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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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Linux Mint Cinnamon 22 Soundblaster AE-9 no sound
by GitHub issues - opened 30 Nov '24

30 Nov '24

alsa-project/alsa-lib issue #424 was opened from Starfoxfs: Hi, i have fresh installed Linux Mint Cinnamon 22 Kernel 6.8.0.49-generic I can´t get the Soundblaster AE-9 to work. Creative Labs has no Drivers for Linux and points to the alsa community. Is there any change to get support for the Soundblaster AE-9 in the future ? I think the big Problem is the external DAC for Headphones from the Soundblaster AE-9. In Windows this "clicks" on startup and activates the DAC, but in Linux this is all dead. Issue URL : https://github.com/alsa-project/alsa-lib/issues/424 Repository URL: https://github.com/alsa-project/alsa-lib

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arecordmidi.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 30 Nov '24

30 Nov '24

Package: alsa-utils Version: 1.2.12-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff: backtrace: file '<stdin>':69 troff:<stdin>:69: warning: trailing space in the line troff: backtrace: file '<stdin>':77 troff:<stdin>:77: warning: trailing space in the line Output from "test-nroff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z ": troff: backtrace: file '<stdin>':69 troff:<stdin>:69: warning: trailing space in the line troff: backtrace: file '<stdin>':77 troff:<stdin>:77: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.10-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

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[tiwai-sound:for-next 6/10] sound/core/ump.c:1259:69: warning: '%s' directive output may be truncated writing up to 11 bytes into a region of size between 4 and 22
by kernel test robot 29 Nov '24

29 Nov '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 9b5f8ee43e48c25fbe1a10163ec04343d750acd0 commit: e29e504e7890b9ee438ca6370d0180d607c473f9 [6/10] ALSA: ump: Indicate the inactive group in legacy substream names config: alpha-allyesconfig (https://download.01.org/0day-ci/archive/20241130/202411300103.FrGuTAYp-lkp@…) compiler: alpha-linux-gcc (GCC) 14.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241130/202411300103.FrGuTAYp-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202411300103.FrGuTAYp-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/ump.c: In function 'fill_substream_names': >> sound/core/ump.c:1259:69: warning: '%s' directive output may be truncated writing up to 11 bytes into a region of size between 4 and 22 [-Wformat-truncation=] 1259 | snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", | ^~ sound/core/ump.c:1259:17: note: 'snprintf' output between 11 and 40 bytes into a destination of size 32 1259 | snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 1260 | idx + 1, name, | ~~~~~~~~~~~~~~ 1261 | ump->groups[idx].active ? "" : " [Inactive]"); | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +1259 sound/core/ump.c 1246 1247 static void fill_substream_names(struct snd_ump_endpoint *ump, 1248 struct snd_rawmidi *rmidi, int dir) 1249 { 1250 struct snd_rawmidi_substream *s; 1251 const char *name; 1252 int idx; 1253 1254 list_for_each_entry(s, &rmidi->streams[dir].substreams, list) { 1255 idx = ump->legacy_mapping[s->number]; 1256 name = ump->groups[idx].name; 1257 if (!*name) 1258 name = ump->info.name; > 1259 snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", 1260 idx + 1, name, 1261 ump->groups[idx].active ? "" : " [Inactive]"); 1262 } 1263 } 1264 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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amd-soundwire: add support for AMD generic legacy machine driver
by GitHub pull_request - opened 29 Nov '24

29 Nov '24

alsa-project/alsa-ucm-conf pull request #478 was opened from perexg: The card components string is similar to Intel's SOF driver like, so we can reuse sof-soundwire configuration and it is logical to reduce future maintenance, because the chips connected to soundwire bus are similar for multiple PCI bridges. Example of the components string: Components : ' cfg-amp:1 hs:rt722 spk:rt722 mic:rt722' Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/478 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/478.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.13-merge-window
by Mark Brown 28 Nov '24

28 Nov '24

The following changes since commit 82ff5abc2edcfba0c0f1a1be807795e2876f46e9: ASoC: hdmi-codec: reorder channel allocation list (2024-11-15 13:43:00 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-merge-window for you to fetch changes up to b682aa788e5f9f1ddacdfbb453e49fd3f4e83721: ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 (2024-11-27 13:45:45 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A pile of driver specific quirks and fixes that came in since the merge window. One of the AMD fixes is a bit broken for some systems, I'm expecting an incremental change to fix that but it seems better overall to merge the rest of the fixes. There's also one small documentation update that seemed sensible to apply now, pointing to the dapm-graph tool. ---------------------------------------------------------------- Alex Far (1): ASoC: amd: yc: fix internal mic on Redmi G 2022 Bard Liao (1): ASoC: SOF: ipc3-topology: Convert the topology pin index to ALH dai index Charles Han (1): ASoC: imx-audmix: Add NULL check in imx_audmix_probe Ilya Zverev (1): ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 Mario Limonciello (1): ASoC: amd: Fix build dependencies for `SND_SOC_AMD_PS` Nícolas F. R. A. Prado (1): ASoC: mediatek: Check num_codecs is not zero to avoid panic during probe Uwe Kleine-König (1): ASoC: amd: yc: Add quirk for microphone on Lenovo Thinkpad T14s Gen 6 21M1CTO1WW Venkata Prasad Potturu (1): ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry anish kumar (1): ASoC: doc: dapm: Add location information for dapm-graph tool liujing (1): ASoC: apple: Fix the wrong format specifier Documentation/sound/soc/dapm.rst | 3 ++ sound/soc/amd/Kconfig | 1 + sound/soc/amd/yc/acp6x-mach.c | 39 +++++++++++++++++++++- sound/soc/apple/mca.c | 2 +- sound/soc/fsl/imx-audmix.c | 3 ++ sound/soc/mediatek/mt8188/mt8188-mt6359.c | 9 +++-- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +-- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 9 +++-- sound/soc/sof/ipc3-topology.c | 26 +++++++++++++-- 9 files changed, 86 insertions(+), 10 deletions(-)

2 1

aplay.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 Nov '24

Package: alsa-utils Version: 1.2.12-1 (Debian testing) Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff: backtrace: file '<stdin>':3 troff:<stdin>:3: warning: trailing space in the line troff: backtrace: file '<stdin>':87 troff:<stdin>:87: warning: trailing space in the line troff: backtrace: file '<stdin>':91 troff:<stdin>:91: warning: trailing space in the line troff: backtrace: file '<stdin>':95 troff:<stdin>:95: warning: trailing space in the line troff: backtrace: file '<stdin>':99 troff:<stdin>:99: warning: trailing space in the line troff: backtrace: file '<stdin>':103 troff:<stdin>:103: warning: trailing space in the line troff: backtrace: file '<stdin>':107 troff:<stdin>:107: warning: trailing space in the line troff: backtrace: file '<stdin>':111 troff:<stdin>:111: warning: trailing space in the line troff: backtrace: file '<stdin>':114 troff:<stdin>:114: warning: trailing space in the line troff: backtrace: file '<stdin>':115 troff:<stdin>:115: warning: trailing space in the line troff: backtrace: file '<stdin>':118 troff:<stdin>:118: warning: trailing space in the line troff: backtrace: file '<stdin>':121 troff:<stdin>:121: warning: trailing space in the line troff: backtrace: file '<stdin>':131 troff:<stdin>:131: warning: trailing space in the line troff: backtrace: file '<stdin>':215 troff:<stdin>:215: warning: trailing space in the line troff: backtrace: file '<stdin>':225 troff:<stdin>:225: warning: trailing space in the line troff: backtrace: file '<stdin>':258 troff:<stdin>:258: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.9-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

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[PATCH] ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry
by Venkata Prasad Potturu 27 Nov '24

27 Nov '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 5426f506b584 ("ASoC: amd: Add support for enabling DMIC on acp6x via _DSD") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 438865d5e376..c301dac8b979 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -523,8 +523,14 @@ static int acp6x_probe(struct platform_device *pdev) struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; struct acpi_device *adev; + acpi_handle handle; + acpi_integer dmic_status; int ret; + bool is_dmic_enable, wov_en; + /* IF WOV entry not found, enable dmic based on AcpDmicConnected entry*/ + is_dmic_enable = false; + wov_en = true; /* check the parent device's firmware node has _DSD or not */ adev = ACPI_COMPANION(pdev->dev.parent); if (adev) { @@ -532,9 +538,19 @@ static int acp6x_probe(struct platform_device *pdev) if (!acpi_dev_get_property(adev, "AcpDmicConnected", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == 1) - platform_set_drvdata(pdev, &acp6x_card); + is_dmic_enable = true; } + handle = ACPI_HANDLE(pdev->dev.parent); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; + + if (is_dmic_enable && wov_en) + platform_set_drvdata(pdev, &acp6x_card); + else + return 0; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.39.2

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[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR, partly fixed but HDMI AC3 passthrough
by schorpp 27 Nov '24

27 Nov '24

>>> >>> [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR >>> >>> ALSA versions 1.0.28 >>> >>> Kernel Driver versions 3.12.40 > Partly FIXED in Longterm Stable Kernel 4.19.324 and using HDMI DEV 1 instead of 0. > > 2 CH PCM HDMI to AVR works now but AC3/DTS passthrough still not or is not recognized by AVR or some wrong AES setting but iec958 ALC886 optical toslink AC3/DTS output to AVR works. > > New BUG: No more 5.1 Analog Output on the CM106 USB Box if AC3 stream is selected. No. Sorry, VDR xineliboutput config error, needs speaker setup 5.1 instead of pass through. y tom > > Your ALSA information is located at http://alsa-project.org/db/?f=5ef8b56450583ba8afbcc705afc02228da4a3072 > > y > tom > >>> >>> Hardware: >>> >>> DFI Mainboard MB330-CRM >>> >>> https://www.dfi.com/de/product/index/273#specification >>> >>> Reproducible Always >>> >>> Steps to reproduce and full report: >>> >>> https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… >> Output of alsa-info.sh is here >> >> http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 >> >>> >>> Software crosscheck with CM106 USB Soundbox iec958: works OK. >>> >>> Where to start in the drivers source code to fix it? >>> >>> y >>> tom >>> >>>

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alsa-ucm-conf update from 1.2.12 to 1.2.13 causes HDMI/DsiplayPort outputs to disappear
by GitHub issues - opened 27 Nov '24

27 Nov '24

alsa-project/alsa-ucm-conf issue #477 was opened from mwsltn: Basic Info: Linux Distro: Arch Kernel Version: 6.12.1 I have very reliably been using a USB-C connected external monitor that has internal speakers represented by the output device: HDMI / DisplayPort 1 After running a package update this AM all the HDMI/DisplayPort outputs disappeared. First I downgraded the kernel which didn't help, then checked the pacman logs to see what else had changed and the list included the following: ``` [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-ucm-conf (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-lib (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-utils (1.2.12-1 -> 1.2.13-1) ``` So I downgrade: ``` [2024-11-26T06:51:22-0600] [PACMAN] Running 'pacman -U file:///var/cache/pacman/pkg/alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-lib-1.2.12-1-x86_64.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-utils-1.2.12-1-x86_64.pkg.tar.zst' [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-ucm-conf (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-lib (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-utils (1.2.13-1 -> 1.2.12-1) ``` Problem fixed. Please let me know if I can provide any other info. Originally I had opened an issue on the `alsa-lib` project and others are reporting the same: https://github.com/alsa-project/alsa-lib/issues/423 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/477 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsactl.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 Nov '24

Package: alsa-utils Version: 1.2.12-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff: backtrace: file '<stdin>':93 troff:<stdin>:93: warning: trailing space in the line troff: backtrace: file '<stdin>':224 troff:<stdin>:224: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.9-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

alsabat.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 27 Nov '24

27 Nov '24

Package: alsa-utils Version: 1.2.12-1 (Debian testing) Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? troff: backtrace: file '<stdin>':175 troff:<stdin>:175: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.9-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

RE: Set pcbeep to default value
by Kailang 26 Nov '24

26 Nov '24

Sorry! Attach again. Add ALSA: hda/realtek: to Title. > -----Original Message----- > From: Kailang > Sent: Tuesday, November 26, 2024 10:36 AM > To: 'Takashi Iwai' <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org>; > 'linux-sound(a)vger.kernel.org' <linux-sound(a)vger.kernel.org> > Subject: RE: Set pcbeep to default value > > OK. > > > -----Original Message----- > > From: Takashi Iwai <tiwai(a)suse.de> > > Sent: Monday, November 25, 2024 10:45 PM > > To: Kailang <kailang(a)realtek.com> > > Cc: Takashi Iwai <tiwai(a)suse.de> > > Subject: Re: Set pcbeep to default value > > > > > > External mail. > > > > > > > > On Mon, 25 Nov 2024 03:00:26 +0100, > > Kailang wrote: > > > > > > > > > > > > > -----Original Message----- > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > Sent: Friday, November 22, 2024 5:34 PM > > > > To: Kailang <kailang(a)realtek.com> > > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > > Subject: Re: Set pcbeep to default value > > > > > > > > > > > > External mail. > > > > > > > > > > > > > > > > On Fri, 22 Nov 2024 08:29:56 +0100, Kailang wrote: > > > > > > > > > > Hi Takashi, > > > > > > > > > > Attach patch will set pcbeep to default value for ALC274. > > > > > > > > Could you resubmit with your Signed-off-by tag? > > > > > > Sorry! Attach again. > > > > You seem to have dropped Cc to alsa-devel ML. > > Also, nowadays we use linux-sound(a)vger.kernel.org for the main ML for > > kernel patches. Care to resubmit to there? > > > > > > thanks, > > > > Takashi

