%02x\n", - ump->info.sw_revision[0], - ump->info.sw_revision[1], - ump->info.sw_revision[2], - ump->info.sw_revision[3]); + snd_iprintf(buffer, "SW Revision: 0x%4phN\n", ump->info.sw_revision); } snd_iprintf(buffer, "Static Blocks: %s\n", (ump->info.flags & SNDRV_UMP_EP_INFO_STATIC_BLOCKS) ? "Yes" : "No"); @@ -710,14 +706,11 @@ static int ump_handle_device_info_msg(struct snd_ump_endpoint *ump, ump->info.sw_revision[1] = (buf->device_info.sw_revision >> 16) & 0x7f; ump->info.sw_revision[2] = (buf->device_info.sw_revision >> 8) & 0x7f; ump->info.sw_revision[3] = buf->device_info.sw_revision & 0x7f; - ump_dbg(ump, "EP devinfo: manid=%08x, family=%04x, model=%04x, sw=%02x%02x%02x%02x\n", + ump_dbg(ump, "EP devinfo: manid=%08x, family=%04x, model=%04x, sw=%4phN\n", ump->info.manufacturer_id, ump->info.family_id, ump->info.model_id, - ump->info.sw_revision[0], - ump->info.sw_revision[1], - ump->info.sw_revision[2], - ump->info.sw_revision[3]); + ump->info.sw_revision); return 1; /* finished */ } From bd07676ddade417c7cfefb58fb87c27751395bb3 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 12 Sep 2024 20:03:02 +0800 Subject: [PATCH 390/410] ASoC: Intel: board_helpers: support HDA link initialization MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Add a helper function for machine drivers to initialize HDA external codec DAI link. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-2-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_board_helpers.c | 152 +++++++++++++++++++++ sound/soc/intel/boards/sof_board_helpers.h | 3 + 2 files changed, 155 insertions(+) diff --git a/sound/soc/intel/boards/sof_board_helpers.c b/sound/soc/intel/boards/sof_board_helpers.c index 7519c545cbe2..24f716e42d6a 100644 --- a/sound/soc/intel/boards/sof_board_helpers.c +++ b/sound/soc/intel/boards/sof_board_helpers.c @@ -70,6 +70,64 @@ static int dmic_init(struct snd_soc_pcm_runtime *rtd) return 0; } +/* + * HDA External Codec DAI Link + */ +static const struct snd_soc_dapm_widget hda_widgets[] = { + SND_SOC_DAPM_MIC("Analog In", NULL), + SND_SOC_DAPM_MIC("Digital In", NULL), + SND_SOC_DAPM_MIC("Alt Analog In", NULL), + + SND_SOC_DAPM_HP("Analog Out", NULL), + SND_SOC_DAPM_SPK("Digital Out", NULL), + SND_SOC_DAPM_HP("Alt Analog Out", NULL), +}; + +static const struct snd_soc_dapm_route hda_routes[] = { + { "Codec Input Pin1", NULL, "Analog In" }, + { "Codec Input Pin2", NULL, "Digital In" }, + { "Codec Input Pin3", NULL, "Alt Analog In" }, + + { "Analog Out", NULL, "Codec Output Pin1" }, + { "Digital Out", NULL, "Codec Output Pin2" }, + { "Alt Analog Out", NULL, "Codec Output Pin3" }, + + /* CODEC BE connections */ + { "codec0_in", NULL, "Analog CPU Capture" }, + { "Analog CPU Capture", NULL, "Analog Codec Capture" }, + { "codec1_in", NULL, "Digital CPU Capture" }, + { "Digital CPU Capture", NULL, "Digital Codec Capture" }, + { "codec2_in", NULL, "Alt Analog CPU Capture" }, + { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, + + { "Analog Codec Playback", NULL, "Analog CPU Playback" }, + { "Analog CPU Playback", NULL, "codec0_out" }, + { "Digital Codec Playback", NULL, "Digital CPU Playback" }, + { "Digital CPU Playback", NULL, "codec1_out" }, + { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" }, + { "Alt Analog CPU Playback", NULL, "codec2_out" }, +}; + +static int hda_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_new_controls(&card->dapm, hda_widgets, + ARRAY_SIZE(hda_widgets)); + if (ret) { + dev_err(rtd->dev, "fail to add hda widgets, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, hda_routes, + ARRAY_SIZE(hda_routes)); + if (ret) + dev_err(rtd->dev, "fail to add hda routes, ret %d\n", ret); + + return ret; +} + /* * DAI Link Helpers */ @@ -79,6 +137,11 @@ enum sof_dmic_be_type { SOF_DMIC_16K, }; +enum sof_hda_be_type { + SOF_HDA_ANALOG, + SOF_HDA_DIGITAL, +}; + /* DEFAULT_LINK_ORDER: the order used in sof_rt5682 */ #define DEFAULT_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_CODEC, \ SOF_LINK_DMIC01, \ @@ -95,6 +158,16 @@ static struct snd_soc_dai_link_component dmic_component[] = { } }; +SND_SOC_DAILINK_DEF(hda_analog_cpus, + DAILINK_COMP_ARRAY(COMP_CPU("Analog CPU DAI"))); +SND_SOC_DAILINK_DEF(hda_analog_codecs, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D0", "Analog Codec DAI"))); + +SND_SOC_DAILINK_DEF(hda_digital_cpus, + DAILINK_COMP_ARRAY(COMP_CPU("Digital CPU DAI"))); +SND_SOC_DAILINK_DEF(hda_digital_codecs, + DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D0", "Digital Codec DAI"))); + static struct snd_soc_dai_link_component platform_component[] = { { /* name might be overridden during probe */ @@ -380,6 +453,55 @@ static int set_hdmi_in_link(struct device *dev, struct snd_soc_dai_link *link, return 0; } +static int set_hda_codec_link(struct device *dev, struct snd_soc_dai_link *link, + int be_id, enum sof_hda_be_type be_type) +{ + switch (be_type) { + case SOF_HDA_ANALOG: + dev_dbg(dev, "link %d: hda analog\n", be_id); + + link->name = "Analog Playback and Capture"; + + /* cpus */ + link->cpus = hda_analog_cpus; + link->num_cpus = ARRAY_SIZE(hda_analog_cpus); + + /* codecs */ + link->codecs = hda_analog_codecs; + link->num_codecs = ARRAY_SIZE(hda_analog_codecs); + break; + case SOF_HDA_DIGITAL: + dev_dbg(dev, "link %d: hda digital\n", be_id); + + link->name = "Digital Playback and Capture"; + + /* cpus */ + link->cpus = hda_digital_cpus; + link->num_cpus = ARRAY_SIZE(hda_digital_cpus); + + /* codecs */ + link->codecs = hda_digital_codecs; + link->num_codecs = ARRAY_SIZE(hda_digital_codecs); + break; + default: + dev_err(dev, "invalid be type %d\n", be_type); + return -EINVAL; + } + + /* platforms */ + link->platforms = platform_component; + link->num_platforms = ARRAY_SIZE(platform_component); + + link->id = be_id; + if (be_type == SOF_HDA_ANALOG) + link->init = hda_init; + link->no_pcm = 1; + link->dpcm_capture = 1; + link->dpcm_playback = 1; + + return 0; +} + static int calculate_num_links(struct sof_card_private *ctx) { int num_links = 0; @@ -409,6 +531,10 @@ static int calculate_num_links(struct sof_card_private *ctx) /* HDMI-In */ num_links += hweight32(ctx->ssp_mask_hdmi_in); + /* HDA external codec */ + if (ctx->hda_codec_present) + num_links += 2; + return num_links; } @@ -566,6 +692,32 @@ int sof_intel_board_set_dai_link(struct device *dev, struct snd_soc_card *card, be_id++; } break; + case SOF_LINK_HDA: + /* HDA external codec */ + if (!ctx->hda_codec_present) + continue; + + ret = set_hda_codec_link(dev, &links[idx], be_id, + SOF_HDA_ANALOG); + if (ret) { + dev_err(dev, "fail to set hda analog link, ret %d\n", + ret); + return ret; + } + + idx++; + be_id++; + + ret = set_hda_codec_link(dev, &links[idx], be_id, + SOF_HDA_DIGITAL); + if (ret) { + dev_err(dev, "fail to set hda digital link, ret %d\n", + ret); + return ret; + } + + idx++; + break; case SOF_LINK_NONE: /* caught here if it's not used as terminator in macro */ fallthrough; diff --git a/sound/soc/intel/boards/sof_board_helpers.h b/sound/soc/intel/boards/sof_board_helpers.h index faba847bb7c9..33a9601b770c 100644 --- a/sound/soc/intel/boards/sof_board_helpers.h +++ b/sound/soc/intel/boards/sof_board_helpers.h @@ -57,6 +57,7 @@ enum { SOF_LINK_AMP, SOF_LINK_BT_OFFLOAD, SOF_LINK_HDMI_IN, + SOF_LINK_HDA, }; #define SOF_LINK_ORDER_MASK (0xF) @@ -121,6 +122,7 @@ struct sof_rt5682_private { * @ssp_bt: ssp port number of BT offload BE link * @ssp_mask_hdmi_in: ssp port mask of HDMI-IN BE link * @bt_offload_present: true to create BT offload BE link + * @hda_codec_present: true to create HDA codec BE links * @codec_link: pointer to headset codec dai link * @amp_link: pointer to speaker amplifier dai link * @link_order_overwrite: custom DAI link order @@ -144,6 +146,7 @@ struct sof_card_private { unsigned long ssp_mask_hdmi_in; bool bt_offload_present; + bool hda_codec_present; struct snd_soc_dai_link *codec_link; struct snd_soc_dai_link *amp_link; From b28b23dea31497548010c248398162ef4c25cfd2 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 12 Sep 2024 20:03:03 +0800 Subject: [PATCH 391/410] ASoC: Intel: skl_hda_dsp_generic: use common module for DAI links MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Use intel_board module to create DAI link array for Intel iDisp HDMI, HDA external codec, DMIC01, DMIC16K, and BT audio offload DAI BE links. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-3-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 156 ------------ sound/soc/intel/boards/skl_hda_dsp_common.h | 38 --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 252 +++++-------------- 5 files changed, 70 insertions(+), 379 deletions(-) delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 793a9170513a..cc10ae58b0c7 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -306,6 +306,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH tristate "Skylake+ with HDA Codecs" depends on SND_HDA_CODEC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON + select SND_SOC_INTEL_SOF_BOARD_HELPERS select SND_SOC_DMIC # SND_SOC_HDAC_HDA is already selected help diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 9b24de731332..fcd517d6c279 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,7 +21,7 @@ snd-soc-sof_cs42l42-y := sof_cs42l42.o snd-soc-sof_es8336-y := sof_es8336.o snd-soc-sof_nau8825-y := sof_nau8825.o snd-soc-sof_da7219-y := sof_da7219.o -snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o skl_hda_dsp_common.o +snd-soc-skl_hda_dsp-y := skl_hda_dsp_generic.o snd-soc-ehl-rt5660-y := ehl_rt5660.o snd-soc-sof-ssp-amp-y := sof_ssp_amp.o snd-soc-sof-sdw-y += sof_sdw.o \ diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c deleted file mode 100644 index 5019bdfa5b45..000000000000 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ /dev/null @@ -1,156 +0,0 @@ -// SPDX-License-Identifier: GPL-2.0-only -// Copyright(c) 2015-18 Intel Corporation. - -/* - * Common functions used in different Intel machine drivers - */ -#include -#include -#include -#include -#include -#include -#include -#include "skl_hda_dsp_common.h" - -#include -#include "../../codecs/hdac_hda.h" - -#define NAME_SIZE 32 - -int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) -{ - struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai *dai; - char dai_name[NAME_SIZE]; - - snprintf(dai_name, sizeof(dai_name), "intel-hdmi-hifi%d", - ctx->dai_index); - dai = snd_soc_card_get_codec_dai(card, dai_name); - if (!dai) - return -EINVAL; - - ctx->hdmi.hdmi_comp = dai->component; - - return 0; -} - -SND_SOC_DAILINK_DEF(idisp1_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin"))); -SND_SOC_DAILINK_DEF(idisp1_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1"))); - -SND_SOC_DAILINK_DEF(idisp2_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin"))); -SND_SOC_DAILINK_DEF(idisp2_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2"))); - -SND_SOC_DAILINK_DEF(idisp3_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin"))); -SND_SOC_DAILINK_DEF(idisp3_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3"))); - -SND_SOC_DAILINK_DEF(analog_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("Analog CPU DAI"))); -SND_SOC_DAILINK_DEF(analog_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D0", "Analog Codec DAI"))); - -SND_SOC_DAILINK_DEF(digital_cpu, - DAILINK_COMP_ARRAY(COMP_CPU("Digital CPU DAI"))); -SND_SOC_DAILINK_DEF(digital_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D0", "Digital Codec DAI"))); - -SND_SOC_DAILINK_DEF(dmic_pin, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin"))); - -SND_SOC_DAILINK_DEF(dmic_codec, - DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi"))); - -SND_SOC_DAILINK_DEF(dmic16k, - DAILINK_COMP_ARRAY(COMP_CPU("DMIC16k Pin"))); - -SND_SOC_DAILINK_DEF(bt_offload_pin, - DAILINK_COMP_ARRAY(COMP_CPU(""))); /* initialized in driver probe function */ -SND_SOC_DAILINK_DEF(dummy, - DAILINK_COMP_ARRAY(COMP_DUMMY())); - -SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3"))); - -/* skl_hda_digital audio interface glue - connects codec <--> CPU */ -struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { - /* Back End DAI links */ - { - .name = "iDisp1", - .id = 1, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp1_cpu, idisp1_codec, platform), - }, - { - .name = "iDisp2", - .id = 2, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp2_cpu, idisp2_codec, platform), - }, - { - .name = "iDisp3", - .id = 3, - .dpcm_playback = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(idisp3_cpu, idisp3_codec, platform), - }, - { - .name = "Analog Playback and Capture", - .id = 4, - .dpcm_playback = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(analog_cpu, analog_codec, platform), - }, - { - .name = "Digital Playback and Capture", - .id = 5, - .dpcm_playback = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(digital_cpu, digital_codec, platform), - }, - { - .name = "dmic01", - .id = 6, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform), - }, - { - .name = "dmic16k", - .id = 7, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(dmic16k, dmic_codec, platform), - }, - { - .name = NULL, /* initialized in driver probe function */ - .id = 8, - .dpcm_playback = 1, - .dpcm_capture = 1, - .no_pcm = 1, - SND_SOC_DAILINK_REG(bt_offload_pin, dummy, platform), - }, -}; - -int skl_hda_hdmi_jack_init(struct snd_soc_card *card) -{ - struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - - /* HDMI disabled, do not create controls */ - if (!ctx->hdmi.idisp_codec) - return 0; - - if (!ctx->hdmi.hdmi_comp) - return -EINVAL; - - return hda_dsp_hdmi_build_controls(card, ctx->hdmi.hdmi_comp); -} diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.h b/sound/soc/intel/boards/skl_hda_dsp_common.h deleted file mode 100644 index 40ffbccb2fe0..000000000000 --- a/sound/soc/intel/boards/skl_hda_dsp_common.h +++ /dev/null @@ -1,38 +0,0 @@ -/* SPDX-License-Identifier: GPL-2.0-only */ -/* - * Copyright(c) 2015-18 Intel Corporation. - */ - -/* - * This file defines data structures used in Machine Driver for Intel - * platforms with HDA Codecs. - */ - -#ifndef __SKL_HDA_DSP_COMMON_H -#define __SKL_HDA_DSP_COMMON_H -#include -#include -#include -#include -#include -#include "../../codecs/hdac_hda.h" -#include "hda_dsp_common.h" -#include "sof_hdmi_common.h" - -#define HDA_DSP_MAX_BE_DAI_LINKS 8 - -struct skl_hda_private { - struct snd_soc_card card; - struct sof_hdmi_private hdmi; - int pcm_count; - int dai_index; - const char *platform_name; - bool bt_offload_present; - int ssp_bt; -}; - -extern struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS]; -int skl_hda_hdmi_jack_init(struct snd_soc_card *card); -int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device); - -#endif /* __SOUND_SOC_HDA_DSP_COMMON_H */ diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 8e104874d58c..9edd6d985cf1 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -8,178 +8,23 @@ #include #include #include +#include #include #include #include #include #include -#include "skl_hda_dsp_common.h" - -static const struct snd_soc_dapm_widget skl_hda_widgets[] = { - SND_SOC_DAPM_HP("Analog Out", NULL), - SND_SOC_DAPM_MIC("Analog In", NULL), - SND_SOC_DAPM_HP("Alt Analog Out", NULL), - SND_SOC_DAPM_MIC("Alt Analog In", NULL), - SND_SOC_DAPM_SPK("Digital Out", NULL), - SND_SOC_DAPM_MIC("Digital In", NULL), - SND_SOC_DAPM_MIC("SoC DMIC", NULL), -}; - -static const struct snd_soc_dapm_route skl_hda_map[] = { - { "hifi3", NULL, "iDisp3 Tx"}, - { "iDisp3 Tx", NULL, "iDisp3_out"}, - { "hifi2", NULL, "iDisp2 Tx"}, - { "iDisp2 Tx", NULL, "iDisp2_out"}, - { "hifi1", NULL, "iDisp1 Tx"}, - { "iDisp1 Tx", NULL, "iDisp1_out"}, - - { "Analog Out", NULL, "Codec Output Pin1" }, - { "Digital Out", NULL, "Codec Output Pin2" }, - { "Alt Analog Out", NULL, "Codec Output Pin3" }, - - { "Codec Input Pin1", NULL, "Analog In" }, - { "Codec Input Pin2", NULL, "Digital In" }, - { "Codec Input Pin3", NULL, "Alt Analog In" }, - - /* digital mics */ - {"DMic", NULL, "SoC DMIC"}, - - /* CODEC BE connections */ - { "Analog Codec Playback", NULL, "Analog CPU Playback" }, - { "Analog CPU Playback", NULL, "codec0_out" }, - { "Digital Codec Playback", NULL, "Digital CPU Playback" }, - { "Digital CPU Playback", NULL, "codec1_out" }, - { "Alt Analog Codec Playback", NULL, "Alt Analog CPU Playback" }, - { "Alt Analog CPU Playback", NULL, "codec2_out" }, - - { "codec0_in", NULL, "Analog CPU Capture" }, - { "Analog CPU Capture", NULL, "Analog Codec Capture" }, - { "codec1_in", NULL, "Digital CPU Capture" }, - { "Digital CPU Capture", NULL, "Digital Codec Capture" }, - { "codec2_in", NULL, "Alt Analog CPU Capture" }, - { "Alt Analog CPU Capture", NULL, "Alt Analog Codec Capture" }, -}; +#include "../../codecs/hdac_hda.h" +#include "../../sof/intel/hda.h" +#include "sof_board_helpers.h" static int skl_hda_card_late_probe(struct snd_soc_card *card) { - return skl_hda_hdmi_jack_init(card); + return sof_intel_board_card_late_probe(card); } -static int -skl_hda_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) -{ - struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - int ret = 0; - - dev_dbg(card->dev, "dai link name - %s\n", link->name); - link->platforms->name = ctx->platform_name; - link->nonatomic = 1; - - if (!ctx->hdmi.idisp_codec) - return 0; - - if (strstr(link->name, "HDMI")) { - ret = skl_hda_hdmi_add_pcm(card, ctx->pcm_count); - - if (ret < 0) - return ret; - - ctx->dai_index++; - } - - ctx->pcm_count++; - return ret; -} - -#define IDISP_DAI_COUNT 3 -#define HDAC_DAI_COUNT 2 -#define DMIC_DAI_COUNT 2 -#define BT_DAI_COUNT 1 - -/* there are two routes per iDisp output */ -#define IDISP_ROUTE_COUNT (IDISP_DAI_COUNT * 2) - #define HDA_CODEC_AUTOSUSPEND_DELAY_MS 1000 -static int skl_hda_fill_card_info(struct device *dev, struct snd_soc_card *card, - struct snd_soc_acpi_mach_params *mach_params) -{ - struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_dai_link *dai_link; - struct snd_soc_dai_link *bt_link; - u32 codec_count, codec_mask; - int i, num_links, num_route; - - codec_mask = mach_params->codec_mask; - codec_count = hweight_long(codec_mask); - - if (!codec_count || codec_count > 2 || - (codec_count == 2 && !ctx->hdmi.idisp_codec)) - return -EINVAL; - - if (codec_mask == IDISP_CODEC_MASK) { - /* topology with iDisp as the only HDA codec */ - num_links = IDISP_DAI_COUNT + DMIC_DAI_COUNT + BT_DAI_COUNT; - num_route = IDISP_ROUTE_COUNT; - - /* - * rearrange the dai link array and make the - * dmic dai links follow idsp dai links for only - * num_links of dai links need to be registered - * to ASoC. - */ - for (i = 0; i < (DMIC_DAI_COUNT + BT_DAI_COUNT); i++) { - skl_hda_be_dai_links[IDISP_DAI_COUNT + i] = - skl_hda_be_dai_links[IDISP_DAI_COUNT + - HDAC_DAI_COUNT + i]; - } - } else { - /* topology with external and iDisp HDA codecs */ - num_links = ARRAY_SIZE(skl_hda_be_dai_links); - num_route = ARRAY_SIZE(skl_hda_map); - card->dapm_widgets = skl_hda_widgets; - card->num_dapm_widgets = ARRAY_SIZE(skl_hda_widgets); - if (!ctx->hdmi.idisp_codec) { - card->dapm_routes = &skl_hda_map[IDISP_ROUTE_COUNT]; - num_route -= IDISP_ROUTE_COUNT; - for (i = 0; i < IDISP_DAI_COUNT; i++) { - skl_hda_be_dai_links[i].codecs = &snd_soc_dummy_dlc; - skl_hda_be_dai_links[i].num_codecs = 1; - } - } - } - - if (!ctx->bt_offload_present) { - /* remove last link since bt audio offload is not supported */ - num_links -= BT_DAI_COUNT; - } else { - if (codec_mask == IDISP_CODEC_MASK) - bt_link = &skl_hda_be_dai_links[IDISP_DAI_COUNT + DMIC_DAI_COUNT]; - else - bt_link = &skl_hda_be_dai_links[IDISP_DAI_COUNT + HDAC_DAI_COUNT + DMIC_DAI_COUNT]; - - /* complete the link name and dai name with SSP port number */ - bt_link->name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", - ctx->ssp_bt); - if (!bt_link->name) - return -ENOMEM; - - bt_link->cpus->dai_name = devm_kasprintf(dev, GFP_KERNEL, - "SSP%d Pin", - ctx->ssp_bt); - if (!bt_link->cpus->dai_name) - return -ENOMEM; - } - - card->num_links = num_links; - card->num_dapm_routes = num_route; - - for_each_card_prelinks(card, i, dai_link) - dai_link->platforms->name = mach_params->platform; - - return 0; -} - static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; @@ -203,48 +48,80 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) } } +#define IDISP_HDMI_BE_ID 1 +#define HDA_BE_ID 4 +#define DMIC01_BE_ID 6 +#define DMIC16K_BE_ID 7 +#define BT_OFFLOAD_BE_ID 8 + +#define HDA_LINK_ORDER SOF_LINK_ORDER(SOF_LINK_IDISP_HDMI, \ + SOF_LINK_HDA, \ + SOF_LINK_DMIC01, \ + SOF_LINK_DMIC16K, \ + SOF_LINK_BT_OFFLOAD, \ + SOF_LINK_NONE, \ + SOF_LINK_NONE) + +#define HDA_LINK_IDS SOF_LINK_ORDER(IDISP_HDMI_BE_ID, \ + HDA_BE_ID, \ + DMIC01_BE_ID, \ + DMIC16K_BE_ID, \ + BT_OFFLOAD_BE_ID, \ + 0, \ + 0) + +static unsigned long +skl_hda_get_board_quirk(struct snd_soc_acpi_mach_params *mach_params) +{ + unsigned long board_quirk = 0; + int ssp_bt; + + if (hweight_long(mach_params->bt_link_mask) == 1) { + ssp_bt = fls(mach_params->bt_link_mask) - 1; + board_quirk |= SOF_SSP_PORT_BT_OFFLOAD(ssp_bt) | + SOF_BT_OFFLOAD_PRESENT; + } + + return board_quirk; +} + static int skl_hda_audio_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = pdev->dev.platform_data; - struct skl_hda_private *ctx; + struct sof_card_private *ctx; struct snd_soc_card *card; + unsigned long board_quirk = skl_hda_get_board_quirk(&mach->mach_params); int ret; - dev_dbg(&pdev->dev, "entry\n"); - - ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); - if (!ctx) + card = devm_kzalloc(&pdev->dev, sizeof(struct snd_soc_card), GFP_KERNEL); + if (!card) return -ENOMEM; - card = &ctx->card; card->name = "hda-dsp"; card->owner = THIS_MODULE; - card->dai_link = skl_hda_be_dai_links; - card->dapm_widgets = skl_hda_widgets; - card->dapm_routes = skl_hda_map; - card->add_dai_link = skl_hda_add_dai_link; card->fully_routed = true; card->late_probe = skl_hda_card_late_probe; - snd_soc_card_set_drvdata(card, ctx); + dev_dbg(&pdev->dev, "board_quirk = %lx\n", board_quirk); + + /* initialize ctx with board quirk */ + ctx = sof_intel_board_get_ctx(&pdev->dev, board_quirk); + if (!ctx) + return -ENOMEM; + + if (HDA_EXT_CODEC(mach->mach_params.codec_mask)) + ctx->hda_codec_present = true; if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->hdmi.idisp_codec = true; - if (hweight_long(mach->mach_params.bt_link_mask) == 1) { - ctx->bt_offload_present = true; - ctx->ssp_bt = fls(mach->mach_params.bt_link_mask) - 1; - } + ctx->link_order_overwrite = HDA_LINK_ORDER; + ctx->link_id_overwrite = HDA_LINK_IDS; - ret = skl_hda_fill_card_info(&pdev->dev, card, &mach->mach_params); - if (ret < 0) { - dev_err(&pdev->dev, "Unsupported HDAudio/iDisp configuration found\n"); + /* update dai_link */ + ret = sof_intel_board_set_dai_link(&pdev->dev, card, ctx); + if (ret) return ret; - } - - ctx->pcm_count = card->num_links; - ctx->dai_index = 1; /* hdmi codec dai name starts from index 1 */ - ctx->platform_name = mach->mach_params.platform; card->dev = &pdev->dev; if (!snd_soc_acpi_sof_parent(&pdev->dev)) @@ -258,6 +135,13 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; } + ret = snd_soc_fixup_dai_links_platform_name(card, + mach->mach_params.platform); + if (ret) + return ret; + + snd_soc_card_set_drvdata(card, ctx); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (!ret) skl_set_hda_codec_autosuspend_delay(card); @@ -280,4 +164,4 @@ MODULE_DESCRIPTION("SKL/KBL/BXT/APL HDA Generic Machine driver"); MODULE_AUTHOR("Rakesh Ughreja "); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:skl_hda_dsp_generic"); -MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); +MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_BOARD_HELPERS); From 2c80bcc27557b5db4aca8a0c621fc7d5cd10cf7e Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 12 Sep 2024 20:03:04 +0800 Subject: [PATCH 392/410] ASoC: Intel: ehl_rt5660: do not check common_hdmi_codec_drv MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The variable common_hdmi_codec_drv is always true on SOF platform so we could remove the reference in machine driver. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-4-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/ehl_rt5660.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/ehl_rt5660.c b/sound/soc/intel/boards/ehl_rt5660.c index 26289e8fdd87..90d93e667bd9 100644 --- a/sound/soc/intel/boards/ehl_rt5660.c +++ b/sound/soc/intel/boards/ehl_rt5660.c @@ -256,8 +256,7 @@ static void hdmi_link_init(struct snd_soc_card *card, { int i; - if (mach->mach_params.common_hdmi_codec_drv && - (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) { + if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) { ctx->idisp_codec = true; return; } From f22a351fe2193dac803fc919096b734ff2947958 Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 12 Sep 2024 20:03:05 +0800 Subject: [PATCH 393/410] ASoC: Intel: sof_pcm512x: do not check common_hdmi_codec_drv MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The variable common_hdmi_codec_drv is always true on SOF platform so we could remove the reference in machine driver. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-5-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_pcm512x.