February 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 1/2] mfd: wm8994: Fix driver operation if loaded as modules
by Marek Szyprowski 27 Feb '20

27 Feb '20

WM8994 chip has built-in regulators, which might be used for chip operation. They are controlled by a separate wm8994-regulator driver, which should be loaded before this driver calls regulator_get(), because that driver also provides consumer-supply mapping for the them. If that driver is not yet loaded, regulator core substitute them with dummy regulator, what breaks chip operation, because the built-in regulators are never enabled. Fix this by annotating this driver with MODULE_SOFTDEP() "pre" dependency to "wm8994_regulator" module. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - replaced request_module() call with MODULE_SOFTDEP() annotation --- drivers/mfd/wm8994-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..737dede4a95c 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -690,3 +690,4 @@ module_i2c_driver(wm8994_i2c_driver); MODULE_DESCRIPTION("Core support for the WM8994 audio CODEC"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_SOFTDEP("pre: wm8994_regulator"); -- 2.17.1

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[PATCH for-next] ASoC: tlv320adcx140: Fix MIC_BIAS defines for ADC full scale
by Dan Murphy 27 Feb '20

27 Feb '20

Change the #defines for the ADC full scale bits from MIC_BIAS to ADC_FSCALE. This also changes the error message to incidate ADC full scale value error as opposed to the Mic bias. Reported-by: Ricard Wanderlof <ricardw(a)axis.com> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 12 ++++++------ sound/soc/codecs/tlv320adcx140.h | 8 ++++---- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662c..825ace9b5fa7 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -758,12 +758,12 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u8(adcx140->dev, "ti,vref-source", &vref_source); if (ret) - vref_source = ADCX140_MIC_BIAS_VREF_275V; + vref_source = ADCX140_ADC_FSCALE_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + if (vref_source != ADCX140_ADC_FSCALE_VREF_275V && + vref_source != ADCX140_ADC_FSCALE_VREF_25V && + vref_source != ADCX140_ADC_FSCALE_VREF_1375V) { + dev_err(adcx140->dev, "ADC full scale setting is invalid\n"); return -EINVAL; } @@ -787,7 +787,7 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG, ADCX140_MIC_BIAS_VAL_MSK | - ADCX140_MIC_BIAS_VREF_MSK, bias_source); + ADCX140_ADC_FSCALE_VREF_MSK, bias_source); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); out: diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 6d055e55909e..adb9513900b1 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -117,10 +117,10 @@ #define ADCX140_MIC_BIAS_VAL_AVDD 6 #define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4) -#define ADCX140_MIC_BIAS_VREF_275V 0 -#define ADCX140_MIC_BIAS_VREF_25V 1 -#define ADCX140_MIC_BIAS_VREF_1375V 2 -#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_ADC_FSCALE_VREF_275V 0 +#define ADCX140_ADC_FSCALE_VREF_25V 1 +#define ADCX140_ADC_FSCALE_VREF_1375V 2 +#define ADCX140_ADC_FSCALE_VREF_MSK GENMASK(1, 0) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) -- 2.25.0

2 1

snd_rawmidi_params_set_no_active_sensing()
by GitHub issues - opened 27 Feb '20

27 Feb '20

alsa-project/alsa-lib issue #29 was opened from MusicMaker: Does this filter out receiving 0xFE incoming and outgoing messages when enabled ? Still see incoming active sensing messages after using this. Issue URL : https://github.com/alsa-project/alsa-lib/issues/29 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/8] ASoC: soc-pcm cleanup step5
by Kuninori Morimoto 27 Feb '20

27 Feb '20

Hi Mark These are soc-pcm cleanup step5. These are based on Bard Liao's "ASoC: Add Multi CPU DAI support" patch-set. [7/8] patch might exchange the behavior Kuninori Morimoto (8): 1) ASoC: soc-pcm: use defined stream 2) ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() 3) ASoC: soc-pcm: move dpcm_fe_dai_close() 4) ASoC: soc-pcm: add dpcm_fe_dai_clean() 5) ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() 6) ASoC: soc-pcm: check DAI's activity more simply 7) ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. 8) ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() sound/soc/soc-pcm.c | 107 ++++++++++++++++++++------------------------ 1 file changed, 48 insertions(+), 59 deletions(-) -- 2.17.1

