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Merge branch 'asoc-5.6' into asoc-next

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Mark Brown
2020-01-23 12:36:45 +00:00
父節點 20230620b4 d8e2e0d249
當前提交 a7196caf83
共有 256 個檔案被更改，包括 23959 行新增 和 4063 行删除
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51
sound/soc/intel/boards/Kconfig
查看文件

@@ -13,6 +13,19 @@ menuconfig SND_SOC_INTEL_MACH
if SND_SOC_INTEL_MACH
config SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES
bool "Use more user friendly long card names"
help
Some drivers report the I/O configuration to userspace through the
soundcard's long card name in the control user space AP. An unfortunate
side effect is that this long name may also be used by the GUI,
confusing users with information they don't need.
This option prevents the long name from being modified, and the I/O
configuration will be provided through a different component interface.
Select Y if userspace like UCM (Use Case Manager) uses the component
interface.
If unsure select N.
if SND_SOC_INTEL_HASWELL
config SND_SOC_INTEL_HASWELL_MACH
@@ -31,12 +44,27 @@ endif ## SND_SOC_INTEL_HASWELL
if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL
config SND_SOC_INTEL_BDW_RT5650_MACH
tristate "Broadwell with RT5650 codec"
depends on I2C
depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST
depends on X86_INTEL_LPSS || COMPILE_TEST
select SND_SOC_RT5645
help
This adds the ASoC machine driver for Intel Broadwell platforms with
the RT5650 codec.
Say Y if you have such a device.
If unsure select "N".
config SND_SOC_INTEL_BDW_RT5677_MACH
tristate "Broadwell with RT5677 codec"
depends on I2C
depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST
depends on GPIOLIB || COMPILE_TEST
depends on X86_INTEL_LPSS || COMPILE_TEST
depends on SPI_MASTER
select SPI_PXA2XX
select SND_SOC_RT5677_SPI
select SND_SOC_RT5677
help
This adds support for Intel Broadwell platform based boards with
@@ -261,6 +289,7 @@ config SND_SOC_INTEL_DA7219_MAX98357A_GENERIC
select SND_SOC_DA7219
select SND_SOC_MAX98357A
select SND_SOC_DMIC
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
config SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON
@@ -387,6 +416,7 @@ config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH
select SND_SOC_RT5682
select SND_SOC_MAX98357A
select SND_SOC_DMIC
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for Geminilake platforms
@@ -400,6 +430,7 @@ if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC
config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH
tristate "SKL/KBL/BXT/APL with HDA Codecs"
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
select SND_SOC_DMIC
# SND_SOC_HDAC_HDA is already selected
@@ -419,6 +450,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH
(SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST))
select SND_SOC_RT5682
select SND_SOC_DMIC
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
select SND_SOC_HDAC_HDMI
help
This adds support for ASoC machine driver for SOF platforms
@@ -448,6 +480,7 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
select SND_SOC_RT5682
select SND_SOC_DMIC
select SND_SOC_HDAC_HDMI
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platform with
RT1011 + RT5682 I2S codec.
@@ -456,4 +489,22 @@ config SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH
endif ## SND_SOC_SOF_COMETLAKE_LP && SND_SOC_SOF_HDA_LINK
if SND_SOC_SOF_JASPERLAKE
config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH
tristate "SOF with DA7219 and MAX98373 in I2S Mode"
depends on I2C && ACPI
depends on MFD_INTEL_LPSS || COMPILE_TEST
select SND_SOC_DA7219
select SND_SOC_MAX98373
select SND_SOC_DMIC
select SND_HDA_CODEC_HDMI if SND_SOC_SOF_HDA_AUDIO_CODEC
help
This adds support for ASoC machine driver for SOF platforms
with DA7219 + MAX98373 I2S audio codec.
Say Y if you have such a device.
If unsure select "N".
endif ## SND_SOC_SOF_JASPERLAKE
endif ## SND_SOC_INTEL_MACH

5
sound/soc/intel/boards/Makefile
查看文件

@@ -2,6 +2,7 @@
snd-soc-sst-haswell-objs := haswell.o
snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o
snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o
snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o
snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o
snd-soc-sst-broadwell-objs := broadwell.o
snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o hda_dsp_common.o
@@ -28,6 +29,7 @@ snd-soc-skl_rt286-objs := skl_rt286.o
snd-soc-skl_hda_dsp-objs := skl_hda_dsp_generic.o skl_hda_dsp_common.o hda_dsp_common.o
snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o
snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o
snd-soc-sof_da7219_max98373-objs := sof_da7219_max98373.o hda_dsp_common.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_RT5682_MACH) += snd-soc-sof_rt5682.o
obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o
@@ -37,6 +39,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_COMMON) += snd-soc-sst-bxt-da721
obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o
obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o
obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o
@@ -58,3 +61,5 @@ obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o
obj-$(CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH) += snd-soc-skl_hda_dsp.o
obj-$(CONFIG_SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH) += snd-soc-sof_da7219_max98373.o

