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

hifive-unleashed-5.1
Mark Brown 2015-06-22 10:24:31 +01:00
parent dacdd36911 662e8d917f b43fccf4b5 6876c7c884
commit 861fe71725
13 changed files with 1955 additions and 54 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

13
Documentation/devicetree/bindings/sound/mt8173-max98090.txt 100644
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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 100644
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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 100644
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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
View File

@ -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 100644
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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 100644
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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 100644
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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 100644
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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 100644
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 100644
View File

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),