December 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v4] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v4: - Remove redundant spaces from the commit message - Link to v3: https://lore.kernel.org/all/20241128155926.1774625-1-mmaslanka@google.com/ Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

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[PATCH v3] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

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[PATCH] ASoC: Intel: avs: Remove component->set_jack() on suspend/resume
by Marek Maslanka 02 Dec '24

02 Dec '24

Removed calls to component->set_jack() during suspend/resume to fix an issue where an inserted jack is not detected after resuming from suspend if the jack was inserted before suspend. This problem is caused by calls to the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function during resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

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No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was edited from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : ``` pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered ``` If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. **EDIT** only the alsa-ucm-conf package is faulty somewhere between these two version alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 Workaround: Downgrade to alsa-ucm-conf-1.2.12-1 With this package built by the maintainer @christian-heusel for testing https://pkgbuild.com/\~gromit/alsa-ucm-conf-v1.2.13.r6.g62acef2-1-any.pkg.t… the microphone seens to work but in the logs I have the same error message as above : ``` pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) graph xrun not-triggered (0 suppressed) pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) xrun state:0x7e7d78f97008 pending:6/6 s:69949980861 a:69950023367 f:69950232612 waiting:42506 process:209245 status:triggered ``` although I manage to receive the signal of my microphone. See https://bbs.archlinux.org/viewtopic.php?pid=2211786 for the details of our discussion with @christian-heusel Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Unable to request ucm2 to load BootSequence controls defaults
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #480 was opened from singalsu: I'm stuck with my ucm2 patch to set by default control switches for SOF DRC and beamformer "on" when the controls appear for first time. I don't seem to be able to reset the controls for the SOF sound card to defaults. There's a similar feature in https://github.com/alsa-project/alsa-ucm-conf/blob/62acef2efffdcb75a35f90d2… for "Dmic0 Capture Volume". But I haven't been able to see the volume to go to 70% in my tries to reset the configuration. In Fedora 41 I've tried to remove /var/lib/alsa/asound.state and rename /usr/sbin/alsactl, reboot, and rename it back. But no success. The volume control remains in 100%. Similarly my tries to get DRC and TDFB enabled fail. What is the right trick needed to trigger the BootSequence? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/480 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH AUTOSEL 5.10 4/6] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 02 Dec '24

02 Dec '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 63c1c87993e0e5bb11bced3d8224446a2bc62338 ] This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3aa1cf2624020..3a7f0102b4c5c 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -380,8 +380,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; -- 2.43.0

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No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was edited from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. **EDIT** only the alsa-ucm-conf package is faulty somewhere between these two version alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 Workaround: Downgrade to alsa-ucm-conf-1.2.12-1 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was opened from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek: Enable mute and micmute LED on HP ProBook 430 G8
by Nazar Bilinskyi 01 Dec '24

01 Dec '24

HP ProBook 430 G8 has a mute and micmute LEDs that can be made to work using quirk ALC236_FIXUP_HP_GPIO_LED. Enable already existing quirk. Signed-off-by: Nazar Bilinskyi <nbilinskyi(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2bf5c512e..877c5d20a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10340,6 +10340,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87d3, "HP Laptop 15-gw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x87df, "HP ProBook 430 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f1, "HP ProBook 630 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), -- 2.47.1

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snd_seq_open and pulse audio??
by Mark Lentczner 01 Dec '24

01 Dec '24

My ALSA Seq port manager runs as a service. The service is installed under its own service user, which has the home directory set to some place non-existent and non-writable. When my service starts I see this surprising log message: *amidiminder[136345]: Failed to create secure directory (/var/lib/misc/.config/pulse): No such file or directory* A little investigation reveals that this error is generated by this call, opening the ALSA Seq: serr = snd_seq_open(&seq, "default", SND_SEQ_OPEN_DUPLEX, 0); It's a little surprising that opening ALSA Seq would somehow invoke something in pulse audio, and further that pulse audio would then be trying to create its user directory (.configure/pulse is where pulse audio sticks things under each user's home directory). This seems to be a reprise of this Debian bug from 2013 involving alsa-utils and pulse audio: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=712980 Which was close via this patch to alsa-utils: alsactl: Make homedir configurable <https://mailman.alsa-project.org/hyperkitty/list/alsa-devel@alsa-project.or…> The fix there seems to be adding some mechanism to change the $HOME environment variable to point at /var/run/alsa. I agree with Takashi's reply to that fix: Overriding $HOME sounds scary to me, too! Is there any insight into this... and did that fix stick or was some other method found to deal with this. In my case, I don't understand why just opening the ALSA Seq is causing any pulse audio code to run... but perhaps things are more entwined at a lower level of ALSA lib than I understand. - Mark

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