Merge remote-tracking branches 'asoc/topic/tas5086', 'asoc/topic/tegra', 'asoc/topic/tlv320aic31xx', 'asoc/topic/tlv320dac33' and 'asoc/topic/topology' into asoc-next

2016-09-29 12:44:45 -07:00 · 2016-09-29 12:44:45 -07:00 · 84b720f0fe
commit 84b720f0fe
parent 5b56b9bb6c e56375155e c614a31287 3520646dbb 8891098246 f918e1697b
12 changed files with 585 additions and 80 deletions
--- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt
+++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-sgtl5000.txt
@ -0,0 +1,42 @@
+NVIDIA Tegra audio complex, with SGTL5000 CODEC
+
+Required properties:
+- compatible : "nvidia,tegra-audio-sgtl5000"
+- clocks : Must contain an entry for each entry in clock-names.
+  See ../clocks/clock-bindings.txt for details.
+- clock-names : Must include the following entries:
+  - pll_a
+  - pll_a_out0
+  - mclk (The Tegra cdev1/extern1 clock, which feeds the CODEC's mclk)
+- nvidia,model : The user-visible name of this sound complex.
+- nvidia,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 for sources and
+  sinks are the SGTL5000's pins (as documented in its binding), and the jacks
+  on the board:
+
+  * Headphone Jack
+  * Line In Jack
+  * Mic Jack
+
+- nvidia,i2s-controller : The phandle of the Tegra I2S controller that's
+  connected to the CODEC.
+- nvidia,audio-codec : The phandle of the SGTL5000 audio codec.
+
+Example:
+
+sound {
+	compatible = "toradex,tegra-audio-sgtl5000-apalis_t30",
+		     "nvidia,tegra-audio-sgtl5000";
+	nvidia,model = "Toradex Apalis T30";
+	nvidia,audio-routing =
+		"Headphone Jack", "HP_OUT",
+		"LINE_IN", "Line In Jack",
+		"MIC_IN", "Mic Jack";
+	nvidia,i2s-controller = <&tegra_i2s2>;
+	nvidia,audio-codec = <&sgtl5000>;
+	clocks = <&tegra_car TEGRA30_CLK_PLL_A>,
+		 <&tegra_car TEGRA30_CLK_PLL_A_OUT0>,
+		 <&tegra_car TEGRA30_CLK_EXTERN1>;
+	clock-names = "pll_a", "pll_a_out0", "mclk";
+};
--- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
+++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt
@ -11,6 +11,7 @@ Required properties:
    "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)

 - reg - <int> -  I2C slave address
 - HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply,
@ -37,9 +38,11 @@ CODEC output pins:
  * MICBIAS

 CODEC input pins:
-  * MIC1LP
-  * MIC1RP
-  * MIC1LM
+  * 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.

--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@ -83,7 +83,7 @@
 #define SND_SOC_TPLG_NUM_TEXTS		16

 /* ABI version */
-#define SND_SOC_TPLG_ABI_VERSION	0x4
+#define SND_SOC_TPLG_ABI_VERSION	0x5

 /* Max size of TLV data */
 #define SND_SOC_TPLG_TLV_SIZE		32
@ -105,7 +105,8 @@
 #define SND_SOC_TPLG_TYPE_CODEC_LINK	9
 #define SND_SOC_TPLG_TYPE_BACKEND_LINK	10
 #define SND_SOC_TPLG_TYPE_PDATA		11
-#define SND_SOC_TPLG_TYPE_MAX	SND_SOC_TPLG_TYPE_PDATA
+#define SND_SOC_TPLG_TYPE_BE_DAI	12
+#define SND_SOC_TPLG_TYPE_MAX		SND_SOC_TPLG_TYPE_BE_DAI

 /* vendor block IDs - please add new vendor types to end */
 #define SND_SOC_TPLG_TYPE_VENDOR_FW	1000
@ -124,6 +125,11 @@
 #define SND_SOC_TPLG_TUPLE_TYPE_WORD	4
 #define SND_SOC_TPLG_TUPLE_TYPE_SHORT	5

+/* BE DAI flags */
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES         (1 << 0)
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS      (1 << 1)
+#define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
+
 /*
 * Block Header.
 * This header precedes all object and object arrays below.
@ -251,6 +257,7 @@ struct snd_soc_tplg_stream_caps {
 	__le32 period_size_max;	/* max period size bytes */
 	__le32 buffer_size_min;	/* min buffer size bytes */
 	__le32 buffer_size_max;	/* max buffer size bytes */
+	__le32 sig_bits;        /* number of bits of content */
 } __attribute__((packed));

