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

[PATCH] ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op
by Ranjani Sridharan 02 Mar '20

02 Mar '20

Replace the calls to hda_dsp_set_power_state() with the top-level SOF op snd_sof_set_power_state(). Along with this, modify the hda_dsp_resume() function to return the value of snd_sof_set_power_state() directly. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 0e61c27785a3..79ce52c32ef1 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -662,7 +662,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) hda_codec_i915_display_power(sdev, true); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_state.state, target_state.substate); @@ -686,8 +686,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - hda_dsp_set_power_state(sdev, &target_state); - return ret; + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -702,7 +701,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -730,7 +729,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) @@ -753,7 +752,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) hda_codec_i915_display_power(sdev, false); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_dsp_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_dsp_state.state, @@ -781,7 +780,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return ret; } - return hda_dsp_set_power_state(sdev, &target_dsp_state); + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) @@ -849,7 +848,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) return; /* This can fail but error cannot be propagated */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) dev_err_ratelimited(sdev->dev, "error: failed to set DSP state %d substate %d\n", -- 2.17.1

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[PATCH] ASoc: amd: Add DMIC switch capability to machine driver
by Akshu Agrawal 02 Mar '20

02 Mar '20

Switch between DMIC0 and DMIC1 based on recording device selected. This is done by toggling the dmic select gpio. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 53 ++++++++++++++++++++++++---- 1 file changed, 47 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 96fbcd29e3ed..511b8b1722aa 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -12,6 +12,7 @@ #include <sound/jack.h> #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> @@ -27,6 +28,7 @@ static struct snd_soc_jack pco_jack; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; +static struct gpio_desc *dmic_sel; static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { @@ -176,7 +178,7 @@ static int acp3x_max_startup(struct snd_pcm_substream *substream) return rt5682_clk_enable(substream); } -static int acp3x_ec_startup(struct snd_pcm_substream *substream) +static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -185,6 +187,23 @@ static int acp3x_ec_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 0); + + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_dmic1_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); + if (dmic_sel) + gpiod_set_value(dmic_sel, 1); return rt5682_clk_enable(substream); } @@ -204,8 +223,13 @@ static const struct snd_soc_ops acp3x_max_play_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap_ops = { - .startup = acp3x_ec_startup, +static const struct snd_soc_ops acp3x_ec_cap0_ops = { + .startup = acp3x_ec_dmic0_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap1_ops = { + .startup = acp3x_ec_dmic1_startup, .shutdown = rt5682_shutdown, }; @@ -246,12 +270,21 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { SND_SOC_DAILINK_REG(acp3x_bt, max, platform), }, { - .name = "acp3x-ec-capture", - .stream_name = "Capture", + .name = "acp3x-ec-dmic0-capture", + .stream_name = "Capture DMIC0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap0_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, + { + .name = "acp3x-ec-dmic1-capture", + .stream_name = "Capture DMIC1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_capture = 1, - .ops = &acp3x_ec_cap_ops, + .ops = &acp3x_ec_cap1_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, }; @@ -302,6 +335,14 @@ static int acp3x_probe(struct platform_device *pdev) acp3x_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); + + dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); + if (IS_ERR(dmic_sel)) { + dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + PTR_ERR(dmic_sel)); + return PTR_ERR(dmic_sel); + } + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); if (ret) { dev_err(&pdev->dev, -- 2.17.1

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Applied "ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver" to the asoc tree
by Mark Brown 02 Mar '20

02 Mar '20

The patch ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver 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 a79ee2e095c0a60c32d5b1fce39d58e0fc4d9ec5 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Mon, 2 Mar 2020 15:05:22 +0800 Subject: [PATCH] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver snd_soc_dai_driver should set ops in rt1015_dai driver. Also make the two variable static to fix sparse warnings. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20200302070522.48104-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c118d030bd2d..100b8c89d537 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.20.1

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alsa-gobject: add support for i386 build job to workflow of Github Actions
by GitHub pull_request - opened 02 Mar '20

02 Mar '20

alsa-project/alsa-gobject pull request #9 was opened from takaswie: In lxd repository for Ubuntu Cloud Images[1], Ubuntu i386 container is available for applications of Intel386 ABI. This commit uses the container to test build for Intel386 ABI in workflow of Github Actions. I note that no official Fedora i386 container is in dockerhub registry[2]. It's needed to find the way to build application for Intel386 ABI on AMD64 architecture (perhaps multiarch) if we are going to support it. [1] https://cloud-images.ubuntu.com/releases/ [2] https://hub.docker.com/_/fedora Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/9 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/9.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH -next] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver
by YueHaibing 02 Mar '20

02 Mar '20

snd_soc_dai_driver should set ops in rt1015_dai driver. Also make the two variable static to fix sparse warnings. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1015.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b41..100b8c8 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, @@ -857,6 +857,7 @@ struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.7.4

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RE: [alsa-devel] No sound with Realtek ALC298 on a Samsung Galaxy book
by Kailang 02 Mar '20

02 Mar '20

> -----Original Message----- > From: Aurélien Croc <aurelien(a)ap2c.com> > Sent: Saturday, February 29, 2020 1:40 AM > To: alsa-devel(a)alsa-project.org > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] No sound with Realtek ALC298 on a Samsung Galaxy > book > > > Sorry. > > Samsung didn't request this platform need to support Linux. > > I can understand that you can't work on my problem if your client (Samsung) > didn't paid for that. However is it possible to get ALC298 specs in order to > understand how-to configure it (in particular a document which explains its > COEF registers and their meaning) and I will do the job. > Hi, OK. I could try to request spec from our AE. > Thank you in advance. > Regards, > > Aurélien > > > > ------Please consider the environment before printing this e-mail.