2 1

alsa-lib update from 1.2.12 to 1.2.13 causes HDMI/DsiplayPort outputs to disappear
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-lib issue #423 was opened from mwsltn: Basic Info: Linux Distro: Arch Kernel Version: 6.12.1 I have very reliably been using a USB-C connected external monitor that has internal speakers represented by the output device: `HDMI / DisplayPort 1` After running a package update this AM all the HDMI/DisplayPort outputs disappeared. First I downgraded the kernel which didn't help, then checked the pacman logs to see what else had changed and the list included the following: ``` [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-ucm-conf (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-lib (1.2.12-1 -> 1.2.13-1) [2024-11-26T05:07:01-0600] [ALPM] upgraded alsa-utils (1.2.12-1 -> 1.2.13-1) ``` So I downgrade: ``` [2024-11-26T06:51:22-0600] [PACMAN] Running 'pacman -U file:///var/cache/pacman/pkg/alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-lib-1.2.12-1-x86_64.pkg.tar.zst file:///var/cache/pacman/pkg/alsa-utils-1.2.12-1-x86_64.pkg.tar.zst' [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-ucm-conf (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-lib (1.2.13-1 -> 1.2.12-1) [2024-11-26T06:51:24-0600] [ALPM] downgraded alsa-utils (1.2.13-1 -> 1.2.12-1) ``` Problem fixed. Please let me know if I can provide any other info. Issue URL : https://github.com/alsa-project/alsa-lib/issues/423 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

kernel snd seq bugs: SND_SEQ_EVENT_CLIENT_CHANGE & SND_SEQ_EVENT_PORT_CHANGE
by Mark Lentczner 26 Nov '24

26 Nov '24

I'm the author of amidiminder - a utility that keeps track of desired connections between ALSA Seq (MIDI) ports. I've found a bug in the kernel part of the ALSA Sequencer, and am not sure where or how to report it. Summary: The events: * SND_SEQ_EVENT_CLIENT_CHANGE is never sent * SND_SEQ_EVENT_PORT_CHANGE is not sent on common port changes (name and port info) I have a lot of detail on these bugs, what other software causes them to be a problem (PureData), and where the missing code in the kernel sound/core/seq files is. Before I dump all the details in this forum.... where is the right place for me to report and/or discuss this? - Mark

2 3

Build failures with GCC 15 (-std=gnu23)
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-tools issue #25 was edited from kostadinsh: Some of the programs in alsa-tools fail to build with upcoming GCC 15 which uses `-std=gnu23` by default. This can be reproduced on older GCC and Clang versions by adding `-std=gnu23` to CFLAGS. Downstream Gentoo bug report: https://bugs.gentoo.org/943946 hdajackretask: ``` apply-changes.c:358:10: error: conflicting types for ‘reset_changes_boot’; have ‘gboolean(GError **)’ {aka ‘int(struct _GError **)’} 358 | gboolean reset_changes_boot(GError** err) | ^~~~~~~~~~~~~~~~~~ In file included from apply-changes.c:12: apply-changes.h:12:10: note: previous declaration of ‘reset_changes_boot’ with type ‘gboolean(void)’ {aka ‘int(void)’} 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ main-gtk.c: In function ‘reset_boot_clicked’: main-gtk.c:375:5: error: too many arguments to function ‘reset_changes_boot’ 375 | reset_changes_boot(&err); | ^~~~~~~~~~~~~~~~~~ In file included from main-gtk.c:11: apply-changes.h:12:10: note: declared here 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ ``` envy24control: ``` new_process.c: In function ‘new_process’: new_process.c:67:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 67 | int_stat = signal(SIGINT, SIG_IGN); | ^ new_process.c:68:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 68 | quit_stat = signal(SIGQUIT, SIG_IGN); | ^ new_process.c:69:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 69 | usr2_stat = signal(SIGUSR2, SIG_IGN); | ^ new_process.c:81:24: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 81 | signal(SIGINT, int_stat); | ^~~~~~~~ | | | void (*)(void) In file included from envy24control.h:4: /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:82:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 82 | signal(SIGQUIT, quit_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:83:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 83 | signal(SIGUSR2, usr2_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/25 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Build failures with GCC 15 (-std=gnu23)
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-tools issue #25 was opened from kostadinsh: Some of the programs in alsa-tools fail to build with upcoming GCC 15 which uses `-std=gnu23` by default. This can be reproduced on older GCC and Clang versions by adding `-std=gnu23` to CFLAGS. Downstream Gentoo bug report: https://bugs.gentoo.org/943946 hdajackretask: ``` main-gtk.c: In function ‘reset_boot_clicked’: main-gtk.c:375:5: error: too many arguments to function ‘reset_changes_boot’ 375 | reset_changes_boot(&err); | ^~~~~~~~~~~~~~~~~~ In file included from main-gtk.c:11: apply-changes.h:12:10: note: declared here 12 | gboolean reset_changes_boot(); | ^~~~~~~~~~~~~~~~~~ ``` envy24control: ``` new_process.c: In function ‘new_process’: new_process.c:67:18: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 67 | int_stat = signal(SIGINT, SIG_IGN); | ^ new_process.c:68:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 68 | quit_stat = signal(SIGQUIT, SIG_IGN); | ^ new_process.c:69:19: error: assignment to ‘void (*)(void)’ from incompatible pointer type ‘__sighandler_t’ {aka ‘void (*)(int)’} [-Wincompatible-pointer-types] 69 | usr2_stat = signal(SIGUSR2, SIG_IGN); | ^ new_process.c:81:24: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 81 | signal(SIGINT, int_stat); | ^~~~~~~~ | | | void (*)(void) In file included from envy24control.h:4: /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:82:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 82 | signal(SIGQUIT, quit_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ new_process.c:83:25: error: passing argument 2 of ‘signal’ from incompatible pointer type [-Wincompatible-pointer-types] 83 | signal(SIGUSR2, usr2_stat); | ^~~~~~~~~ | | | void (*)(void) /usr/include/signal.h:88:57: note: expected ‘__sighandler_t’ {aka ‘void (*)(int)’} but argument is of type ‘void (*)(void)’ 88 | extern __sighandler_t signal (int __sig, __sighandler_t __handler) | ~~~~~~~~~~~~~~~^~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/25 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

No built-in microphone in Thinkpad X1 Carbon (Gen 12) after update to v1.2.13
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #476 was opened from iyanmv: I think the title is self-explanatory. After updating from v1.2.12 to v1.2.13 in Arch Linux, the built-in microphone on my laptop is not working. Downgrading to previous versions fixes the issue. In addition, I also noticed that when attached to an external monitor using a docking station, the speakers of the monitor are not listed as before, and only the laptop's built-in speaker appears. I'm running linux 6.12.1 and KDE Plasma 6.2.3. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/476 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

how do I create configure a ucm2 conf file for my card just like an acp file
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #475 was opened from beh-10257: 1_ how do I know if my card is using acp or ucm profile 2_ can ucm config file be availble in user directories 3_ how do i test a ucm profile 4_ if my card is using ucm is there a way to cooverride that file 5_ for more context https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4421 6_ as a tldr I want to take advanatage of indep_hp=yes (IDK why this is not enabled by default its a great feature compared to other platforms after all) either way I want to make hw:0,0 as speaker and hw:0,2 as headphones can a ucm profile replace the indep_hp hint and udev rule and just porceed to seperate them directly also indep_hp seperates speaker and headphones output is there a way to seperate internal mic and headphone speaker also is there a way to put priority for headphones when the jack port is actually put or even disable it entirely and just default back to speaker automatically thats all thanks thats all and thanks further editing will be filled once I have more questions and thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/475 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC225 update depop procedure
by Kailang 26 Nov '24

26 Nov '24

Hi Takashi, ALC225 update depop procedure. Many Thanks. BR, Kailang

2 1

Enable speaker pins for Medion E15443 platform
by Kailang 26 Nov '24

26 Nov '24

Hi Takashi, Attach was fixed speaker no sound issue. BR, Kailang

2 1

Device selection regression in 1.2.13 for LNL
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was edited from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. According to `aplay -l`, the audio card is sofhdadsp. Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Device selection regression in 1.2.13 for LNL
by GitHub issues - edited 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was edited from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. According to `aplay -l`, the audio card is sofhdadsp . Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Device selection regression in 1.2.13 for LNL
by GitHub issues - opened 26 Nov '24

26 Nov '24

alsa-project/alsa-ucm-conf issue #474 was opened from CarelessCaution: On Intel Lunar Lake, there seems to be a regression in alsa-ucm-conf that affects device selection in KDE Plasma and probably in other software. In 1.2.12, there are the options Play HiFi Quality Music and Pro Audio for both input and output. However, when upgrading to 1.2.13, there is only the option for Pro Audio. This problem is immediately remedied by downgrading just the alsa-ucm-conf package on Arch Linux. Before (1.2.12): ![before](https://github.com/user-attachments/assets/c7fa81e0-97f1-45ae-a14e-8549f85cdd27) After (1.2.13): ![after](https://github.com/user-attachments/assets/5867886b-6686-4265-ad68-50c7f53e1a47) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/474 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: Set pcbeep to default value
by Kailang 26 Nov '24

26 Nov '24

OK. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Monday, November 25, 2024 10:45 PM > To: Kailang <kailang(a)realtek.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Subject: Re: Set pcbeep to default value > > > External mail. > > > > On Mon, 25 Nov 2024 03:00:26 +0100, > Kailang wrote: > > > > > > > > > -----Original Message----- > > > From: Takashi Iwai <tiwai(a)suse.de> > > > Sent: Friday, November 22, 2024 5:34 PM > > > To: Kailang <kailang(a)realtek.com> > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > Subject: Re: Set pcbeep to default value > > > > > > > > > External mail. > > > > > > > > > > > > On Fri, 22 Nov 2024 08:29:56 +0100, > > > Kailang wrote: > > > > > > > > Hi Takashi, > > > > > > > > Attach patch will set pcbeep to default value for ALC274. > > > > > > Could you resubmit with your Signed-off-by tag? > > > > Sorry! Attach again. > > You seem to have dropped Cc to alsa-devel ML. > Also, nowadays we use linux-sound(a)vger.kernel.org for the main ML for kernel > patches. Care to resubmit to there? > > > thanks, > > Takashi

1 0

GoXLR not working in 1.2.13
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #473 was opened from AruAki: GoXLR not splitting into multiple channels. `alsaucm -c hw:1 dump text` returns ``` ALSA lib parser.c:270:(parse_syntax_field) Incompatible syntax 7 in USB-Audio.conf ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:1 use case configuration -22 alsaucm: error failed to open sound card hw:1: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/473 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RFC PATCH 0/2] Add support for stacked and parallel memories
by Amit Kumar Mahapatra 25 Nov '24