3 20

Applied "ASoC: tas2562: Fix sample rate error message" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: tas2562: Fix sample rate error message has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 38b6a714941a285319734b9ccc30d34e107d5d7e Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:05 -0600 Subject: [PATCH] ASoC: tas2562: Fix sample rate error message Fix error message for setting the sample rate. It says bitwidth but should say sample rate. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200226130305.12043-3-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5b4803a76719..be52886a5edb 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -252,7 +252,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream, ret = tas2562_set_samplerate(tas2562, params_rate(params)); if (ret) - dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + dev_err(tas2562->dev, "set sample rate failed, %d\n", ret); return ret; } -- 2.20.1

1 0

[PATCH v5 0/6] ASoC: Add Multi CPU DAI support
by Bard Liao 26 Feb '20

26 Feb '20

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v5: - rebase to latest kernel base Bard Liao (2): ASoC: Return error if the function does not support multi-cpu ASoC: pcm: check if cpu-dai supports a given stream Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 + sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 168 +++++----- sound/soc/soc-dapm.c | 133 ++++---- sound/soc/soc-generic-dmaengine-pcm.c | 18 + sound/soc/soc-pcm.c | 463 ++++++++++++++++++-------- 6 files changed, 531 insertions(+), 271 deletions(-) -- 2.17.1

2 10

Applied "ASoC: Add dapm_add_valid_dai_widget helper" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: Add dapm_add_valid_dai_widget helper has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6c4b13b51aa36aab023dd0bf24bf5582c9ba091e Mon Sep 17 00:00:00 2001 From: Shreyas NC <shreyas.nc(a)intel.com> Date: Tue, 25 Feb 2020 21:39:14 +0800 Subject: [PATCH] ASoC: Add dapm_add_valid_dai_widget helper Adding a helper to connect widget for a specific cpu and codec dai The helper will help dapm_connect_dai_link_widgets() to reduce indents. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200225133917.21314-4-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 119 +++++++++++++++++++++++-------------------- 1 file changed, 63 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 58c318c9debb..539a1eaebeac 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4277,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) +static void dapm_add_valid_dai_widget(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - int i; if (rtd->dai_link->params) { playback_cpu = cpu_dai->capture_widget; @@ -4298,67 +4297,75 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture_cpu = capture; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI playback if widgets are valid */ - codec = codec_dai->playback_widget; - - if (playback_cpu && codec) { - if (!playback) { - substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(playback)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + /* connect BE DAI playback if widgets are valid */ + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + playback = snd_soc_dapm_new_dai(card, substream, + "playback"); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + goto capture; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); } - } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI capture if widgets are valid */ - codec = codec_dai->capture_widget; - - if (codec && capture_cpu) { - if (!capture) { - substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(capture)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); + + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); + } + +capture: + /* connect BE DAI capture if widgets are valid */ + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; + capture = snd_soc_dapm_new_dai(card, substream, + "capture"); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + return; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); } + + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); + + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, + codec_dai, rtd->cpu_dais[0]); +} + static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { -- 2.20.1

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Applied "ASoC: Add multiple CPU DAI support in DAPM" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: Add multiple CPU DAI support in DAPM has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de6214a33633d8ce1c1490336f8e798e75ccd004 Mon Sep 17 00:00:00 2001 From: Shreyas NC <shreyas.nc(a)intel.com> Date: Tue, 25 Feb 2020 21:39:15 +0800 Subject: [PATCH] ASoC: Add multiple CPU DAI support in DAPM DAPM handles DAIs during soc_dapm_stream_event() and during addition and creation of DAI widgets i.e., dapm_add_valid_dai_widget() and dapm_connect_dai_link_widgets(). Extend these functions to handle multiple cpu dai. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200225133917.21314-5-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 539a1eaebeac..6ce024d52170 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4361,9 +4361,19 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, struct snd_soc_dai *codec_dai; int i; - for_each_rtd_codec_dai(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, - codec_dai, rtd->cpu_dais[0]); + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } + } static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, @@ -4424,9 +4434,11 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int i; - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); for_each_rtd_codec_dai(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); -- 2.20.1

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[PATCH 0/3] SoundWire: ASoC interfaces for multi-cpu dais and DisCo helpers
by Pierre-Louis Bossart 26 Feb '20