 /*
@ -285,6 +292,8 @@ struct snd_soc_tplg_manifest {
 	__le32 graph_elems;	/* number of graph elements */
 	__le32 pcm_elems;	/* number of PCM elements */
 	__le32 dai_link_elems;	/* number of DAI link elements */
+	__le32 be_dai_elems;	/* number of BE DAI elements */
+	__le32 reserved[20];	/* reserved for new ABI element types */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));

@ -450,4 +459,26 @@ struct snd_soc_tplg_link_config {
 	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
 	__le32 num_streams;     /* number of streams */
 } __attribute__((packed));
+
+/*
+ * Describes SW/FW specific features of BE DAI.
+ *
+ * File block representation for BE DAI :-
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_hdr           |  1  |
+ * +-----------------------------------+-----+
+ * | struct snd_soc_tplg_be_dai        |  N  |
+ * +-----------------------------------+-----+
+ */
+struct snd_soc_tplg_be_dai {
+	__le32 size;            /* in bytes of this structure */
+	char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
+	__le32 dai_id;          /* unique ID - used to match */
+	__le32 playback;        /* supports playback mode */
+	__le32 capture;         /* supports capture mode */
+	struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
+	__le32 flag_mask;       /* bitmask of flags to configure */
+	__le32 flags;           /* SND_SOC_TPLG_DAI_FLGBIT_* */
+	struct snd_soc_tplg_private priv;
+} __attribute__((packed));
 #endif
--- a/sound/soc/codecs/tas5086.c
+++ b/sound/soc/codecs/tas5086.c
@ -387,7 +387,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream,
 	val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios),
 			     priv->mclk / priv->rate);
 	if (val < 0) {
-		dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n");
+		dev_err(codec->dev, "Invalid MCLK / Fs ratio\n");
 		return -EINVAL;
 	}

--- a/sound/soc/codecs/tlv320aic31xx.c
+++ b/sound/soc/codecs/tlv320aic31xx.c
@ -273,10 +273,20 @@ static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0);
 /*
 * controls to be exported to the user space
 */
-static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
+static const struct snd_kcontrol_new common31xx_snd_controls[] = {
 	SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL,
 			   AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv),

+	SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
+		     AIC31XX_HPRGAIN, 2, 1, 0),
+	SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
+			 AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
+
+	SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
+			 AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
+};
+
+static const struct snd_kcontrol_new aic31xx_snd_controls[] = {
 	SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1,
 		       adc_fgain_tlv),

@ -286,14 +296,6 @@ static const struct snd_kcontrol_new aic31xx_snd_controls[] = {

 	SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0,
 		       119, 0, mic_pga_tlv),
-
-	SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN,
-		     AIC31XX_HPRGAIN, 2, 1, 0),
-	SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN,
-			 AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv),
-
-	SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL,
-			 AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv),
 };

 static const struct snd_kcontrol_new aic311x_snd_controls[] = {
@ -397,17 +399,28 @@ static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }

-static const struct snd_kcontrol_new left_output_switches[] = {
+static const struct snd_kcontrol_new aic31xx_left_output_switches[] = {
 	SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
 	SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0),
 	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0),
 };

-static const struct snd_kcontrol_new right_output_switches[] = {
+static const struct snd_kcontrol_new aic31xx_right_output_switches[] = {
 	SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
 	SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0),
 };

+static const struct snd_kcontrol_new dac31xx_left_output_switches[] = {
+	SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0),
+	SOC_DAPM_SINGLE("From AIN1", AIC31XX_DACMIXERROUTE, 5, 1, 0),
+	SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new dac31xx_right_output_switches[] = {
+	SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0),
+	SOC_DAPM_SINGLE("From AIN2", AIC31XX_DACMIXERROUTE, 1, 1, 0),
+};
+
 static const struct snd_kcontrol_new p_term_mic1lp =
 	SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum);

@ -457,7 +470,7 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }

-static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget common31xx_dapm_widgets[] = {
 	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),

 	SND_SOC_DAPM_MUX("DAC Left Input",
@ -473,14 +486,7 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
 			   AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event,
 			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),