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 5de883b9563e..8d237f67bd06 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -371,8 +371,7 @@ static int sof_audio_probe(struct platform_device *pdev) sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(2); } else { dmic_be_num = 2; - if (mach->mach_params.common_hdmi_codec_drv && - (mach->mach_params.codec_mask & IDISP_CODEC_MASK)) + if (mach->mach_params.codec_mask & IDISP_CODEC_MASK) ctx->idisp_codec = true; /* links are always present in topology */ From dfa1a7f456f10018229d0d5b3c36dd36a9b5344f Mon Sep 17 00:00:00 2001 From: Brent Lu Date: Thu, 12 Sep 2024 20:03:06 +0800 Subject: [PATCH 394/410] ASoC: SOF: Intel: hda: remove common_hdmi_codec_drv MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Do not set common_hdmi_codec_drv in SOF platform driver since no machine driver needs it. Remove member variable common_hdmi_codec_drv from snd_soc_acpi_mach_params structure. Signed-off-by: Brent Lu Reviewed-by: Péter Ujfalusi Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-6-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- include/sound/soc-acpi.h | 2 -- sound/soc/sof/intel/hda.c | 1 - 2 files changed, 3 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index c1552bc6e31c..60d3b86a4660 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -62,7 +62,6 @@ static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg) * @platform: string used for HDAudio codec support * @codec_mask: used for HDAudio support * @dmic_num: number of SoC- or chipset-attached PDM digital microphones - * @common_hdmi_codec_drv: use commom HDAudio HDMI codec driver * @link_mask: SoundWire links enabled on the board * @links: array of SoundWire link _ADR descriptors, null terminated * @i2s_link_mask: I2S/TDM links enabled on the board @@ -80,7 +79,6 @@ struct snd_soc_acpi_mach_params { const char *platform; u32 codec_mask; u32 dmic_num; - bool common_hdmi_codec_drv; u32 link_mask; const struct snd_soc_acpi_link_adr *links; u32 i2s_link_mask; diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index d0a41e8e6334..70fc08c8fc99 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1051,7 +1051,6 @@ static void hda_generic_machine_select(struct snd_sof_dev *sdev, if (*mach) { mach_params = &(*mach)->mach_params; mach_params->codec_mask = bus->codec_mask; - mach_params->common_hdmi_codec_drv = true; } } #else From 47d94c13d5f1f9f9c2bc29e26ebbd4efe912256c Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Thu, 12 Sep 2024 20:03:07 +0800 Subject: [PATCH 395/410] ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for ARL. Added match table entry on arl machines to support HDMI-In capture with rt5682 I2S audio codec. also added the respective quirk configuration in rt5682 machine driver. Signed-off-by: Balamurugan C Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-7-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 7 +++++++ sound/soc/intel/common/soc-acpi-intel-arl-match.c | 13 +++++++++++++ 2 files changed, 20 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 23a40b913290..bc581fea0e3a 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -870,6 +870,13 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_PORT_BT_OFFLOAD(2) | SOF_BT_OFFLOAD_PRESENT), }, + { + .name = "arl_rt5682_c1_h02", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_SSP_PORT_CODEC(1) | + /* SSP 0 and SSP 2 are used for HDMI IN */ + SOF_SSP_MASK_HDMI_CAPTURE(0x5)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index 9936a4b4eee7..c13afff84923 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -7,6 +7,7 @@ #include #include +#include static const struct snd_soc_acpi_endpoint single_endpoint = { .num = 0, @@ -289,6 +290,11 @@ static const struct snd_soc_acpi_codecs arl_essx_83x6 = { .codecs = { "ESSX8316", "ESSX8326", "ESSX8336"}, }; +static const struct snd_soc_acpi_codecs arl_rt5682_hp = { + .num_codecs = 2, + .codecs = {RT5682_ACPI_HID, RT5682S_ACPI_HID}, +}; + static const struct snd_soc_acpi_codecs arl_lt6911_hdmi = { .num_codecs = 1, .codecs = {"INTC10B0"} @@ -310,6 +316,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .comp_ids = &arl_rt5682_hp, + .drv_name = "arl_rt5682_c1_h02", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &arl_lt6911_hdmi, + .sof_tplg_filename = "sof-arl-rt5682-ssp1-hdmi-ssp02.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_machines); From 322706e16988f6156ddd8fdcc6d06f87efc058f6 Mon Sep 17 00:00:00 2001 From: Balamurugan C Date: Thu, 12 Sep 2024 20:03:08 +0800 Subject: [PATCH 396/410] ASoC: Intel: ARL: Add entry for HDMI-In capture support to non-I2S codec boards. Adding HDMI-In capture support for the ARL products which doesn't have onboard I2S codec. But need to support HDMI-In capture via I2S and audio playback through HDMI/DP monitor. Signed-off-by: Balamurugan C Reviewed-by: Pierre-Louis Bossart Signed-off-by: Bard Liao Link: https://patch.msgid.link/20240912120308.134762-8-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_ssp_amp.c | 6 ++++++ sound/soc/intel/common/soc-acpi-intel-arl-match.c | 6 ++++++ 2 files changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index f51f1008e016..6ff8895a294a 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -210,6 +210,12 @@ static const struct platform_device_id board_ids[] = { /* SSP 0 and SSP 2 are used for HDMI IN */ SOF_HDMI_PLAYBACK_PRESENT), }, + { + .name = "arl_lt6911_hdmi_ssp", + .driver_data = (kernel_ulong_t)(SOF_SSP_MASK_HDMI_CAPTURE(0x5) | + /* SSP 0 and SSP 2 are used for HDMI IN */ + SOF_HDMI_PLAYBACK_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-arl-match.c b/sound/soc/intel/common/soc-acpi-intel-arl-match.c index c13afff84923..c97c961187dd 100644 --- a/sound/soc/intel/common/soc-acpi-intel-arl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-arl-match.c @@ -323,6 +323,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_arl_machines[] = { .quirk_data = &arl_lt6911_hdmi, .sof_tplg_filename = "sof-arl-rt5682-ssp1-hdmi-ssp02.tplg", }, + /* place amp-only boards in the end of table */ + { + .id = "INTC10B0", + .drv_name = "arl_lt6911_hdmi_ssp", + .sof_tplg_filename = "sof-arl-hdmi-ssp02.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_arl_machines); From d69f11e8c57e9459c9e60bffc0f2c6c3aa02f4b1 Mon Sep 17 00:00:00 2001 From: Muhammad Usama Anjum Date: Wed, 11 Sep 2024 17:36:22 +0500 Subject: [PATCH 397/410] ASoc: mediatek: mt8365: Remove unneeded assignment The ret is being assigned, but not being used. Remove the assignment. One of the reviewer mentioned that dev_warn should be replaced with dev_info. Make this change as well. Fixes: 1bf6dbd75f76 ("ASoc: mediatek: mt8365: Add a specific soundcard for EVK") Signed-off-by: Muhammad Usama Anjum Reviewed-by: AngeloGioacchino Del Regno Link: https://patch.msgid.link/20240911123629.125686-1-usama.anjum@collabora.com Reviewed-by: Alexandre Mergnat Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/mediatek/mt8365/mt8365-mt6357.c b/sound/soc/mediatek/mt8365/mt8365-mt6357.c index 1b8d1656101b..42cbdfdfadb5 100644 --- a/sound/soc/mediatek/mt8365/mt8365-mt6357.c +++ b/sound/soc/mediatek/mt8365/mt8365-mt6357.c @@ -257,8 +257,7 @@ static int mt8365_mt6357_gpio_probe(struct snd_soc_card *card) priv->pin_states[i] = pinctrl_lookup_state(priv->pinctrl, mt8365_mt6357_pin_str[i]); if (IS_ERR(priv->pin_states[i])) { - ret = PTR_ERR(priv->pin_states[i]); - dev_warn(card->dev, "No pin state for %s\n", + dev_info(card->dev, "No pin state for %s\n", mt8365_mt6357_pin_str[i]); } else { ret = pinctrl_select_state(priv->pinctrl, From a51c925c11d7b855167e64b63eb4378e5adfc11d Mon Sep 17 00:00:00 2001 From: Joshua Pius Date: Thu, 12 Sep 2024 15:26:28 +0000 Subject: [PATCH 398/410] ALSA: usb-audio: Add logitech Audio profile quirk Specify shortnames for the following Logitech Devices: Rally bar, Rally bar mini, Tap, MeetUp and Huddle. Signed-off-by: Joshua Pius Link: https://patch.msgid.link/20240912152635.1859737-1-joshuapius@google.com Signed-off-by: Takashi Iwai --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 778de9244f1e..9c411b82a218 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -384,6 +384,12 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* Creative/Toshiba Multimedia Center SB-0500 */ DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + /* Logitech Audio Devices */ + DEVICE_NAME(0x046d, 0x0867, "Logitech, Inc.", "Logi-MeetUp"), + DEVICE_NAME(0x046d, 0x0874, "Logitech, Inc.", "Logi-Tap-Audio"), + DEVICE_NAME(0x046d, 0x087c, "Logitech, Inc.", "Logi-Huddle"), + DEVICE_NAME(0x046d, 0x0898, "Logitech, Inc.", "Logi-RB-Audio"), + DEVICE_NAME(0x046d, 0x08d2, "Logitech, Inc.", "Logi-RBM-Audio"), DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), From c880a5146642e9d35f88aaa353ae98ffd4fc3f99 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Sep 2024 17:52:24 +0200 Subject: [PATCH 399/410] ALSA: memalloc: Use proper DMA mapping API for x86 WC buffer allocations The x86 WC page allocation assumes incorrectly the DMA address directly taken from the page. Also it checks the DMA ops inappropriately for switching to the own method. This patch rewrites the stuff to use the proper DMA mapping API instead. Link: https://patch.msgid.link/20240912155227.4078-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 51 +++++++++++++++++++++++++++---------------- 1 file changed, 32 insertions(+), 19 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index b21dba4b374a..39e97f6fe8ac 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -496,41 +496,54 @@ static const struct snd_malloc_ops snd_dma_dev_ops = { /* * Write-combined pages */ -/* x86-specific allocations */ #ifdef CONFIG_SND_DMA_SGBUF -#define x86_fallback(dmab) (!get_dma_ops(dmab->dev.dev)) -#else -#define x86_fallback(dmab) false -#endif - +/* x86-specific allocations */ +static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) +{ + void *p = do_alloc_pages(dmab->dev.dev, size, &dmab->addr, true); + + if (!p) + return NULL; + dmab->addr = dma_map_single(dmab->dev.dev, p, size, DMA_BIDIRECTIONAL); + if (dmab->addr == DMA_MAPPING_ERROR) { + do_free_pages(dmab->area, size, true); + return NULL; + } + return p; +} + +static void snd_dma_wc_free(struct snd_dma_buffer *dmab) +{ + dma_unmap_single(dmab->dev.dev, dmab->addr, dmab->bytes, + DMA_BIDIRECTIONAL); + do_free_pages(dmab->area, dmab->bytes, true); +} + +static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return dma_mmap_coherent(dmab->dev.dev, area, + dmab->area, dmab->addr, dmab->bytes); +} +#else static void *snd_dma_wc_alloc(struct snd_dma_buffer *dmab, size_t size) { - if (x86_fallback(dmab)) - return do_alloc_pages(dmab->dev.dev, size, &dmab->addr, true); return dma_alloc_wc(dmab->dev.dev, size, &dmab->addr, DEFAULT_GFP); } static void snd_dma_wc_free(struct snd_dma_buffer *dmab) { - if (x86_fallback(dmab)) { - do_free_pages(dmab->area, dmab->bytes, true); - return; - } dma_free_wc(dmab->dev.dev, dmab->bytes, dmab->area, dmab->addr); } static int snd_dma_wc_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { -#ifdef CONFIG_SND_DMA_SGBUF - if (x86_fallback(dmab)) { - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); - return snd_dma_continuous_mmap(dmab, area); - } -#endif return dma_mmap_wc(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } +#endif static const struct snd_malloc_ops snd_dma_wc_ops = { .alloc = snd_dma_wc_alloc, @@ -804,7 +817,7 @@ static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) dmab->dev.type = type; /* restore the type */ /* if IOMMU is present but failed, give up */ - if (!x86_fallback(dmab)) + if (get_dma_ops(dmab->dev.dev)) return NULL; /* try fallback */ return snd_dma_sg_fallback_alloc(dmab, size); From 0b9f2bd00fc3677e38ae5e46ff79b8d48d9cb02e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 12 Sep 2024 17:52:25 +0200 Subject: [PATCH 400/410] ALSA: memalloc: Use proper DMA mapping API for x86 S/G buffer allocations The fallback S/G buffer allocation for x86 used the addresses deduced from the page allocations blindly. It broke the allocations on IOMMU and made us to work around with a hackish DMA ops check. For cleaning up those messes, this patch switches to the proper DMA mapping API usages with the standard sg-table instead. By introducing the sg-table, the address table isn't needed, but for keeping the original allocation sizes for freeing, replace it with the array keeping the number of pages. The get_addr callback is changed to use the existing one for non-contiguous buffers. (Also it's the reason sg_table is put at the beginning of struct snd_dma_sg_fallback.) And finally, the hackish workaround that checks the DMA ops is dropped now. Link: https://patch.msgid.link/20240912155227.4078-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/memalloc.c | 78 ++++++++++++++++++++----------------------- 1 file changed, 36 insertions(+), 42 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 39e97f6fe8ac..13b71069ae18 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -680,43 +680,43 @@ static const struct snd_malloc_ops snd_dma_noncontig_ops = { #ifdef CONFIG_SND_DMA_SGBUF /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { + struct sg_table sgt; /* used by get_addr - must be the first item */ size_t count; struct page **pages; - /* DMA address array; the first page contains #pages in ~PAGE_MASK */ - dma_addr_t *addrs; + unsigned int *npages; }; static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG; size_t i, size; - if (sgbuf->pages && sgbuf->addrs) { + if (sgbuf->pages && sgbuf->npages) { i = 0; while (i < sgbuf->count) { - if (!sgbuf->pages[i] || !sgbuf->addrs[i]) - break; - size = sgbuf->addrs[i] & ~PAGE_MASK; - if (WARN_ON(!size)) + size = sgbuf->npages[i]; + if (!size) break; do_free_pages(page_address(sgbuf->pages[i]), - size << PAGE_SHIFT, false); + size << PAGE_SHIFT, wc); i += size; } } kvfree(sgbuf->pages); - kvfree(sgbuf->addrs); + kvfree(sgbuf->npages); kfree(sgbuf); } /* fallback manual S/G buffer allocations */ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) { + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG; struct snd_dma_sg_fallback *sgbuf; struct page **pagep, *curp; - size_t chunk, npages; - dma_addr_t *addrp; + size_t chunk; dma_addr_t addr; + unsigned int idx, npages; void *p; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); @@ -725,16 +725,16 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) size = PAGE_ALIGN(size); sgbuf->count = size >> PAGE_SHIFT; sgbuf->pages = kvcalloc(sgbuf->count, sizeof(*sgbuf->pages), GFP_KERNEL); - sgbuf->addrs = kvcalloc(sgbuf->count, sizeof(*sgbuf->addrs), GFP_KERNEL); - if (!sgbuf->pages || !sgbuf->addrs) + sgbuf->npages = kvcalloc(sgbuf->count, sizeof(*sgbuf->npages), GFP_KERNEL); + if (!sgbuf->pages || !sgbuf->npages) goto error; pagep = sgbuf->pages; - addrp = sgbuf->addrs; - chunk = (PAGE_SIZE - 1) << PAGE_SHIFT; /* to fit in low bits in addrs */ + chunk = size; + idx = 0; while (size > 0) { chunk = min(size, chunk); - p = do_alloc_pages(dmab->dev.dev, chunk, &addr, false); + p = do_alloc_pages(dmab->dev.dev, chunk, &addr, wc); if (!p) { if (chunk <= PAGE_SIZE) goto error; @@ -746,27 +746,33 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) size -= chunk; /* fill pages */ npages = chunk >> PAGE_SHIFT; - *addrp = npages; /* store in lower bits */ + sgbuf->npages[idx] = npages; + idx += npages; curp = virt_to_page(p); - while (npages--) { + while (npages--) *pagep++ = curp++; - *addrp++ |= addr; - addr += PAGE_SIZE; - } } + if (sg_alloc_table_from_pages(&sgbuf->sgt, sgbuf->pages, sgbuf->count, + 0, sgbuf->count << PAGE_SHIFT, GFP_KERNEL)) + goto error; + + if (dma_map_sgtable(dmab->dev.dev, &sgbuf->sgt, DMA_BIDIRECTIONAL, 0)) + goto error_dma_map; + p = vmap(sgbuf->pages, sgbuf->count, VM_MAP, PAGE_KERNEL); if (!p) - goto error; - - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - set_pages_array_wc(sgbuf->pages, sgbuf->count); + goto error_vmap; dmab->private_data = sgbuf; /* store the first page address for convenience */ - dmab->addr = sgbuf->addrs[0] & PAGE_MASK; + dmab->addr = snd_sgbuf_get_addr(dmab, 0); return p; + error_vmap: + dma_unmap_sgtable(dmab->dev.dev, &sgbuf->sgt, DMA_BIDIRECTIONAL, 0); + error_dma_map: + sg_free_table(&sgbuf->sgt); error: __snd_dma_sg_fallback_free(dmab, sgbuf); return NULL; @@ -776,21 +782,12 @@ static void snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab) { struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - set_pages_array_wb(sgbuf->pages, sgbuf->count); vunmap(dmab->area); + dma_unmap_sgtable(dmab->dev.dev, &sgbuf->sgt, DMA_BIDIRECTIONAL, 0); + sg_free_table(&sgbuf->sgt); __snd_dma_sg_fallback_free(dmab, dmab->private_data); } -static dma_addr_t snd_dma_sg_fallback_get_addr(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - size_t index = offset >> PAGE_SHIFT; - - return (sgbuf->addrs[index] & PAGE_MASK) | (offset & ~PAGE_MASK); -} - static int snd_dma_sg_fallback_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { @@ -816,10 +813,6 @@ static void *snd_dma_sg_alloc(struct snd_dma_buffer *dmab, size_t size) return p; dmab->dev.type = type; /* restore the type */ - /* if IOMMU is present but failed, give up */ - if (get_dma_ops(dmab->dev.dev)) - return NULL; - /* try fallback */ return snd_dma_sg_fallback_alloc(dmab, size); } @@ -827,7 +820,8 @@ static const struct snd_malloc_ops snd_dma_sg_ops = { .alloc = snd_dma_sg_alloc, .free = snd_dma_sg_fallback_free, .mmap = snd_dma_sg_fallback_mmap, - .get_addr = snd_dma_sg_fallback_get_addr, + /* reuse noncontig helper */ + .get_addr = snd_dma_noncontig_get_addr, /* reuse vmalloc helpers */ .get_page = snd_dma_vmalloc_get_page, .get_chunk_size = snd_dma_vmalloc_get_chunk_size, From f6e2e7397d00192bda11166d5fb3e2e67a8cf92e Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Thu, 12 Sep 2024 16:41:10 +0800 Subject: [PATCH 401/410] ASoC: mediatek: mt7986-afe-pcm: Remove redundant error message In the function mt7986_afe_pcm_dev_probe, when get irq failed, the function platform_get_irq() logs an error message, so remove redundant one here. Reviewed-by: AngeloGioacchino Del Regno Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240912084110.1854-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index d400bea0ef9c..7db090414d59 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -529,10 +529,9 @@ static int mt7986_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id < 0) { - ret = irq_id; - return dev_err_probe(dev, ret, "No irq found\n"); - } + if (irq_id < 0) + return irq_id; + ret = devm_request_irq(dev, irq_id, mt7986_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); if (ret) From 5740434e1e0f51db1282436a7783658e6c139fd1 Mon Sep 17 00:00:00 2001 From: Joshua Grisham Date: Fri, 13 Sep 2024 10:00:55 +0200 Subject: [PATCH 402/410] ALSA: hda/realtek: Add support for Galaxy Book2 Pro (NP950XEE) Adds support for GB2Pro Arc variant (NP950XEE) based on successful test and information provided by Github user drewdrew0 [1]. [1]: https://github.com/thesofproject/linux/issues/4055#issuecomment-2346890020 Signed-off-by: Joshua Grisham Link: https://patch.msgid.link/20240913080055.10807-1-josh@joshuagrisham.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ef6d9ac28a61..4ca66234e561 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10639,6 +10639,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc870, "Samsung Galaxy Book2 Pro (NP950XED)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), + SND_PCI_QUIRK(0x144d, 0xc872, "Samsung Galaxy Book2 Pro (NP950XEE)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), SND_PCI_QUIRK(0x144d, 0xc886, "Samsung Galaxy Book3 Pro (NP964XFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), From 2ed1a4a5c0058dfd78f5037576d668a37d0ec609 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Fri, 13 Sep 2024 15:06:22 +0300 Subject: [PATCH 403/410] ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls Avoid removing these controls, as doing so can cause issues if the stream is initiated from another control. Ensure these controls remain intact when the stream is started or finished. Instead of removing them, return an -EBUSY error code to indicate that the controller is busy, especially when the audio filter and the SINC filter are in use. [andrei.simion@microchip.com: Reword the commit title and the commit message. Replace spinlock and busy variable with atomic_t busy_stream.] Signed-off-by: Codrin Ciubotariu Signed-off-by: Andrei Simion Link: https://patch.msgid.link/20240913120621.79088-1-andrei.simion@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-pdmc.c | 57 ++++++++++--------------------------- 1 file changed, 15 insertions(+), 42 deletions(-) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index d97d153ee375..939cd44ebc8a 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -124,6 +124,7 @@ struct mchp_pdmc { int mic_no; int sinc_order; bool audio_filter_en; + atomic_t busy_stream; }; static const char *const mchp_pdmc_sinc_filter_order_text[] = { @@ -167,6 +168,10 @@ static int mchp_pdmc_sinc_order_put(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (val == dd->sinc_order) return 0; @@ -193,6 +198,9 @@ static int mchp_pdmc_af_put(struct snd_kcontrol *kcontrol, struct mchp_pdmc *dd = snd_soc_component_get_drvdata(component); bool af = uvalue->value.integer.value[0] ? true : false; + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (dd->audio_filter_en == af) return 0; @@ -379,52 +387,10 @@ static const struct snd_kcontrol_new mchp_pdmc_snd_controls[] = { }, }; -static int mchp_pdmc_close(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - return snd_soc_add_component_controls(component, mchp_pdmc_snd_controls, - ARRAY_SIZE(mchp_pdmc_snd_controls)); -} - -static int mchp_pdmc_open(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - int i; - - /* remove controls that can't be changed at runtime */ - for (i = 0; i < ARRAY_SIZE(mchp_pdmc_snd_controls); i++) { - const struct snd_kcontrol_new *control = &mchp_pdmc_snd_controls[i]; - struct snd_ctl_elem_id id; - int err; - - if (component->name_prefix) - snprintf(id.name, sizeof(id.name), "%s %s", component->name_prefix, - control->name); - else - strscpy(id.name, control->name, sizeof(id.name)); - - id.numid = 0; - id.iface = control->iface; - id.device = control->device; - id.subdevice = control->subdevice; - id.index = control->index; - err = snd_ctl_remove_id(component->card->snd_card, &id); - if (err < 0) - dev_err(component->dev, "%d: Failed to remove %s\n", err, - control->name); - } - - return 0; -} - static const struct snd_soc_component_driver mchp_pdmc_dai_component = { .name = "mchp-pdmc", .controls = mchp_pdmc_snd_controls, .num_controls = ARRAY_SIZE(mchp_pdmc_snd_controls), - .open = &mchp_pdmc_open, - .close = &mchp_pdmc_close, - .legacy_dai_naming = 1, - .trigger_start = SND_SOC_TRIGGER_ORDER_LDC, }; static const unsigned int mchp_pdmc_1mic[] = {1}; @@ -587,6 +553,11 @@ static int mchp_pdmc_hw_params(struct snd_pcm_substream *substream, cfgr_val |= MCHP_PDMC_CFGR_BSSEL(i); } + /* + * from these point forward, we consider the controller busy, so the + * audio filter and SINC order can't be changed + */ + atomic_set(&dd->busy_stream, 1); for (osr_start = dd->audio_filter_en ? 64 : 8; osr_start <= 256 && best_diff_rate; osr_start *= 2) { long round_rate; @@ -1143,6 +1114,8 @@ static void mchp_pdmc_remove(struct platform_device *pdev) { struct mchp_pdmc *dd = platform_get_drvdata(pdev); + atomic_set(&dd->busy_stream, 0); + if (!pm_runtime_status_suspended(dd->dev)) mchp_pdmc_runtime_suspend(dd->dev); From f5c05fd7e9d20a3a8f3401b467fec2d24f49ea5a Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 13 Sep 2024 14:36:27 +0530 Subject: [PATCH 404/410] ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures Rename SoundWire endpoint and dai link structures with asoc tag to make it generic. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Link: https://patch.msgid.link/20240913090631.1834543-2-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f10ed95770ea..6eeddced1f6f 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -617,7 +617,7 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -struct sof_sdw_endpoint { +struct asoc_sdw_endpoint { struct list_head list; u32 link_mask; @@ -629,7 +629,7 @@ struct sof_sdw_endpoint { const struct asoc_sdw_dai_info *dai_info; }; -struct sof_sdw_dailink { +struct asoc_sdw_dailink { bool initialised; u8 group_id; @@ -660,8 +660,8 @@ static int count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *nu return 0; } -static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks, - const struct snd_soc_acpi_endpoint *new) +static struct asoc_sdw_dailink *find_dailink(struct asoc_sdw_dailink *dailinks, + const struct snd_soc_acpi_endpoint *new) { while (dailinks->initialised) { if (new->aggregated && dailinks->group_id == new->group_id) @@ -678,8 +678,8 @@ static struct sof_sdw_dailink *find_dailink(struct sof_sdw_dailink *dailinks, } static int parse_sdw_endpoints(struct snd_soc_card *card, - struct sof_sdw_dailink *sof_dais, - struct sof_sdw_endpoint *sof_ends, + struct asoc_sdw_dailink *sof_dais, + struct asoc_sdw_endpoint *sof_ends, int *num_devs) { struct device *dev = card->dev; @@ -687,7 +687,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; const struct snd_soc_acpi_link_adr *adr_link; - struct sof_sdw_endpoint *sof_end = sof_ends; + struct asoc_sdw_endpoint *sof_end = sof_ends; int num_dais = 0; int i, j; int ret; @@ -738,7 +738,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, for (j = 0; j < adr_dev->num_endpoints; j++) { const struct snd_soc_acpi_endpoint *adr_end; const struct asoc_sdw_dai_info *dai_info; - struct sof_sdw_dailink *sof_dai; + struct asoc_sdw_dailink *sof_dai; int stream; adr_end = &adr_dev->endpoints[j]; @@ -799,14 +799,14 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, } static int create_sdw_dailink(struct snd_soc_card *card, - struct sof_sdw_dailink *sof_dai, + struct asoc_sdw_dailink *sof_dai, struct snd_soc_dai_link **dai_links, int *be_id, struct snd_soc_codec_conf **codec_conf) { struct device *dev = card->dev; struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct intel_mc_ctx *intel_ctx = (struct intel_mc_ctx *)ctx->private; - struct sof_sdw_endpoint *sof_end; + struct asoc_sdw_endpoint *sof_end; int stream; int ret; @@ -845,7 +845,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, continue; sof_end = list_first_entry(&sof_dai->endpoints, - struct sof_sdw_endpoint, list); + struct asoc_sdw_endpoint, list); *be_id = sof_end->dai_info->dailink[stream]; if (*be_id < 0) { @@ -936,7 +936,7 @@ static int create_sdw_dailink(struct snd_soc_card *card, static int create_sdw_dailinks(struct snd_soc_card *card, struct snd_soc_dai_link **dai_links, int *be_id, - struct sof_sdw_dailink *sof_dais, + struct asoc_sdw_dailink *sof_dais, struct snd_soc_codec_conf **codec_conf) { struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -1104,8 +1104,8 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; struct snd_soc_codec_conf *codec_conf; struct asoc_sdw_codec_info *ssp_info; - struct sof_sdw_endpoint *sof_ends; - struct sof_sdw_dailink *sof_dais; + struct asoc_sdw_endpoint *sof_ends; + struct asoc_sdw_dailink *sof_dais; int num_devs = 0; int num_ends = 0; struct snd_soc_dai_link *dai_links; From 23f020bd607b7aec5f301699227ed196430fbc40 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 13 Sep 2024 14:36:28 +0530 Subject: [PATCH 405/410] ASoC: intel: sof_sdw: rename soundwire parsing helper functions Rename SoundWire parsing helper functions with 'asoc_sdw' tag to make it generic. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Link: https://patch.msgid.link/20240913090631.1834543-3-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_sdw.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6eeddced1f6f..222cf4a37707 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -640,7 +640,7 @@ struct asoc_sdw_dailink { static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; -static int count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) +static int asoc_sdw_count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) { struct device *dev = card->dev; struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); @@ -660,8 +660,8 @@ static int count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *nu return 0; } -static struct asoc_sdw_dailink *find_dailink(struct asoc_sdw_dailink *dailinks, - const struct snd_soc_acpi_endpoint *new) +static struct asoc_sdw_dailink *asoc_sdw_find_dailink(struct asoc_sdw_dailink *dailinks, + const struct snd_soc_acpi_endpoint *new) { while (dailinks->initialised) { if (new->aggregated && dailinks->group_id == new->group_id) @@ -677,10 +677,10 @@ static struct asoc_sdw_dailink *find_dailink(struct asoc_sdw_dailink *dailinks, return dailinks; } -static int parse_sdw_endpoints(struct snd_soc_card *card, - struct asoc_sdw_dailink *sof_dais, - struct asoc_sdw_endpoint *sof_ends, - int *num_devs) +static int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, + struct asoc_sdw_dailink *sof_dais, + struct asoc_sdw_endpoint *sof_ends, + int *num_devs) { struct device *dev = card->dev; struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); @@ -743,7 +743,7 @@ static int parse_sdw_endpoints(struct snd_soc_card *card, adr_end = &adr_dev->endpoints[j]; dai_info = &codec_info->dais[adr_end->num]; - sof_dai = find_dailink(sof_dais, adr_end); + sof_dai = asoc_sdw_find_dailink(sof_dais, adr_end); if (dai_info->quirk && !(dai_info->quirk & sof_sdw_quirk)) continue; @@ -1115,7 +1115,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) unsigned long ssp_mask; int ret; - ret = count_sdw_endpoints(card, &num_devs, &num_ends); + ret = asoc_sdw_count_sdw_endpoints(card, &num_devs, &num_ends); if (ret < 0) { dev_err(dev, "failed to count devices/endpoints: %d\n", ret); return ret; @@ -1133,7 +1133,7 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) goto err_dai; } - ret = parse_sdw_endpoints(card, sof_dais, sof_ends, &num_devs); + ret = asoc_sdw_parse_sdw_endpoints(card, sof_dais, sof_ends, &num_devs); if (ret < 0) goto err_end; From 7860df5b29945cfab40dd667f576af31401d7c43 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 13 Sep 2024 14:36:29 +0530 Subject: [PATCH 406/410] ASoC: sdw_util/intel: move soundwire endpoint and dai link structures Move Soundwire endpoint and dai link structures from Intel generic machine driver code to common place holder(soc_sdw_utils.h). These structures will be used in other platform SoundWire machine driver code. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Link: https://patch.msgid.link/20240913090631.1834543-4-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 21 +++++++++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 21 --------------------- 2 files changed, 21 insertions(+), 21 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index e366b7968c2d..e3482720a3eb 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -93,6 +93,27 @@ struct asoc_sdw_mc_private { int codec_info_list_count; }; +struct asoc_sdw_endpoint { + struct list_head list; + + u32 link_mask; + const char *codec_name; + const char *name_prefix; + bool include_sidecar; + + struct asoc_sdw_codec_info *codec_info; + const struct asoc_sdw_dai_info *dai_info; +}; + +struct asoc_sdw_dailink { + bool initialised; + + u8 group_id; + u32 link_mask[SNDRV_PCM_STREAM_LAST + 1]; + int num_devs[SNDRV_PCM_STREAM_LAST + 1]; + struct list_head endpoints; +}; + extern struct asoc_sdw_codec_info codec_info_list[]; int asoc_sdw_get_codec_info_list_count(void); diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 222cf4a37707..6b30659f0e25 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -617,27 +617,6 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -struct asoc_sdw_endpoint { - struct list_head list; - - u32 link_mask; - const char *codec_name; - const char *name_prefix; - bool include_sidecar; - - struct asoc_sdw_codec_info *codec_info; - const struct asoc_sdw_dai_info *dai_info; -}; - -struct asoc_sdw_dailink { - bool initialised; - - u8 group_id; - u32 link_mask[SNDRV_PCM_STREAM_LAST + 1]; - int num_devs[SNDRV_PCM_STREAM_LAST + 1]; - struct list_head endpoints; -}; - static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; static int asoc_sdw_count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) From 13b24f84782d6c0373f62eb645353883d94d1dcd Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 13 Sep 2024 14:36:30 +0530 Subject: [PATCH 407/410] ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions Move SoundWire endpoint parsing helper functions to common place holder. These functions will be used by other platform machine driver code. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Link: https://patch.msgid.link/20240913090631.1834543-5-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- include/sound/soc_sdw_utils.h | 10 ++ sound/soc/intel/boards/sof_sdw.c | 158 --------------------------- sound/soc/sdw_utils/soc_sdw_utils.c | 161 ++++++++++++++++++++++++++++ 3 files changed, 171 insertions(+), 158 deletions(-) diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index e3482720a3eb..f68c1f193b3b 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -161,6 +161,16 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d int (*init)(struct snd_soc_pcm_runtime *rtd), const struct snd_soc_ops *ops); +int asoc_sdw_count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends); + +struct asoc_sdw_dailink *asoc_sdw_find_dailink(struct asoc_sdw_dailink *dailinks, + const struct snd_soc_acpi_endpoint *new); + +int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, + struct asoc_sdw_dailink *soc_dais, + struct asoc_sdw_endpoint *soc_ends, + int *num_devs); + int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd); /* DMIC support */ diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 6b30659f0e25..5196d96f5c0e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -619,164 +619,6 @@ static const struct snd_soc_ops sdw_ops = { static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; -static int asoc_sdw_count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) -{ - struct device *dev = card->dev; - struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); - struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; - const struct snd_soc_acpi_link_adr *adr_link; - int i; - - for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { - *num_devs += adr_link->num_adr; - - for (i = 0; i < adr_link->num_adr; i++) - *num_ends += adr_link->adr_d[i].num_endpoints; - } - - dev_dbg(dev, "Found %d devices with %d endpoints\n", *num_devs, *num_ends); - - return 0; -} - -static struct asoc_sdw_dailink *asoc_sdw_find_dailink(struct asoc_sdw_dailink *dailinks, - const struct snd_soc_acpi_endpoint *new) -{ - while (dailinks->initialised) { - if (new->aggregated && dailinks->group_id == new->group_id) - return dailinks; - - dailinks++; - } - - INIT_LIST_HEAD(&dailinks->endpoints); - dailinks->group_id = new->group_id; - dailinks->initialised = true; - - return dailinks; -} - -static int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, - struct asoc_sdw_dailink *sof_dais, - struct asoc_sdw_endpoint *sof_ends, - int *num_devs) -{ - struct device *dev = card->dev; - struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); - struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); - struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; - const struct snd_soc_acpi_link_adr *adr_link; - struct asoc_sdw_endpoint *sof_end = sof_ends; - int num_dais = 0; - int i, j; - int ret; - - for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { - int num_link_dailinks = 0; - - if (!is_power_of_2(adr_link->mask)) { - dev_err(dev, "link with multiple mask bits: 0x%x\n", - adr_link->mask); - return -EINVAL; - } - - for (i = 0; i < adr_link->num_adr; i++) { - const struct snd_soc_acpi_adr_device *adr_dev = &adr_link->adr_d[i]; - struct asoc_sdw_codec_info *codec_info; - const char *codec_name; - - if (!adr_dev->name_prefix) { - dev_err(dev, "codec 0x%llx does not have a name prefix\n", - adr_dev->adr); - return -EINVAL; - } - - codec_info = asoc_sdw_find_codec_info_part(adr_dev->adr); - if (!codec_info) - return -EINVAL; - - ctx->ignore_internal_dmic |= codec_info->ignore_internal_dmic; - - codec_name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, i); - if (!codec_name) - return -ENOMEM; - - dev_dbg(dev, "Adding prefix %s for %s\n", - adr_dev->name_prefix, codec_name); - - sof_end->name_prefix = adr_dev->name_prefix; - - if (codec_info->count_sidecar && codec_info->add_sidecar) { - ret = codec_info->count_sidecar(card, &num_dais, num_devs); - if (ret) - return ret; - - sof_end->include_sidecar = true; - } - - for (j = 0; j < adr_dev->num_endpoints; j++) { - const struct snd_soc_acpi_endpoint *adr_end; - const struct asoc_sdw_dai_info *dai_info; - struct asoc_sdw_dailink *sof_dai; - int stream; - - adr_end = &adr_dev->endpoints[j]; - dai_info = &codec_info->dais[adr_end->num]; - sof_dai = asoc_sdw_find_dailink(sof_dais, adr_end); - - if (dai_info->quirk && !(dai_info->quirk & sof_sdw_quirk)) - continue; - - dev_dbg(dev, - "Add dev: %d, 0x%llx end: %d, %s, %c/%c to %s: %d\n", - ffs(adr_link->mask) - 1, adr_dev->adr, - adr_end->num, type_strings[dai_info->dai_type], - dai_info->direction[SNDRV_PCM_STREAM_PLAYBACK] ? 'P' : '-', - dai_info->direction[SNDRV_PCM_STREAM_CAPTURE] ? 