327
sound/soc/intel/boards/bdw-rt5650.c 一般檔案
查看文件

@@ -0,0 +1,327 @@
// SPDX-License-Identifier: GPL-2.0-only
/*
* ASoC machine driver for Intel Broadwell platforms with RT5650 codec
*
* Copyright 2019, The Chromium OS Authors. All rights reserved.
*/
#include <linux/delay.h>
#include <linux/gpio/consumer.h>
#include <linux/module.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-acpi.h>
#include "../common/sst-dsp.h"
#include "../haswell/sst-haswell-ipc.h"
#include "../../codecs/rt5645.h"
struct bdw_rt5650_priv {
struct gpio_desc *gpio_hp_en;
struct snd_soc_component *component;
};
static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = {
SND_SOC_DAPM_HP("Headphone", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("DMIC Pair1", NULL),
SND_SOC_DAPM_MIC("DMIC Pair2", NULL),
};
static const struct snd_soc_dapm_route bdw_rt5650_map[] = {
/* Speakers */
{"Speaker", NULL, "SPOL"},
{"Speaker", NULL, "SPOR"},
/* Headset jack connectors */
{"Headphone", NULL, "HPOL"},
{"Headphone", NULL, "HPOR"},
{"IN1P", NULL, "Headset Mic"},
{"IN1N", NULL, "Headset Mic"},
/* Digital MICs
* DMIC Pair1 are the two DMICs connected on the DMICN1 connector.
* DMIC Pair2 are the two DMICs connected on the DMICN2 connector.
* Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2
* are on the right side.
*/
{"DMIC L1", NULL, "DMIC Pair1"},
{"DMIC R1", NULL, "DMIC Pair1"},
{"DMIC L2", NULL, "DMIC Pair2"},
{"DMIC R2", NULL, "DMIC Pair2"},
/* CODEC BE connections */
{"SSP0 CODEC IN", NULL, "AIF1 Capture"},
{"AIF1 Playback", NULL, "SSP0 CODEC OUT"},
};
static const struct snd_kcontrol_new bdw_rt5650_controls[] = {
SOC_DAPM_PIN_SWITCH("Speaker"),
SOC_DAPM_PIN_SWITCH("Headphone"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("DMIC Pair1"),
SOC_DAPM_PIN_SWITCH("DMIC Pair2"),
};
static struct snd_soc_jack headphone_jack;
static struct snd_soc_jack mic_jack;
static struct snd_soc_jack_pin headphone_jack_pin = {
.pin = "Headphone",
.mask = SND_JACK_HEADPHONE,
};
static struct snd_soc_jack_pin mic_jack_pin = {
.pin = "Headset Mic",
.mask = SND_JACK_MICROPHONE,
};
static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/* The ADSP will covert the FE rate to 48k, max 4-channels */
rate->min = rate->max = 48000;
channels->min = 2;
channels->max = 4;
/* set SSP0 to 24 bit */
snd_mask_set_format(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT),
SNDRV_PCM_FORMAT_S24_LE);
return 0;
}
static int bdw_rt5650_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;
/* Workaround: set codec PLL to 19.2MHz that PLL source is
* from MCLK(24MHz) to conform 2.4MHz DMIC clock.
*/
ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK,
24000000, 19200000);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec pll: %d\n", ret);
return ret;
}
/* The actual MCLK freq is 24MHz. The codec is told that MCLK is
* 24.576MHz to satisfy the requirement of rl6231_get_clk_info.
* ASRC is enabled on AD and DA filters to ensure good audio quality.
*/
ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000,
SND_SOC_CLOCK_IN);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec sysclk configuration\n");
return ret;
}
return ret;
}
static struct snd_soc_ops bdw_rt5650_ops = {
.hw_params = bdw_rt5650_hw_params,
};
#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component =
snd_soc_rtdcom_lookup(rtd, DRV_NAME);
struct sst_pdata *pdata = dev_get_platdata(component->dev);
struct sst_hsw *broadwell = pdata->dsp;
int ret;
/* Set ADSP SSP port settings
* clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz
*/
ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0,
SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4);
if (ret < 0) {
dev_err(rtd->dev, "error: failed to set device config\n");
return ret;
}
return 0;
}
#endif
static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd)
{
struct bdw_rt5650_priv *bdw_rt5650 =
snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_component *component = rtd->codec_dai->component;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
int ret;
/* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1.
* The ASRC clock source is clk_i2s1_asrc.
*/
rt5645_sel_asrc_clk_src(component,
RT5645_DA_STEREO_FILTER |
RT5645_DA_MONO_L_FILTER |
RT5645_DA_MONO_R_FILTER |
RT5645_AD_STEREO_FILTER |
RT5645_AD_MONO_L_FILTER |
RT5645_AD_MONO_R_FILTER,
RT5645_CLK_SEL_I2S1_ASRC);
/* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret);
return ret;
}
/* Create and initialize headphone jack */
if (snd_soc_card_jack_new(rtd->card, "Headphone Jack",
SND_JACK_HEADPHONE, &headphone_jack,
&headphone_jack_pin, 1)) {
dev_err(component->dev, "Can't create headphone jack\n");
}
/* Create and initialize mic jack */
if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE,
&mic_jack, &mic_jack_pin, 1)) {
dev_err(component->dev, "Can't create mic jack\n");
}
rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL);
bdw_rt5650->component = component;
return 0;
}
/* broadwell digital audio interface glue - connects codec <--> CPU */
SND_SOC_DAILINK_DEF(dummy,
DAILINK_COMP_ARRAY(COMP_DUMMY()));
SND_SOC_DAILINK_DEF(fe,
DAILINK_COMP_ARRAY(COMP_CPU("System Pin")));
SND_SOC_DAILINK_DEF(platform,
DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio")));
SND_SOC_DAILINK_DEF(be,
DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1")));
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
#else
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
static struct snd_soc_dai_link bdw_rt5650_dais[] = {
/* Front End DAI links */
{
.name = "System PCM",
.stream_name = "System Playback",
.dynamic = 1,
#if !IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
.init = bdw_rt5650_rtd_init,
#endif
.trigger = {
SND_SOC_DPCM_TRIGGER_POST,
SND_SOC_DPCM_TRIGGER_POST
},
.dpcm_playback = 1,
.dpcm_capture = 1,
SND_SOC_DAILINK_REG(fe, dummy, platform),
},
/* Back End DAI links */
{
/* SSP0 - Codec */
.name = "Codec",
.id = 0,
.no_pcm = 1,
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ignore_suspend = 1,
.ignore_pmdown_time = 1,
.be_hw_params_fixup = broadwell_ssp0_fixup,
.ops = &bdw_rt5650_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
.init = bdw_rt5650_init,
SND_SOC_DAILINK_REG(ssp0_port, be, platform),
},
};
/* ASoC machine driver for Broadwell DSP + RT5650 */
static struct snd_soc_card bdw_rt5650_card = {
.name = "bdw-rt5650",
.owner = THIS_MODULE,
.dai_link = bdw_rt5650_dais,
.num_links = ARRAY_SIZE(bdw_rt5650_dais),
.dapm_widgets = bdw_rt5650_widgets,
.num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets),
.dapm_routes = bdw_rt5650_map,
.num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map),
.controls = bdw_rt5650_controls,
.num_controls = ARRAY_SIZE(bdw_rt5650_controls),
.fully_routed = true,
};
static int bdw_rt5650_probe(struct platform_device *pdev)
{
struct bdw_rt5650_priv *bdw_rt5650;
struct snd_soc_acpi_mach *mach;
int ret;
bdw_rt5650_card.dev = &pdev->dev;
/* Allocate driver private struct */
bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv),
GFP_KERNEL);
if (!bdw_rt5650)
return -ENOMEM;
/* override plaform name, if required */
mach = (&pdev->dev)->platform_data;
ret = snd_soc_fixup_dai_links_platform_name(&bdw_rt5650_card,
mach->mach_params.platform);
if (ret)
return ret;
snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650);
return devm_snd_soc_register_card(&pdev->dev, &bdw_rt5650_card);
}
static struct platform_driver bdw_rt5650_audio = {
.probe = bdw_rt5650_probe,
.driver = {
.name = "bdw-rt5650",
.pm = &snd_soc_pm_ops,
},
};
module_platform_driver(bdw_rt5650_audio)
/* Module information */
MODULE_AUTHOR("Ben Zhang <benzh@chromium.org>");
MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bdw-rt5650");

