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Kod Konular Değişiklik İstekleri İşlemler Paketler Projeler Sürüm Wiki Aktivite

Merge remote-tracking branch 'asoc/topic/intel' into asoc-next

Kaynağa Gözat
Bu işleme şunda yer alıyor:
Mark Brown
2017-04-30 22:15:41 +09:00
ebeveyn d872f04606 081dc8ab46
işleme 0c2964cb38
33 değiştirilmiş dosya ile 1409 ekleme ve 428 silme
Yama Dosyasını İndir Diff Dosyasını İndir Tüm dosyaları genişlet Tüm dosyaları daralt

4
sound/soc/intel/boards/Makefile
Dosyayı Görüntüle

@@ -10,6 +10,8 @@ snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o
snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o
snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o
snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o
snd-soc-sst-byt-cht-da7213-objs := bytcht_da7213.o
snd-soc-sst-byt-cht-nocodec-objs := bytcht_nocodec.o
snd-soc-skl_rt286-objs := skl_rt286.o
snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
@@ -26,6 +28,8 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o
obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o
obj-$(CONFIG_SND_SOC_INTEL_BYT_CHT_DA7213_MACH) += snd-soc-sst-byt-cht-da7213.o
obj-$(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

5
sound/soc/intel/boards/bdw-rt5677.c
Dosyayı Görüntüle

@@ -193,13 +193,12 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd)
RT5677_CLK_SEL_I2S1_ASRC);
/* Request rt5677 GPIO for headphone amp control */
bdw_rt5677->gpio_hp_en = devm_gpiod_get_index(codec->dev,
"headphone-enable", 0, 0);
bdw_rt5677->gpio_hp_en = devm_gpiod_get(codec->dev, "headphone-enable",
GPIOD_OUT_LOW);
if (IS_ERR(bdw_rt5677->gpio_hp_en)) {
dev_err(codec->dev, "Can't find HP_AMP_SHDN_L gpio\n");
return PTR_ERR(bdw_rt5677->gpio_hp_en);
}
gpiod_direction_output(bdw_rt5677->gpio_hp_en, 0);
/* Create and initialize headphone jack */
if (!snd_soc_card_jack_new(rtd->card, "Headphone Jack",

3
sound/soc/intel/boards/broadwell.c
Dosyayı Görüntüle

@@ -269,9 +269,6 @@ static struct snd_soc_card broadwell_rt286 = {
static int broadwell_audio_probe(struct platform_device *pdev)
{
broadwell_rt286.dev = &pdev->dev;
snd_soc_set_dmi_name(&broadwell_rt286, NULL);
return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286);
}

97
sound/soc/intel/boards/bxt_da7219_max98357a.c
Dosyayı Görüntüle

@@ -55,6 +55,54 @@ enum {
BXT_DPCM_AUDIO_HDMI3_PB,
};
static inline struct snd_soc_dai *bxt_get_codec_dai(struct snd_soc_card *card)
{
struct snd_soc_pcm_runtime *rtd;
list_for_each_entry(rtd, &card->rtd_list, list) {
if (!strncmp(rtd->codec_dai->name, BXT_DIALOG_CODEC_DAI,
strlen(BXT_DIALOG_CODEC_DAI)))
return rtd->codec_dai;
}
return NULL;
}
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 = bxt_get_codec_dai(card);
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_sysclk(codec_dai,
DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN);
if (ret)
dev_err(card->dev, "can't set codec sysclk configuration\n");
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"),
@@ -69,6 +117,8 @@ static const struct snd_soc_dapm_widget broxton_widgets[] = {
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),
};
static const struct snd_soc_dapm_route broxton_map[] = {
@@ -109,6 +159,9 @@ static const struct snd_soc_dapm_route broxton_map[] = {
/* DMIC */
{"dmic01_hifi", NULL, "DMIC01 Rx"},
{"DMIC01 Rx", NULL, "DMIC AIF"},
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
};
static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
@@ -243,49 +296,6 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = {
.startup = bxt_fe_startup,
};
static int broxton_da7219_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;
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai,
DA7219_CLKSRC_MCLK, 19200000, SND_SOC_CLOCK_IN);
if (ret < 0)
dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7219_SYSCLK_PLL_SRM, 0, DA7219_PLL_FREQ_OUT_98304);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static int broxton_da7219_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7219_SYSCLK_MCLK, 0, 0);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static const struct snd_soc_ops broxton_da7219_ops = {
.hw_params = broxton_da7219_hw_params,
.hw_free = broxton_da7219_hw_free,
};
static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
@@ -467,7 +477,6 @@ static struct snd_soc_dai_link broxton_dais[] = {
SND_SOC_DAIFMT_CBS_CFS,
.ignore_pmdown_time = 1,
.be_hw_params_fixup = broxton_ssp_fixup,
.ops = &broxton_da7219_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
},