-	/* Output Mixers */
-	SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
-			   left_output_switches,
-			   ARRAY_SIZE(left_output_switches)),
-	SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
-			   right_output_switches,
-			   ARRAY_SIZE(right_output_switches)),
-
+	/* HP */
 	SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0,
 			    &aic31xx_dapm_hpl_switch),
 	SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0,
@ -494,10 +500,34 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
 			       NULL, 0, aic31xx_dapm_power_event,
 			       SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU),

-	/* ADC */
-	SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
-			   aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
-			   SND_SOC_DAPM_POST_PMD),
+	/* Mic Bias */
+	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
+			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+static const struct snd_soc_dapm_widget dac31xx_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("AIN1"),
+	SND_SOC_DAPM_INPUT("AIN2"),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+			   dac31xx_left_output_switches,
+			   ARRAY_SIZE(dac31xx_left_output_switches)),
+	SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+			   dac31xx_right_output_switches,
+			   ARRAY_SIZE(dac31xx_right_output_switches)),
+};
+
+static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC1LP"),
+	SND_SOC_DAPM_INPUT("MIC1RP"),
+	SND_SOC_DAPM_INPUT("MIC1LM"),

 	/* Input Selection to MIC_PGA */
 	SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0,
@ -507,24 +537,25 @@ static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = {
 	SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0,
 			 &p_term_mic1lm),

+	/* ADC */
+	SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0,
+			   aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU |
+			   SND_SOC_DAPM_POST_PMD),
+
 	SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0,
 			 &m_term_mic1lm),
+
 	/* Enabling & Disabling MIC Gain Ctl */
 	SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA,
 			 7, 1, NULL, 0),

-	/* Mic Bias */
-	SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event,
-			    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
-
-	/* Outputs */
-	SND_SOC_DAPM_OUTPUT("HPL"),
-	SND_SOC_DAPM_OUTPUT("HPR"),
-
-	/* Inputs */
-	SND_SOC_DAPM_INPUT("MIC1LP"),
-	SND_SOC_DAPM_INPUT("MIC1RP"),
-	SND_SOC_DAPM_INPUT("MIC1LM"),
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0,
+			   aic31xx_left_output_switches,
+			   ARRAY_SIZE(aic31xx_left_output_switches)),
+	SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0,
+			   aic31xx_right_output_switches,
+			   ARRAY_SIZE(aic31xx_right_output_switches)),
 };

 static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = {
@ -554,7 +585,7 @@ static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = {
 };

 static const struct snd_soc_dapm_route
-aic31xx_audio_map[] = {
+common31xx_audio_map[] = {
 	/* DAC Input Routing */
 	{"DAC Left Input", "Left Data", "DAC IN"},
 	{"DAC Left Input", "Right Data", "DAC IN"},
@ -565,6 +596,31 @@ aic31xx_audio_map[] = {
 	{"DAC Left", NULL, "DAC Left Input"},
 	{"DAC Right", NULL, "DAC Right Input"},

+	/* HPL path */
+	{"HP Left", "Switch", "Output Left"},
+	{"HPL Driver", NULL, "HP Left"},
+	{"HPL", NULL, "HPL Driver"},
+
+	/* HPR path */
+	{"HP Right", "Switch", "Output Right"},
+	{"HPR Driver", NULL, "HP Right"},
+	{"HPR", NULL, "HPR Driver"},
+};
+
+static const struct snd_soc_dapm_route
+dac31xx_audio_map[] = {
+	/* Left Output */
+	{"Output Left", "From Left DAC", "DAC Left"},
+	{"Output Left", "From AIN1", "AIN1"},
+	{"Output Left", "From AIN2", "AIN2"},
+
+	/* Right Output */
+	{"Output Right", "From Right DAC", "DAC Right"},
+	{"Output Right", "From AIN2", "AIN2"},
+};
+
+static const struct snd_soc_dapm_route
+aic31xx_audio_map[] = {
 	/* Mic input */
 	{"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"},
 	{"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"},
@ -595,16 +651,6 @@ aic31xx_audio_map[] = {
 	/* Right Output */
 	{"Output Right", "From Right DAC", "DAC Right"},
 	{"Output Right", "From MIC1RP", "MIC1RP"},
-
-	/* HPL path */
-	{"HP Left", "Switch", "Output Left"},
-	{"HPL Driver", NULL, "HP Left"},
-	{"HPL", NULL, "HPL Driver"},
-
-	/* HPR path */
-	{"HP Right", "Switch", "Output Right"},
-	{"HPR Driver", NULL, "HP Right"},
-	{"HPR", NULL, "HPR Driver"},
 };

 static const struct snd_soc_dapm_route
@ -633,6 +679,13 @@ static int aic31xx_add_controls(struct snd_soc_codec *codec)
 	int ret = 0;
 	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);