10
sound/soc/intel/boards/bdw-rt5677.c
查看文件

@@ -295,6 +295,14 @@ SND_SOC_DAILINK_DEF(platform,
SND_SOC_DAILINK_DEF(be,
DAILINK_COMP_ARRAY(COMP_CODEC("i2c-RT5677CE:00", "rt5677-aif1")));
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
#else
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
/* Wake on voice interface */
SND_SOC_DAILINK_DEFS(dsp,
DAILINK_COMP_ARRAY(COMP_CPU("spi-RT5677AA:00")),
@@ -342,7 +350,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = {
.dpcm_playback = 1,
.dpcm_capture = 1,
.init = bdw_rt5677_init,
SND_SOC_DAILINK_REG(dummy, be, dummy),
SND_SOC_DAILINK_REG(ssp0_port, be, platform),
},
};

10
sound/soc/intel/boards/broadwell.c
查看文件

@@ -164,6 +164,14 @@ SND_SOC_DAILINK_DEF(platform,
SND_SOC_DAILINK_DEF(codec,
DAILINK_COMP_ARRAY(COMP_CODEC("i2c-INT343A:00", "rt286-aif1")));
#if IS_ENABLED(CONFIG_SND_SOC_SOF_BROADWELL)
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp0-port")));
#else
SND_SOC_DAILINK_DEF(ssp0_port,
DAILINK_COMP_ARRAY(COMP_DUMMY()));
#endif
/* broadwell digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link broadwell_rt286_dais[] = {
/* Front End DAI links */
@@ -218,7 +226,7 @@ static struct snd_soc_dai_link broadwell_rt286_dais[] = {
.ops = &broadwell_rt286_ops,
.dpcm_playback = 1,
.dpcm_capture = 1,
SND_SOC_DAILINK_REG(dummy, codec, dummy),
SND_SOC_DAILINK_REG(ssp0_port, codec, platform),
},
};

10
sound/soc/intel/boards/bxt_da7219_max98357a.c
查看文件

@@ -161,13 +161,13 @@ static int broxton_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = DUAL_CHANNEL;
chan->min = chan->max = DUAL_CHANNEL;
/* set SSP to 24 bit */
snd_mask_none(fmt);
@@ -313,12 +313,12 @@ static const struct snd_soc_ops broxton_da7219_fe_ops = {
static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
if (params_channels(params) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

8
sound/soc/intel/boards/bxt_rt298.c
查看文件

@@ -206,13 +206,13 @@ static int broxton_ssp5_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
/* set SSP5 to 24 bit */
snd_mask_none(fmt);
@@ -255,9 +255,9 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = {
static int broxton_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

21
sound/soc/intel/boards/bytcht_es8316.c
查看文件

@@ -234,14 +234,6 @@ static int byt_cht_es8316_init(struct snd_soc_pcm_runtime *runtime)
return 0;
}
static const struct snd_soc_pcm_stream byt_cht_es8316_dai_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
static int byt_cht_es8316_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
@@ -360,7 +352,10 @@ static struct snd_soc_dai_link byt_cht_es8316_dais[] = {
/* SoC card */
static char codec_name[SND_ACPI_I2C_ID_LEN];
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
static char long_name[50]; /* = "bytcht-es8316-*-spk-*-mic" */
#endif
static char components_string[32]; /* = "cfg-spk:* cfg-mic:* */
static int byt_cht_es8316_suspend(struct snd_soc_card *card)
{
@@ -573,11 +568,19 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev)
}
}
/* register the soc card */
snprintf(components_string, sizeof(components_string),
"cfg-spk:%s cfg-mic:%s",
(quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "1" : "2",
mic_name[BYT_CHT_ES8316_MAP(quirk)]);
byt_cht_es8316_card.components = components_string;
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
snprintf(long_name, sizeof(long_name), "bytcht-es8316-%s-spk-%s-mic",
(quirk & BYT_CHT_ES8316_MONO_SPEAKER) ? "mono" : "stereo",
mic_name[BYT_CHT_ES8316_MAP(quirk)]);
byt_cht_es8316_card.long_name = long_name;
#endif
/* register the soc card */
snd_soc_card_set_drvdata(&byt_cht_es8316_card, priv);
ret = devm_snd_soc_register_card(dev, &byt_cht_es8316_card);

116
sound/soc/intel/boards/bytcr_rt5640.c
查看文件

@@ -933,14 +933,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime)
return 0;
}
static const struct snd_soc_pcm_stream byt_rt5640_dai_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
@@ -948,65 +940,43 @@ static int byt_rt5640_codec_fixup(struct snd_soc_pcm_runtime *rtd,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
int ret;
int ret, bits;
/* The DSP will covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) ||
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
/* set SSP0 to 16-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S16_LE);
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch 16-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, 16);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
bits = 16;
} else {
/* 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;
}
bits = 24;
}
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch. 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, bits);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
return 0;
}
@@ -1084,9 +1054,10 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = {
/* SoC card */
static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN];
static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */
static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */
#endif
static char byt_rt5640_components[32]; /* = "cfg-spk:* cfg-mic:*" */
static int byt_rt5640_suspend(struct snd_soc_card *card)
{
@@ -1266,28 +1237,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
log_quirks(&pdev->dev);
if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) ||
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
/* fixup codec aif name */
snprintf(byt_rt5640_codec_aif_name,
sizeof(byt_rt5640_codec_aif_name),
"%s", "rt5640-aif2");
byt_rt5640_dais[dai_index].codecs->dai_name =
byt_rt5640_codec_aif_name;
}
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2))
byt_rt5640_dais[dai_index].codecs->dai_name = "rt5640-aif2";
if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) ||
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) {
/* fixup cpu dai name name */
snprintf(byt_rt5640_cpu_dai_name,
sizeof(byt_rt5640_cpu_dai_name),
"%s", "ssp0-port");
byt_rt5640_dais[dai_index].cpus->dai_name =
byt_rt5640_cpu_dai_name;
}
(byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2))
byt_rt5640_dais[dai_index].cpus->dai_name = "ssp0-port";
if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) {
priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -1309,12 +1264,19 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev)
}
}
snprintf(byt_rt5640_components, sizeof(byt_rt5640_components),
"cfg-spk:%s cfg-mic:%s",
(byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ? "1" : "2",
map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]);
byt_rt5640_card.components = byt_rt5640_components;
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
snprintf(byt_rt5640_long_name, sizeof(byt_rt5640_long_name),
"bytcr-rt5640-%s-spk-%s-mic",
(byt_rt5640_quirk & BYT_RT5640_MONO_SPEAKER) ?
"mono" : "stereo",
map_name[BYT_RT5640_MAP(byt_rt5640_quirk)]);
byt_rt5640_card.long_name = byt_rt5640_long_name;
#endif
/* override plaform name, if required */
platform_name = mach->mach_params.platform;