3
sound/soc/intel/boards/bxt_rt298.c
Dosyayı Görüntüle

@@ -274,12 +274,15 @@ static int bxt_fe_startup(struct snd_pcm_substream *substream)
* 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);

283
sound/soc/intel/boards/bytcht_da7213.c Normal dosya
Dosyayı Görüntüle

@@ -0,0 +1,283 @@
/*
* bytcht-da7213.c - ASoc Machine driver for Intel Baytrail and
* Cherrytrail-based platforms, with Dialog DA7213 codec
*
* Copyright (C) 2017 Intel Corporation
* Author: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* 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/acpi.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <asm/platform_sst_audio.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../../codecs/da7213.h"
#include "../atom/sst-atom-controls.h"
#include "../common/sst-acpi.h"
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Mic"),
SOC_DAPM_PIN_SWITCH("Aux In"),
};
static const struct snd_soc_dapm_widget dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Mic", NULL),
SND_SOC_DAPM_LINE("Aux In", NULL),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"Headphone Jack", NULL, "HPL"},
{"Headphone Jack", NULL, "HPR"},
{"AUXL", NULL, "Aux In"},
{"AUXR", NULL, "Aux In"},
/* Assume Mic1 is linked to Headset and Mic2 to on-board mic */
{"MIC1", NULL, "Headset Mic"},
{"MIC2", NULL, "Mic"},
/* SOC-codec link */
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"Playback", NULL, "ssp2 Tx"},
{"ssp2 Rx", NULL, "Capture"},
};
static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
int ret;
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The DSP will convert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0) {
dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
return 0;
}
static int aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_single(substream->runtime,
SNDRV_PCM_HW_PARAM_RATE, 48000);
}
static int aif1_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;
int ret;
ret = snd_soc_dai_set_sysclk(codec_dai, DA7213_CLKSRC_MCLK,
19200000, SND_SOC_CLOCK_IN);
if (ret < 0)
dev_err(codec_dai->dev, "can't set codec sysclk configuration\n");
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7213_SYSCLK_PLL_SRM, 0, DA7213_PLL_FREQ_OUT_98304000);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to start PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static int aif1_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
ret = snd_soc_dai_set_pll(codec_dai, 0,
DA7213_SYSCLK_MCLK, 0, 0);
if (ret < 0) {
dev_err(codec_dai->dev, "failed to stop PLL: %d\n", ret);
return -EIO;
}
return ret;
}
static const struct snd_soc_ops aif1_ops = {
.startup = aif1_startup,
};
static const struct snd_soc_ops ssp2_ops = {
.hw_params = aif1_hw_params,
.hw_free = aif1_hw_free,
};
static struct snd_soc_dai_link dailink[] = {
[MERR_DPCM_AUDIO] = {
.name = "Audio Port",
.stream_name = "Audio",
.cpu_dai_name = "media-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &aif1_ops,
},
[MERR_DPCM_DEEP_BUFFER] = {
.name = "Deep-Buffer Audio Port",
.stream_name = "Deep-Buffer Audio",
.cpu_dai_name = "deepbuffer-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.ops = &aif1_ops,
},
[MERR_DPCM_COMPR] = {
.name = "Compressed Port",
.stream_name = "Compress",
.cpu_dai_name = "compress-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
},
/* CODEC<->CODEC link */
/* back ends */
{
.name = "SSP2-Codec",
.id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "da7213-hifi",
.codec_name = "i2c-DLGS7213:00",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = codec_fixup,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &ssp2_ops,
},
};
/* SoC card */
static struct snd_soc_card bytcht_da7213_card = {
.name = "bytcht-da7213",
.owner = THIS_MODULE,
.dai_link = dailink,
.num_links = ARRAY_SIZE(dailink),
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.dapm_widgets = dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static char codec_name[16]; /* i2c-<HID>:00 with HID being 8 chars */
static int bytcht_da7213_probe(struct platform_device *pdev)
{
int ret_val = 0;
int i;
struct snd_soc_card *card;
struct sst_acpi_mach *mach;
const char *i2c_name = NULL;
int dai_index = 0;
mach = (&pdev->dev)->platform_data;
card = &bytcht_da7213_card;
card->dev = &pdev->dev;
/* fix index of codec dai */
dai_index = MERR_DPCM_COMPR + 1;
for (i = 0; i < ARRAY_SIZE(dailink); i++) {
if (!strcmp(dailink[i].codec_name, "i2c-DLGS7213:00")) {
dai_index = i;
break;
}
}
/* fixup codec name based on HID */
i2c_name = sst_acpi_find_name_from_hid(mach->id);
if (i2c_name != NULL) {
snprintf(codec_name, sizeof(codec_name),
"%s%s", "i2c-", i2c_name);
dailink[dai_index].codec_name = codec_name;
}
ret_val = devm_snd_soc_register_card(&pdev->dev, card);
if (ret_val) {
dev_err(&pdev->dev,
"snd_soc_register_card failed %d\n", ret_val);
return ret_val;
}
platform_set_drvdata(pdev, card);
return ret_val;
}
static struct platform_driver bytcht_da7213_driver = {
.driver = {
.name = "bytcht_da7213",
},
.probe = bytcht_da7213_probe,
};
module_platform_driver(bytcht_da7213_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail+DA7213 Machine driver");
MODULE_AUTHOR("Pierre-Louis Bossart");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytcht_da7213");