'C' : '-', - adr_end->aggregated ? "group" : "solo", - adr_end->group_id); - - if (adr_end->num >= codec_info->dai_num) { - dev_err(dev, - "%d is too many endpoints for codec: 0x%x\n", - adr_end->num, codec_info->part_id); - return -EINVAL; - } - - for_each_pcm_streams(stream) { - if (dai_info->direction[stream] && - dai_info->dailink[stream] < 0) { - dev_err(dev, - "Invalid dailink id %d for codec: 0x%x\n", - dai_info->dailink[stream], - codec_info->part_id); - return -EINVAL; - } - - if (dai_info->direction[stream]) { - num_dais += !sof_dai->num_devs[stream]; - sof_dai->num_devs[stream]++; - sof_dai->link_mask[stream] |= adr_link->mask; - } - } - - num_link_dailinks += !!list_empty(&sof_dai->endpoints); - list_add_tail(&sof_end->list, &sof_dai->endpoints); - - sof_end->link_mask = adr_link->mask; - sof_end->codec_name = codec_name; - sof_end->codec_info = codec_info; - sof_end->dai_info = dai_info; - sof_end++; - } - } - - ctx->append_dai_type |= (num_link_dailinks > 1); - } - - return num_dais; -} - static int create_sdw_dailink(struct snd_soc_card *card, struct asoc_sdw_dailink *sof_dai, struct snd_soc_dai_link **dai_links, diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index d59ccb56642c..a6070f822eb9 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -1005,5 +1005,166 @@ int asoc_sdw_init_simple_dai_link(struct device *dev, struct snd_soc_dai_link *d } EXPORT_SYMBOL_NS(asoc_sdw_init_simple_dai_link, SND_SOC_SDW_UTILS); +int asoc_sdw_count_sdw_endpoints(struct snd_soc_card *card, int *num_devs, int *num_ends) +{ + struct device *dev = card->dev; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + int i; + + for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { + *num_devs += adr_link->num_adr; + + for (i = 0; i < adr_link->num_adr; i++) + *num_ends += adr_link->adr_d[i].num_endpoints; + } + + dev_dbg(dev, "Found %d devices with %d endpoints\n", *num_devs, *num_ends); + + return 0; +} +EXPORT_SYMBOL_NS(asoc_sdw_count_sdw_endpoints, SND_SOC_SDW_UTILS); + +struct asoc_sdw_dailink *asoc_sdw_find_dailink(struct asoc_sdw_dailink *dailinks, + const struct snd_soc_acpi_endpoint *new) +{ + while (dailinks->initialised) { + if (new->aggregated && dailinks->group_id == new->group_id) + return dailinks; + + dailinks++; + } + + INIT_LIST_HEAD(&dailinks->endpoints); + dailinks->group_id = new->group_id; + dailinks->initialised = true; + + return dailinks; +} +EXPORT_SYMBOL_NS(asoc_sdw_find_dailink, SND_SOC_SDW_UTILS); + +int asoc_sdw_parse_sdw_endpoints(struct snd_soc_card *card, + struct asoc_sdw_dailink *soc_dais, + struct asoc_sdw_endpoint *soc_ends, + int *num_devs) +{ + struct device *dev = card->dev; + struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); + struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; + const struct snd_soc_acpi_link_adr *adr_link; + struct asoc_sdw_endpoint *soc_end = soc_ends; + int num_dais = 0; + int i, j; + int ret; + + for (adr_link = mach_params->links; adr_link->num_adr; adr_link++) { + int num_link_dailinks = 0; + + if (!is_power_of_2(adr_link->mask)) { + dev_err(dev, "link with multiple mask bits: 0x%x\n", + adr_link->mask); + return -EINVAL; + } + + for (i = 0; i < adr_link->num_adr; i++) { + const struct snd_soc_acpi_adr_device *adr_dev = &adr_link->adr_d[i]; + struct asoc_sdw_codec_info *codec_info; + const char *codec_name; + + if (!adr_dev->name_prefix) { + dev_err(dev, "codec 0x%llx does not have a name prefix\n", + adr_dev->adr); + return -EINVAL; + } + + codec_info = asoc_sdw_find_codec_info_part(adr_dev->adr); + if (!codec_info) + return -EINVAL; + + ctx->ignore_internal_dmic |= codec_info->ignore_internal_dmic; + + codec_name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, i); + if (!codec_name) + return -ENOMEM; + + dev_dbg(dev, "Adding prefix %s for %s\n", + adr_dev->name_prefix, codec_name); + + soc_end->name_prefix = adr_dev->name_prefix; + + if (codec_info->count_sidecar && codec_info->add_sidecar) { + ret = codec_info->count_sidecar(card, &num_dais, num_devs); + if (ret) + return ret; + + soc_end->include_sidecar = true; + } + + for (j = 0; j < adr_dev->num_endpoints; j++) { + const struct snd_soc_acpi_endpoint *adr_end; + const struct asoc_sdw_dai_info *dai_info; + struct asoc_sdw_dailink *soc_dai; + int stream; + + adr_end = &adr_dev->endpoints[j]; + dai_info = &codec_info->dais[adr_end->num]; + soc_dai = asoc_sdw_find_dailink(soc_dais, adr_end); + + if (dai_info->quirk && !(dai_info->quirk & ctx->mc_quirk)) + continue; + + dev_dbg(dev, + "Add dev: %d, 0x%llx end: %d, dai: %d, %c/%c to %s: %d\n", + ffs(adr_link->mask) - 1, adr_dev->adr, + adr_end->num, dai_info->dai_type, + dai_info->direction[SNDRV_PCM_STREAM_PLAYBACK] ? 'P' : '-', + dai_info->direction[SNDRV_PCM_STREAM_CAPTURE] ? 'C' : '-', + adr_end->aggregated ? "group" : "solo", + adr_end->group_id); + + if (adr_end->num >= codec_info->dai_num) { + dev_err(dev, + "%d is too many endpoints for codec: 0x%x\n", + adr_end->num, codec_info->part_id); + return -EINVAL; + } + + for_each_pcm_streams(stream) { + if (dai_info->direction[stream] && + dai_info->dailink[stream] < 0) { + dev_err(dev, + "Invalid dailink id %d for codec: 0x%x\n", + dai_info->dailink[stream], + codec_info->part_id); + return -EINVAL; + } + + if (dai_info->direction[stream]) { + num_dais += !soc_dai->num_devs[stream]; + soc_dai->num_devs[stream]++; + soc_dai->link_mask[stream] |= adr_link->mask; + } + } + + num_link_dailinks += !!list_empty(&soc_dai->endpoints); + list_add_tail(&soc_end->list, &soc_dai->endpoints); + + soc_end->link_mask = adr_link->mask; + soc_end->codec_name = codec_name; + soc_end->codec_info = codec_info; + soc_end->dai_info = dai_info; + soc_end++; + } + } + + ctx->append_dai_type |= (num_link_dailinks > 1); + } + + return num_dais; +} +EXPORT_SYMBOL_NS(asoc_sdw_parse_sdw_endpoints, SND_SOC_SDW_UTILS); + MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("SoundWire ASoC helpers"); From 6d8348ddc56ed43ba39d1e8adda13299201f32ed Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 13 Sep 2024 14:36:31 +0530 Subject: [PATCH 408/410] ASoC: amd: acp: refactor SoundWire machine driver code Refactor Soundwire machine driver code by using common SoundWire endpoint parsing helper functions. Signed-off-by: Vijendar Mukunda Reviewed-by: Bard Liao Link: https://patch.msgid.link/20240913090631.1834543-6-Vijendar.Mukunda@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 544 ++++++++------------------- 1 file changed, 156 insertions(+), 388 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index b1cd173f607d..6c50c8276538 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -64,197 +64,6 @@ static const struct snd_soc_ops sdw_ops = { .shutdown = asoc_sdw_shutdown, }; -/* - * get BE dailink number and CPU DAI number based on sdw link adr. - * Since some sdw slaves may be aggregated, the CPU DAI number - * may be larger than the number of BE dailinks. - */ -static int get_dailink_info(struct device *dev, - const struct snd_soc_acpi_link_adr *adr_link, - int *sdw_be_num, int *codecs_num) -{ - bool group_visited[AMD_SDW_MAX_GROUPS]; - int i; - int j; - - *sdw_be_num = 0; - - if (!adr_link) - return -EINVAL; - - for (i = 0; i < AMD_SDW_MAX_GROUPS; i++) - group_visited[i] = false; - - for (; adr_link->num_adr; adr_link++) { - const struct snd_soc_acpi_endpoint *endpoint; - struct asoc_sdw_codec_info *codec_info; - int stream; - u64 adr; - - /* make sure the link mask has a single bit set */ - if (!is_power_of_2(adr_link->mask)) - return -EINVAL; - - for (i = 0; i < adr_link->num_adr; i++) { - adr = adr_link->adr_d[i].adr; - codec_info = asoc_sdw_find_codec_info_part(adr); - if (!codec_info) - return -EINVAL; - - *codecs_num += codec_info->dai_num; - - if (!adr_link->adr_d[i].name_prefix) { - dev_err(dev, "codec 0x%llx does not have a name prefix\n", - adr_link->adr_d[i].adr); - return -EINVAL; - } - - endpoint = adr_link->adr_d[i].endpoints; - if (endpoint->aggregated && !endpoint->group_id) { - dev_err(dev, "invalid group id on link %x\n", - adr_link->mask); - return -EINVAL; - } - - for (j = 0; j < codec_info->dai_num; j++) { - /* count DAI number for playback and capture */ - for_each_pcm_streams(stream) { - if (!codec_info->dais[j].direction[stream]) - continue; - - /* count BE for each non-aggregated slave or group */ - if (!endpoint->aggregated || - !group_visited[endpoint->group_id]) - (*sdw_be_num)++; - } - } - - if (endpoint->aggregated) - group_visited[endpoint->group_id] = true; - } - } - return 0; -} - -static int fill_sdw_codec_dlc(struct device *dev, - const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_dai_link_component *codec, - int adr_index, int dai_index) -{ - u64 adr = adr_link->adr_d[adr_index].adr; - struct asoc_sdw_codec_info *codec_info; - - codec_info = asoc_sdw_find_codec_info_part(adr); - if (!codec_info) - return -EINVAL; - - codec->name = asoc_sdw_get_codec_name(dev, codec_info, adr_link, adr_index); - if (!codec->name) - return -ENOMEM; - - codec->dai_name = codec_info->dais[dai_index].dai_name; - dev_err(dev, "codec->dai_name:%s\n", codec->dai_name); - return 0; -} - -static int set_codec_init_func(struct snd_soc_card *card, - const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_dai_link *dai_links, - bool playback, int group_id, int adr_index, int dai_index) -{ - int i = adr_index; - - do { - /* - * Initialize the codec. If codec is part of an aggregated - * group (group_id>0), initialize all codecs belonging to - * same group. - * The first link should start with adr_link->adr_d[adr_index] - * because that is the device that we want to initialize and - * we should end immediately if it is not aggregated (group_id=0) - */ - for ( ; i < adr_link->num_adr; i++) { - struct asoc_sdw_codec_info *codec_info; - - codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; - - /* The group_id is > 0 iff the codec is aggregated */ - if (adr_link->adr_d[i].endpoints->group_id != group_id) - continue; - if (codec_info->dais[dai_index].init) - codec_info->dais[dai_index].init(card, - dai_links, - codec_info, - playback); - - if (!group_id) - return 0; - } - - i = 0; - adr_link++; - } while (adr_link->mask); - - return 0; -} - -/* - * check endpoint status in slaves and gather link ID for all slaves in - * the same group to generate different CPU DAI. Now only support - * one sdw link with all slaves set with only single group id. - * - * one slave on one sdw link with aggregated = 0 - * one sdw BE DAI <---> one-cpu DAI <---> one-codec DAI - * - * two or more slaves on one sdw link with aggregated = 1 - * one sdw BE DAI <---> one-cpu DAI <---> multi-codec DAIs - */ -static int get_slave_info(const struct snd_soc_acpi_link_adr *adr_link, - struct device *dev, int *cpu_dai_id, int *cpu_dai_num, - int *codec_num, unsigned int *group_id, - int adr_index) -{ - int i; - - if (!adr_link->adr_d[adr_index].endpoints->aggregated) { - cpu_dai_id[0] = ffs(adr_link->mask) - 1; - *cpu_dai_num = 1; - *codec_num = 1; - *group_id = 0; - return 0; - } - - *codec_num = 0; - *cpu_dai_num = 0; - *group_id = adr_link->adr_d[adr_index].endpoints->group_id; - - /* Count endpoints with the same group_id in the adr_link */ - for (; adr_link && adr_link->num_adr; adr_link++) { - unsigned int link_codecs = 0; - - for (i = 0; i < adr_link->num_adr; i++) { - if (adr_link->adr_d[i].endpoints->aggregated && - adr_link->adr_d[i].endpoints->group_id == *group_id) - link_codecs++; - } - - if (link_codecs) { - *codec_num += link_codecs; - - if (*cpu_dai_num >= ACP63_SDW_MAX_CPU_DAIS) { - dev_err(dev, "cpu_dai_id array overflowed\n"); - return -EINVAL; - } - - cpu_dai_id[(*cpu_dai_num)++] = ffs(adr_link->mask) - 1; - } - } - - return 0; -} - static int get_acp63_cpu_pin_id(u32 sdw_link_id, int be_id, int *cpu_pin_id, struct device *dev) { switch (sdw_link_id) { @@ -306,116 +115,64 @@ static int get_acp63_cpu_pin_id(u32 sdw_link_id, int be_id, int *cpu_pin_id, str static const char * const type_strings[] = {"SimpleJack", "SmartAmp", "SmartMic"}; static int create_sdw_dailink(struct snd_soc_card *card, + struct asoc_sdw_dailink *sof_dai, struct snd_soc_dai_link **dai_links, - const struct snd_soc_acpi_link_adr *adr_link, - struct snd_soc_codec_conf **codec_conf, - int *be_id, int adr_index, int dai_index) + int *be_id, struct snd_soc_codec_conf **codec_conf) { + struct device *dev = card->dev; struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct amd_mc_ctx *amd_ctx = (struct amd_mc_ctx *)ctx->private; - struct device *dev = card->dev; - const struct snd_soc_acpi_link_adr *adr_link_next; - struct snd_soc_dai_link_ch_map *sdw_codec_ch_maps; - struct snd_soc_dai_link_component *codecs; - struct snd_soc_dai_link_component *cpus; - struct asoc_sdw_codec_info *codec_info; - int cpu_dai_id[ACP63_SDW_MAX_CPU_DAIS]; - int cpu_dai_num; - unsigned int group_id; - unsigned int sdw_link_id; - int codec_dlc_index = 0; - int codec_num; - int stream; - int i = 0; - int j, k; - int ret; + struct asoc_sdw_endpoint *sof_end; int cpu_pin_id; + int stream; + int ret; - ret = get_slave_info(adr_link, dev, cpu_dai_id, &cpu_dai_num, &codec_num, - &group_id, adr_index); - if (ret) - return ret; - codecs = devm_kcalloc(dev, codec_num, sizeof(*codecs), GFP_KERNEL); - if (!codecs) - return -ENOMEM; - - sdw_codec_ch_maps = devm_kcalloc(dev, codec_num, - sizeof(*sdw_codec_ch_maps), GFP_KERNEL); - if (!sdw_codec_ch_maps) - return -ENOMEM; - - /* generate codec name on different links in the same group */ - j = adr_index; - for (adr_link_next = adr_link; adr_link_next && adr_link_next->num_adr && - i < cpu_dai_num; adr_link_next++) { - /* skip the link excluded by this processed group */ - if (cpu_dai_id[i] != ffs(adr_link_next->mask) - 1) - continue; - - /* j reset after loop, adr_index only applies to first link */ - for (k = 0 ; (j < adr_link_next->num_adr) && (k < codec_num) ; j++, k++) { - const struct snd_soc_acpi_endpoint *endpoints; - - endpoints = adr_link_next->adr_d[j].endpoints; - if (group_id && (!endpoints->aggregated || - endpoints->group_id != group_id)) - continue; - - /* sanity check */ - if (*codec_conf >= card->codec_conf + card->num_configs) { - dev_err(dev, "codec_conf array overflowed\n"); - return -EINVAL; - } - - ret = fill_sdw_codec_dlc(dev, adr_link_next, - &codecs[codec_dlc_index], - j, dai_index); - if (ret) - return ret; - (*codec_conf)->dlc = codecs[codec_dlc_index]; - (*codec_conf)->name_prefix = adr_link_next->adr_d[j].name_prefix; - - sdw_codec_ch_maps[codec_dlc_index].cpu = i; - sdw_codec_ch_maps[codec_dlc_index].codec = codec_dlc_index; - - codec_dlc_index++; + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (sof_end->name_prefix) { + (*codec_conf)->dlc.name = sof_end->codec_name; + (*codec_conf)->name_prefix = sof_end->name_prefix; (*codec_conf)++; } - j = 0; - /* check next link to create codec dai in the processed group */ - i++; + if (sof_end->include_sidecar) { + ret = sof_end->codec_info->add_sidecar(card, dai_links, codec_conf); + if (ret) + return ret; + } } - /* find codec info to create BE DAI */ - codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[adr_index].adr); - if (!codec_info) - return -EINVAL; - - ctx->ignore_internal_dmic |= codec_info->ignore_internal_dmic; - - sdw_link_id = (adr_link->adr_d[adr_index].adr) >> 48; for_each_pcm_streams(stream) { - char *name, *cpu_name; - int playback, capture; static const char * const sdw_stream_name[] = { "SDW%d-PIN%d-PLAYBACK", "SDW%d-PIN%d-CAPTURE", "SDW%d-PIN%d-PLAYBACK-%s", "SDW%d-PIN%d-CAPTURE-%s", }; + struct snd_soc_dai_link_ch_map *codec_maps; + struct snd_soc_dai_link_component *codecs; + struct snd_soc_dai_link_component *cpus; + int num_cpus = hweight32(sof_dai->link_mask[stream]); + int num_codecs = sof_dai->num_devs[stream]; + int playback, capture; + int i = 0, j = 0; + char *name; - if (!codec_info->dais[dai_index].direction[stream]) + if (!sof_dai->num_devs[stream]) continue; - *be_id = codec_info->dais[dai_index].dailink[stream]; + sof_end = list_first_entry(&sof_dai->endpoints, + struct asoc_sdw_endpoint, list); + + *be_id = sof_end->dai_info->dailink[stream]; if (*be_id < 0) { dev_err(dev, "Invalid dailink id %d\n", *be_id); return -EINVAL; } + switch (amd_ctx->acp_rev) { case ACP63_PCI_REV: - ret = get_acp63_cpu_pin_id(sdw_link_id, *be_id, &cpu_pin_id, dev); + ret = get_acp63_cpu_pin_id(ffs(sof_end->link_mask - 1), + *be_id, &cpu_pin_id, dev); if (ret) return ret; break; @@ -425,55 +182,105 @@ static int create_sdw_dailink(struct snd_soc_card *card, /* create stream name according to first link id */ if (ctx->append_dai_type) { name = devm_kasprintf(dev, GFP_KERNEL, - sdw_stream_name[stream + 2], sdw_link_id, cpu_pin_id, - type_strings[codec_info->dais[dai_index].dai_type]); + sdw_stream_name[stream + 2], + ffs(sof_end->link_mask) - 1, + cpu_pin_id, + type_strings[sof_end->dai_info->dai_type]); } else { name = devm_kasprintf(dev, GFP_KERNEL, - sdw_stream_name[stream], sdw_link_id, cpu_pin_id); + sdw_stream_name[stream], + ffs(sof_end->link_mask) - 1, + cpu_pin_id); } if (!name) return -ENOMEM; - cpus = devm_kcalloc(dev, cpu_dai_num, sizeof(*cpus), GFP_KERNEL); + cpus = devm_kcalloc(dev, num_cpus, sizeof(*cpus), GFP_KERNEL); if (!cpus) return -ENOMEM; - /* - * generate CPU DAI name base on the sdw link ID and - * cpu pin id according to sdw dai driver. - */ - for (k = 0; k < cpu_dai_num; k++) { - cpu_name = devm_kasprintf(dev, GFP_KERNEL, - "SDW%d Pin%d", sdw_link_id, cpu_pin_id); - if (!cpu_name) + + codecs = devm_kcalloc(dev, num_codecs, sizeof(*codecs), GFP_KERNEL); + if (!codecs) + return -ENOMEM; + + codec_maps = devm_kcalloc(dev, num_codecs, sizeof(*codec_maps), GFP_KERNEL); + if (!codec_maps) + return -ENOMEM; + + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (!sof_end->dai_info->direction[stream]) + continue; + + int link_num = ffs(sof_end->link_mask) - 1; + + cpus[i].dai_name = devm_kasprintf(dev, GFP_KERNEL, + "SDW%d Pin%d", + link_num, cpu_pin_id); + dev_dbg(dev, "cpu[%d].dai_name:%s\n", i, cpus[i].dai_name); + if (!cpus[i].dai_name) return -ENOMEM; - cpus[k].dai_name = cpu_name; + codec_maps[j].cpu = i; + codec_maps[j].codec = j; + + codecs[j].name = sof_end->codec_name; + codecs[j].dai_name = sof_end->dai_info->dai_name; + j++; } + WARN_ON(j != num_codecs); + playback = (stream == SNDRV_PCM_STREAM_PLAYBACK); capture = (stream == SNDRV_PCM_STREAM_CAPTURE); - asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, - playback, capture, - cpus, cpu_dai_num, - platform_component, ARRAY_SIZE(platform_component), - codecs, codec_num, + + asoc_sdw_init_dai_link(dev, *dai_links, be_id, name, playback, capture, + cpus, num_cpus, platform_component, + ARRAY_SIZE(platform_component), codecs, num_codecs, asoc_sdw_rtd_init, &sdw_ops); + /* * SoundWire DAILINKs use 'stream' functions and Bank Switch operations * based on wait_for_completion(), tag them as 'nonatomic'. */ (*dai_links)->nonatomic = true; - (*dai_links)->ch_maps = sdw_codec_ch_maps; + (*dai_links)->ch_maps = codec_maps; - ret = set_codec_init_func(card, adr_link, *dai_links, - playback, group_id, adr_index, dai_index); - if (ret < 0) { - dev_err(dev, "failed to init codec 0x%x\n", codec_info->part_id); - return ret; + list_for_each_entry(sof_end, &sof_dai->endpoints, list) { + if (sof_end->dai_info->init) + sof_end->dai_info->init(card, *dai_links, + sof_end->codec_info, + playback); } (*dai_links)++; } + + return 0; +} + +static int create_sdw_dailinks(struct snd_soc_card *card, + struct snd_soc_dai_link **dai_links, int *be_id, + struct asoc_sdw_dailink *sof_dais, + struct snd_soc_codec_conf **codec_conf) +{ + int ret; + + /* generate DAI links by each sdw link */ + while (sof_dais->initialised) { + int current_be_id; + + ret = create_sdw_dailink(card, sof_dais, dai_links, + ¤t_be_id, codec_conf); + if (ret) + return ret; + + /* Update the be_id to match the highest ID used for SDW link */ + if (*be_id < current_be_id) + *be_id = current_be_id; + + sof_dais++; + } + return 0; } @@ -504,132 +311,93 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) int sdw_be_num = 0, dmic_num = 0; struct asoc_sdw_mc_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_acpi_mach_params *mach_params = &mach->mach_params; - const struct snd_soc_acpi_link_adr *adr_link = mach_params->links; struct snd_soc_codec_conf *codec_conf; - int codec_conf_num = 0; - bool group_generated[AMD_SDW_MAX_GROUPS] = { }; + struct asoc_sdw_endpoint *sof_ends; + struct asoc_sdw_dailink *sof_dais; struct snd_soc_dai_link *dai_links; - struct asoc_sdw_codec_info *codec_info; + int num_devs = 0; + int num_ends = 0; int num_links; - int i, j, be_id = 0; + int be_id = 0; int ret; - ret = get_dailink_info(dev, adr_link, &sdw_be_num, &codec_conf_num); + ret = asoc_sdw_count_sdw_endpoints(card, &num_devs, &num_ends); if (ret < 0) { - dev_err(dev, "failed to get sdw link info %d\n", ret); + dev_err(dev, "failed to count devices/endpoints: %d\n", ret); return ret; } + /* One per DAI link, worst case is a DAI link for every endpoint */ + sof_dais = kcalloc(num_ends, sizeof(*sof_dais), GFP_KERNEL); + if (!sof_dais) + return -ENOMEM; + + /* One per endpoint, ie. each DAI on each codec/amp */ + sof_ends = kcalloc(num_ends, sizeof(*sof_ends), GFP_KERNEL); + if (!sof_ends) { + ret = -ENOMEM; + goto err_dai; + } + + ret = asoc_sdw_parse_sdw_endpoints(card, sof_dais, sof_ends, &num_devs); + if (ret < 0) + goto err_end; + + sdw_be_num = ret; + /* enable dmic */ if (sof_sdw_quirk & ASOC_SDW_ACP_DMIC || mach_params->dmic_num) dmic_num = 1; dev_dbg(dev, "sdw %d, dmic %d", sdw_be_num, dmic_num); + codec_conf = devm_kcalloc(dev, num_devs, sizeof(*codec_conf), GFP_KERNEL); + if (!codec_conf) { + ret = -ENOMEM; + goto err_end; + } + /* allocate BE dailinks */ num_links = sdw_be_num + dmic_num; dai_links = devm_kcalloc(dev, num_links, sizeof(*dai_links), GFP_KERNEL); - if (!dai_links) - return -ENOMEM; - - /* allocate codec conf, will be populated when dailinks are created */ - codec_conf = devm_kcalloc(dev, codec_conf_num, sizeof(*codec_conf), - GFP_KERNEL); - if (!codec_conf) - return -ENOMEM; + if (!dai_links) { + ret = -ENOMEM; + goto err_end; + } + card->codec_conf = codec_conf; + card->num_configs = num_devs; card->dai_link = dai_links; card->num_links = num_links; - card->codec_conf = codec_conf; - card->num_configs = codec_conf_num; /* SDW */ - if (!sdw_be_num) - goto DMIC; - - for (; adr_link->num_adr; adr_link++) { - /* - * If there are two or more different devices on the same sdw link, we have to - * append the codec type to the dai link name to prevent duplicated dai link name. - * The same type devices on the same sdw link will be in the same - * snd_soc_acpi_adr_device array. They won't be described in different adr_links. - */ - for (i = 0; i < adr_link->num_adr; i++) { - /* find codec info to get dai_num */ - codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; - if (codec_info->dai_num > 1) { - ctx->append_dai_type = true; - goto out; - } - for (j = 0; j < i; j++) { - if ((SDW_PART_ID(adr_link->adr_d[i].adr) != - SDW_PART_ID(adr_link->adr_d[j].adr)) || - (SDW_MFG_ID(adr_link->adr_d[i].adr) != - SDW_MFG_ID(adr_link->adr_d[j].adr))) { - ctx->append_dai_type = true; - goto out; - } - } - } - } -out: - - /* generate DAI links by each sdw link */ - for (adr_link = mach_params->links ; adr_link->num_adr; adr_link++) { - for (i = 0; i < adr_link->num_adr; i++) { - const struct snd_soc_acpi_endpoint *endpoint; - - endpoint = adr_link->adr_d[i].endpoints; - - /* this group has been generated */ - if (endpoint->aggregated && - group_generated[endpoint->group_id]) - continue; - - /* find codec info to get dai_num */ - codec_info = asoc_sdw_find_codec_info_part(adr_link->adr_d[i].adr); - if (!codec_info) - return -EINVAL; - - for (j = 0; j < codec_info->dai_num ; j++) { - int current_be_id; - - ret = create_sdw_dailink(card, &dai_links, adr_link, - &codec_conf, ¤t_be_id, - i, j); - if (ret < 0) { - dev_err(dev, - "failed to create dai link %d on 0x%x\n", - j, codec_info->part_id); - return ret; - } - /* Update the be_id to match the highest ID used for SDW link */ - if (be_id < current_be_id) - be_id = current_be_id; - } - - if (endpoint->aggregated) - group_generated[endpoint->group_id] = true; - } + if (sdw_be_num) { + ret = create_sdw_dailinks(card, &dai_links, &be_id, + sof_dais, &codec_conf); + if (ret) + goto err_end; } -DMIC: /* dmic */ if (dmic_num > 0) { if (ctx->ignore_internal_dmic) { dev_warn(dev, "Ignoring ACP DMIC\n"); } else { - be_id = SOC_SDW_DMIC_DAI_ID; ret = create_dmic_dailinks(card, &dai_links, &be_id); if (ret) - return ret; + goto err_end; } } + WARN_ON(codec_conf != card->codec_conf + card->num_configs); WARN_ON(dai_links != card->dai_link + card->num_links); - return 0; + +err_end: + kfree(sof_ends); +err_dai: + kfree(sof_dais); + + return ret; } /* SoC card */ From 49e2e353fb0dbef8dced3e8e65365580349c4b14 Mon Sep 17 00:00:00 2001 From: Shenghao Ding Date: Thu, 12 Sep 2024 07:27:37 +0800 Subject: [PATCH 409/410] ASoC: tas2781: Add Calibration Kcontrols for Chromebook Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook. Signed-off-by: Shenghao Ding Link: https://patch.msgid.link/20240911232739.1509-1-shenghao-ding@ti.com Signed-off-by: Mark Brown --- include/sound/tas2781.h | 68 +++ sound/soc/codecs/tas2781-comlib.c | 26 + sound/soc/codecs/tas2781-fmwlib.c | 60 +- sound/soc/codecs/tas2781-i2c.c | 887 +++++++++++++++++++++++++++++- 4 files changed, 1030 insertions(+), 11 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dbda552398b5..8cd6da0480b7 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,12 +49,59 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +117,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_backup; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -81,9 +136,19 @@ struct tasdevice { bool is_loaderr; }; +struct cali_reg { + unsigned