72
sound/soc/intel/boards/bytcr_rt5651.c
查看文件

@@ -656,14 +656,6 @@ static int byt_rt5651_init(struct snd_soc_pcm_runtime *runtime)
return 0;
}
static const struct snd_soc_pcm_stream byt_rt5651_dai_params = {
.formats = SNDRV_PCM_FMTBIT_S24_LE,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
};
static int byt_rt5651_codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
@@ -795,9 +787,10 @@ static struct snd_soc_dai_link byt_rt5651_dais[] = {
/* SoC card */
static char byt_rt5651_codec_name[SND_ACPI_I2C_ID_LEN];
static char byt_rt5651_codec_aif_name[12]; /* = "rt5651-aif[1|2]" */
static char byt_rt5651_cpu_dai_name[10]; /* = "ssp[0|2]-port" */
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
static char byt_rt5651_long_name[50]; /* = "bytcr-rt5651-*-spk-*-mic[-swapped-hp]" */
#endif
static char byt_rt5651_components[50]; /* = "cfg-spk:* cfg-mic:*" */
static int byt_rt5651_suspend(struct snd_soc_card *card)
{
@@ -876,7 +869,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
const char *platform_name;
struct acpi_device *adev;
struct device *codec_dev;
const char *hp_swapped;
bool is_bytcr = false;
int ret_val = 0;
int dai_index = 0;
@@ -996,10 +988,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
if (byt_rt5651_gpios) {
devm_acpi_dev_add_driver_gpios(codec_dev, byt_rt5651_gpios);
priv->ext_amp_gpio = devm_fwnode_get_index_gpiod_from_child(
&pdev->dev, "ext-amp-enable", 0,
codec_dev->fwnode,
GPIOD_OUT_LOW, "speaker-amp");
priv->ext_amp_gpio = devm_fwnode_gpiod_get(&pdev->dev,
codec_dev->fwnode,
"ext-amp-enable",
GPIOD_OUT_LOW,
"speaker-amp");
if (IS_ERR(priv->ext_amp_gpio)) {
ret_val = PTR_ERR(priv->ext_amp_gpio);
switch (ret_val) {
@@ -1015,10 +1008,11 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
return ret_val;
}
}
priv->hp_detect = devm_fwnode_get_index_gpiod_from_child(
&pdev->dev, "hp-detect", 0,
codec_dev->fwnode,
GPIOD_IN, "hp-detect");
priv->hp_detect = devm_fwnode_gpiod_get(&pdev->dev,
codec_dev->fwnode,
"hp-detect",
GPIOD_IN,
"hp-detect");
if (IS_ERR(priv->hp_detect)) {
ret_val = PTR_ERR(priv->hp_detect);
switch (ret_val) {
@@ -1041,26 +1035,12 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
log_quirks(&pdev->dev);
if ((byt_rt5651_quirk & BYT_RT5651_SSP2_AIF2) ||
(byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) {
/* fixup codec aif name */
snprintf(byt_rt5651_codec_aif_name,
sizeof(byt_rt5651_codec_aif_name),
"%s", "rt5651-aif2");
byt_rt5651_dais[dai_index].codecs->dai_name =
byt_rt5651_codec_aif_name;
}
(byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2))
byt_rt5651_dais[dai_index].codecs->dai_name = "rt5651-aif2";
if ((byt_rt5651_quirk & BYT_RT5651_SSP0_AIF1) ||
(byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2)) {
/* fixup cpu dai name name */
snprintf(byt_rt5651_cpu_dai_name,
sizeof(byt_rt5651_cpu_dai_name),
"%s", "ssp0-port");
byt_rt5651_dais[dai_index].cpus->dai_name =
byt_rt5651_cpu_dai_name;
}
(byt_rt5651_quirk & BYT_RT5651_SSP0_AIF2))
byt_rt5651_dais[dai_index].cpus->dai_name = "ssp0-port";
if (byt_rt5651_quirk & BYT_RT5651_MCLK_EN) {
priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -1080,17 +1060,23 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev)
}
}
if (byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED)
hp_swapped = "-hp-swapped";
else
hp_swapped = "";
snprintf(byt_rt5651_components, sizeof(byt_rt5651_components),
"cfg-spk:%s cfg-mic:%s%s",
(byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ? "1" : "2",
mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
(byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ?
" cfg-hp:lrswap" : "");
byt_rt5651_card.components = byt_rt5651_components;
#if !IS_ENABLED(CONFIG_SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES)
snprintf(byt_rt5651_long_name, sizeof(byt_rt5651_long_name),
"bytcr-rt5651-%s-spk-%s-mic%s",
(byt_rt5651_quirk & BYT_RT5651_MONO_SPEAKER) ?
"mono" : "stereo",
mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)], hp_swapped);
mic_name[BYT_RT5651_MAP(byt_rt5651_quirk)],
(byt_rt5651_quirk & BYT_RT5651_HP_LR_SWAPPED) ?
"-hp-swapped" : "");
byt_rt5651_card.long_name = byt_rt5651_long_name;
#endif
/* override plaform name, if required */
platform_name = mach->mach_params.platform;