25 Nov '24

Hello Everyone, Following an email discussion with Miquel regarding the binding changes and overall architecture for implementing support for stacked and parallel memories, I’m sharing this RFC to initiate a discussion on the proposed updates to current bindings and to finalize the implementation architecture. Before diving into the main topic, here is some background on stacked and parallel memories. The AMD QSPI controller supports two advanced connection modes(Stacked and Parallel) which allow the controller to treat two different flashes as one storage. Stacked: Flashes share the same SPI bus, but different CS line, controller driver asserts the CS of the flash to which it needs to communicate. Stacked mode is a software abstraction rather than a controller feature or capability. At any given time, the controller communicates with one of the two connected flash devices, as determined by the requested address and data length. If an operation starts on one flash and ends on the other, the core needs to split it into two separate operations and adjust the data length accordingly. Parallel(Multi-CS): Both the flashes have their separate SPI bus, CS of both the flashes will be asserted/de-asserted at the same time. In this mode data will be split across both the flashes by enabling the STRIPE setting in the controller. Parallel mode is a controller feature where if the STRIPE bit is set then the controller internally handles the data split during data write to the flashes and while reading data from the flash the controller internally merges data from both the flashes before writing to the controller FIFO. If STRIPE is not enabled, then same data will be sent to both the devices. In parallel mode both the flashes should be identical. For more information on the modes please feel free to go through the controller flash interface below [1]. Mirochip QSPI controller[2] also supports "Dual Parallel 8-bit IO mode", but they call it "Twin Quad Mode". Initially in [3] [4] [5] Miquel had tried to extend MTD-CONCAT driver to support Stacked mode, but the bindings were not accepted. So, the MTD-CONCAT approach was dropped and the DT bindings that got accepted [6] [7] [8] that describes the two flash devices as being one. SPI core changes to support the above bindings were added [9]. While adding the support in SPI-NOR Tudor provided additional feedback, leading to a discussion on updating the current stacked and parallel DT bindings. Proposed Solution: The solution has two parts: 1. Update MTD-CONCAT Update MTD-CONCAT to create virtual concatinated mtd devices as defined in the device tree. 2. Add a New Layer Add a new layer between the SPI-NOR and MTD layers to support stacked and parallel configurations. This new layer will be part of spi-nor, located in mtd/spi-nor/, can be included/excluded via Kconfig, will be maintained by AMD and will: - During probing, store information from all connected flashes in stacked or parallel mode and present them as a single device to the MTD layer. - Register callbacks and manage MTD device registration within the new layer instead of spi-nor/core.c. - Make minimal changes in spi-nor/core.c, as stacked and parallel handling will be managed by the new layer on top of SPI-NOR. - Handle odd byte count requests from the MTD layer during flash operations in parallel mode. [1] https://docs.amd.com/r/en-US/am011-versal-acap-trm/QSPI-Flash-Device-Interf… [2] https://ww1.microchip.com/downloads/aemDocuments/documents/MPU32/ProductDoc… [3] https://lore.kernel.org/all/20191113171505.26128-4-miquel.raynal@bootlin.co… [4] https://lore.kernel.org/all/20191127105522.31445-5-miquel.raynal@bootlin.co… [5]https://lore.kernel.org/all/20211112152411.818321-1-miquel.raynal@bootlin… [6] https://github.com/torvalds/linux/commit/f89504300e94524d5d5846ff8b728592ac… [7] https://github.com/torvalds/linux/commit/eba5368503b4291db7819512600fa014ea… [8] https://github.com/torvalds/linux/commit/e2edd1b64f1c79e8abda365149ed62a2a9… [9]https://github.com/torvalds/linux/commit/4d8ff6b0991d5e86b17b235fc46ec62e… Thanks, Amit Amit Kumar Mahapatra (2): dt-bindings: mtd: Add bindings for describing concatinated MTD devices dt-bindings: spi: Update stacked and parallel bindings .../mtd/partitions/fixed-partitions.yaml | 18 +++++++++++++++ .../bindings/mtd/partitions/partitions.yaml | 6 +++++ .../bindings/spi/spi-controller.yaml | 23 +++++++++++++++++-- .../bindings/spi/spi-peripheral-props.yaml | 9 +++----- 4 files changed, 48 insertions(+), 8 deletions(-) -- 2.34.1

6 15

ABI break in 1.2.12 -> 1.2.13
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-lib issue #422 was edited from bbhtt: This seems a bit unusual for a patch version bump. Any idea what this is about? Thanks! ``` ┌───────────────────────────┐ │ ABI Break: libasound.so.2 │ └───────────────────────────┘ Functions changes summary: 1 Removed, 3 Changed (250 filtered out), 35 Added functions Variables changes summary: 0 Removed, 0 Changed, 0 Added variable 1 Removed function: [D] 'function void snd_ump_block_info_set_block_id(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_block_id@@ALSA_1.2.10} 35 Added functions: [A] 'function const unsigned char* snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t*)' {snd_pcm_hw_params_get_sync@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_block(snd_seq_t*, int, const snd_ump_block_info_t*)' {snd_seq_create_ump_block@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_endpoint(snd_seq_t*, const snd_ump_endpoint_info_t*, unsigned int)' {snd_seq_create_ump_endpoint@@ALSA_1.2.13} [A] 'function int snd_seq_has_queue_tempo_base(snd_seq_t*)' {snd_seq_has_queue_tempo_base@@ALSA_0.9} [A] 'function int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t*)' {snd_seq_port_info_get_ump_is_midi1@@ALSA_1.2.13} [A] 'function void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t*, int)' {snd_seq_port_info_set_ump_is_midi1@@ALSA_1.2.13} [A] 'function unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t*)' {snd_seq_queue_tempo_get_tempo_base@@ALSA_1.2.13} [A] 'function void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t*, unsigned int)' {snd_seq_queue_tempo_set_tempo_base@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_clear(snd_ump_block_info_t*)' {snd_ump_block_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_active(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_active@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_card(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_device(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_direction(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_direction@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_first_group(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_first_group@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_flags(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_midi_ci_version(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_midi_ci_version@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_name(snd_ump_block_info_t*, const char*)' {snd_ump_block_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_num_groups(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_num_groups@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_sysex8_streams(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_sysex8_streams@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_ui_hint(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_ui_hint@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_clear(snd_ump_endpoint_info_t*)' {snd_ump_endpoint_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_card(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_device(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_family_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_family_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_flags(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_manufacturer_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_manufacturer_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_model_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_model_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_name(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_num_blocks(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_num_blocks@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_product_id(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_product_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol_caps(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol_caps@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_sw_revision(snd_ump_endpoint_info_t*, const unsigned char*)' {snd_ump_endpoint_info_set_sw_revision@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_version(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_version@@ALSA_1.2.13} [A] 'function int snd_ump_packet_length(unsigned int)' {snd_ump_packet_length@@ALSA_1.2.13} 3 functions with some indirect sub-type change: [C] 'function int __old_snd_pcm_hw_params_get_access(const snd_pcm_hw_params_t*)' at pcm.c:8062:1 has some indirect sub-type changes: Please note that the exported symbol of this function is snd_pcm_hw_params_get_access(a)ALSA_0.9 parameter 1 of type 'const snd_pcm_hw_params_t*' has sub-type changes: in pointed to type 'const snd_pcm_hw_params_t': in unqualified underlying type 'typedef snd_pcm_hw_params_t' at pcm.h:64:1: underlying type 'struct snd_pcm_hw_params' at asound.h:414:1 changed: type size hasn't changed 1 data member insertion: 'unsigned char sync[16]', at offset 4352 (in bits) at asound.h:421:1 1 data member change: type of 'unsigned char reserved[64]' changed: type name changed from 'unsigned char[64]' to 'unsigned char[48]' array type size changed from 512 to 384 array type subrange 1 changed length from 64 to 48 array subrange changed: upper bound of '<anonymous range>[48]' change from '63' to '47' and offset changed from 4352 to 4480 (in bits) (by +128 bits) [C] 'function int snd_ctl_pcm_info(snd_ctl_t*, snd_pcm_info_t*)' at control.c:1219:1 has some indirect sub-type changes: parameter 2 of type 'snd_pcm_info_t*' has sub-type changes: in pointed to type 'typedef snd_pcm_info_t' at pcm.h:48:1: underlying type 'struct snd_pcm_info' at asound.h:345:1 changed: type size hasn't changed 1 data member change: type of 'snd_pcm_sync_id sync' changed: entity changed from 'union snd_pcm_sync_id' to 'unsigned char[16]' and name of 'snd_pcm_info::sync' changed to 'snd_pcm_info::pad1' at asound.h:349:1 [C] 'function int snd_seq_get_queue_tempo(snd_seq_t*, int, snd_seq_queue_tempo_t*)' at seq.c:3880:1 has some indirect sub-type changes: parameter 3 of type 'snd_seq_queue_tempo_t*' has sub-type changes: in pointed to type 'typedef snd_seq_queue_tempo_t' at seq.h:441:1: underlying type 'struct snd_seq_queue_tempo' at asequencer.h:540:1 changed: type size hasn't changed 1 data member insertion: 'unsigned short int tempo_base', at offset 160 (in bits) at asequencer.h:532:1 1 data member change: type of 'char reserved[24]' changed: type name changed from 'char[24]' to 'char[22]' array type size changed from 192 to 176 array type subrange 1 changed length from 24 to 22 array subrange changed: upper bound of '<anonymous range>[22]' change from '23' to '21' and offset changed from 160 to 176 (in bits) (by +16 bits) ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/422 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ABI break in 1.2.12 -> 1.2.13
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-lib issue #422 was opened from bbhtt: This seems a bit unusual for a patch version bump. ``` ┌───────────────────────────┐ │ ABI Break: libasound.so.2 │ └───────────────────────────┘ Functions changes summary: 1 Removed, 3 Changed (250 filtered out), 35 Added functions Variables changes summary: 0 Removed, 0 Changed, 0 Added variable 1 Removed function: [D] 'function void snd_ump_block_info_set_block_id(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_block_id@@ALSA_1.2.10} 35 Added functions: [A] 'function const unsigned char* snd_pcm_hw_params_get_sync(const snd_pcm_hw_params_t*)' {snd_pcm_hw_params_get_sync@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_block(snd_seq_t*, int, const snd_ump_block_info_t*)' {snd_seq_create_ump_block@@ALSA_1.2.13} [A] 'function int snd_seq_create_ump_endpoint(snd_seq_t*, const snd_ump_endpoint_info_t*, unsigned int)' {snd_seq_create_ump_endpoint@@ALSA_1.2.13} [A] 'function int snd_seq_has_queue_tempo_base(snd_seq_t*)' {snd_seq_has_queue_tempo_base@@ALSA_0.9} [A] 'function int snd_seq_port_info_get_ump_is_midi1(const snd_seq_port_info_t*)' {snd_seq_port_info_get_ump_is_midi1@@ALSA_1.2.13} [A] 'function void snd_seq_port_info_set_ump_is_midi1(snd_seq_port_info_t*, int)' {snd_seq_port_info_set_ump_is_midi1@@ALSA_1.2.13} [A] 'function unsigned int snd_seq_queue_tempo_get_tempo_base(const snd_seq_queue_tempo_t*)' {snd_seq_queue_tempo_get_tempo_base@@ALSA_1.2.13} [A] 'function void snd_seq_queue_tempo_set_tempo_base(snd_seq_queue_tempo_t*, unsigned int)' {snd_seq_queue_tempo_set_tempo_base@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_clear(snd_ump_block_info_t*)' {snd_ump_block_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_active(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_active@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_card(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_device(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_direction(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_direction@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_first_group(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_first_group@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_flags(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_midi_ci_version(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_midi_ci_version@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_name(snd_ump_block_info_t*, const char*)' {snd_ump_block_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_num_groups(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_num_groups@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_sysex8_streams(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_sysex8_streams@@ALSA_1.2.13} [A] 'function void snd_ump_block_info_set_ui_hint(snd_ump_block_info_t*, unsigned int)' {snd_ump_block_info_set_ui_hint@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_clear(snd_ump_endpoint_info_t*)' {snd_ump_endpoint_info_clear@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_card(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_card@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_device(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_device@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_family_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_family_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_flags(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_flags@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_manufacturer_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_manufacturer_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_model_id(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_model_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_name(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_name@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_num_blocks(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_num_blocks@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_product_id(snd_ump_endpoint_info_t*, const char*)' {snd_ump_endpoint_info_set_product_id@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_protocol_caps(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_protocol_caps@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_sw_revision(snd_ump_endpoint_info_t*, const unsigned char*)' {snd_ump_endpoint_info_set_sw_revision@@ALSA_1.2.13} [A] 'function void snd_ump_endpoint_info_set_version(snd_ump_endpoint_info_t*, unsigned int)' {snd_ump_endpoint_info_set_version@@ALSA_1.2.13} [A] 'function int snd_ump_packet_length(unsigned int)' {snd_ump_packet_length@@ALSA_1.2.13} 3 functions with some indirect sub-type change: [C] 'function int __old_snd_pcm_hw_params_get_access(const snd_pcm_hw_params_t*)' at pcm.c:8062:1 has some indirect sub-type changes: Please note that the exported symbol of this function is snd_pcm_hw_params_get_access(a)ALSA_0.9 parameter 1 of type 'const snd_pcm_hw_params_t*' has sub-type changes: in pointed to type 'const snd_pcm_hw_params_t': in unqualified underlying type 'typedef snd_pcm_hw_params_t' at pcm.h:64:1: underlying type 'struct snd_pcm_hw_params' at asound.h:414:1 changed: type size hasn't changed 1 data member insertion: 'unsigned char sync[16]', at offset 4352 (in bits) at asound.h:421:1 1 data member change: type of 'unsigned char reserved[64]' changed: type name changed from 'unsigned char[64]' to 'unsigned char[48]' array type size changed from 512 to 384 array type subrange 1 changed length from 64 to 48 array subrange changed: upper bound of '<anonymous range>[48]' change from '63' to '47' and offset changed from 4352 to 4480 (in bits) (by +128 bits) [C] 'function int snd_ctl_pcm_info(snd_ctl_t*, snd_pcm_info_t*)' at control.c:1219:1 has some indirect sub-type changes: parameter 2 of type 'snd_pcm_info_t*' has sub-type changes: in pointed to type 'typedef snd_pcm_info_t' at pcm.h:48:1: underlying type 'struct snd_pcm_info' at asound.h:345:1 changed: type size hasn't changed 1 data member change: type of 'snd_pcm_sync_id sync' changed: entity changed from 'union snd_pcm_sync_id' to 'unsigned char[16]' and name of 'snd_pcm_info::sync' changed to 'snd_pcm_info::pad1' at asound.h:349:1 [C] 'function int snd_seq_get_queue_tempo(snd_seq_t*, int, snd_seq_queue_tempo_t*)' at seq.c:3880:1 has some indirect sub-type changes: parameter 3 of type 'snd_seq_queue_tempo_t*' has sub-type changes: in pointed to type 'typedef snd_seq_queue_tempo_t' at seq.h:441:1: underlying type 'struct snd_seq_queue_tempo' at asequencer.h:540:1 changed: type size hasn't changed 1 data member insertion: 'unsigned short int tempo_base', at offset 160 (in bits) at asequencer.h:532:1 1 data member change: type of 'char reserved[24]' changed: type name changed from 'char[24]' to 'char[22]' array type size changed from 192 to 176 array type subrange 1 changed length from 24 to 22 array subrange changed: upper bound of '<anonymous range>[22]' change from '23' to '21' and offset changed from 160 to 176 (in bits) (by +16 bits) ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/422 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.13 will lead to mute of speakers and headphone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was edited from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… edit: reading more seems like problem is with the new HDMI macro, because other report problems with HDMI output. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.13 will lead to mute of speakers and headphone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - edited 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was edited from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