26 Feb '20

The first two patches prepare the support of multi-cpu dais for synchronized playback and capture. We remove an unused set of prototypes and add a get_sdw_stream() callback prototype currently missing (the implementation will come later as part of the synchronized playback) The last exposes macros used internally, so that they can be reused to extract information from the _ADR 64-bit values in SOF platform drivers and related machine drivers. I think it's simpler if all these simple patches are merged through the SoundWire tree. With the additional changes to remove the platform drivers and the merge of interrupt handling, that will result in a single immutable tag provided to Mark Brown. Pierre-Louis Bossart (3): soundwire: cadence: remove useless prototypes ASoC: soc-dai: add get_sdw_stream() callback soundwire: add helper macros for devID fields drivers/soundwire/bus.c | 21 +++++---------------- drivers/soundwire/cadence_master.h | 8 -------- include/linux/soundwire/sdw.h | 23 +++++++++++++++++++++++ include/sound/soc-dai.h | 21 +++++++++++++++++++++ 4 files changed, 49 insertions(+), 24 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: amd: Add machine driver for Raven based platform
by Akshu Agrawal 26 Feb '20

26 Feb '20

Add machine driver for Raven based platform using RT5682 + MAX9836 + CROS_EC codecs Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- This patch is dependent on https://patchwork.kernel.org/patch/11381839/ sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp3x-rt5682-max9836.c | 334 +++++++++++++++++++++++++++ 3 files changed, 346 insertions(+) create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f40517717c4..b29ef1373946 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -26,3 +26,13 @@ config SND_SOC_AMD_ACP3x depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform + +config SND_SOC_AMD_RV_RT5682_MACH + tristate "AMD RV support for RT5682" + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_CROS_EC_CODEC + select I2C_CROS_EC_TUNNEL + depends on SND_SOC_AMD_ACP3x && I2C + help + This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index c4ddc6adb6f0..e6f3d9b469f3 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -2,8 +2,10 @@ acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o +snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ +obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c new file mode 100644 index 000000000000..96fbcd29e3ed --- /dev/null +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -0,0 +1,334 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec. +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/acpi.h> + +#include "raven/acp3x.h" +#include "../codecs/rt5682.h" + +#define PCO_PLAT_CLK 48000000 +#define RT5682_PLL_FREQ (48000 * 512) +#define DUAL_CHANNEL 2 + +static struct snd_soc_jack pco_jack; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; + +static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = codec_dai->component; + + dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); + + /* set rt5682 dai fmt */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + dev_err(rtd->card->dev, + "Failed to set rt5682 dai fmt: %d\n", ret); + return ret; + } + + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + PCO_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + + /* Set tdm/i2s1 master bclk ratio */ + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 tdm bclk ratio: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk"); + rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pco_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &pco_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* RT5682 will support only 48K output with 48M mclk */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int acp3x_5682_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + + return rt5682_clk_enable(substream); +} + +static void rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + +static const struct snd_soc_ops acp3x_5682_ops = { + .startup = acp3x_5682_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_max_play_ops = { + .startup = acp3x_max_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap_ops = { + .startup = acp3x_ec_startup, + .shutdown = rt5682_shutdown, +}; + +SND_SOC_DAILINK_DEF(acp3x_i2s, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); +SND_SOC_DAILINK_DEF(acp3x_bt, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2"))); + +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); +SND_SOC_DAILINK_DEF(max, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(cros_ec, + DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); + +static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { + { + .name = "acp3x-5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = acp3x_5682_init, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &acp3x_5682_ops, + SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), + }, + { + .name = "acp3x-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &acp3x_max_play_ops, + SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + }, + { + .name = "acp3x-ec-capture", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, +}; + +static const struct snd_soc_dapm_widget acp3x_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_audio_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_card = { + .name = "acp3xalc5682m98357", + .owner = THIS_MODULE, + .dai_link = acp3x_dai_5682_98357, + .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dapm_widgets = acp3x_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), + .dapm_routes = acp3x_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), + .controls = acp3x_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_controls), +}; + +static int acp3x_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + struct acp3x_platform_info *machine; + + machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card = &acp3x_card; + acp3x_card.dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + if (ret) { + dev_err(&pdev->dev, + "devm_snd_soc_register_card(%s) failed: %d\n", + acp3x_card.name, ret); + return ret; + } + return 0; +} + +static const struct acpi_device_id acp3x_audio_acpi_match[] = { + { "AMDI5682", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); + +static struct platform_driver acp3x_audio = { + .driver = { + .name = "acp3x-alc5682-max98357", + .acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = acp3x_probe, +}; + +module_platform_driver(acp3x_audio); + +MODULE_AUTHOR("akshu.agrawal(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_LICENSE("GPL v2"); -- 2.17.1

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