int r0_reg; + unsigned int r0_low_reg; + unsigned int invr0_reg; + unsigned int pow_reg; + unsigned int tlimit_reg; +}; + struct calidata { unsigned char *data; unsigned long total_sz; + struct cali_reg cali_reg_array; + unsigned int cali_dat_sz_per_dev; }; struct tasdevice_priv { @@ -119,6 +184,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -145,6 +211,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 664c371796d6..1e0b3aa95749 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -88,6 +88,32 @@ out: return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f0710..3de0132c345d 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,20 +2151,61 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (!cal) + } + if (!priv->is_user_space_calidata) return; + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); + return; + } + k++; - load_calib_data(priv, &cal->dev_data); + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->r0_low_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->invr0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->pow_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->tlimit_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit_reg err = %d\n", i, rc); + return; + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2259,9 +2300,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..8a8d97dd7f25 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -33,6 +33,69 @@ #include #include +#define X2563_CL_STT_VAL(xreg, xval) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, } + +#define X2563_CL_STT_4BYTS(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, } + +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + X2563_CL_STT_VAL(TAS2563_IDLE, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_ENFF_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_DISTCK_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_TE_SCTHR_REG, 0x7f, 0xff, 0xff, 0xff), + X2563_CL_STT_4BYTS(TAS2563_PRM_PLT_FLAG_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_SINEGAIN_REG, 0x0a, 0x3d, 0x70, 0xa4), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA2_REG, 0x00, 0x06, 0xd3, 0x72), + X2563_CL_STT_4BYTS(TAS2563_TE_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_DT_REG, 0x00, 0x36, 0x91, 0x5e), +}; + +#define X2781_CL_STT_VAL(xreg, xval, xlocked) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, \ + .is_locked = xlocked, } + +#define X2781_CL_STT_4BYTS_UNLOCKED(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, \ + .is_locked = false, } + +#define X2781_CL_STT_LEN_UNLOCKED(xreg) \ +{ .reg = xreg, \ + .val_len = 4, \ + .is_locked = false, } + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), + X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), + X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), + X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), + X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), + X2781_CL_STT_VAL(TAS2781_PRM_NG_CFG0_REG, 0xb9, false), + X2781_CL_STT_VAL(TAS2781_PRM_IDLE_CH_DET_REG, 0x92, false), + /* + * This register is pilot tone threshold, different with the + * calibration tool version, it will be updated in + * tas2781_calib_start_put(), set to 1mA. + */ + X2781_CL_STT_4BYTS_UNLOCKED(0, 0x00, 0x00, 0x00, 0x56), + X2781_CL_STT_4BYTS_UNLOCKED(TAS2781_PRM_PLT_FLAG_REG, + 0x40, 0x00, 0x00, 0x00), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN_REG), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN2_REG), +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -141,6 +204,557 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = cali_data->data; + unsigned int i = 0; + unsigned int j, k; + int rc; + + guard(mutex)(&priv->codec_lock); + if (!priv->is_user_space_calidata) + return -1; + + if (!p->r0_reg) + return -1; + + dst[i++] = bytes_ext->max; + dst[i++] = 'r'; + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_low_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->invr0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->pow_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->tlimit_reg); + + for (j = 0, k = 0; j < priv->ndev; j++) { + if (j == data[k]) { + dst[i++] = j; + k++; + } else { + dev_err(priv->dev, "chn %d device %u not match\n", + j, data[k]); + k += 21; + continue; + } + rc = tasdevice_dev_bulk_read(priv, j, p->r0_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_rd err = %d\n", + j, rc); + i += 20; + k += 20; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_data is not same\n", j); + k += 4; + i += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->r0_low_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low bulk_rd err = %d\n", + j, rc); + i += 16; + k += 16; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_low is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->invr0_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0 bulk_rd err = %d\n", + j, rc); + i += 12; + k += 12; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d invr0 is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->pow_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_rd err = %d\n", + j, rc); + i += 8; + k += 8; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d pow_reg is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->tlimit_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit bulk_rd err = %d\n", + j, rc); + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d tlimit is not same\n", j); + i += 4; + k += 4; + } + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, + int *reg, unsigned char *dat) +{ + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + switch (p[j].reg) { + case 0: { + if (!reg[0]) + continue; + p[j].reg = reg[0]; + } + break; + case TAS2781_PRM_PLT_FLAG_REG: + p[j].reg = reg[1]; + break; + case TAS2781_PRM_SINEGAIN_REG: + p[j].reg = reg[2]; + break; + case TAS2781_PRM_SINEGAIN2_REG: + p[j].reg = reg[3]; + break; + } + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + } + + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int i, reg[4]; + int j = 0; + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id != TAS2781 || bytes_ext->max != dat[0] || + dat[1] != 'r') { + dev_err(priv->dev, "%s: package fmt or chipid incorrect\n", + __func__); + return 0; + } + j += 2; + /* refresh pilot tone and SineGain register */ + for (i = 0; i < ARRAY_SIZE(reg); i++) { + reg[i] = TASDEVICE_REG(dat[j], dat[j + 1], dat[j + 2]); + j += 3; + } + + for (i = 0; i < priv->ndev; i++) { + int k = i * 9 + j; + + if (dat[k] != i) { + dev_err(priv->dev, "%s:no cal-setting for dev %d\n", + __func__, i); + continue; + } + sngl_calib_start(priv, i, reg, dat + k); + } + return 1; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id == TAS2563) + tas2563_calib_stop_put(priv); + else + tas2781_calib_stop_put(priv); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *src = ucontrol->value.bytes.data; + unsigned char *dst = cali_data->data; + int rc = 1, i = 0; + int j; + + guard(mutex)(&priv->codec_lock); + if (src[0] != bytes_ext->max || src[1] != 'r') { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + for (j = 0; j < priv->ndev; j++) { + if (src[17 + j * 21] != j) { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + } + i += 2; + priv->is_user_space_calidata = true; + + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + + memcpy(dst, &src[i], cali_data->total_sz); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + dst[0] = bytes_ext->max; + guard(mutex)(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[1] = i; + rc = tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + if (rc < 0) + dev_err(tas_priv->dev, "%s, get value error\n", + __func__); + else + dst[2] = val; + + break; + } + } + + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + + if (tas_priv->chip_id == TAS2563) + reg = TAS2563_PRM_R0_REG; + else if (cali_data->cali_reg_array.r0_reg) + reg = cali_data->cali_reg_array.r0_reg; + else + return -1; + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +855,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 6, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 6, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 6, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 6, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +874,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +885,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +907,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,6 +988,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -421,11 +1090,47 @@ static int tasdevice_configuration_put( return ret; } +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + guard(mutex)(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + + return rc; +} + static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -466,6 +1171,30 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", + GFP_KERNEL); + if (!active_dev_num) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); + if (!chip_id) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -473,6 +1202,149 @@ out: return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) +{ + struct calidata *cali_data = &priv->cali_data; + struct tasdevice *tasdev = priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int nctrls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + nctrls = ARRAY_SIZE(tas2781_cali_controls); + for (i = 0; i < priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + nctrls = ARRAY_SIZE(tas2563_cali_controls); + for (i = 0; i < priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls); + if (rc < 0) { + dev_err(priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (priv->chip_id == TAS2781) + nctrls = 2; + else + nctrls = 1; + + /* + * Alloc kcontrol via devm_kzalloc(), which don't manually + * free the kcontrol。 + */ + cali_ctrls = devm_kcalloc(priv->dev, nctrls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + ext_cali_data = devm_kzalloc(priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!ext_cali_data) + return -ENOMEM; + + cali_name = devm_kstrdup(priv->dev, "Speaker Calibrated Data", + GFP_KERNEL); + if (!cali_name) + return -ENOMEM; + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz_per_dev = 20; + /* + * Data structure for tas2563/tas2781 calibrated data: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register array for calibrated data (15 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Calibrated data for Device #i (20 bytes) + * } + */ + ext_cali_data->max = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; + priv->cali_data.total_sz = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1); + priv->cali_data.data = devm_kzalloc(priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + ext_cali_start = devm_kzalloc(priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!ext_cali_start) + return -ENOMEM; + + cali_start_name = devm_kstrdup(priv->dev, + "Calibration Start", GFP_KERNEL); + if (!cali_start_name) + return -ENOMEM; + /* + * package structure for tas2781 ftc start: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register for pilot threshold, pilot tone + * and sine gain (12 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Sine gain for Device #i (8 bytes) + * } + */ + ext_cali_start->max = 14 + priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + nctrls < i ? nctrls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -519,6 +1391,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -720,6 +1598,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_backup); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); From 2772ee6de6cf94e5f2a0c0ce6067d0796a4170ba Mon Sep 17 00:00:00 2001 From: Tang Bin Date: Sun, 8 Sep 2024 22:02:59 +0800 Subject: [PATCH 410/410] ASoC: topology: Fix redundant logical jump In the function soc_tplg_dai_config, the logical jump of 'goto err' is redundant, so remove it. Signed-off-by: Tang Bin Link: https://patch.msgid.link/20240908140259.3859-1-tangbin@cmss.chinamobile.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index af5d42b57be7..af3158cdc8d5 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1894,7 +1894,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; ret = set_stream_info(tplg, stream, caps); if (ret < 0) - goto err; + return ret; } if (d->capture) { @@ -1902,7 +1902,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; ret = set_stream_info(tplg, stream, caps); if (ret < 0) - goto err; + return ret; } if (d->flag_mask) @@ -1914,13 +1914,10 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->dev, "ASoC: DAI loading failed\n"); - goto err; + return ret; } return 0; - -err: - return ret; } /* load physical DAI elements */