1.2.13 will lead to mute of speakers and head phone jack on MTL laptop Lenovo Youga Pro 7 14IMH9
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #472 was opened from phush0: Changing version from 1.2.12 to 1.2.13 will lead to not working sound during plug/unplug of the head phone jack or mute during reboot on Arch install. Other ppl with similar laptops have reported same problem. On my version codec is ALC287 with bass speaker output. I can unmute speaker or headphones trough alsamixer, but plugin headphone and the unplugin will lead to both mute. Sometimes reboot will lead to same problem and I have to operate channels by hand again. Similar problems: [https://discuss.cachyos.org/t/sound-problems-kde/4350](https://discuss.cach… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/472 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR, partly fixed but fix broke CM106
by schorpp 25 Nov '24

25 Nov '24

>> >> [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR >> >> ALSA versions 1.0.28 >> >> Kernel Driver versions 3.12.40 Partly FIXED in Longterm Stable Kernel 4.19.324 and using HDMI DEV 1 instead of 0. 2 CH PCM HDMI to AVR works now but AC3/DTS passthrough still not or is not recognized by AVR or some wrong AES setting but iec958 optical out AC3/DTS output to AVR works. New BUG: No more 5.1 Analog Output on the CM106 USB Box if AC3 stream is selected. Your ALSA information is located at http://alsa-project.org/db/?f=5ef8b56450583ba8afbcc705afc02228da4a3072 y tom >> >> Hardware: >> >> DFI Mainboard MB330-CRM >> >> https://www.dfi.com/de/product/index/273#specification >> >> Reproducible Always >> >> Steps to reproduce and full report: >> >> https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… > Output of alsa-info.sh is here > > http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 > >> >> Software crosscheck with CM106 USB Soundbox iec958: works OK. >> >> Where to start in the drivers source code to fix it? >> >> y >> tom >> >>

1 0

Motu D828 (2024) not working again
by GitHub issues - opened 25 Nov '24

25 Nov '24

alsa-project/alsa-ucm-conf issue #471 was opened from beatboxa: I had previously gotten my MOTU D828 working with UCM. I recently used Ubuntu 24.04's software updater, and now my UCM appears to have stopped working. The UCM files are still present in the directories where they previously worked. The output of spa-acp-tool -c 0 -vvvv info now begins with the following: ``` UCM is not supported for this USB device (MOTU 828 at usb-0000:59:00.0-6, high speed @ USB07fd:000e) error: failed to import hw:0 use case configuration -6 UCM is not supported for this USB device (MOTU 828 at usb-0000:59:00.0-6, high speed @ USB07fd:000e) error: failed to import 828 use case configuration -6 UCM not available for card 828 Check for file: /home/beatboxa/.config/alsa-card-profile/mixer/profile-sets/default.conf Check for file: /home/beatboxa/.config/alsa-card-profile/profile-sets/default.conf Check for file: /etc/alsa-card-profile/mixer/profile-sets/default.conf Check for file: /etc/alsa-card-profile/profile-sets/default.conf Loading profile set: /usr/share/alsa-card-profile/mixer/profile-sets/default.conf Parsing configuration file '/usr/share/alsa-card-profile/mixer/profile-sets/default.conf' Parsing configuration file '/usr/share/alsa-card-profile/mixer/profile-sets/9999-custom.conf' Looking at profile input:analog-stereo Checking for recording on Analog Stereo (analog-stereo) Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... ALSA device open 'front:0' capture: 0x5804b63f3d90 snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Float 32 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Float 32 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 32 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 32 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 24 bit Little Endian) failed: Invalid argument snd_pcm_hw_params_set_format(Signed 24 bit Big Endian) failed: Invalid argument snd_pcm_hw_params_set_channels(2) failed: Invalid argument ALSA device close 0x5804b63f3d90 ``` And goes on from there. Previously, my configuration worked, and is tracked in this thread: #https://github.com/alsa-project/alsa-ucm-conf/pull/416 Is there some little thing I need to change to get UCM working again? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/471 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. Running alsactl restore as a runit service, I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro (kernel 6.6) on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. I don't run Pulseaudio or Pipewire, just plain ALSA. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro (kernel 6.6) on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. I don't run Pulseaudio or Pipewire, just plain ALSA. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

amd-soundwire: add support for AMD generic soundwire legacy machine driver
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #470 was edited from vijendarmukunda: Add support for AMD generic legacy(No DSP) machine driver for ACP6.3 platform using RT722 codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

amd-soundwire: add support for AMD generic legacy machine driver
by GitHub pull_request - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #470 was opened from vijendarmukunda: Add support for AMD generic legacy(No DSP) machine driver for ACP6.3 platform using RT722 codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/470.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound and alsactl init errors after updating to 1.2.13
by GitHub issues - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was edited from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

No sound and alsactl init errors after updating to 1.12.13
by GitHub issues - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-utils issue #283 was opened from wastlnd: I think I have kind of a regression after updating to the latest version. I have this error at startup: ``` Found hardware: "acp" "" "" "" "" Hardware is initialized using a generic method ``` And I have no sound at all. I have tried to apply the latest commit fixing the udev rule, but it didn't work. Downgrading to 1.2.12 fixes the issue. I am running an Arch based distro on AMD Ryzen 7 5825U CPU with Radeon graphics. My soundcard is Realtek and I have an AMD microphone array. Issue URL : https://github.com/alsa-project/alsa-utils/issues/283 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

UCM2 Profile for Roland BridgeCastX
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was edited from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

UCM Configuration for Roland BridgeCastX
by GitHub pull_request - edited 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was edited from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Configuration for Roland BridgeCastX
by GitHub pull_request - opened 24 Nov '24

24 Nov '24

alsa-project/alsa-ucm-conf pull request #469 was opened from Konna1337: Added Configuration for the X version of Bridge Cast. Config is based on the non X version but it's wired differently and has two more Hardware Channels. It's for the 2.0 Firmware of the Device. The "Game" channel has a virtual 5.1 mode which doesn't work properly and isn't configured. - Added condition to USB-Audio.conf - Added BridgeCastXV2.conf - Added BridgeCastXV2-Hifi.conf with configuration for channels Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/469.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

TLVs with (size % 4) != 0 make amixer and alsactl crash
by GitHub issues - opened 23 Nov '24

23 Nov '24

alsa-project/alsa-utils issue #282 was opened from geoffreybennett: While trying to figure out how TLVs work, I found that if the length field is not a multiple of 4 then amixer and alsactl keep reading past the end of the buffer until they segfault, because size is unsigned. One example here: ``` diff --git a/amixer/amixer.c b/amixer/amixer.c index 8b8000b..03cc9cb 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -587,7 +587,7 @@ static void decode_tlv(unsigned int spaces, unsigned int *tlv, unsigned int tlv_ #endif default: printf("unk-%u-", type); - while (size > 0) { + while (size >= sizeof(unsigned int)) { printf("0x%08x,", tlv[idx++]); size -= sizeof(unsigned int); } ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/282 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR
by schorpp 23 Nov '24

23 Nov '24

> > [BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR > > ALSA versions 1.0.25...current > > Kernel Driver versions 3.12.40...6.x > > Hardware: > > DFI Mainboard MB330-CRM > > https://www.dfi.com/de/product/index/273#specification > > Reproducible Always > > Steps to reproduce and full report: > > https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… Output of alsa-info.sh is here http://alsa-project.org/db/?f=fc4734deddd9a92331692bf94b15cf84b7722cd4 > > Software crosscheck with CM106 USB Soundbox iec958: works OK. > > Where to start in the drivers source code to fix it? > > y > tom > >

1 0

x870-A no S/PDIF output audio (ALC4080)
by GitHub issues - opened 23 Nov '24

23 Nov '24

alsa-project/alsa-ucm-conf issue #468 was opened from SentryII: Hi all, I've been searching for hours for a fix for this and im posting this as a last resort I've checked under If.realtek-alc4080 in the USB-Audio.conf and the X870-A is not listed IDVendor is 0b05 IDProduct is 1b9b 0b05:1b9b Line-Out works fine, issue is with Soundcard 2 USB-Audio Which is using optical / S/PDIF I've attached the alsa-info output below Thank you for your time [alsa-info.txt](https://github.com/user-attachments/files/17876970/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/468 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH] ASoC: qcom: apq8016_sbc: Fix second argument of apq8016_dai_init()
by Stephan Gerhold 22 Nov '24

22 Nov '24

Hi, On Mon, Nov 18, 2024 at 12:20:58AM +0100, Bert Karwatzki wrote: > Since commit a78a42fb48b8 the second argument of apq8016_dai_init() has > to be an lpass id returned by qdsp6_dai_get_lpass_id(). > > Fixes: a78a42fb48b8 ("ASoC: qcom: apq8016_sbc: Allow routing audio through QDSP6") > > Signed-off-by: Bert Karwatzki <spasswolf(a)web.de> > --- > sound/soc/qcom/apq8016_sbc.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c > index 3023cf180a75..ddfcd4616895 100644 > --- a/sound/soc/qcom/apq8016_sbc.c > +++ b/sound/soc/qcom/apq8016_sbc.c > @@ -150,7 +150,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) > { > struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); > > - return apq8016_dai_init(rtd, cpu_dai->id); > + return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); > } > Hm no, the code is correct the way it is. There are two cases handled by this driver: - ADSP bypass (lpass-* drivers, qcom,apq8016-sbc-sndcard): This one uses the LPASS IDs as DAI IDs, so cpu_dai->id already contains the correct value. - ADSP (qdsp6 drivers, qcom,msm8916-qdsp6-sndcard): This one uses the QDSP6 IDs (e.g. PRIMARY_MI2S_RX/TX instead of MI2S_PRIMARY), so qdsp6_dai_get_lpass_id() is used to translate that to the LPASS IDs. The code you are changing here handles the first case (ADSP bypass), the DAI IDs do not need any translation there. Thanks, Stephan

2 1

Set pcbeep to default value
by Kailang 22 Nov '24

22 Nov '24

Hi Takashi, Attach patch will set pcbeep to default value for ALC274. BR, Kailang

2 1

Re: [PATCH] ASoC: qcom: apq8016_sbc: Fix second argument of apq8016_dai_init()
by Mark Brown 21 Nov '24

21 Nov '24

On Mon, Nov 18, 2024 at 12:20:58AM +0100, Bert Karwatzki wrote: > Since commit a78a42fb48b8 the second argument of apq8016_dai_init() has > to be an lpass id returned by qdsp6_dai_get_lpass_id(). This breaks the build: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c: In function ‘apq8016_sbc_dai_in it’: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:153:38: error: implicit declarat ion of function ‘qdsp6_dai_get_lpass_id’ [-Werror=implicit-function-declaration] 153 | return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); | ^~~~~~~~~~~~~~~~~~~~~~ /build/stage/linux/sound/soc/qcom/apq8016_sbc.c: At top level: /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:165:12: error: static declaratio n of ‘qdsp6_dai_get_lpass_id’ follows non-static declaration 165 | static int qdsp6_dai_get_lpass_id(struct snd_soc_dai *cpu_dai) | ^~~~~~~~~~~~~~~~~~~~~~ /build/stage/linux/sound/soc/qcom/apq8016_sbc.c:153:38: note: previous implicit declaration of ‘qdsp6_dai_get_lpass_id’ with type ‘int()’ 153 | return apq8016_dai_init(rtd, qdsp6_dai_get_lpass_id(cpu_dai)); | ^~~~~~~~~~~~~~~~~~~~~~ cc1: all warnings being treated as errors