208
sound/soc/intel/boards/bytcht_nocodec.c Normal dosya
Dosyayı Görüntüle

@@ -0,0 +1,208 @@
/*
* bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up
* to make I2S signals observable on the Low-Speed connector. Audio codec
* is not managed by ASoC/DAPM
*
* Copyright (C) 2015-2017 Intel Corp
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; version 2 of the License.
*
* 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/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../atom/sst-atom-controls.h"
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_MIC("Mic", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
};
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Mic"),
SOC_DAPM_PIN_SWITCH("Speaker"),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"ssp2 Rx", NULL, "Mic"},
{"Speaker", NULL, "ssp2 Tx"},
};
static int codec_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 *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
int ret;
/* The DSP will convert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0) {
dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
return 0;
}
static unsigned int rates_48000[] = {
48000,
};
static struct snd_pcm_hw_constraint_list constraints_48000 = {
.count = ARRAY_SIZE(rates_48000),
.list = rates_48000,
};
static int aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_48000);
}
static struct snd_soc_ops aif1_ops = {
.startup = aif1_startup,
};
static struct snd_soc_dai_link dais[] = {
[MERR_DPCM_AUDIO] = {
.name = "Audio Port",
.stream_name = "Audio",
.cpu_dai_name = "media-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.ignore_suspend = 1,
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &aif1_ops,
},
[MERR_DPCM_DEEP_BUFFER] = {
.name = "Deep-Buffer Audio Port",
.stream_name = "Deep-Buffer Audio",
.cpu_dai_name = "deepbuffer-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
.ignore_suspend = 1,
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.ops = &aif1_ops,
},
[MERR_DPCM_COMPR] = {
.name = "Compressed Port",
.stream_name = "Compress",
.cpu_dai_name = "compress-cpu-dai",
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.platform_name = "sst-mfld-platform",
},
/* CODEC<->CODEC link */
/* back ends */
{
.name = "SSP2-LowSpeed Connector",
.id = 1,
.cpu_dai_name = "ssp2-port",
.platform_name = "sst-mfld-platform",
.no_pcm = 1,
.codec_dai_name = "snd-soc-dummy-dai",
.codec_name = "snd-soc-dummy",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = codec_fixup,
.ignore_suspend = 1,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
},
};
/* SoC card */
static struct snd_soc_card bytcht_nocodec_card = {
.name = "bytcht-nocodec",
.owner = THIS_MODULE,
.dai_link = dais,
.num_links = ARRAY_SIZE(dais),
.dapm_widgets = widgets,
.num_dapm_widgets = ARRAY_SIZE(widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.fully_routed = true,
};
static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
/* register the soc card */
bytcht_nocodec_card.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card);
if (ret_val) {
dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
ret_val);
return ret_val;
}
platform_set_drvdata(pdev, &bytcht_nocodec_card);
return ret_val;
}
static struct platform_driver snd_bytcht_nocodec_mc_driver = {
.driver = {
.name = "bytcht_nocodec",
},
.probe = snd_bytcht_nocodec_mc_probe,
};
module_platform_driver(snd_bytcht_nocodec_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver");
MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytcht_nocodec");