26
sound/soc/intel/boards/cht_bsw_rt5645.c
查看文件

@@ -515,8 +515,6 @@ static struct cht_acpi_card snd_soc_cards[] = {
};
static char cht_rt5645_codec_name[SND_ACPI_I2C_ID_LEN];
static char cht_rt5645_codec_aif_name[12]; /* = "rt5645-aif[1|2]" */
static char cht_rt5645_cpu_dai_name[10]; /* = "ssp[0|2]-port" */
struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */
u64 aif_value; /* 1: AIF1, 2: AIF2 */
@@ -641,28 +639,12 @@ static int snd_cht_mc_probe(struct platform_device *pdev)
log_quirks(&pdev->dev);
if ((cht_rt5645_quirk & CHT_RT5645_SSP2_AIF2) ||
(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) {
/* fixup codec aif name */
snprintf(cht_rt5645_codec_aif_name,
sizeof(cht_rt5645_codec_aif_name),
"%s", "rt5645-aif2");
cht_dailink[dai_index].codecs->dai_name =
cht_rt5645_codec_aif_name;
}
(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2))
cht_dailink[dai_index].codecs->dai_name = "rt5645-aif2";
if ((cht_rt5645_quirk & CHT_RT5645_SSP0_AIF1) ||
(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2)) {
/* fixup cpu dai name name */
snprintf(cht_rt5645_cpu_dai_name,
sizeof(cht_rt5645_cpu_dai_name),
"%s", "ssp0-port");
cht_dailink[dai_index].cpus->dai_name =
cht_rt5645_cpu_dai_name;
}
(cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2))
cht_dailink[dai_index].cpus->dai_name = "ssp0-port";
/* override plaform name, if required */
platform_name = mach->mach_params.platform;

8
sound/soc/intel/boards/cml_rt1011_rt5682.c
查看文件

@@ -405,19 +405,19 @@ static struct snd_soc_dai_link cml_rt1011_rt5682_dailink[] = {
static struct snd_soc_codec_conf rt1011_conf[] = {
{
.dev_name = "i2c-10EC1011:00",
.dlc = COMP_CODEC_CONF("i2c-10EC1011:00"),
.name_prefix = "WL",
},
{
.dev_name = "i2c-10EC1011:01",
.dlc = COMP_CODEC_CONF("i2c-10EC1011:01"),
.name_prefix = "WR",
},
{
.dev_name = "i2c-10EC1011:02",
.dlc = COMP_CODEC_CONF("i2c-10EC1011:02"),
.name_prefix = "TL",
},
{
.dev_name = "i2c-10EC1011:03",
.dlc = COMP_CODEC_CONF("i2c-10EC1011:03"),
.name_prefix = "TR",
},
};

18
sound/soc/intel/boards/glk_rt5682_max98357a.c
查看文件

@@ -118,13 +118,13 @@ static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = DUAL_CHANNEL;
chan->min = chan->max = DUAL_CHANNEL;
/* set SSP to 24 bit */
snd_mask_none(fmt);
@@ -249,16 +249,6 @@ static const struct snd_pcm_hw_constraint_list constraints_rates = {
.mask = 0,
};
static const unsigned int channels[] = {
DUAL_CHANNEL,
};
static const struct snd_pcm_hw_constraint_list constraints_channels = {
.count = ARRAY_SIZE(channels),
.list = channels,
.mask = 0,
};
static unsigned int channels_quad[] = {
QUAD_CHANNEL,
};
@@ -272,13 +262,13 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = {
static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/*
* set BE channel constraint as user FE channels
*/
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

2
sound/soc/intel/boards/hda_dsp_common.c
查看文件

@@ -63,7 +63,7 @@ int hda_dsp_hdmi_build_controls(struct snd_soc_card *card,
"%s: mapping HDMI converter %d to PCM %d (%p)\n",
__func__, i, hpcm->device, spcm);
} else {
hpcm->pcm = 0;
hpcm->pcm = NULL;
hpcm->device = SNDRV_PCM_INVALID_DEVICE;
dev_warn(card->dev,
"%s: no PCM in topology for HDMI converter %d\n\n",

23
sound/soc/intel/boards/kbl_da7219_max98357a.c
查看文件

@@ -141,13 +141,13 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = DUAL_CHANNEL;
chan->min = chan->max = DUAL_CHANNEL;
/* set SSP to 24 bit */
snd_mask_none(fmt);
@@ -305,7 +305,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = {
static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/*
@@ -313,9 +313,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
*/
if (params_channels(params) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}
@@ -336,19 +336,6 @@ static struct snd_soc_ops kabylake_dmic_ops = {
.startup = kabylake_dmic_startup,
};
static const unsigned int rates_16000[] = {
16000,
};
static const struct snd_pcm_hw_constraint_list constraints_16000 = {
.count = ARRAY_SIZE(rates_16000),
.list = rates_16000,
};
static const unsigned int ch_mono[] = {
1,
};
SND_SOC_DAILINK_DEF(dummy,
DAILINK_COMP_ARRAY(COMP_DUMMY()));

27
sound/soc/intel/boards/kbl_da7219_max98927.c
查看文件

@@ -176,7 +176,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *runtime = substream->private_data;
int ret = 0, j;
int ret, j;
for (j = 0; j < runtime->num_codecs; j++) {
struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
@@ -279,7 +279,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_soc_dpcm *dpcm = container_of(
@@ -298,7 +298,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
!strcmp(rtd->card->name, "kblmax98373")) {
/* The ADSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = DUAL_CHANNEL;
chan->min = chan->max = DUAL_CHANNEL;
/* set SSP to 24 bit */
snd_mask_none(fmt);
@@ -313,7 +313,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
!strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
!strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
snd_mask_none(fmt);
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
}
@@ -341,6 +341,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
kabylake_ssp1_map,
ARRAY_SIZE(kabylake_ssp1_map));
if (ret)
return ret;
/*
* Headset buttons map to the google Reference headset.
* These can be configured by userspace.
@@ -491,7 +494,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = {
static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
/*
@@ -499,9 +502,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
*/
if (params_channels(params) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}
@@ -571,12 +574,12 @@ static struct snd_soc_ops skylake_refcap_ops = {
static struct snd_soc_codec_conf max98927_codec_conf[] = {
{
.dev_name = MAX98927_DEV0_NAME,
.dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME),
.name_prefix = "Right",
},
{
.dev_name = MAX98927_DEV1_NAME,
.dlc = COMP_CODEC_CONF(MAX98927_DEV1_NAME),
.name_prefix = "Left",
},
};
@@ -584,12 +587,12 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = {
static struct snd_soc_codec_conf max98373_codec_conf[] = {
{
.dev_name = MAX98373_DEV0_NAME,
.dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME),
.name_prefix = "Right",
},
{
.dev_name = MAX98373_DEV1_NAME,
.dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME),
.name_prefix = "Left",
},
};
@@ -1092,7 +1095,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
struct kbl_codec_private *ctx;
struct snd_soc_dai_link *kbl_dai_link;
struct snd_soc_dai_link_component **codecs;
int i = 0;
int i;
ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)

