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Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/ml26124' and 'asoc/topic/omap' into asoc-next

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This commit is contained in:
Mark Brown 2015-06-22 10:24:31 +01:00
commit 861fe71725
13 changed files with 1955 additions and 54 deletions
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13
Documentation/devicetree/bindings/sound/mt8173-max98090.txt Normal file
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@ -0,0 +1,13 @@
MT8173 with MAX98090 CODEC
Required properties:
- compatible : "mediatek,mt8173-max98090"
- mediatek,audio-codec: the phandle of the MAX98090 audio codec
Example:
sound {
compatible = "mediatek,mt8173-max98090";
mediatek,audio-codec = <&max98090>;
};

13
Documentation/devicetree/bindings/sound/mt8173-rt5650-rt5676.txt Normal file
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@ -0,0 +1,13 @@
MT8173 with RT5650 RT5676 CODECS
Required properties:
- compatible : "mediatek,mt8173-rt5650-rt5676"
- mediatek,audio-codec: the phandles of rt5650 and rt5676 codecs
Example:
sound {
compatible = "mediatek,mt8173-rt5650-rt5676";
mediatek,audio-codec = <&rt5650 &rt5676>;
};

45
Documentation/devicetree/bindings/sound/mtk-afe-pcm.txt Normal file
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@ -0,0 +1,45 @@
Mediatek AFE PCM controller
Required properties:
- compatible = "mediatek,mt8173-afe-pcm";
- reg: register location and size
- interrupts: Should contain AFE interrupt
- clock-names: should have these clock names:
"infra_sys_audio_clk",
"top_pdn_audio",
"top_pdn_aud_intbus",
"bck0",
"bck1",
"i2s0_m",
"i2s1_m",
"i2s2_m",
"i2s3_m",
"i2s3_b";
Example:
afe: mt8173-afe-pcm@11220000 {
compatible = "mediatek,mt8173-afe-pcm";
reg = <0 0x11220000 0 0x1000>;
interrupts = <GIC_SPI 134 IRQ_TYPE_EDGE_FALLING>;
clocks = <&infracfg INFRA_AUDIO>,
<&topckgen TOP_AUDIO_SEL>,
<&topckgen TOP_AUD_INTBUS_SEL>,
<&topckgen TOP_APLL1_DIV0>,
<&topckgen TOP_APLL2_DIV0>,
<&topckgen TOP_I2S0_M_CK_SEL>,
<&topckgen TOP_I2S1_M_CK_SEL>,
<&topckgen TOP_I2S2_M_CK_SEL>,
<&topckgen TOP_I2S3_M_CK_SEL>,
<&topckgen TOP_I2S3_B_CK_SEL>;
clock-names = "infra_sys_audio_clk",
"top_pdn_audio",
"top_pdn_aud_intbus",
"bck0",
"bck1",
"i2s0_m",
"i2s1_m",
"i2s2_m",
"i2s3_m",
"i2s3_b";
};

1
sound/soc/Kconfig
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@ -45,6 +45,7 @@ source "sound/soc/nuc900/Kconfig"
source "sound/soc/omap/Kconfig"
source "sound/soc/kirkwood/Kconfig"
source "sound/soc/intel/Kconfig"
source "sound/soc/mediatek/Kconfig"
source "sound/soc/mxs/Kconfig"
source "sound/soc/pxa/Kconfig"
source "sound/soc/qcom/Kconfig"

1
sound/soc/Makefile
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@ -24,6 +24,7 @@ obj-$(CONFIG_SND_SOC) += dwc/
obj-$(CONFIG_SND_SOC) += fsl/
obj-$(CONFIG_SND_SOC) += jz4740/
obj-$(CONFIG_SND_SOC) += intel/
obj-$(CONFIG_SND_SOC) += mediatek/
obj-$(CONFIG_SND_SOC) += mxs/
obj-$(CONFIG_SND_SOC) += nuc900/
obj-$(CONFIG_SND_SOC) += omap/