1 0

Tiny sound on ASUS Zenbook UM5606 out of the box
by GitHub issues - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-ucm-conf issue #467 was opened from IShashkin: ASUS Zenbook UM5606 doesn't use subwoofers out of the box. So sound is tiny on the laptop. There is a workaround described here https://wiki.archlinux.org/title/ASUS_Zenbook_UM5606#Audio but it's broken in the last alsa release. alsa-info.sh --no-upload [alsa-info.txt.zip](https://github.com/user-attachments/files/17834654/alsa-… Let me know if you need more info. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/467 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

sof-soundwire: cs42l43: Correct CapturePCM and routing
by GitHub pull_request - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-ucm-conf pull request #466 was opened from charleskeepax: For headset microphone capture the correct PCM device to use is "hw:${CardId},1", "hw:${CardId},4" is for built in microphone capture. Adjust the routing as well since :0,1 is connected to data port 2 on the codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/466 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/466.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC225 ALC256 update depop procedure
by Kailang 20 Nov '24

20 Nov '24

Hi Takashi, ALC225 ALC256 update depop procedure. Many Thanks. BR, Kailang

2 1

alsactl: 90-alsa-restore.rules - fix AMD acp-pdm-mach link name acp-dmic-codec
by GitHub pull_request - edited 20 Nov '24

20 Nov '24

alsa-project/alsa-utils pull request #281 was edited from pma1: The link name of AMD ACP digital microphones driver acp-pdm-mach is acp-dmic-codec. fix it to make UCM initialize it again. Fixes: 8116639 ("alsactl: 90-alsa-restore.rules - add support for AMD ACP digital microphone") ``` # arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: acppdmmach [acp-pdm-mach], device 0: (null) dmic-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # ls -la /sys/class/sound/controlC2/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 3 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:47 controlC2 lrwxrwxrwx 1 root root 0 Nov 20 15:47 device -> ../../../acp-pdm-mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 id -r--r--r-- 1 root root 4096 Nov 20 15:47 number drwxr-xr-x 3 root root 0 Nov 20 15:47 pcmC2D0c drwxr-xr-x 2 root root 0 Nov 20 15:50 power lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../../../class/sound -rw-r--r-- 1 root root 4096 Nov 20 15:47 uevent # ls -la /sys/class/sound/controlC2/device/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 4 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:50 acp-dmic-codec lrwxrwxrwx 1 root root 0 Nov 20 15:47 driver -> ../../../../../../bus/platform/drivers/acp_mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 driver_override -r--r--r-- 1 root root 4096 Nov 20 15:50 modalias drwxr-xr-x 2 root root 0 Nov 20 15:50 power drwxr-xr-x 3 root root 0 Nov 20 15:50 sound lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../bus/platform -rw-r--r-- 1 root root 4096 Nov 20 15:50 uevent ``` After the change of link name acp-dmic-codec, the micmute led can work on ThinkPad T14s Gen 6. Request URL : https://github.com/alsa-project/alsa-utils/pull/281 Patch URL : https://github.com/alsa-project/alsa-utils/pull/281.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsactl: 90-alsa-restore.rules - fix AMD acp-pdm-mach link name acp-dmic-codec
by GitHub pull_request - opened 20 Nov '24

20 Nov '24

alsa-project/alsa-utils pull request #281 was opened from pma1: The link name of AMD ACP digital microphones driver acp-pdm-mach is acp-dmic-codec. fix it to make UCM initialize it again. Fixes: 8116639 ("alsactl: 90-alsa-restore.rules - add support for AMD ACP digital microphone") # arecord -l **** List of CAPTURE Hardware Devices **** card 1: Generic_1 [HD-Audio Generic], device 0: ALC257 Analog [ALC257 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: acppdmmach [acp-pdm-mach], device 0: (null) dmic-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # ls -la /sys/class/sound/controlC2/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 3 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:47 controlC2 lrwxrwxrwx 1 root root 0 Nov 20 15:47 device -> ../../../acp-pdm-mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 id -r--r--r-- 1 root root 4096 Nov 20 15:47 number drwxr-xr-x 3 root root 0 Nov 20 15:47 pcmC2D0c drwxr-xr-x 2 root root 0 Nov 20 15:50 power lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../../../class/sound -rw-r--r-- 1 root root 4096 Nov 20 15:47 uevent # ls -la /sys/class/sound/controlC2/device/device/ total 0 drwxr-xr-x 5 root root 0 Nov 20 15:50 . drwxr-xr-x 4 root root 0 Nov 20 15:50 .. drwxr-xr-x 3 root root 0 Nov 20 15:50 acp-dmic-codec lrwxrwxrwx 1 root root 0 Nov 20 15:47 driver -> ../../../../../../bus/platform/drivers/acp_mach -rw-r--r-- 1 root root 4096 Nov 20 15:50 driver_override -r--r--r-- 1 root root 4096 Nov 20 15:50 modalias drwxr-xr-x 2 root root 0 Nov 20 15:50 power drwxr-xr-x 3 root root 0 Nov 20 15:50 sound lrwxrwxrwx 1 root root 0 Nov 20 15:47 subsystem -> ../../../../../../bus/platform -rw-r--r-- 1 root root 4096 Nov 20 15:50 uevent After the change of link name acp-dmic-codec, the micmute led can work on ThinkPad T14s Gen 6. Request URL : https://github.com/alsa-project/alsa-utils/pull/281 Patch URL : https://github.com/alsa-project/alsa-utils/pull/281.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

sof-soundwire: cs42l43-spk: Correct PlaybackPCM and routing
by GitHub pull_request - opened 19 Nov '24

19 Nov '24

alsa-project/alsa-ucm-conf pull request #465 was opened from ujfalusi: For speaker the correct PCM device to use is "hw:${CardId},2", the "hw:${CardId},0" is for headset playback. Adjust the routing as well since with :0,2 the DP6RX1/2 needs to be selected for the speaker. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/465 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/465.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v6.13
by Mark Brown 18 Nov '24