109
sound/soc/intel/boards/bytcr_rt5640.c
Dosyayı Görüntüle

@@ -19,6 +19,7 @@
#include <linux/init.h>
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/platform_device.h>
#include <linux/acpi.h>
#include <linux/device.h>
@@ -56,35 +57,88 @@ enum {
struct byt_rt5640_private {
struct clk *mclk;
};
static bool is_bytcr;
static unsigned long byt_rt5640_quirk = BYT_RT5640_MCLK_EN;
static unsigned int quirk_override;
module_param_named(quirk, quirk_override, uint, 0444);
MODULE_PARM_DESC(quirk, "Board-specific quirk override");
static void log_quirks(struct device *dev)
{
if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC1_MAP)
dev_info(dev, "quirk DMIC1_MAP enabled");
if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_DMIC2_MAP)
dev_info(dev, "quirk DMIC2_MAP enabled");
if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN1_MAP)
dev_info(dev, "quirk IN1_MAP enabled");
if (BYT_RT5640_MAP(byt_rt5640_quirk) == BYT_RT5640_IN3_MAP)
dev_info(dev, "quirk IN3_MAP enabled");
if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN)
dev_info(dev, "quirk DMIC enabled");
int map;
bool has_dmic = false;
bool has_mclk = false;
bool has_ssp0 = false;
bool has_ssp0_aif1 = false;
bool has_ssp0_aif2 = false;
bool has_ssp2_aif2 = false;
map = BYT_RT5640_MAP(byt_rt5640_quirk);
switch (map) {
case BYT_RT5640_DMIC1_MAP:
dev_info(dev, "quirk DMIC1_MAP enabled\n");
has_dmic = true;
break;
case BYT_RT5640_DMIC2_MAP:
dev_info(dev, "quirk DMIC2_MAP enabled\n");
has_dmic = true;
break;
case BYT_RT5640_IN1_MAP:
dev_info(dev, "quirk IN1_MAP enabled\n");
break;
case BYT_RT5640_IN3_MAP:
dev_info(dev, "quirk IN3_MAP enabled\n");
break;
default:
dev_err(dev, "quirk map 0x%x is not supported, microphone input will not work\n", map);
break;
}
if (byt_rt5640_quirk & BYT_RT5640_DMIC_EN) {
if (has_dmic)
dev_info(dev, "quirk DMIC enabled\n");
else
dev_err(dev, "quirk DMIC enabled but no DMIC input set, will be ignored\n");
}
if (byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER)
dev_info(dev, "quirk MONO_SPEAKER enabled");
if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC)
dev_info(dev, "quirk DIFF_MIC enabled");
if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2)
dev_info(dev, "quirk SSP2_AIF2 enabled");
if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1)
dev_info(dev, "quirk SSP0_AIF1 enabled");
if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)
dev_info(dev, "quirk SSP0_AIF2 enabled");
if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN)
dev_info(dev, "quirk MCLK_EN enabled");
if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ)
dev_info(dev, "quirk MCLK_25MHZ enabled");
dev_info(dev, "quirk MONO_SPEAKER enabled\n");
if (byt_rt5640_quirk & BYT_RT5640_DIFF_MIC) {
if (!has_dmic)
dev_info(dev, "quirk DIFF_MIC enabled\n");
else
dev_info(dev, "quirk DIFF_MIC enabled but DMIC input selected, will be ignored\n");
}
if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) {
dev_info(dev, "quirk SSP0_AIF1 enabled\n");
has_ssp0 = true;
has_ssp0_aif1 = true;
}
if (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2) {
dev_info(dev, "quirk SSP0_AIF2 enabled\n");
has_ssp0 = true;
has_ssp0_aif2 = true;
}
if (byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) {
dev_info(dev, "quirk SSP2_AIF2 enabled\n");
has_ssp2_aif2 = true;
}
if (is_bytcr && !has_ssp0)
dev_err(dev, "Invalid routing, bytcr detected but no SSP0-based quirk, audio cannot work with SSP2 on bytcr\n");
if (has_ssp0_aif1 && has_ssp0_aif2)
dev_err(dev, "Invalid routing, SSP0 cannot be connected to both AIF1 and AIF2\n");
if (has_ssp0 && has_ssp2_aif2)
dev_err(dev, "Invalid routing, cannot have both SSP0 and SSP2 connected to codec\n");
if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) {
dev_info(dev, "quirk MCLK_EN enabled\n");
has_mclk = true;
}
if (byt_rt5640_quirk & BYT_RT5640_MCLK_25MHZ) {
if (has_mclk)
dev_info(dev, "quirk MCLK_25MHZ enabled\n");
else
dev_err(dev, "quirk MCLK_25MHZ enabled but quirk MCLK not selected, will be ignored\n");
}
}
@@ -128,7 +182,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w,
ret = clk_prepare_enable(priv->mclk);
if (ret < 0) {
dev_err(card->dev,
"could not configure MCLK state");
"could not configure MCLK state\n");
return ret;
}
}
@@ -710,8 +764,8 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
int i;
int dai_index;
struct byt_rt5640_private *priv;
bool is_bytcr = false;
is_bytcr = false;
priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_ATOMIC);
if (!priv)
return -ENOMEM;
@@ -806,6 +860,11 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
/* check quirks before creating card */
dmi_check_system(byt_rt5640_quirk_table);
if (quirk_override) {
dev_info(&pdev->dev, "Overriding quirk 0x%x => 0x%x\n",
(unsigned int)byt_rt5640_quirk, quirk_override);
byt_rt5640_quirk = quirk_override;
}
log_quirks(&pdev->dev);
if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) ||