58
sound/soc/codecs/ml26124.c
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@ -341,6 +341,7 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
struct snd_soc_codec *codec = dai->codec;
struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec);
int i = get_coeff(priv->mclk, params_rate(hw_params));
int srate;
if (i < 0)
return i;
@ -370,53 +371,16 @@ static int ml26124_hw_params(struct snd_pcm_substream *substream,
BIT(0) | BIT(1), 0);
}
switch (params_rate(hw_params)) {
case 16000:
snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
get_srate(params_rate(hw_params)));
snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
coeff_div[i].pllnl);
snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
coeff_div[i].pllnh);
snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
coeff_div[i].pllml);
snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
coeff_div[i].pllmh);
snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
coeff_div[i].plldiv);
break;
case 32000:
snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
get_srate(params_rate(hw_params)));
snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
coeff_div[i].pllnl);
snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
coeff_div[i].pllnh);
snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
coeff_div[i].pllml);
snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
coeff_div[i].pllmh);
snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
coeff_div[i].plldiv);
break;
case 48000:
snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf,
get_srate(params_rate(hw_params)));
snd_soc_update_bits(codec, ML26124_PLLNL, 0xff,
coeff_div[i].pllnl);
snd_soc_update_bits(codec, ML26124_PLLNH, 0x1,
coeff_div[i].pllnh);
snd_soc_update_bits(codec, ML26124_PLLML, 0xff,
coeff_div[i].pllml);
snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f,
coeff_div[i].pllmh);
snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f,
coeff_div[i].plldiv);
break;
default:
pr_err("%s:this rate is no support for ml26124\n", __func__);
return -EINVAL;
}
srate = get_srate(params_rate(hw_params));
if (srate < 0)
return srate;
snd_soc_update_bits(codec, ML26124_SMPLING_RATE, 0xf, srate);
snd_soc_update_bits(codec, ML26124_PLLNL, 0xff, coeff_div[i].pllnl);
snd_soc_update_bits(codec, ML26124_PLLNH, 0x1, coeff_div[i].pllnh);
snd_soc_update_bits(codec, ML26124_PLLML, 0xff, coeff_div[i].pllml);
snd_soc_update_bits(codec, ML26124_PLLMH, 0x3f, coeff_div[i].pllmh);
snd_soc_update_bits(codec, ML26124_PLLDIV, 0x1f, coeff_div[i].plldiv);
return 0;
}

30
sound/soc/mediatek/Kconfig Normal file
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@ -0,0 +1,30 @@
config SND_SOC_MEDIATEK
tristate "ASoC support for Mediatek chip"
depends on ARCH_MEDIATEK
help
This adds ASoC platform driver support for Mediatek chip
that can be used with other codecs.
Select Y if you have such device.
Ex: MT8173
config SND_SOC_MT8173_MAX98090
tristate "ASoC Audio driver for MT8173 with MAX98090 codec"
depends on SND_SOC_MEDIATEK
select SND_SOC_MAX98090
help
This adds ASoC driver for Mediatek MT8173 boards
with the MAX98090 audio codec.
Select Y if you have such device.
If unsure select "N".
config SND_SOC_MT8173_RT5650_RT5676
tristate "ASoC Audio driver for MT8173 with RT5650 RT5676 codecs"
depends on SND_SOC_MEDIATEK
select SND_SOC_RT5645
select SND_SOC_RT5677
help
This adds ASoC driver for Mediatek MT8173 boards
with the RT5650 and RT5676 codecs.
Select Y if you have such device.
If unsure select "N".