+	if (!(aic31xx->pdata.codec_type & DAC31XX_BIT))
+		ret = snd_soc_add_codec_controls(
+			codec, aic31xx_snd_controls,
+			ARRAY_SIZE(aic31xx_snd_controls));
+	if (ret)
+		return ret;
+
 	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT)
 		ret = snd_soc_add_codec_controls(
 			codec, aic311x_snd_controls,
@ -651,6 +704,30 @@ static int aic31xx_add_widgets(struct snd_soc_codec *codec)
 	struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;

+	if (aic31xx->pdata.codec_type & DAC31XX_BIT) {
+		ret = snd_soc_dapm_new_controls(
+			dapm, dac31xx_dapm_widgets,
+			ARRAY_SIZE(dac31xx_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, dac31xx_audio_map,
+					      ARRAY_SIZE(dac31xx_audio_map));
+		if (ret)
+			return ret;
+	} else {
+		ret = snd_soc_dapm_new_controls(
+			dapm, aic31xx_dapm_widgets,
+			ARRAY_SIZE(aic31xx_dapm_widgets));
+		if (ret)
+			return ret;
+
+		ret = snd_soc_dapm_add_routes(dapm, aic31xx_audio_map,
+					      ARRAY_SIZE(aic31xx_audio_map));
+		if (ret)
+			return ret;
+	}
+
 	if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) {
 		ret = snd_soc_dapm_new_controls(
 			dapm, aic311x_dapm_widgets,
@ -1115,12 +1192,12 @@ static struct snd_soc_codec_driver soc_codec_driver_aic31xx = {
 	.suspend_bias_off	= true,

 	.component_driver = {
-		.controls		= aic31xx_snd_controls,
-		.num_controls		= ARRAY_SIZE(aic31xx_snd_controls),
-		.dapm_widgets		= aic31xx_dapm_widgets,
-		.num_dapm_widgets	= ARRAY_SIZE(aic31xx_dapm_widgets),
-		.dapm_routes		= aic31xx_audio_map,
-		.num_dapm_routes	= ARRAY_SIZE(aic31xx_audio_map),
+		.controls		= common31xx_snd_controls,
+		.num_controls		= ARRAY_SIZE(common31xx_snd_controls),
+		.dapm_widgets		= common31xx_dapm_widgets,
+		.num_dapm_widgets	= ARRAY_SIZE(common31xx_dapm_widgets),
+		.dapm_routes		= common31xx_audio_map,
+		.num_dapm_routes	= ARRAY_SIZE(common31xx_audio_map),
 	},
 };

@ -1131,19 +1208,34 @@ static const struct snd_soc_dai_ops aic31xx_dai_ops = {
 	.digital_mute	= aic31xx_dac_mute,
 };

+static struct snd_soc_dai_driver dac31xx_dai_driver[] = {
+	{
+		.name = "tlv32dac31xx-hifi",
+		.playback = {
+			.stream_name	 = "Playback",
+			.channels_min	 = 2,
+			.channels_max	 = 2,
+			.rates		 = AIC31XX_RATES,
+			.formats	 = AIC31XX_FORMATS,
+		},
+		.ops = &aic31xx_dai_ops,
+		.symmetric_rates = 1,
+	}
+};
+
 static struct snd_soc_dai_driver aic31xx_dai_driver[] = {
 	{
 		.name = "tlv320aic31xx-hifi",
 		.playback = {
 			.stream_name	 = "Playback",
-			.channels_min	 = 1,
+			.channels_min	 = 2,
 			.channels_max	 = 2,
 			.rates		 = AIC31XX_RATES,
 			.formats	 = AIC31XX_FORMATS,
 		},
 		.capture = {
 			.stream_name	 = "Capture",
-			.channels_min	 = 1,
+			.channels_min	 = 2,
 			.channels_max	 = 2,
 			.rates		 = AIC31XX_RATES,
 			.formats	 = AIC31XX_FORMATS,
@ -1261,9 +1353,16 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c,
 	if (ret)
 		return ret;