18 Nov '24

The following changes since commit 2ae6da569e34e1d26c5275442d17ffd75fd343b3: ASoC: max9768: Fix event generation for playback mute (2024-11-12 13:16:37 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.13 for you to fetch changes up to 82ff5abc2edcfba0c0f1a1be807795e2876f46e9: ASoC: hdmi-codec: reorder channel allocation list (2024-11-15 13:43:00 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.13 This release was mainly about new drivers, there's a very large batch of new drivers and devices including quite a few from newer vendors which is great to see. Other than the new drivers and the usual routine fixes and enhancements the bulk of the work has been Morimoto-san's continuing work on simplifiying APIs, plus a few other bits: - More API simplifications from Morimoto-san. - Renaming of the sh directory to Renesas to reflect the focus on other architectures. - Factoring out of some of the common code for Realtek devices. - Support for Allwinner H616, AMD ACP 6.3 systems, AWInic AW88081, Cirrus Logic CS32L84, Everest ES8328, Iron Devices SMA1307, Longsoon I2S, NeoFidelity NTP8918 and NTP8835, Philips UDA1342, Qualcomm SM8750, RealTek RT721, and ST Microelectronics STM32MP25. ---------------------------------------------------------------- Advait Dhamorikar (1): ASoC: tlv320adc3xxx: Fix unsigned int compared against 0 Alexey Klimov (1): ASoC: qcom: sm8250: add handling of secondary MI2S clock Amadeusz Sławiński (2): ASoC: Intel: Remove unused code ASoC: Intel: avs: Fix return status of avs_pcm_hw_constraints_init() Andy Shevchenko (1): ASoC: codecs: wcd937x: Remove unused of_gpio.h Anne Onciulescu (1): ASoC: SOF: sof-of-dev: add parameter to override tplg/fw_filename Arnd Bergmann (3): ASoC: loongson: make loongson-i2s.o a separate module ASoC: cs42l84: remove incorrect of_match_ptr() ASoC: sma1307: fix uninitialized variable refence Balamurugan C (1): ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for MTL. Bard Liao (6): soundwire: sdw_intel: include linux/acpi.h ASoC: rt-sdw-common: fix rt_sdca_index_update_bits function parameter description ASoC: Intel: Kconfig: make SND_SOC_ACPI_INTEL_MATCH depend on ACPI ASoC: Intel: Kconfig: select SND_SOC_SDCA by SND_SOC_ACPI_INTEL_SDCA_QUIRKS ASoC: sdca: test adev before calling acpi_dev_for_each_child ASoC: Intel: soc-acpi-intel-lnl-match: add rt712_vb + rt1320 support Biju Das (1): ASoC: sh: rz-ssi: Use SSIFCR_FIFO_RST macro Binbin Zhou (7): ASoC: dt-bindings: Add Everest ES8323 Codec ASoC: codecs: Add support for ES8323 ASoC: dt-bindings: Add NXP uda1342 Codec ASoC: codecs: Add uda1342 codec driver ASoC: dt-bindings: Add Loongson I2S controller ASoC: loongson: Add I2S controller driver as platform device ASoC: loongson: Fix build warning when !CONFIG_PCI Chancel Liu (2): ASoC: imx-card: Set mclk for codec ASoC: imx-card: Add CS42888 support Charles Keepax (4): ASoC: sdw_utils: Add support for exclusion DAI quirks ASoC: sdw_utils: Add a quirk to allow the cs42l43 mic DAI to be ignored ASoC: Intel: sof_sdw: Add quirk for cs42l43 system using host DMICs ASoC: Intel: sof_sdw: Add quirks for some new Lenovo laptops Claudiu Beznea (3): ASoC: da7213: Populate max_register to regmap_config ASoC: da7213: Return directly the value of regcache_sync() ASoC: da7213: Add suspend to RAM support Codrin Ciubotariu (4): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation ASoC: atmel: mchp-spdiftx: Remove interface name from stream_name ASoC: atmel: mchp-spdifrx: Remove interface name from stream_name Colin Ian King (3): ASoC: codecs: aw88395: Fix spelling mistake "unsupport" -> "unsupported" ASoC: codecs: aw88399: Fix spelling mistake "unsupport" -> "unsupported" ASoC: SOF: ipc4-topology: remove redundant assignment to variable ret Cristian Ciocaltea (1): ASoC: dt-bindings: everest,es8328: Document audio graph port Daniel Baluta (1): ASoC: SOF: ipc3: Use standard dev_dbg API Deep Harsora (1): ASoC: Intel: sof_sdw: Add missing quirks from some new Dell Dharageswari R (2): ASoC: Intel: sof_rt5682: Add support for ptl_max98360a_rt5682 ASoC: intel: sof_sdw: add RT722 SDCA card for PTL platform Dr. David Alan Gilbert (1): ASoC: soc-devres: Remove unused devm_snd_soc_register_dai Everest K.C (2): ASoC: rt721-sdca: Clean logically deadcode in rt721-sdca.c ASoC: rt722-sdca: Remove logically deadcode in rt722-sdca.c Fabio Estevam (1): ASoC: dt-bindings: fsl_spdif: Document imx6sl/sx compatible fallback Fei Shao (3): ASoC: dt-bindings: mediatek,mt8188-mt6359: Add mediatek,adsp property ASoC: dt-bindings: everest,es8326: Document interrupt property ASoC: dt-bindings: maxim,max98390: Reference common DAI properties Frank Li (2): ASoC: dt-bindings: fsl-esai: Add power-domains for fsl,imx8qm-esai ASoC: dt-bindings: fsl-esai: allow fsl,imx8qm-esai fallback to fsl,imx6ull-esai Geert Uytterhoeven (2): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio Hao Bui (2): ASoC: da7213: Avoid setting PLL when closing audio stream ASoC: da7213: Extend support for the MCK in range [2, 50] MHz Harshit Mogalapalli (1): ASoC: codecs: fix error code in ntp8835_i2c_probe() Heiko Stuebner (1): ASoC: dt-bindings: rockchip,rk3036-codec: convert to yaml Hendrik v. Raven (2): ASoc: simple-mux: add idle-state support ASoC: dt-bindings: simple-mux: add idle-state property Herve Codina (1): ASoC: fsl: fsl_qmc_audio: Remove the logging when parsing channels Hongbo Li (5): ASoC: improve macro definition on TWL4030_OUTPUT_PGA ASoC: remove unused substream in macro soc_component_mark_pop ASoC: remove unused substream in macro soc_dai_mark_pop ASoC: remove unused substream in macro soc_link_mark_pop ASoC: stm: fix macro definition on STM_SAI_HAS_EXT_SYNC Igor Prusov (6): dt-bindings: vendor-prefixes: Add NeoFidelity, Inc ASoC: codecs: Add NeoFidelity Firmware helpers ASoC: dt-bindings: Add NeoFidelity NTP8918 ASoC: codecs: Add NeoFidelity NTP8918 codec ASoC: dt-bindings: Add NeoFidelity NTP8835 ASoC: codecs: Add NeoFidelity NTP8835 codec Ingyu Jang (2): ASoC: uniphier: Handle regmap_write errors in aio_src_set_param() ASoC: uniphier: Handle regmap_write errors in aio_iecout_set_enable() Jack Yu (6): ASoC: rt-sdw-common: Common functions for Realtek soundwire driver ASoC: rt721-sdca: Add RT721 SDCA driver ASoC: rt721-sdca: Fix issue of warning message ASoC: Intel: Add rt721-sdca support for PTL platform ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable ASoC: rt721-sdca: change interrupt mask from XU to GE James Calligeros (1): ASoC: cs42l84: leverage ring sense IRQs to correctly detect headsets Javier Carrasco (1): ASoC: constify snd_soc_component_driver struct Jerome Brunet (1): ASoC: meson: axg-iface: set continuous rates Jiapeng Chong (1): ASoC: cs42l84: Remove unused including <linux/version.h> John Watts (1): ASoC: test-component: Support continuous rates for test component Jonas Karlman (1): ASoC: hdmi-codec: reorder channel allocation list Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 Krzysztof Kozlowski (5): ASoC: qcom: sm8250: correct typo in shutdown function name ASoC: dt-bindings: qcom: Add SM8750 LPASS macro codecs ASoC: qcom: x1e80100: Support boards with two speakers ASoC: dt-bindings: qcom,sm8250: Add SM8750 sound card ASoC: qcom: sc8280xp Add SM8750 sound card Kuninori Morimoto (24): ASoC: makes snd_soc_set_runtime_hwparams() inline ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check ASoC: rename rtd->num to rtd->id ASoC: fsl: switch to use rtd->id from rtd->num ASoC: meson: switch to use rtd->id from rtd->num ASoC: sh: switch to use rtd->id from rtd->num ASoC: generic: switch to use rtd->id from rtd->num ASoC: remove rtd->num ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() ASoC: cleanup function parameter for rtd and its id ASoC: add symmetric_ prefix for dai->rate/channels/sample_bits ASoC: simple-card-utils: care simple_util_dai for dummy DAI Lad Prabhakar (6): ASoC: rsnd: Refactor port handling with helper for endpoint node selection ASoC: Rename "sh" to "renesas" ASoC: renesas, rsnd: Update file path ASoC: audio-graph-card2: Update comment with renamed file path MAINTAINERS: Add entry for Renesas R-Car and FSI ASoC drivers MAINTAINERS: Add entry for Renesas RZ ASoC driver Liu Jing (1): ASoC: rx651: Use card->dev in replace of the &pdev->dev argument in the dev_err function Mac Chiang (3): ASoC: sdw_utils: Add quirk to exclude amplifier function ASoC: intel: sof_sdw: add quirk for Dell SKU ASoC: Intel: soc-acpi-intel-arl-match: Add rt722 and rt1320 support Macpaul Lin (1): ASoC: dt-bindings: mt6359: Update generic node name and dmic-mode Marcus Cooper (2): ASoC: sun4i-codec: Add support for different DAC FIFOC addresses to quirks ASoC: sun4i-codec: Add playback only flag to quirks Marek Vasut (3): ASoC: max98088: Remove duplicate DACs ASoC: max98088: Add left/right DAC volume control ASoC: max98088: Add headphone mixer switch Mario Limonciello (1): ASoC: amd: acp: Add stream name to ACP PDM DMIC devices Mark Brown (41): ASoC: bcm2835-i2s: Use maple tree register cache ASoC: fsl: Use maple tree register cache ASoC: fsl_micfil: fix and improvement ASoC: amd: acp: refactor acp version ASoC: improve some macro definitions and usages Updates for Atmel SSC DAI ASoC: Updates for mchp-spdif(tx/rx) {ASoC,media}: constify snd_soc_component_driver ASoC: Add NTP8918 and NTP8835 codecs support ASoC: Clean up {hp,mic}-det-gpio handling ASoC: Intel: sof_rt5682: add supports for new ASoC: Merge up v6.12 Fixes and improvements related to amd soundwire ASoC: Intel: add rt722/rt721 support for PTL platform ASoC: SOF: ipc4-topology: Rework the module audio ASoC: Some issues about loongson i2s ASoC/SoundWire: Simply interrupt enabling for Intel ASoC: imx-card: add cs42888 codec support ASoC: Merge up fixes ASoC: add support for some new Lenovo laptops with ASoC/soundwire: add initial support for SDCA ASoC: fsl_xcvr: enable some interrupts ASoC: SOF: core: Module parameter updates ASoC: remove dpcm_xxx flags ASoC: add Allwinner H616 audio codec support ASoC: add CS42L84 codec driver ASoC: Intel: add rt722 and rt1320 support ASoC: codecs: adau1373: drop platform data ASoC: Rename "sh" to "renesas" ASoC: cleasnup rtd and its ID Soundwire dai link init logic refactor ASoC: codecs: Add aw88081 amplifier driver ASoC: Intel: Kconfig: fix undefined symbol: ASoC: SOF: ipc3-loader: Handle PROBE_INFO ext_manifest Add a driver for the Iron Device SMA1307 Amp ASoC: Intel: Fix Kconfig for ACPI=n ASoC: stm32: sai: add stm32mp25 support ASoC: stm32: i2s: add stm32mp25 support ASoC: Merge up fixes Add generic AMD Soundwire machine driver for Legacy(No ASoc: simple-mux: Allow to specify an idle-state Markus Elfring (1): ASoC: tas5805m: Improve a size determination in tas5805m_i2c_probe() Martin Povišer (2): ASoC: cs42l84: Add new codec driver ASoC: dt-bindings: Add CS42L84 codec Masahiro Yamada (1): ASoC: codecs: wcd9335: remove unnecessary MODULE_ALIAS() Murad Masimov (1): ASoC: amd: acp: drop bogus NULL check from i2s_irq_handler Naveen Manohar (2): ASoC: intel/sdw_utils: refactor RT multifunction sdca speaker codecs ASoC: Intel: soc-acpi-intel-ptl-match: Add rt721 support Neil Armstrong (1): ASoC: dt-bindings: realtek,rt5640: Convert to dtschema Nuno Sa (5): ASoC: codecs: adau1372: add match table ASoC: codecs: adau1373: add some kconfig text ASoC: dt-bindings: document the adau1373 Codec ASoC: codecs: adau1373: drop patform_data ASoC: codecs: adau1373: add powerdown gpio Olivier Moysan (6): ASoC: dt-bindings: add stm32mp25 support for sai ASoC: stm32: sai: add stm32mp25 support ASoC: dt-bindings: add stm32mp25 support for i2s ASoC: stm32: i2s: add stm32mp25 support ASoC: dt-bindings: stm32: add missing port property ASoC: stm32: dfsdm: change rate upper limits Peter Ujfalusi (16): ASoC: SOF: ipc4-topology: Drop the 'index' from 'Pin index' of format print ASoC: SOF: ipc4-topology: Use local variables in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Remove redundant check in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Simplify match format print in sof_ipc4_init_input_audio_fmt() ASoC: SOF: ipc4-topology: Use local variables in sof_ipc4_init_output_audio_fmt() ASoC: SOF: ipc4-topology: Simplify code to deal with process modules without output ASoC: SOF: ipc4-topology: Concentrate prints inside of sof_ipc4_init_output_audio_fmt() ASoC: SOF: ipc4-topology: Add helper function to print the module's in/out audio format ASoC: SOF: core: Add debug module parameters to set IPC and boot timeout ASoC: SOF: core: Add fw, tplg and ipc type override parameters ASoC: SOF: ext_manifest: Add missing ext_manifest type for PROBE_INFO ASoC: SOF: ipc3-loader: 'Handle' PROBE_INFO ext_manifest type when parsing ASoC: SOF: Intel: hda-stream: Always use at least two BDLE for transfers ASoC: SOF: Intel: hda: Add support for persistent Code Loader DMA buffers ASoC: Intel: Kconfig: Only select SND_SOC_SDCA if ACPI is enabled ASoC: Intel: Kconfig: Revert make SND_SOC_ACPI_INTEL_MATCH depend on ACPI Pierre-Louis Bossart (12): ASoC: SOF: Intel: hda-mlink: expose unlocked interrupt enable routine ASoC/SoundWire: Intel: lnl: enable interrupts after first power-up/before last power-down ASoC/soundwire: remove sdw_slave_extended_id ASoC: SDCA: add initial module soundwire: slave: lookup SDCA version and functions ASoC: SDCA: add quirk function for RT712_VB match ASoC: rt712-sdca: detect the SMART_MIC function during the probe stage ASoC: soc-acpi: introduce new 'machine check' callback ASoC: sdw_utils: add SmartMic DAI for RT712 VB ASoC: sdw_utils: add SmartMic DAI for RT713 VB ASoC: Intel: soc-acpi: add is_device_rt712_vb() helper ASoC: SOF: Intel: hda: use machine_check() for SoundWire Qiu-ji Chen (1): ASoC: codecs: Fix atomicity violation in snd_soc_component_get_drvdata() Ranjani Sridharan (1): ASoC: SOF: Intel: hda: handle only paused streams in hda_dai_suspend() Ritu Chaudhary (1): ASoC: tegra: Add support for S24_LE audio format Ryan Roberts (1): ASoC: soc-utils: Remove PAGE_SIZE compile-time constant assumption Ryan Walklin (2): ASoC: sun4i-codec: support allwinner H616 codec ASoC: dt-bindings: allwinner: add H616 sun4i audio codec binding Shengjiu Wang (6): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt ASoC: fsl_xcvr: enable interrupt of cmdc status update ASoC: fsl_xcvr: reset RX dpath after wrong preamble ASoC: fsl_mqs: Support accessing registers by scmi interface Shuming Fan (4): ASoC: rt1320: reads patch code from firmware file ASoC: rt1320: fix the range of patch code address ASoC: rt1320: add mic function ASoC: rt722: change the interrupt mask for jack type detection Stanislav Jakubek (2): ASoC: dt-bindings: sprd,pcm-platform: convert to YAML ASoC: dt-bindings: sprd,sc9860-mcdt: convert to YAML Suraj Sonawane (2): ASoC: bcm63xx-pcm-whistler: fix uninit-value in i2s_dma_isr ASoc: SOF: ipc4-pcm: fix uninit-value in sof_ipc4_pcm_dai_link_fixup_rate Tang Bin (7): ASoC: tas2781: Fix redundant parameter assignment ASoC: codecs: Fix error check in es8323_i2c_probe ASoC: mediatek: mt8188: Remove unnecessary variable assignments ASoC: mediatek: mt8188: remove unnecessary variable assignment ASoC: ux500: Remove redundant casts ASoC: mediatek: mt8183: Remove unnecessary variable assignments ASoC: sma1307: Fix invalid logical judgement Venkata Prasad Potturu (1): ASoC: amd: acp: Fix for ACP SOF dmic tplg component load failure Vijendar Mukunda (20): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: update mach_params subsystem_rev field ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver ASoC: amd: acp: fix for inconsistent indenting ASoC: amd: acp: fix for cpu dai index logic ASoC: amd: acp: refactor sof_card_dai_links_create() function ASoC: sdw_utils/intel/amd: refactor dai link init logic ASoC: sdw_utils: Update stream_name in dai_links structure ASoC: amd: acp: add rt722 based soundwire machines ASoC: amd: acp: add RT711, RT714 & RT1316 support for acp 6.3 platform ASoC: amd: ps: add soundwire machines for acp6.3 platform ASoC: amd: acp: move get_acp63_cpu_pin_id() to common file ASoC: amd: acp: add soundwire machine driver for legacy stack ASoC: amd: ps: fix the pcm device numbering for acp 6.3 platform Vitaliy Shevtsov (1): ASoC: nau8821: check regmap_raw_read/regmap_raw_write for failure Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88081" ASoC: codecs: Add aw88081 amplifier driver Yang Li (1): ASoC: codecs: Remove unneeded semicolon Zhang Yi (2): ASoC: codecs: ES8326: Modify the configuration of and micbias ASoC: codecs: ES8326: Reduce pop noise Zhu Jun (1): ASoC: aw88399: Correct error handling in aw_dev_get_dsp_status function anish kumar (2): ASoC: doc: update clock api details ASoC: machine: update documentation .mailmap | 17 +- CREDITS | 58 +- Documentation/admin-guide/LSM/ipe.rst | 7 +- Documentation/admin-guide/kernel-parameters.txt | 2 +- Documentation/admin-guide/mm/transhuge.rst | 2 +- Documentation/admin-guide/pm/cpufreq.rst | 20 +- Documentation/core-api/protection-keys.rst | 38 +- .../bindings/display/elgin,jg10309-01.yaml | 54 + .../bindings/display/mediatek/mediatek,dpi.yaml | 24 +- .../bindings/display/mediatek/mediatek,split.yaml | 19 + .../devicetree/bindings/firmware/arm,scmi.yaml | 2 +- .../devicetree/bindings/iio/adc/adi,ad7380.yaml | 21 + .../devicetree/bindings/iio/dac/adi,ad5686.yaml | 53 +- .../devicetree/bindings/iio/dac/adi,ad5696.yaml | 3 +- .../interrupt-controller/fsl,ls-extirq.yaml | 26 +- .../devicetree/bindings/misc/fsl,qoriq-mc.yaml | 2 +- .../devicetree/bindings/net/brcm,unimac-mdio.yaml | 1 + .../devicetree/bindings/net/xlnx,axi-ethernet.yaml | 2 +- .../bindings/phy/qcom,sc8280xp-qmp-pcie-phy.yaml | 5 +- .../devicetree/bindings/sound/adi,adau1373.yaml | 111 + .