5
sound/soc/mediatek/Makefile Normal file
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@ -0,0 +1,5 @@
# MTK Platform Support
obj-$(CONFIG_SND_SOC_MEDIATEK) += mtk-afe-pcm.o
# Machine support
obj-$(CONFIG_SND_SOC_MT8173_MAX98090) += mt8173-max98090.o
obj-$(CONFIG_SND_SOC_MT8173_RT5650_RT5676) += mt8173-rt5650-rt5676.o

213
sound/soc/mediatek/mt8173-max98090.c Normal file
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@ -0,0 +1,213 @@
/*
* mt8173-max98090.c -- MT8173 MAX98090 ALSA SoC machine driver
*
* Copyright (c) 2015 MediaTek Inc.
* Author: Koro Chen <koro.chen@mediatek.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that 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.
*/
#include <linux/module.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <linux/gpio.h>
#include "../codecs/max98090.h"
static struct snd_soc_jack mt8173_max98090_jack;
static struct snd_soc_jack_pin mt8173_max98090_jack_pins[] = {
{
.pin = "Headphone",
.mask = SND_JACK_HEADPHONE,
},
{
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
},
};
static const struct snd_soc_dapm_widget mt8173_max98090_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
static const struct snd_soc_dapm_route mt8173_max98090_routes[] = {
{"Speaker", NULL, "SPKL"},
{"Speaker", NULL, "SPKR"},
{"DMICL", NULL, "Int Mic"},
{"Headphone", NULL, "HPL"},
{"Headphone", NULL, "HPR"},
{"Headset Mic", NULL, "MICBIAS"},
{"IN34", NULL, "Headset Mic"},
};
static const struct snd_kcontrol_new mt8173_max98090_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
SOC_DAPM_PIN_SWITCH("Int Mic"),
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
static int mt8173_max98090_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;
return snd_soc_dai_set_sysclk(codec_dai, 0, params_rate(params) * 256,
SND_SOC_CLOCK_IN);
}
static struct snd_soc_ops mt8173_max98090_ops = {
.hw_params = mt8173_max98090_hw_params,
};
static int mt8173_max98090_init(struct snd_soc_pcm_runtime *runtime)
{
int ret;
struct snd_soc_card *card = runtime->card;
struct snd_soc_codec *codec = runtime->codec;
/* enable jack detection */
ret = snd_soc_card_jack_new(card, "Headphone", SND_JACK_HEADPHONE,
&mt8173_max98090_jack, NULL, 0);
if (ret) {
dev_err(card->dev, "Can't snd_soc_jack_new %d\n", ret);
return ret;
}
ret = snd_soc_jack_add_pins(&mt8173_max98090_jack,
ARRAY_SIZE(mt8173_max98090_jack_pins),
mt8173_max98090_jack_pins);
if (ret) {
dev_err(card->dev, "Can't snd_soc_jack_add_pins %d\n", ret);
return ret;
}
return max98090_mic_detect(codec, &mt8173_max98090_jack);
}
/* Digital audio interface glue - connects codec <---> CPU */