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conf: namehint - add noargs to the hint section
by GitHub pull_request - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-lib pull request #31 was opened from perexg: Do not list simple surround devices in the namehint section by default. Fixes: https://github.com/alsa-project/alsa-lib/issues/27 Request URL : https://github.com/alsa-project/alsa-lib/pull/31 Patch URL : https://github.com/alsa-project/alsa-lib/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Remove duplicate surround entries. Fix for #27
by GitHub pull_request - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-lib pull request #30 was opened from DarkWav: Try to remove duplicate surround entries to fix crashes with certain apps by changing the config a bit. This would have multiple benefits. 1. It allows all the bonus devices (such as samplerate) gained from Version 1.2.2 to stay. 2. It still keeps surround devices around, just removes duplicate entries if they exist. 3. It fixes Issue #27 and all Unity3D-based games. I'm not sure about side effects tho, but they seem to be reasonably low from what I've tested. Thanks for taking a look. Request URL : https://github.com/alsa-project/alsa-lib/pull/30 Patch URL : https://github.com/alsa-project/alsa-lib/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158
by GitHub issues - edited 01 Mar '20

01 Mar '20

alsa-project/alsa-ucm-conf issue #14 was edited from JeremyFiggins: Hi, I have the following hardware: > ~> cat /proc/asound/cards > 0 [NVidia ]: HDA-Intel - HDA NVidia > HDA NVidia at 0xf7080000 irq 90 > 1 [Generic ]: HDA-Intel - HD-Audio Generic > HDAudio-Gigabyte-ALC1220DualCodecs and I'm running: > ~> cat /etc/os-release > NAME="openSUSE Tumbleweed" > VERSION="20200228" > ID="opensuse-tumbleweed" > ID_LIKE="opensuse suse" > VERSION_ID="20200228" > PRETTY_NAME="openSUSE Tumbleweed" > ANSI_COLOR="0;32" > CPE_NAME="cpe:/o:opensuse:tumbleweed:20200228" > BUG_REPORT_URL="https://bugs.opensuse.org" > HOME_URL="https://www.opensuse.org/" > LOGO="distributor-logo" Audio was working until a recent update. Now, I have no audio and when starting PulseAudio, I get: > ~> pulseaudio > W: [pulseaudio] pid.c: Stale PID file, overwriting. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. > E: [pulseaudio] alsa-ucm.c: Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158, function device_add_hw_mute_jack(). Aborting. > Aborted (core dumped) I tracked it down to /usr/share/alsa/ucm2/HDA-Intel/HiFi-dual.conf. If I revert the two commits here (https://github.com/alsa-project/alsa-ucm-conf/commits/master/ucm2/HDA-Intel…) and then use the version of HiFi-dual.conf before it was moved, then PulseAudio starts normally and I have sound. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/14 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158
by GitHub issues - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-ucm-conf issue #14 was opened from JeremyFiggins: Hi, I have the following hardware: > ~> cat /proc/asound/cards > 0 [NVidia ]: HDA-Intel - HDA NVidia > HDA NVidia at 0xf7080000 irq 90 > 1 [Generic ]: HDA-Intel - HD-Audio Generic > HDAudio-Gigabyte-ALC1220DualCodecs and I'm running: > ~> cat /etc/os-release > NAME="openSUSE Tumbleweed" > # VERSION="20200228" > ID="opensuse-tumbleweed" > ID_LIKE="opensuse suse" > VERSION_ID="20200228" > PRETTY_NAME="openSUSE Tumbleweed" > ANSI_COLOR="0;32" > CPE_NAME="cpe:/o:opensuse:tumbleweed:20200228" > BUG_REPORT_URL="https://bugs.opensuse.org" > HOME_URL="https://www.opensuse.org/" > LOGO="distributor-logo" Audio was working until a recent update. Now, I have no audio and when starting PulseAudio, I get: > ~> pulseaudio > W: [pulseaudio] pid.c: Stale PID file, overwriting. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. > E: [pulseaudio] alsa-ucm.c: Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158, function device_add_hw_mute_jack(). Aborting. > Aborted (core dumped) I tracked it down to /usr/share/alsa/ucm2/HDA-Intel/HiFi-dual.conf. If I revert the two commits here (https://github.com/alsa-project/alsa-ucm-conf/commits/master/ucm2/HDA-Intel…) and then use the version of HiFi-dual.conf before it was moved, then PulseAudio starts normally and I have sound. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/14 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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