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 53 +- .../devicetree/bindings/sound/audio-graph.yaml | 6 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 4 +- .../devicetree/bindings/sound/cirrus,cs42l84.yaml | 56 + .../devicetree/bindings/sound/everest,es8316.yaml | 4 +- .../devicetree/bindings/sound/everest,es8326.yaml | 4 + .../devicetree/bindings/sound/everest,es8328.yaml | 4 + .../devicetree/bindings/sound/fsl,esai.yaml | 28 +- .../devicetree/bindings/sound/fsl,spdif.yaml | 27 +- .../devicetree/bindings/sound/inno-rk3036.txt | 20 - .../bindings/sound/irondevice,sma1307.yaml | 53 + .../bindings/sound/loongson,ls2k1000-i2s.yaml | 68 + .../devicetree/bindings/sound/maxim,max98390.yaml | 8 +- .../bindings/sound/mediatek,mt8188-mt6359.yaml | 7 + .../devicetree/bindings/sound/mt6359.yaml | 10 +- .../bindings/sound/neofidelity,ntp8835.yaml | 73 + .../bindings/sound/neofidelity,ntp8918.yaml | 70 + .../devicetree/bindings/sound/nxp,uda1342.yaml | 42 + .../bindings/sound/qcom,lpass-rx-macro.yaml | 1 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 1 + .../bindings/sound/qcom,lpass-va-macro.yaml | 1 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 1 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 1 + .../devicetree/bindings/sound/realtek,rt5640.yaml | 146 + .../devicetree/bindings/sound/renesas,rsnd.txt | 2 +- .../bindings/sound/rockchip,rk3036-codec.yaml | 58 + 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| 5 +- Documentation/networking/tcp_ao.rst | 20 +- Documentation/process/maintainer-netdev.rst | 17 + Documentation/process/maintainer-soc.rst | 42 +- Documentation/rust/arch-support.rst | 2 +- Documentation/scheduler/sched-ext.rst | 2 +- Documentation/sound/soc/clocking.rst | 12 + Documentation/sound/soc/dpcm.rst | 11 +- Documentation/sound/soc/machine.rst | 26 + Documentation/userspace-api/mseal.rst | 307 +- Documentation/virt/kvm/api.rst | 16 +- Documentation/virt/kvm/locking.rst | 2 +- MAINTAINERS | 247 +- Makefile | 2 +- arch/Kconfig | 26 +- arch/arm/boot/dts/broadcom/bcm2837-rpi-cm3-io3.dts | 2 +- arch/arm/boot/dts/rockchip/rk3036-kylin.dts | 4 +- arch/arm/boot/dts/rockchip/rk3036.dtsi | 14 +- arch/arm64/Kconfig | 1 + arch/arm64/boot/dts/freescale/imx8-ss-lvds0.dtsi | 12 +- arch/arm64/boot/dts/freescale/imx8-ss-vpu.dtsi | 4 +- .../dts/freescale/imx8mp-phyboard-pollux-rdk.dts | 12 + .../freescale/imx8mp-skov-revb-mi1010ait-1cp1.dts | 1 + arch/arm64/boot/dts/freescale/imx8mp.dtsi | 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265 +- kernel/sched/ext.h | 2 +- kernel/sched/fair.c | 52 +- kernel/sched/sched.h | 39 +- kernel/sched/stats.h | 48 +- kernel/sched/syscalls.c | 13 +- kernel/signal.c | 3 +- kernel/task_work.c | 15 +- kernel/time/posix-clock.c | 3 + kernel/time/tick-sched.c | 6 + kernel/trace/bpf_trace.c | 42 +- kernel/trace/fgraph.c | 41 +- kernel/trace/ring_buffer.c | 53 +- kernel/trace/trace.c | 10 +- kernel/trace/trace_eprobe.c | 7 +- kernel/trace/trace_fprobe.c | 6 +- kernel/trace/trace_kprobe.c | 6 +- kernel/trace/trace_probe.c | 2 +- kernel/trace/trace_selftest.c | 2 +- kernel/trace/trace_uprobe.c | 13 +- kernel/ucount.c | 9 +- lib/Kconfig.debug | 2 +- lib/buildid.c | 5 + lib/codetag.c | 3 + lib/crypto/mpi/mpi-mul.c | 2 +- lib/iov_iter.c | 25 +- lib/maple_tree.c | 14 +- lib/objpool.c | 18 +- lib/slub_kunit.c | 2 +- mm/Kconfig | 1 - mm/damon/core.c | 42 +- mm/damon/tests/sysfs-kunit.h | 1 + mm/filemap.c | 2 +- mm/gup.c | 33 +- mm/huge_memory.c | 71 +- mm/internal.h | 55 +- mm/kasan/init.c | 8 +- mm/kasan/kasan_test_c.c | 27 - mm/khugepaged.c | 6 +- mm/memcontrol-v1.c | 25 + mm/memcontrol.c | 9 +- mm/memory.c | 32 +- mm/migrate.c | 9 +- mm/mlock.c | 9 +- mm/mmap.c | 229 +- mm/mprotect.c | 2 +- mm/mremap.c | 11 +- mm/nommu.c | 9 +- mm/numa_memblks.c | 2 +- mm/page_alloc.c | 26 +- mm/page_io.c | 4 +- mm/pagewalk.c | 16 +- mm/rmap.c | 9 +- mm/secretmem.c | 4 +- mm/shmem.c | 12 +- mm/shrinker.c | 8 +- mm/slab_common.c | 33 +- mm/sparse-vmemmap.c | 5 + mm/swap.c | 4 +- mm/swapfile.c | 58 +- mm/vma.c | 37 +- mm/vma.h | 32 +- mm/vmscan.c | 110 +- mm/zswap.c | 1 - net/9p/Kconfig | 2 + net/9p/client.c | 12 +- net/bluetooth/af_bluetooth.c | 25 + net/bluetooth/bnep/core.c | 3 +- net/bluetooth/hci_conn.c | 3 + net/bluetooth/hci_core.c | 24 +- net/bluetooth/hci_sync.c | 30 +- net/bluetooth/iso.c | 24 +- net/bluetooth/rfcomm/sock.c | 2 - net/bluetooth/sco.c | 18 +- net/bpf/test_run.c | 1 + net/bridge/br_netfilter_hooks.c | 5 + net/bridge/br_netlink.c | 6 +- net/bridge/br_private.h | 5 +- net/bridge/br_vlan.c | 19 +- net/core/dev.c | 4 + net/core/dst.c | 17 +- net/core/filter.c | 50 +- net/core/rtnetlink.c | 33 +- net/core/sock_map.c | 12 + net/dsa/user.c | 11 +- net/ipv4/devinet.c | 35 +- net/ipv4/inet_connection_sock.c | 21 +- net/ipv4/ip_tunnel.c | 2 +- net/ipv4/netfilter/nft_fib_ipv4.c | 4 +- net/ipv4/tcp_bpf.c | 7 +- net/ipv4/tcp_input.c | 42 +- net/ipv4/tcp_output.c | 5 +- net/ipv4/udp.c | 4 +- net/ipv4/xfrm4_policy.c | 40 +- net/ipv6/netfilter/nf_reject_ipv6.c | 15 +- net/ipv6/netfilter/nft_fib_ipv6.c | 5 +- net/ipv6/udp.c | 4 +- net/ipv6/xfrm6_policy.c | 31 +- net/l2tp/l2tp_netlink.c | 4 +- net/mac80211/Kconfig | 2 +- net/mac80211/cfg.c | 25 +- net/mac80211/ieee80211_i.h | 10 +- net/mac80211/key.c | 42 +- net/mac80211/link.c | 7 +- net/mac80211/main.c | 2 + net/mctp/af_mctp.c | 6 +- net/mctp/device.c | 30 +- net/mctp/neigh.c | 31 +- net/mctp/route.c | 33 +- net/mpls/af_mpls.c | 32 +- net/mptcp/mib.c | 3 + net/mptcp/mib.h | 3 + net/mptcp/mptcp_pm_gen.c | 1 - net/mptcp/pm_netlink.c | 6 +- net/mptcp/pm_userspace.c | 3 +- net/mptcp/protocol.c | 26 +- net/mptcp/protocol.h | 1 + net/mptcp/subflow.c | 17 +- net/netfilter/nf_bpf_link.c | 7 +- net/netfilter/nf_tables_api.c | 41 +- net/netfilter/nft_payload.c | 3 + net/netfilter/x_tables.c | 2 +- net/netfilter/xt_CHECKSUM.c | 33 +- net/netfilter/xt_CLASSIFY.c | 16 +- net/netfilter/xt_CONNSECMARK.c | 36 +- net/netfilter/xt_CT.c | 106 +- net/netfilter/xt_IDLETIMER.c | 59 +- net/netfilter/xt_LED.c | 39 +- net/netfilter/xt_NFLOG.c | 36 +- net/netfilter/xt_RATEEST.c | 39 +- net/netfilter/xt_SECMARK.c | 27 +- net/netfilter/xt_TRACE.c | 36 +- net/netfilter/xt_addrtype.c | 15 +- net/netfilter/xt_cluster.c | 33 +- net/netfilter/xt_connbytes.c | 4 +- net/netfilter/xt_connlimit.c | 39 +- net/netfilter/xt_connmark.c | 28 +- net/netfilter/xt_mark.c | 42 +- net/netlink/af_netlink.c | 3 +- net/netlink/genetlink.c | 28 +- net/phonet/pn_netlink.c | 28 +- net/rxrpc/ar-internal.h | 2 +- net/rxrpc/conn_client.c | 4 + net/rxrpc/io_thread.c | 10 +- net/rxrpc/local_object.c | 2 +- net/rxrpc/sendmsg.c | 10 +- net/sched/act_api.c | 23 +- net/sched/cls_api.c | 1 + net/sched/sch_api.c | 9 +- net/sched/sch_generic.c | 8 +- net/sched/sch_taprio.c | 21 +- net/sctp/sm_statefuns.c | 2 +- net/sctp/socket.c | 18 +- net/smc/af_smc.c | 4 +- net/smc/smc_ib.c | 8 +- net/smc/smc_inet.c | 11 + net/smc/smc_pnet.c | 6 +- net/smc/smc_wr.c | 6 +- net/socket.c | 7 +- net/sunrpc/svc.c | 11 +- net/sunrpc/xprtrdma/ib_client.c | 1 + net/sunrpc/xprtsock.c | 1 + net/vmw_vsock/virtio_transport.c | 8 +- net/vmw_vsock/virtio_transport_common.c | 14 +- net/vmw_vsock/vsock_bpf.c | 8 - net/wireless/core.c | 8 + net/wireless/nl80211.c | 8 +- net/wireless/scan.c | 4 + net/xfrm/xfrm_device.c | 11 +- net/xfrm/xfrm_policy.c | 53 +- net/xfrm/xfrm_user.c | 10 +- rust/kernel/device.rs | 15 +- rust/kernel/firmware.rs | 2 +- scripts/Kconfig.include | 3 + scripts/Makefile.compiler | 14 +- scripts/Makefile.package | 7 +- scripts/kconfig/menu.c | 13 +- scripts/mod/file2alias.c | 12 +- scripts/mod/sumversion.c | 5 +- scripts/package/builddeb | 2 +- scripts/package/install-extmod-build | 6 +- scripts/package/mkdebian | 10 +- scripts/rustc-llvm-version.sh | 22 + security/ipe/Kconfig | 19 + security/ipe/policy.c | 18 +- security/keys/keyring.c | 7 +- security/keys/trusted-keys/trusted_dcp.c | 9 +- security/security.c | 10 +- sound/Kconfig | 2 +- sound/core/ump.c | 2 +- sound/firewire/amdtp-stream.c | 3 + sound/firewire/tascam/amdtp-tascam.c | 2 +- sound/hda/intel-sdw-acpi.c | 33 +- sound/pci/hda/Kconfig | 2 +- sound/pci/hda/patch_conexant.c | 21 +- sound/pci/hda/patch_cs8409.c | 5 +- sound/pci/hda/patch_realtek.c | 162 +- sound/soc/Kconfig | 3 +- sound/soc/Makefile | 3 +- sound/soc/amd/acp-da7219-max98357a.c | 20 +- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/Kconfig | 29 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-i2s.c | 38 +- sound/soc/amd/acp/acp-legacy-common.c | 24 +- sound/soc/amd/acp/acp-legacy-mach.c | 8 +- sound/soc/amd/acp/acp-mach-common.c | 63 +- sound/soc/amd/acp/acp-mach.h | 12 +- sound/soc/amd/acp/acp-pci.c | 7 +- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 18 +- sound/soc/amd/acp/acp-rembrandt.c | 4 +- sound/soc/amd/acp/acp-renoir.c | 4 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 486 ++ sound/soc/amd/acp/acp-sdw-mach-common.c | 64 + sound/soc/amd/acp/acp-sdw-sof-mach.c | 104 +- sound/soc/amd/acp/acp-sof-mach.c | 6 +- sound/soc/amd/acp/acp63.c | 4 +- sound/soc/amd/acp/acp70.c | 14 +- sound/soc/amd/acp/acp_common.h | 19 + sound/soc/amd/acp/amd-acp63-acpi-match.c | 54 + sound/soc/amd/acp/amd.h | 9 +- sound/soc/amd/acp/soc_amd_sdw_common.h | 4 + sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 + sound/soc/amd/vangogh/acp5x-mach.c | 6 - sound/soc/atmel/atmel_ssc_dai.c | 5 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/bcm2835-i2s.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 6 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/Kconfig | 69 +- sound/soc/codecs/Makefile | 20 + sound/soc/codecs/adau1372-i2c.c | 1 + sound/soc/codecs/adau1372-spi.c | 1 + sound/soc/codecs/adau1372.c | 8 + sound/soc/codecs/adau1372.h | 1 + sound/soc/codecs/adau1373.c | 200 +- sound/soc/codecs/aw88081.c | 1087 ++++ sound/soc/codecs/aw88081.h | 286 + sound/soc/codecs/aw88395/aw88395_device.c | 2 +- sound/soc/codecs/aw88395/aw88395_lib.c | 2 +- sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/cs42l84.c | 1111 ++++ sound/soc/codecs/cs42l84.h | 210 + sound/soc/codecs/da7213.c | 27 +- sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/da7219.c | 9 +- sound/soc/codecs/es8323.c | 792 +++ sound/soc/codecs/es8323.h | 78 + sound/soc/codecs/es8326.c | 20 +- sound/soc/codecs/hdmi-codec.c | 140 +- sound/soc/codecs/max98088.c | 86 +- sound/soc/codecs/nau8821.c | 9 +- sound/soc/codecs/ntp8835.c | 480 ++ sound/soc/codecs/ntp8918.c | 397 ++ sound/soc/codecs/ntpfw.c | 137 + sound/soc/codecs/ntpfw.h | 23 + sound/soc/codecs/pcm186x.c | 4 +- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/rt-sdw-common.c | 238 + sound/soc/codecs/rt-sdw-common.h | 66 + sound/soc/codecs/rt1320-sdw.c | 3668 ++--------- sound/soc/codecs/rt1320-sdw.h | 6 + sound/soc/codecs/rt712-sdca-sdw.c | 1 + sound/soc/codecs/rt712-sdca.c | 38 +- sound/soc/codecs/rt712-sdca.h | 1 + sound/soc/codecs/rt721-sdca-sdw.c | 546 ++ sound/soc/codecs/rt721-sdca-sdw.h | 150 + sound/soc/codecs/rt721-sdca.c | 1545 +++++ sound/soc/codecs/rt721-sdca.h | 269 + sound/soc/codecs/rt722-sdca-sdw.c | 12 +- sound/soc/codecs/rt722-sdca.c | 15 +- sound/soc/codecs/simple-mux.c | 39 +- sound/soc/codecs/sma1307.c | 2049 ++++++ sound/soc/codecs/sma1307.h | 444 ++ sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 35 +- sound/soc/codecs/tas5805m.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 2 +- sound/soc/codecs/twl4030.c | 12 +- sound/soc/codecs/uda1342.c | 347 + sound/soc/codecs/uda1342.h | 78 + sound/soc/codecs/wcd9335.c | 1 - sound/soc/codecs/wcd937x.c | 1 - sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl-asoc-card.c | 32 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 51 +- sound/soc/fsl/fsl_mqs.c | 41 + sound/soc/fsl/fsl_qmc_audio.c | 2 - sound/soc/fsl/fsl_xcvr.c | 94 +- sound/soc/fsl/fsl_xcvr.h | 5 + sound/soc/fsl/imx-audmix.c | 18 +- sound/soc/fsl/imx-card.c | 69 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 16 +- sound/soc/generic/test-component.c | 2 +- sound/soc/intel/Kconfig | 8 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 +- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_rt5682.c | 15 + sound/soc/intel/boards/sof_sdw.c | 117 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/intel/common/Makefile | 6 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 63 + sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 65 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 58 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 75 +- .../soc/intel/common/soc-acpi-intel-sdca-quirks.c | 42 + .../soc/intel/common/soc-acpi-intel-sdca-quirks.h | 14 + sound/soc/intel/common/sst-dsp-priv.h | 101 - sound/soc/intel/common/sst-dsp.c | 250 - sound/soc/intel/common/sst-dsp.h | 61 - sound/soc/intel/common/sst-ipc.c | 294 - sound/soc/intel/common/sst-ipc.h | 86 - sound/soc/loongson/Kconfig | 32 +- sound/soc/loongson/Makefile | 9 +- sound/soc/loongson/loongson_i2s.c | 5 + sound/soc/loongson/loongson_i2s_plat.c | 185 + sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 +- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 34 +- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 7 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 1 - sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 5 +- sound/soc/mediatek/mt8188/mt8188-dai-pcm.c | 2 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 +- sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 6 +- sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +- sound/soc/meson/axg-card.c | 6 +- sound/soc/meson/axg-tdm-interface.c | 12 +- sound/soc/meson/axg-tdm.h | 2 - sound/soc/meson/gx-card.c | 2 +- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sm8250.c | 12 +- sound/soc/qcom/x1e80100.c | 40 +- sound/soc/{sh => renesas}/Kconfig | 0 sound/soc/{sh => renesas}/Makefile | 0 sound/soc/{sh => renesas}/dma-sh7760.c | 0 sound/soc/{sh => renesas}/fsi.c | 0 sound/soc/{sh => renesas}/hac.c | 0 sound/soc/{sh => renesas}/migor.c | 0 sound/soc/{sh => renesas}/rcar/Makefile | 0 sound/soc/{sh => renesas}/rcar/adg.c | 0 sound/soc/{sh => renesas}/rcar/cmd.c | 0 sound/soc/{sh => renesas}/rcar/core.c | 32 +- sound/soc/{sh => renesas}/rcar/ctu.c | 0 sound/soc/{sh => renesas}/rcar/debugfs.c | 0 sound/soc/{sh => renesas}/rcar/dma.c | 0 sound/soc/{sh => renesas}/rcar/dvc.c | 0 sound/soc/{sh => renesas}/rcar/gen.c | 0 sound/soc/{sh => renesas}/rcar/mix.c | 0 sound/soc/{sh => renesas}/rcar/rsnd.h | 0 sound/soc/{sh => renesas}/rcar/src.c | 0 sound/soc/{sh => renesas}/rcar/ssi.c | 0 sound/soc/{sh => renesas}/rcar/ssiu.c | 0 sound/soc/{sh => renesas}/rz-ssi.c | 6 +- sound/soc/{sh => renesas}/sh7760-ac97.c | 0 sound/soc/{sh => renesas}/siu.h | 0 sound/soc/{sh => renesas}/siu_dai.c | 0 sound/soc/{sh => renesas}/siu_pcm.c | 0 sound/soc/{sh => renesas}/ssi.c | 0 sound/soc/samsung/odroid.c | 11 +- sound/soc/sdca/Kconfig | 11 + sound/soc/sdca/Makefile | 5 + sound/soc/sdca/sdca_device.c | 67 + sound/soc/sdca/sdca_functions.c | 177 + sound/soc/sdw_utils/Makefile | 3 +- sound/soc/sdw_utils/soc_sdw_rt712_sdca.c | 48 - sound/soc/sdw_utils/soc_sdw_rt722_sdca.c | 41 - sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 90 + sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c | 8 + sound/soc/sdw_utils/soc_sdw_utils.c | 80 +- sound/soc/soc-acpi.c | 30 +- sound/soc/soc-component.c | 14 +- sound/soc/soc-compress.c | 28 +- sound/soc/soc-core.c | 70 +- sound/soc/soc-dai.c | 14 +- sound/soc/soc-devres.c | 37 - sound/soc/soc-link.c | 10 +- sound/soc/soc-pcm.c | 152 +- sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/soc-utils.c | 4 +- sound/soc/sof/amd/acp-common.c | 3 +- sound/soc/sof/core.c | 64 + sound/soc/sof/intel/hda-dai.c | 4 + sound/soc/sof/intel/hda-dsp.c | 5 - sound/soc/sof/intel/hda-loader.c | 100 +- sound/soc/sof/intel/hda-mlink.c | 18 + sound/soc/sof/intel/hda-stream.c | 32 +- sound/soc/sof/intel/hda.c | 27 +- sound/soc/sof/intel/hda.h | 14 +- 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2 1