static struct snd_soc_dai_link mt8173_max98090_dais[] = {
/* Front End DAI links */
{
.name = "MAX98090 Playback",
.stream_name = "MAX98090 Playback",
.cpu_dai_name = "DL1",
.platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dynamic = 1,
.dpcm_playback = 1,
},
{
.name = "MAX98090 Capture",
.stream_name = "MAX98090 Capture",
.cpu_dai_name = "VUL",
.platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dynamic = 1,
.dpcm_capture = 1,
},
/* Back End DAI links */
{
.name = "Codec",
.cpu_dai_name = "I2S",
.platform_name = "11220000.mt8173-afe-pcm",
.no_pcm = 1,
.codec_dai_name = "HiFi",
.init = mt8173_max98090_init,
.ops = &mt8173_max98090_ops,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.dpcm_playback = 1,
.dpcm_capture = 1,
},
};
static struct snd_soc_card mt8173_max98090_card = {
.name = "mt8173-max98090",
.dai_link = mt8173_max98090_dais,
.num_links = ARRAY_SIZE(mt8173_max98090_dais),
.controls = mt8173_max98090_controls,
.num_controls = ARRAY_SIZE(mt8173_max98090_controls),
.dapm_widgets = mt8173_max98090_widgets,
.num_dapm_widgets = ARRAY_SIZE(mt8173_max98090_widgets),
.dapm_routes = mt8173_max98090_routes,
.num_dapm_routes = ARRAY_SIZE(mt8173_max98090_routes),
};
static int mt8173_max98090_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_max98090_card;
struct device_node *codec_node;
int ret, i;
codec_node = of_parse_phandle(pdev->dev.of_node,
"mediatek,audio-codec", 0);
if (!codec_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
for (i = 0; i < card->num_links; i++) {
if (mt8173_max98090_dais[i].codec_name)
continue;
mt8173_max98090_dais[i].codec_of_node = codec_node;
}
card->dev = &pdev->dev;
ret = snd_soc_register_card(card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
static int mt8173_max98090_dev_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
snd_soc_unregister_card(card);
return 0;
}
static const struct of_device_id mt8173_max98090_dt_match[] = {
{ .compatible = "mediatek,mt8173-max98090", },
{ }
};
MODULE_DEVICE_TABLE(of, mt8173_max98090_dt_match);
static struct platform_driver mt8173_max98090_driver = {
.driver = {
.name = "mt8173-max98090",
.owner = THIS_MODULE,
.of_match_table = mt8173_max98090_dt_match,
#ifdef CONFIG_PM
.pm = &snd_soc_pm_ops,
#endif
},
.probe = mt8173_max98090_dev_probe,
.remove = mt8173_max98090_dev_remove,
};
module_platform_driver(mt8173_max98090_driver);
/* Module information */
MODULE_DESCRIPTION("MT8173 MAX98090 ALSA SoC machine driver");
MODULE_AUTHOR("Koro Chen <koro.chen@mediatek.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:mt8173-max98090");