4
sound/soc/intel/boards/kbl_rt5660.c
查看文件

@@ -138,13 +138,13 @@ static int kabylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will convert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = DUAL_CHANNEL;
chan->min = chan->max = DUAL_CHANNEL;
/* set SSP0 to 24 bit */
snd_mask_none(fmt);

14
sound/soc/intel/boards/kbl_rt5663_max98927.c
查看文件

@@ -229,11 +229,11 @@ static const struct snd_soc_dapm_route kabylake_5663_map[] = {
static struct snd_soc_codec_conf max98927_codec_conf[] = {
{
.dev_name = MAXIM_DEV0_NAME,
.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
.name_prefix = "Right",
},
{
.dev_name = MAXIM_DEV1_NAME,
.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
.name_prefix = "Left",
},
};
@@ -398,7 +398,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_soc_dpcm *dpcm = container_of(
@@ -413,7 +413,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
!strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
!strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
snd_mask_none(fmt);
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
}
@@ -454,13 +454,13 @@ static struct snd_soc_ops kabylake_rt5663_ops = {
static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

14
sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
查看文件

@@ -193,11 +193,11 @@ static const struct snd_soc_dapm_route kabylake_map[] = {
static struct snd_soc_codec_conf max98927_codec_conf[] = {
{
.dev_name = MAXIM_DEV0_NAME,
.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
.name_prefix = "Right",
},
{
.dev_name = MAXIM_DEV1_NAME,
.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
.name_prefix = "Left",
},
};
@@ -333,7 +333,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
struct snd_soc_dpcm *dpcm = container_of(
@@ -348,15 +348,15 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
!strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
!strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
snd_mask_none(fmt);
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
} else if (!strcmp(fe_dai_link->name, "Kbl Audio DMIC cap")) {
if (params_channels(params) == 2 ||
DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
}
/*
* The speaker on the SSP0 supports S16_LE and not S24_LE.
@@ -761,7 +761,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
{
struct kbl_codec_private *ctx;
struct snd_soc_acpi_mach *mach;
int ret = 0;
int ret;
ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
if (!ctx)

8
sound/soc/intel/boards/skl_hda_dsp_generic.c
查看文件

@@ -100,6 +100,8 @@ static struct snd_soc_card hda_soc_card = {
.late_probe = skl_hda_card_late_probe,
};
static char hda_soc_components[30];
#define IDISP_DAI_COUNT 3
#define HDAC_DAI_COUNT 2
#define DMIC_DAI_COUNT 2
@@ -183,6 +185,12 @@ static int skl_hda_audio_probe(struct platform_device *pdev)
hda_soc_card.dev = &pdev->dev;
snd_soc_card_set_drvdata(&hda_soc_card, ctx);
if (mach->mach_params.dmic_num > 0) {
snprintf(hda_soc_components, sizeof(hda_soc_components),
"cfg-dmics:%d", mach->mach_params.dmic_num);
hda_soc_card.components = hda_soc_components;
}
return devm_snd_soc_register_card(&pdev->dev, &hda_soc_card);
}

10
sound/soc/intel/boards/skl_nau88l25_max98357a.c
查看文件

@@ -139,13 +139,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
@@ -315,13 +315,13 @@ static const struct snd_soc_ops skylake_nau8825_ops = {
static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

14
sound/soc/intel/boards/skl_nau88l25_ssm4567.c
查看文件

@@ -147,11 +147,11 @@ static const struct snd_soc_dapm_route skylake_map[] = {
static struct snd_soc_codec_conf ssm4567_codec_conf[] = {
{
.dev_name = "i2c-INT343B:00",
.dlc = COMP_CODEC_CONF("i2c-INT343B:00"),
.name_prefix = "Left",
},
{
.dev_name = "i2c-INT343B:01",
.dlc = COMP_CODEC_CONF("i2c-INT343B:01"),
.name_prefix = "Right",
},
};
@@ -317,13 +317,13 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The ADSP will covert the FE rate to 48k, stereo */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
@@ -334,12 +334,12 @@ static int skylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
if (params_channels(params) == 2 || DMIC_CH(dmic_constraints) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

10
sound/soc/intel/boards/skl_rt286.c
查看文件

@@ -211,13 +211,13 @@ static int skylake_ssp0_fixup(struct snd_soc_pcm_runtime *rtd,
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
/* The output is 48KHz, stereo, 16bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
chan->min = chan->max = 2;
/* set SSP0 to 24 bit */
snd_mask_none(fmt);
@@ -247,12 +247,12 @@ static const struct snd_soc_ops skylake_rt286_ops = {
static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *channels = hw_param_interval(params,
struct snd_interval *chan = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
if (params_channels(params) == 2)
channels->min = channels->max = 2;
chan->min = chan->max = 2;
else
channels->min = channels->max = 4;
chan->min = chan->max = 4;
return 0;
}