-	return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx,
-				     aic31xx_dai_driver,
-				     ARRAY_SIZE(aic31xx_dai_driver));
+	if (aic31xx->pdata.codec_type & DAC31XX_BIT)
+		return snd_soc_register_codec(&i2c->dev,
+				&soc_codec_driver_aic31xx,
+				dac31xx_dai_driver,
+				ARRAY_SIZE(dac31xx_dai_driver));
+	else
+		return snd_soc_register_codec(&i2c->dev,
+				&soc_codec_driver_aic31xx,
+				aic31xx_dai_driver,
+				ARRAY_SIZE(aic31xx_dai_driver));
 }

 static int aic31xx_i2c_remove(struct i2c_client *i2c)
@ -1279,6 +1378,7 @@ static const struct i2c_device_id aic31xx_i2c_id[] = {
 	{ "tlv320aic3110", AIC3110 },
 	{ "tlv320aic3120", AIC3120 },
 	{ "tlv320aic3111", AIC3111 },
+	{ "tlv320dac3100", DAC3100 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id);
--- a/sound/soc/codecs/tlv320aic31xx.h
+++ b/sound/soc/codecs/tlv320aic31xx.h
@ -24,12 +24,14 @@

 #define AIC31XX_STEREO_CLASS_D_BIT	0x1
 #define AIC31XX_MINIDSP_BIT		0x2
+#define DAC31XX_BIT			0x4

 enum aic31xx_type {
 	AIC3100	= 0,
 	AIC3110 = AIC31XX_STEREO_CLASS_D_BIT,
 	AIC3120 = AIC31XX_MINIDSP_BIT,
 	AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT),
+	DAC3100 = DAC31XX_BIT,
 };

 struct aic31xx_pdata {
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@ -90,7 +90,6 @@ static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = {

 struct tlv320dac33_priv {
 	struct mutex mutex;
-	struct workqueue_struct *dac33_wq;
 	struct work_struct work;
 	struct snd_soc_codec *codec;
 	struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES];
@ -771,7 +770,7 @@ static irqreturn_t dac33_interrupt_handler(int irq, void *dev)

 	/* Do not schedule the workqueue in Mode7 */
 	if (dac33->fifo_mode != DAC33_FIFO_MODE7)
-		queue_work(dac33->dac33_wq, &dac33->work);
+		schedule_work(&dac33->work);

 	return IRQ_HANDLED;
 }
@ -1127,7 +1126,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (dac33->fifo_mode) {
 			dac33->state = DAC33_PREFILL;
-			queue_work(dac33->dac33_wq, &dac33->work);
+			schedule_work(&dac33->work);
 		}
 		break;
 	case SNDRV_PCM_TRIGGER_STOP:
@ -1135,7 +1134,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (dac33->fifo_mode) {
 			dac33->state = DAC33_FLUSH;
-			queue_work(dac33->dac33_wq, &dac33->work);
+			schedule_work(&dac33->work);
 		}
 		break;
 	default:
@ -1410,14 +1409,6 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 			dac33->irq = -1;
 		}
 		if (dac33->irq != -1) {
-			/* Setup work queue */
-			dac33->dac33_wq =
-				create_singlethread_workqueue("tlv320dac33");
-			if (dac33->dac33_wq == NULL) {
-				free_irq(dac33->irq, codec);
-				return -ENOMEM;
-			}
-
 			INIT_WORK(&dac33->work, dac33_work);
 		}
 	}
@ -1437,7 +1428,7 @@ static int dac33_soc_remove(struct snd_soc_codec *codec)

 	if (dac33->irq >= 0) {
 		free_irq(dac33->irq, dac33->codec);
-		destroy_workqueue(dac33->dac33_wq);
+		flush_work(&dac33->work);
 	}
 	return 0;
 }
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@ -48,9 +48,10 @@
 #define SOC_TPLG_PASS_PCM_DAI		4
 #define SOC_TPLG_PASS_GRAPH		5
 #define SOC_TPLG_PASS_PINS		6
+#define SOC_TPLG_PASS_BE_DAI		7

