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

Re: linux-next: Tree for Feb 27 (sound/soc/codecs/: SND_SOC_MAX98357A)
by Randy Dunlap 27 Feb '20

27 Feb '20

On 2/26/20 8:22 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200226: > on x86_64: # CONFIG_GPIOLIB is not set SND_SOC_MAX98357A depends on GPIOLIB, which is not enabled. Some driver(s) is selecting GPIOLIB in error. WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && GPIOLIB [=n] Selected by [m]: - SND_SOC_AMD_RV_RT5682_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP3x [=m] && I2C [=y] - SND_SOC_STORM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

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[PATCH] ASoC: SOF - topology - do not change the link trigger order for pre-1.4 firmware
by Jaroslav Kysela 27 Feb '20

27 Feb '20

This patch is for SOF v1.3 firmware. The DSP firmware will crash (DSP oops) without this patch. The 1.4.1 firmare has this issue fixed. The ABI version is used for the comparison, because the firmware version for the firmware files before 1.4.2 was not set properly (git hash was used). BugLink: https://github.com/thesofproject/sof/issues/2102 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/topology.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9f4f8868b386..58ff4766b47b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3111,6 +3111,7 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_private *private = &cfg->priv; struct sof_ipc_dai_config config; struct snd_soc_tplg_hw_config *hw_config; + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; int num_hw_configs; int ret; int i = 0; @@ -3128,9 +3129,12 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, if (!link->no_pcm) { link->nonatomic = true; - /* set trigger order */ - link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; - link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + /* this causes DSP panic on firmware v1.3 */ + if (v->abi_version > SOF_ABI_VER(3, 7, 0)) { + /* set trigger order */ + link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; + link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + } /* nothing more to do for FE dai links */ return 0; -- 2.24.1

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Re: [PATCH] zte:zx-spdif:remove redundant dev_err message
by Mark Brown 27 Feb '20

27 Feb '20

On Thu, Feb 27, 2020 at 11:07:01PM +0800, tangbin wrote: > devm_ioremap_resource has already contains error message, so remove > the redundant dev_err message Please submit patches using subject lines reflecting the style for the subsystem, this makes it easier for people to identify relevant patches. Look at what existing commits in the area you're changing are doing and make sure your subject lines visually resemble what they're doing. There's no need to resubmit to fix this alone.

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[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

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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

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[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

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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

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[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

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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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