371
sound/soc/intel/boards/sof_da7219_max98373.c 一般檔案
查看文件

@@ -0,0 +1,371 @@
// SPDX-License-Identifier: GPL-2.0
// Copyright(c) 2019 Intel Corporation.
/*
* Intel SOF Machine driver for DA7219 + MAX98373 codec
*/
#include <linux/input.h>
#include <linux/module.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <linux/platform_device.h>
#include <sound/soc.h>
#include <sound/soc-acpi.h>
#include "../../codecs/da7219.h"
#include "../../codecs/da7219-aad.h"
#include "hda_dsp_common.h"
#define DIALOG_CODEC_DAI "da7219-hifi"
#define MAX98373_CODEC_DAI "max98373-aif1"
#define MAXIM_DEV0_NAME "i2c-MX98373:00"
#define MAXIM_DEV1_NAME "i2c-MX98373:01"
struct hdmi_pcm {
struct list_head head;
struct snd_soc_dai *codec_dai;
int device;
};
struct card_private {
struct snd_soc_jack headset;
struct list_head hdmi_pcm_list;
struct snd_soc_jack hdmi[3];
};
static int platform_clock_control(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
struct snd_soc_dapm_context *dapm = w->dapm;
struct snd_soc_card *card = dapm->card;
struct snd_soc_dai *codec_dai;
int ret = 0;
codec_dai = snd_soc_card_get_codec_dai(card, DIALOG_CODEC_DAI);
if (!codec_dai) {
dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n");
return -EIO;
}
if (SND_SOC_DAPM_EVENT_OFF(event)) {
ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK,
0, 0);
if (ret)
dev_err(card->dev, "failed to stop PLL: %d\n", ret);
} else if (SND_SOC_DAPM_EVENT_ON(event)) {
ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL_SRM,
0, DA7219_PLL_FREQ_OUT_98304);
if (ret)
dev_err(card->dev, "failed to start PLL: %d\n", ret);
}
return ret;
}
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Headphone Jack"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Left Spk"),
SOC_DAPM_PIN_SWITCH("Right Spk"),
};
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_SPK("Left Spk", NULL),
SND_SOC_DAPM_SPK("Right Spk", NULL),
SND_SOC_DAPM_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 audio_map[] = {
{ "Headphone Jack", NULL, "HPL" },
{ "Headphone Jack", NULL, "HPR" },
{ "Left Spk", NULL, "Left BE_OUT" },
{ "Right Spk", NULL, "Right BE_OUT" },
{ "MIC", NULL, "Headset Mic" },
{ "Headphone Jack", NULL, "Platform Clock" },
{ "Headset Mic", NULL, "Platform Clock" },
};
static struct snd_soc_jack headset;
static int da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_component *component = rtd->codec_dai->component;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_jack *jack;
int ret;
/* Configure sysclk for codec */
ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 24000000,
SND_SOC_CLOCK_IN);
if (ret) {
dev_err(rtd->dev, "can't set codec sysclk configuration\n");
return ret;
}
/*
* Headset buttons map to the google Reference headset.
* These can be configured by userspace.
*/
ret = snd_soc_card_jack_new(rtd->card, "Headset Jack",
SND_JACK_HEADSET | SND_JACK_BTN_0 |
SND_JACK_BTN_1 | SND_JACK_BTN_2 |
SND_JACK_BTN_3 | SND_JACK_LINEOUT,
&headset, NULL, 0);
if (ret) {
dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret);
return ret;
}
jack = &headset;
snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
da7219_aad_jack_det(component, jack);
return ret;
}
static int ssp1_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *runtime = substream->private_data;
int ret, j;
for (j = 0; j < runtime->num_codecs; j++) {
struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) {
/* vmon_slot_no = 0 imon_slot_no = 1 for TX slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 3, 4, 16);
if (ret < 0) {
dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret);
return ret;
}
}
if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) {
/* vmon_slot_no = 2 imon_slot_no = 3 for TX slots */
ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC, 3, 4, 16);
if (ret < 0) {
dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret);
return ret;
}
}
}
return 0;
}
static struct snd_soc_ops ssp1_ops = {
.hw_params = ssp1_hw_params,
};
static struct snd_soc_codec_conf max98373_codec_conf[] = {
{
.dlc = COMP_CODEC_CONF(MAXIM_DEV0_NAME),
.name_prefix = "Right",
},
{
.dlc = COMP_CODEC_CONF(MAXIM_DEV1_NAME),
.name_prefix = "Left",
},
};
static int hdmi_init(struct snd_soc_pcm_runtime *rtd)
{
struct card_private *ctx = snd_soc_card_get_drvdata(rtd->card);
struct snd_soc_dai *dai = rtd->codec_dai;
struct hdmi_pcm *pcm;
pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL);
if (!pcm)
return -ENOMEM;
pcm->device = dai->id;
pcm->codec_dai = dai;
list_add_tail(&pcm->head, &ctx->hdmi_pcm_list);
return 0;
}
static int card_late_probe(struct snd_soc_card *card)
{
struct card_private *ctx = snd_soc_card_get_drvdata(card);
struct snd_soc_acpi_mach *mach = (card->dev)->platform_data;
struct hdmi_pcm *pcm;
if (mach->mach_params.common_hdmi_codec_drv) {
pcm = list_first_entry(&ctx->hdmi_pcm_list, struct hdmi_pcm,
head);
return hda_dsp_hdmi_build_controls(card,
pcm->codec_dai->component);
}
return -EINVAL;
}
SND_SOC_DAILINK_DEF(ssp0_pin,
DAILINK_COMP_ARRAY(COMP_CPU("SSP0 Pin")));
SND_SOC_DAILINK_DEF(ssp0_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", DIALOG_CODEC_DAI)));
SND_SOC_DAILINK_DEF(ssp1_pin,
DAILINK_COMP_ARRAY(COMP_CPU("SSP1 Pin")));
SND_SOC_DAILINK_DEF(ssp1_amps,
DAILINK_COMP_ARRAY(
/* Left */ COMP_CODEC(MAXIM_DEV0_NAME, MAX98373_CODEC_DAI),
/* Right */ COMP_CODEC(MAXIM_DEV1_NAME, MAX98373_CODEC_DAI)));
SND_SOC_DAILINK_DEF(dmic_pin,
DAILINK_COMP_ARRAY(COMP_CPU("DMIC01 Pin")));
SND_SOC_DAILINK_DEF(dmic_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec", "dmic-hifi")));
SND_SOC_DAILINK_DEF(idisp1_pin,
DAILINK_COMP_ARRAY(COMP_CPU("iDisp1 Pin")));
SND_SOC_DAILINK_DEF(idisp1_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi1")));
SND_SOC_DAILINK_DEF(idisp2_pin,
DAILINK_COMP_ARRAY(COMP_CPU("iDisp2 Pin")));
SND_SOC_DAILINK_DEF(idisp2_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi2")));
SND_SOC_DAILINK_DEF(idisp3_pin,
DAILINK_COMP_ARRAY(COMP_CPU("iDisp3 Pin")));
SND_SOC_DAILINK_DEF(idisp3_codec,
DAILINK_COMP_ARRAY(COMP_CODEC("ehdaudio0D2", "intel-hdmi-hifi3")));
SND_SOC_DAILINK_DEF(platform, /* subject to be overridden during probe */
DAILINK_COMP_ARRAY(COMP_PLATFORM("0000:00:1f.3")));
static struct snd_soc_dai_link dais[] = {
/* Back End DAI links */
{
.name = "SSP1-Codec",
.id = 0,
.ignore_pmdown_time = 1,
.no_pcm = 1,
.dpcm_playback = 1,
.dpcm_capture = 1, /* IV feedback */
.ops = &ssp1_ops,
SND_SOC_DAILINK_REG(ssp1_pin, ssp1_amps, platform),
},
{
.name = "SSP0-Codec",
.id = 1,
.no_pcm = 1,
.init = da7219_codec_init,
.ignore_pmdown_time = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform),
},
{
.name = "dmic01",
.id = 2,
.ignore_suspend = 1,
.dpcm_capture = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(dmic_pin, dmic_codec, platform),
},
{
.name = "iDisp1",
.id = 3,
.init = hdmi_init,
.dpcm_playback = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(idisp1_pin, idisp1_codec, platform),
},
{
.name = "iDisp2",
.id = 4,
.init = hdmi_init,
.dpcm_playback = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(idisp2_pin, idisp2_codec, platform),
},
{
.name = "iDisp3",
.id = 5,
.init = hdmi_init,
.dpcm_playback = 1,
.no_pcm = 1,
SND_SOC_DAILINK_REG(idisp3_pin, idisp3_codec, platform),
},
};
static struct snd_soc_card card_da7219_m98373 = {
.name = "da7219max",
.owner = THIS_MODULE,
.dai_link = dais,
.num_links = ARRAY_SIZE(dais),
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.dapm_widgets = widgets,
.num_dapm_widgets = ARRAY_SIZE(widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.codec_conf = max98373_codec_conf,
.num_configs = ARRAY_SIZE(max98373_codec_conf),
.fully_routed = true,
.late_probe = card_late_probe,
};
static int audio_probe(struct platform_device *pdev)
{
static struct snd_soc_card *card;
struct snd_soc_acpi_mach *mach;
struct card_private *ctx;
int ret;
ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC);
if (!ctx)
return -ENOMEM;
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
card = (struct snd_soc_card *)pdev->id_entry->driver_data;
card->dev = &pdev->dev;
mach = (&pdev->dev)->platform_data;
ret = snd_soc_fixup_dai_links_platform_name(card,
mach->mach_params.platform);
if (ret)
return ret;
snd_soc_card_set_drvdata(card, ctx);
return devm_snd_soc_register_card(&pdev->dev, card);
}
static const struct platform_device_id board_ids[] = {
{
.name = "sof_da7219_max98373",
.driver_data = (kernel_ulong_t)&card_da7219_m98373,
},
{ }
};
static struct platform_driver audio = {
.probe = audio_probe,
.driver = {
.name = "sof_da7219_max98373",
.pm = &snd_soc_pm_ops,
},
.id_table = board_ids,
};
module_platform_driver(audio)
/* Module information */
MODULE_DESCRIPTION("ASoC Intel(R) SOF Machine driver");
MODULE_AUTHOR("Yong Zhi <yong.zhi@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:sof_da7219_max98373");