 #define SOC_TPLG_PASS_START	SOC_TPLG_PASS_MANIFEST
-#define SOC_TPLG_PASS_END	SOC_TPLG_PASS_PINS
+#define SOC_TPLG_PASS_END	SOC_TPLG_PASS_BE_DAI

 struct soc_tplg {
 	const struct firmware *fw;
@ -1475,6 +1476,7 @@ widget:
 	if (widget == NULL) {
 		dev_err(tplg->dev, "ASoC: failed to create widget %s controls\n",
 			w->name);
+		ret = -ENOMEM;
 		goto hdr_err;
 	}

@ -1554,6 +1556,25 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream,
 	stream->rate_min = caps->rate_min;
 	stream->rate_max = caps->rate_max;
 	stream->formats = caps->formats;
+	stream->sig_bits = caps->sig_bits;
+}
+
+static void set_dai_flags(struct snd_soc_dai_driver *dai_drv,
+			  unsigned int flag_mask, unsigned int flags)
+{
+	if (flag_mask & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES)
+		dai_drv->symmetric_rates =
+			flags & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES ? 1 : 0;
+
+	if (flag_mask & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS)
+		dai_drv->symmetric_channels =
+			flags & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS ?
+			1 : 0;
+
+	if (flag_mask & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS)
+		dai_drv->symmetric_samplebits =
+			flags & SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS ?
+			1 : 0;
 }

 static int soc_tplg_dai_create(struct soc_tplg *tplg,
@ -1690,8 +1711,96 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 	return 0;
 }

+/* *
+ * soc_tplg_be_dai_config - Find and configure an existing BE DAI.
+ * @tplg: topology context
+ * @be: topology BE DAI configs.
+ *
+ * The BE dai should already be registered by the platform driver. The
+ * platform driver should specify the BE DAI name and ID for matching.
+ */
+static int soc_tplg_be_dai_config(struct soc_tplg *tplg,
+				  struct snd_soc_tplg_be_dai *be)
+{
+	struct snd_soc_dai_link_component dai_component = {0};
+	struct snd_soc_dai *dai;
+	struct snd_soc_dai_driver *dai_drv;
+	struct snd_soc_pcm_stream *stream;
+	struct snd_soc_tplg_stream_caps *caps;
+	int ret;
+
+	dai_component.dai_name = be->dai_name;
+	dai = snd_soc_find_dai(&dai_component);
+	if (!dai) {
+		dev_err(tplg->dev, "ASoC: BE DAI %s not registered\n",
+			be->dai_name);
+		return -EINVAL;
+	}
+
+	if (be->dai_id != dai->id) {
+		dev_err(tplg->dev, "ASoC: BE DAI %s id mismatch\n",
+			be->dai_name);
+		return -EINVAL;
+	}
+
+	dai_drv = dai->driver;
+	if (!dai_drv)
+		return -EINVAL;
+
+	if (be->playback) {
+		stream = &dai_drv->playback;
+		caps = &be->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
+		set_stream_info(stream, caps);
+	}
+
+	if (be->capture) {
+		stream = &dai_drv->capture;
+		caps = &be->caps[SND_SOC_TPLG_STREAM_CAPTURE];
+		set_stream_info(stream, caps);
+	}
+
+	if (be->flag_mask)
+		set_dai_flags(dai_drv, be->flag_mask, be->flags);
+
+	/* pass control to component driver for optional further init */
+	ret = soc_tplg_dai_load(tplg, dai_drv);
+	if (ret < 0) {
+		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int soc_tplg_be_dai_elems_load(struct soc_tplg *tplg,
+				      struct snd_soc_tplg_hdr *hdr)
+{
+	struct snd_soc_tplg_be_dai *be;
+	int count = hdr->count;
+	int i;
+
+	if (tplg->pass != SOC_TPLG_PASS_BE_DAI)
+		return 0;
+
+	/* config the existing BE DAIs */
+	for (i = 0; i < count; i++) {
+		be = (struct snd_soc_tplg_be_dai *)tplg->pos;
+		if (be->size != sizeof(*be)) {
+			dev_err(tplg->dev, "ASoC: invalid BE DAI size\n");
+			return -EINVAL;
+		}
+
+		soc_tplg_be_dai_config(tplg, be);
+		tplg->pos += (sizeof(*be) + be->priv.size);
+	}
+
+	dev_dbg(tplg->dev, "ASoC: Configure %d BE DAIs\n", count);
+	return 0;
+}
+
+
 static int soc_tplg_manifest_load(struct soc_tplg *tplg,
-	struct snd_soc_tplg_hdr *hdr)
+				  struct snd_soc_tplg_hdr *hdr)
 {
 	struct snd_soc_tplg_manifest *manifest;