278
sound/soc/mediatek/mt8173-rt5650-rt5676.c Normal file
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@ -0,0 +1,278 @@
/*
* mt8173-rt5650-rt5676.c -- MT8173 machine driver with RT5650/5676 codecs
*
* Copyright (c) 2015 MediaTek Inc.
* Author: Koro Chen <koro.chen@mediatek.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that 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.
*/
#include <linux/module.h>
#include <linux/gpio.h>
#include <linux/of_gpio.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include "../codecs/rt5645.h"
#include "../codecs/rt5677.h"
#define MCLK_FOR_CODECS 12288000
static const struct snd_soc_dapm_widget mt8173_rt5650_rt5676_widgets[] = {
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
static const struct snd_soc_dapm_route mt8173_rt5650_rt5676_routes[] = {
{"Speaker", NULL, "SPOL"},
{"Speaker", NULL, "SPOR"},
{"Speaker", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650 */
{"Sub DMIC L1", NULL, "Int Mic"}, /* DMIC from 5676 */
{"Sub DMIC R1", NULL, "Int Mic"},
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"Headphone", NULL, "Sub AIF2TX"}, /* IF2 ADC to 5650 */
{"Headset Mic", NULL, "micbias1"},
{"Headset Mic", NULL, "micbias2"},
{"IN1P", NULL, "Headset Mic"},
{"IN1N", NULL, "Headset Mic"},
{"Sub AIF2RX", NULL, "Headset Mic"}, /* IF2 DAC from 5650 */
};
static const struct snd_kcontrol_new mt8173_rt5650_rt5676_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
SOC_DAPM_PIN_SWITCH("Int Mic"),
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
};
static int mt8173_rt5650_rt5676_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
int i, ret;
for (i = 0; i < rtd->num_codecs; i++) {
struct snd_soc_dai *codec_dai = rtd->codec_dais[i];
/* pll from mclk 12.288M */
ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS,
params_rate(params) * 512);
if (ret)
return ret;
/* sysclk from pll */
ret = snd_soc_dai_set_sysclk(codec_dai, 1,
params_rate(params) * 512,
SND_SOC_CLOCK_IN);
if (ret)
return ret;
}
return 0;
}
static struct snd_soc_ops mt8173_rt5650_rt5676_ops = {
.hw_params = mt8173_rt5650_rt5676_hw_params,
};
static struct snd_soc_jack mt8173_rt5650_rt5676_jack;
static int mt8173_rt5650_rt5676_init(struct snd_soc_pcm_runtime *runtime)
{
struct snd_soc_card *card = runtime->card;
struct snd_soc_codec *codec = runtime->codec_dais[0]->codec;
struct snd_soc_codec *codec_sub = runtime->codec_dais[1]->codec;
int ret;
rt5645_sel_asrc_clk_src(codec,
RT5645_DA_STEREO_FILTER |
RT5645_AD_STEREO_FILTER,
RT5645_CLK_SEL_I2S1_ASRC);
rt5677_sel_asrc_clk_src(codec_sub,
RT5677_DA_STEREO_FILTER |
RT5677_AD_STEREO1_FILTER,
RT5677_CLK_SEL_I2S1_ASRC);
rt5677_sel_asrc_clk_src(codec_sub,
RT5677_AD_STEREO2_FILTER |
RT5677_I2S2_SOURCE,
RT5677_CLK_SEL_I2S2_ASRC);
/* enable jack detection */
ret = snd_soc_card_jack_new(card, "Headset Jack",
SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
SND_JACK_BTN_0 | SND_JACK_BTN_1 |
SND_JACK_BTN_2 | SND_JACK_BTN_3,
&mt8173_rt5650_rt5676_jack, NULL, 0);
if (ret) {
dev_err(card->dev, "Can't new Headset Jack %d\n", ret);
return ret;
}
return rt5645_set_jack_detect(codec,
&mt8173_rt5650_rt5676_jack,
&mt8173_rt5650_rt5676_jack,
&mt8173_rt5650_rt5676_jack);
}
static struct snd_soc_dai_link_component mt8173_rt5650_rt5676_codecs[] = {
{
.dai_name = "rt5645-aif1",
},
{
.dai_name = "rt5677-aif1",
},
};
/* Digital audio interface glue - connects codec <---> CPU */
static struct snd_soc_dai_link mt8173_rt5650_rt5676_dais[] = {
/* Front End DAI links */
{
.name = "rt5650_rt5676 Playback",
.stream_name = "rt5650_rt5676 Playback",
.cpu_dai_name = "DL1",
.platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dynamic = 1,
.dpcm_playback = 1,
},
{
.name = "rt5650_rt5676 Capture",
.stream_name = "rt5650_rt5676 Capture",
.cpu_dai_name = "VUL",
.platform_name = "11220000.mt8173-afe-pcm",
.codec_name = "snd-soc-dummy",
.codec_dai_name = "snd-soc-dummy-dai",
.trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST},
.dynamic = 1,
.dpcm_capture = 1,
},
/* Back End DAI links */
{
.name = "Codec",
.cpu_dai_name = "I2S",
.platform_name = "11220000.mt8173-afe-pcm",
.no_pcm = 1,
.codecs = mt8173_rt5650_rt5676_codecs,
.num_codecs = 2,
.init = mt8173_rt5650_rt5676_init,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &mt8173_rt5650_rt5676_ops,
.ignore_pmdown_time = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
},
{ /* rt5676 <-> rt5650 intercodec link: Sets rt5676 I2S2 as master */
.name = "rt5650_rt5676 intercodec",
.stream_name = "rt5650_rt5676 intercodec",
.cpu_dai_name = "snd-soc-dummy-dai",
.platform_name = "snd-soc-dummy",
.no_pcm = 1,
.codec_dai_name = "rt5677-aif2",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
},
};
static struct snd_soc_codec_conf mt8173_rt5650_rt5676_codec_conf[] = {
{
.name_prefix = "Sub",
},
};
static struct snd_soc_card mt8173_rt5650_rt5676_card = {
.name = "mtk-rt5650-rt5676",
.dai_link = mt8173_rt5650_rt5676_dais,
.num_links = ARRAY_SIZE(mt8173_rt5650_rt5676_dais),
.codec_conf = mt8173_rt5650_rt5676_codec_conf,
.num_configs = ARRAY_SIZE(mt8173_rt5650_rt5676_codec_conf),
.controls = mt8173_rt5650_rt5676_controls,
.num_controls = ARRAY_SIZE(mt8173_rt5650_rt5676_controls),
.dapm_widgets = mt8173_rt5650_rt5676_widgets,
.num_dapm_widgets = ARRAY_SIZE(mt8173_rt5650_rt5676_widgets),
.dapm_routes = mt8173_rt5650_rt5676_routes,
.num_dapm_routes = ARRAY_SIZE(mt8173_rt5650_rt5676_routes),
};
static int mt8173_rt5650_rt5676_dev_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &mt8173_rt5650_rt5676_card;
int ret;
mt8173_rt5650_rt5676_codecs[0].of_node =
of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 0);
if (!mt8173_rt5650_rt5676_codecs[0].of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
mt8173_rt5650_rt5676_codecs[1].of_node =
of_parse_phandle(pdev->dev.of_node, "mediatek,audio-codec", 1);
if (!mt8173_rt5650_rt5676_codecs[1].of_node) {
dev_err(&pdev->dev,
"Property 'audio-codec' missing or invalid\n");
return -EINVAL;
}
mt8173_rt5650_rt5676_codec_conf[0].of_node =
mt8173_rt5650_rt5676_codecs[1].of_node;
mt8173_rt5650_rt5676_dais[3].codec_of_node =
mt8173_rt5650_rt5676_codecs[1].of_node;
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
ret = snd_soc_register_card(card);
if (ret)
dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n",
__func__, ret);
return ret;
}
static int mt8173_rt5650_rt5676_dev_remove(struct platform_device *pdev)
{
struct snd_soc_card *card = platform_get_drvdata(pdev);
snd_soc_unregister_card(card);
return 0;
}
static const struct of_device_id mt8173_rt5650_rt5676_dt_match[] = {
{ .compatible = "mediatek,mt8173-rt5650-rt5676", },
{ }
};
MODULE_DEVICE_TABLE(of, mt8173_rt5650_rt5676_dt_match);
static struct platform_driver mt8173_rt5650_rt5676_driver = {
.driver = {
.name = "mtk-rt5650-rt5676",
.owner = THIS_MODULE,
.of_match_table = mt8173_rt5650_rt5676_dt_match,
#ifdef CONFIG_PM
.pm = &snd_soc_pm_ops,
#endif
},
.probe = mt8173_rt5650_rt5676_dev_probe,
.remove = mt8173_rt5650_rt5676_dev_remove,
};
module_platform_driver(mt8173_rt5650_rt5676_driver);
/* Module information */
MODULE_DESCRIPTION("MT8173 RT5650 and RT5676 SoC machine driver");
MODULE_AUTHOR("Koro Chen <koro.chen@mediatek.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:mtk-rt5650-rt5676");