43
sound/soc/intel/boards/sof_rt5682.c
查看文件

@@ -35,6 +35,10 @@
#define SOF_RT5682_SSP_AMP(quirk) \
(((quirk) << SOF_RT5682_SSP_AMP_SHIFT) & SOF_RT5682_SSP_AMP_MASK)
#define SOF_RT5682_MCLK_BYTCHT_EN BIT(9)
#define SOF_RT5682_NUM_HDMIDEV_SHIFT 10
#define SOF_RT5682_NUM_HDMIDEV_MASK (GENMASK(12, 10))
#define SOF_RT5682_NUM_HDMIDEV(quirk) \
((quirk << SOF_RT5682_NUM_HDMIDEV_SHIFT) & SOF_RT5682_NUM_HDMIDEV_MASK)
/* Default: MCLK on, MCLK 19.2M, SSP0 */
static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN |
@@ -594,6 +598,19 @@ static int sof_audio_probe(struct platform_device *pdev)
if (!ctx)
return -ENOMEM;
if (pdev->id_entry && pdev->id_entry->driver_data)
sof_rt5682_quirk = (unsigned long)pdev->id_entry->driver_data;
dmi_check_system(sof_rt5682_quirk_table);
mach = (&pdev->dev)->platform_data;
/* A speaker amp might not be present when the quirk claims one is.
* Detect this via whether the machine driver match includes quirk_data.
*/
if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data)
sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT;
if (soc_intel_is_byt() || soc_intel_is_cht()) {
is_legacy_cpu = 1;
dmic_be_num = 0;
@@ -604,11 +621,13 @@ static int sof_audio_probe(struct platform_device *pdev)
SOF_RT5682_SSP_CODEC(2);
} else {
dmic_be_num = 2;
hdmi_num = 3;
hdmi_num = (sof_rt5682_quirk & SOF_RT5682_NUM_HDMIDEV_MASK) >>
SOF_RT5682_NUM_HDMIDEV_SHIFT;
/* default number of HDMI DAI's */
if (!hdmi_num)
hdmi_num = 3;
}
dmi_check_system(sof_rt5682_quirk_table);
/* need to get main clock from pmc */
if (sof_rt5682_quirk & SOF_RT5682_MCLK_BYTCHT_EN) {
ctx->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3");
@@ -652,7 +671,6 @@ static int sof_audio_probe(struct platform_device *pdev)
INIT_LIST_HEAD(&ctx->hdmi_pcm_list);
sof_audio_card_rt5682.dev = &pdev->dev;
mach = (&pdev->dev)->platform_data;
/* set platform name for each dailink */
ret = snd_soc_fixup_dai_links_platform_name(&sof_audio_card_rt5682,
@@ -683,6 +701,21 @@ static int sof_rt5682_remove(struct platform_device *pdev)
return 0;
}
static const struct platform_device_id board_ids[] = {
{
.name = "sof_rt5682",
},
{
.name = "tgl_max98357a_rt5682",
.driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN |
SOF_RT5682_SSP_CODEC(0) |
SOF_SPEAKER_AMP_PRESENT |
SOF_RT5682_SSP_AMP(1) |
SOF_RT5682_NUM_HDMIDEV(4)),
},
{ }
};
static struct platform_driver sof_audio = {
.probe = sof_audio_probe,
.remove = sof_rt5682_remove,
@@ -690,6 +723,7 @@ static struct platform_driver sof_audio = {
.name = "sof_rt5682",
.pm = &snd_soc_pm_ops,
},
.id_table = board_ids,
};
module_platform_driver(sof_audio)
@@ -699,3 +733,4 @@ MODULE_AUTHOR("Bard Liao <bard.liao@intel.com>");
MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r@intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:sof_rt5682");
MODULE_ALIAS("platform:tgl_max98357a_rt5682");