@ -1793,6 +1902,8 @@ static int soc_tplg_load_header(struct soc_tplg *tplg,
 		return soc_tplg_dapm_widget_elems_load(tplg, hdr);
 	case SND_SOC_TPLG_TYPE_PCM:
 		return soc_tplg_pcm_elems_load(tplg, hdr);
+	case SND_SOC_TPLG_TYPE_BE_DAI:
+		return soc_tplg_be_dai_elems_load(tplg, hdr);
 	case SND_SOC_TPLG_TYPE_MANIFEST:
 		return soc_tplg_manifest_load(tplg, hdr);
 	default:
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@ -138,3 +138,14 @@ config SND_SOC_TEGRA_RT5677
 	help
 	  Say Y or M here if you want to add support for SoC audio on Tegra
 	  boards using the RT5677 codec, such as Ryu.
+
+config SND_SOC_TEGRA_SGTL5000
+	tristate "SoC Audio support for Tegra boards using a SGTL5000 codec"
+	depends on SND_SOC_TEGRA && I2C && GPIOLIB
+	select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC
+	select SND_SOC_TEGRA30_I2S if ARCH_TEGRA_3x_SOC
+	select SND_SOC_SGTL5000
+	help
+	  Say Y or M here if you want to add support for SoC audio on Tegra
+	  boards using the SGTL5000 codec, such as Apalis T30, Apalis TK1 or
+	  Colibri T30.
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@ -26,6 +26,7 @@ snd-soc-tegra-wm9712-objs := tegra_wm9712.o
 snd-soc-tegra-trimslice-objs := trimslice.o
 snd-soc-tegra-alc5632-objs := tegra_alc5632.o
 snd-soc-tegra-max98090-objs := tegra_max98090.o
+snd-soc-tegra-sgtl5000-objs := tegra_sgtl5000.o