109
sound/soc/mediatek/mtk-afe-common.h Normal file
View File

@ -0,0 +1,109 @@
/*
* mtk_afe_common.h -- Mediatek audio driver common definitions
*
* Copyright (c) 2015 MediaTek Inc.
* Author: Koro Chen <koro.chen@mediatek.com>
* Sascha Hauer <s.hauer@pengutronix.de>
* Hidalgo Huang <hidalgo.huang@mediatek.com>
* Ir Lian <ir.lian@mediatek.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 and
* only version 2 as published by the Free Software Foundation.
*
* This program is distributed in the hope that 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.
*/
#ifndef _MTK_AFE_COMMON_H_
#define _MTK_AFE_COMMON_H_
#include <linux/clk.h>
#include <linux/regmap.h>
enum {
MTK_AFE_MEMIF_DL1,
MTK_AFE_MEMIF_DL2,
MTK_AFE_MEMIF_VUL,
MTK_AFE_MEMIF_DAI,
MTK_AFE_MEMIF_AWB,
MTK_AFE_MEMIF_MOD_DAI,
MTK_AFE_MEMIF_HDMI,
MTK_AFE_MEMIF_NUM,
MTK_AFE_IO_MOD_PCM1 = MTK_AFE_MEMIF_NUM,
MTK_AFE_IO_MOD_PCM2,
MTK_AFE_IO_PMIC,
MTK_AFE_IO_I2S,
MTK_AFE_IO_2ND_I2S,
MTK_AFE_IO_HW_GAIN1,
MTK_AFE_IO_HW_GAIN2,
MTK_AFE_IO_MRG_O,
MTK_AFE_IO_MRG_I,
MTK_AFE_IO_DAIBT,
MTK_AFE_IO_HDMI,
};
enum {
MTK_AFE_IRQ_1,
MTK_AFE_IRQ_2,
MTK_AFE_IRQ_3,
MTK_AFE_IRQ_4,
MTK_AFE_IRQ_5,
MTK_AFE_IRQ_6,
MTK_AFE_IRQ_7,
MTK_AFE_IRQ_8,
MTK_AFE_IRQ_NUM,
};
enum {
MTK_CLK_INFRASYS_AUD,
MTK_CLK_TOP_PDN_AUD,
MTK_CLK_TOP_PDN_AUD_BUS,
MTK_CLK_I2S0_M,
MTK_CLK_I2S1_M,
MTK_CLK_I2S2_M,
MTK_CLK_I2S3_M,
MTK_CLK_I2S3_B,
MTK_CLK_BCK0,
MTK_CLK_BCK1,
MTK_CLK_NUM
};
struct mtk_afe;
struct snd_pcm_substream;
struct mtk_afe_memif_data {
int id;
const char *name;
int reg_ofs_base;
int reg_ofs_cur;
int fs_shift;
int mono_shift;
int enable_shift;
int irq_reg_cnt;
int irq_cnt_shift;
int irq_en_shift;
int irq_fs_shift;
int irq_clr_shift;
};
struct mtk_afe_memif {
unsigned int phys_buf_addr;
int buffer_size;
unsigned int hw_ptr; /* Previous IRQ's HW ptr */
struct snd_pcm_substream *substream;
const struct mtk_afe_memif_data *data;
const struct mtk_afe_irq_data *irqdata;
};
struct mtk_afe {
/* address for ioremap audio hardware register */
void __iomem *base_addr;
struct device *dev;
struct regmap *regmap;
struct mtk_afe_memif memif[MTK_AFE_MEMIF_NUM];
struct clk *clocks[MTK_CLK_NUM];
};
#endif

1233
sound/soc/mediatek/mtk-afe-pcm.c Normal file
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File diff suppressed because it is too large Load Diff

10
sound/soc/omap/rx51.c
View File

@ -245,6 +245,8 @@ static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
static const struct snd_soc_dapm_route audio_map[] = {
{"Ext Spk", NULL, "HPLOUT"},
{"Ext Spk", NULL, "HPROUT"},
{"Ext Spk", NULL, "HPLCOM"},
{"Ext Spk", NULL, "HPRCOM"},
{"Headphone Jack", NULL, "LLOUT"},
{"Headphone Jack", NULL, "RLOUT"},
{"FM Transmitter", NULL, "LLOUT"},
@ -288,15 +290,8 @@ static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_card *card = rtd->card;
struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Set up NC codec pins */
snd_soc_dapm_nc_pin(dapm, "MIC3L");
snd_soc_dapm_nc_pin(dapm, "MIC3R");
snd_soc_dapm_nc_pin(dapm, "LINE1R");
err = tpa6130a2_add_controls(codec);
if (err < 0) {
dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
@ -383,6 +378,7 @@ static struct snd_soc_card rx51_sound_card = {
.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
.codec_conf = rx51_codec_conf,
.num_configs = ARRAY_SIZE(rx51_codec_conf),
.fully_routed = true,
.controls = aic34_rx51_controls,
.num_controls = ARRAY_SIZE(aic34_rx51_controls),