 obj-$(CONFIG_SND_SOC_TEGRA_RT5640) += snd-soc-tegra-rt5640.o
 obj-$(CONFIG_SND_SOC_TEGRA_RT5677) += snd-soc-tegra-rt5677.o
@ -35,3 +36,4 @@ obj-$(CONFIG_SND_SOC_TEGRA_WM9712) += snd-soc-tegra-wm9712.o
 obj-$(CONFIG_SND_SOC_TEGRA_TRIMSLICE) += snd-soc-tegra-trimslice.o
 obj-$(CONFIG_SND_SOC_TEGRA_ALC5632) += snd-soc-tegra-alc5632.o
 obj-$(CONFIG_SND_SOC_TEGRA_MAX98090) += snd-soc-tegra-max98090.o
+obj-$(CONFIG_SND_SOC_TEGRA_SGTL5000) += snd-soc-tegra-sgtl5000.o
--- a/sound/soc/tegra/tegra_rt5640.c
+++ b/sound/soc/tegra/tegra_rt5640.c
@ -1,5 +1,5 @@
 /*
-* tegra_rt5640.c - Tegra machine ASoC driver for boards using WM8903 codec.
+* tegra_rt5640.c - Tegra machine ASoC driver for boards using RT5640 codec.
 *
 * Copyright (c) 2013, NVIDIA CORPORATION.  All rights reserved.
 *
--- a/sound/soc/tegra/tegra_sgtl5000.c
+++ b/sound/soc/tegra/tegra_sgtl5000.c
@ -0,0 +1,212 @@
+/*
+ * tegra_sgtl5000.c - Tegra machine ASoC driver for boards using SGTL5000 codec
+ *
+ * Author: Marcel Ziswiler <marcel@ziswiler.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms and conditions of the GNU General Public License,
+ * version 2, as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope it will be useful, but WITHOUT
+ * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ * FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ * more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program.  If not, see <http://www.gnu.org/licenses/>.
+ *
+ * Based on code copyright/by:
+ *
+ * Copyright (C) 2010-2012 - NVIDIA, Inc.
+ * (c) 2009, 2010 Nvidia Graphics Pvt. Ltd.
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "../codecs/sgtl5000.h"
+
+#include "tegra_asoc_utils.h"
+
+#define DRV_NAME "tegra-snd-sgtl5000"
+
+struct tegra_sgtl5000 {
+	struct tegra_asoc_utils_data util_data;
+};
+
+static int tegra_sgtl5000_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_card *card = rtd->card;
+	struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card);
+	int srate, mclk;
+	int err;
+
+	srate = params_rate(params);
+	switch (srate) {
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+		mclk = 11289600;
+		break;
+	default:
+		mclk = 12288000;
+		break;
+	}
+
+	err = tegra_asoc_utils_set_rate(&machine->util_data, srate, mclk);
+	if (err < 0) {
+		dev_err(card->dev, "Can't configure clocks\n");
+		return err;
+	}
+
+	err = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_SYSCLK, mclk,
+				     SND_SOC_CLOCK_IN);
+	if (err < 0) {
+		dev_err(card->dev, "codec_dai clock not set\n");
+		return err;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops tegra_sgtl5000_ops = {
+	.hw_params = tegra_sgtl5000_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tegra_sgtl5000_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static struct snd_soc_dai_link tegra_sgtl5000_dai = {
+	.name = "sgtl5000",
+	.stream_name = "HiFi",
+	.codec_dai_name = "sgtl5000",
+	.ops = &tegra_sgtl5000_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS,
+};
+
+static struct snd_soc_card snd_soc_tegra_sgtl5000 = {
+	.name = "tegra-sgtl5000",
+	.owner = THIS_MODULE,
+	.dai_link = &tegra_sgtl5000_dai,
+	.num_links = 1,
+	.dapm_widgets = tegra_sgtl5000_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tegra_sgtl5000_dapm_widgets),
+	.fully_routed = true,
+};
+
+static int tegra_sgtl5000_driver_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	struct snd_soc_card *card = &snd_soc_tegra_sgtl5000;
+	struct tegra_sgtl5000 *machine;
+	int ret;
+
+	machine = devm_kzalloc(&pdev->dev, sizeof(struct tegra_sgtl5000),
+			       GFP_KERNEL);
+	if (!machine) {
+		dev_err(&pdev->dev, "Can't allocate tegra_sgtl5000 struct\n");
+		return -ENOMEM;
+	}
+
+	card->dev = &pdev->dev;
+	platform_set_drvdata(pdev, card);
+	snd_soc_card_set_drvdata(card, machine);
+
+	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
+	if (ret)
+		goto err;
+
+	ret = snd_soc_of_parse_audio_routing(card, "nvidia,audio-routing");
+	if (ret)
+		goto err;
+
+	tegra_sgtl5000_dai.codec_of_node = of_parse_phandle(np,
+			"nvidia,audio-codec", 0);
+	if (!tegra_sgtl5000_dai.codec_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,audio-codec' missing or invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_sgtl5000_dai.cpu_of_node = of_parse_phandle(np,
+			"nvidia,i2s-controller", 0);
+	if (!tegra_sgtl5000_dai.cpu_of_node) {
+		dev_err(&pdev->dev,
+			"Property 'nvidia,i2s-controller' missing/invalid\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node;
+
+	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
+	if (ret)
+		goto err;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n",
+			ret);
+		goto err_fini_utils;
+	}
+
+	return 0;
+
+err_fini_utils:
+	tegra_asoc_utils_fini(&machine->util_data);
+err:
+	return ret;
+}
+
+static int tegra_sgtl5000_driver_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct tegra_sgtl5000 *machine = snd_soc_card_get_drvdata(card);
+	int ret;
+
+	ret = snd_soc_unregister_card(card);
+
+	tegra_asoc_utils_fini(&machine->util_data);
+
+	return ret;
+}
+
+static const struct of_device_id tegra_sgtl5000_of_match[] = {
+	{ .compatible = "nvidia,tegra-audio-sgtl5000", },
+	{ /* sentinel */ },
+};
+
+static struct platform_driver tegra_sgtl5000_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = tegra_sgtl5000_of_match,
+	},
+	.probe = tegra_sgtl5000_driver_probe,
+	.remove = tegra_sgtl5000_driver_remove,
+};
+module_platform_driver(tegra_sgtl5000_driver);
+
+MODULE_AUTHOR("Marcel Ziswiler <marcel@ziswiler.com>");
+MODULE_DESCRIPTION("Tegra SGTL5000 machine ASoC driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:" DRV_NAME);
+MODULE_DEVICE_TABLE(of, tegra_sgtl5000_of_match);