[PATCH v3] ASoC: hdmi-codec: reorder channel allocation list
by Christian Hewitt 15 Nov '24

15 Nov '24

From: Jonas Karlman <jonas(a)kwiboo.se> The ordering in hdmi_codec_get_ch_alloc_table_idx() results in wrong channel allocation for a number of cases, e.g. when ELD reports FL|FR|LFE|FC|RL|RR or FL|FR|LFE|FC|RL|RR|RC|RLC|RRC: ca_id 0x01 with speaker mask FL|FR|LFE is selected instead of ca_id 0x03 with speaker mask FL|FR|LFE|FC for 4 channels and ca_id 0x04 with speaker mask FL|FR|RC gets selected instead of ca_id 0x0b with speaker mask FL|FR|LFE|FC|RL|RR for 6 channels Fix this by reordering the channel allocation list with most specific speaker masks at the top. Signed-off-by: Jonas Karlman <jonas(a)kwiboo.se> Signed-off-by: Christian Hewitt <christianshewitt(a)gmail.com> --- Changes since v2: - No changes, resending to ensure patch formatting is good Changes since v1: - Squash content from an additional related patch authored by Jonas since the original 2019 v1 submission [0] - Minor rewording in the description (nothing creditworthy) [0] https://patchwork.kernel.org/project/alsa-devel/patch/HE1PR06MB4011885AED9F… NB: This patch has been included with most Allwinner, Amlogic and Rockchip LibreELEC distro images since 2019 but has been somewhat forgotten about (despite the occasional nag from me to Jonas who is super-busy). However in the last month or so adding it to LibreELEC x86-64 and Rasbperry Pi images has also solved several 3.1/4.0 speaker placement problems reported in forums so it's time to send v2 on Jonas' behalf. sound/soc/codecs/hdmi-codec.c | 140 +++++++++++++++++++--------------- 1 file changed, 77 insertions(+), 63 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 74caae52e127..d9df29a26f4f 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -185,84 +185,97 @@ static const struct snd_pcm_chmap_elem hdmi_codec_8ch_chmaps[] = { /* * hdmi_codec_channel_alloc: speaker configuration available for CEA * - * This is an ordered list that must match with hdmi_codec_8ch_chmaps struct + * This is an ordered list where ca_id must exist in hdmi_codec_8ch_chmaps * The preceding ones have better chances to be selected by * hdmi_codec_get_ch_alloc_table_idx(). */ static const struct hdmi_codec_cea_spk_alloc hdmi_codec_channel_alloc[] = { { .ca_id = 0x00, .n_ch = 2, - .mask = FL | FR}, - /* 2.1 */ - { .ca_id = 0x01, .n_ch = 4, - .mask = FL | FR | LFE}, - /* Dolby Surround */ + .mask = FL | FR }, + { .ca_id = 0x03, .n_ch = 4, + .mask = FL | FR | LFE | FC }, { .ca_id = 0x02, .n_ch = 4, .mask = FL | FR | FC }, - /* surround51 */ + { .ca_id = 0x01, .n_ch = 4, + .mask = FL | FR | LFE }, { .ca_id = 0x0b, .n_ch = 6, - .mask = FL | FR | LFE | FC | RL | RR}, - /* surround40 */ - { .ca_id = 0x08, .n_ch = 6, - .mask = FL | FR | RL | RR }, - /* surround41 */ - { .ca_id = 0x09, .n_ch = 6, - .mask = FL | FR | LFE | RL | RR }, - /* surround50 */ + .mask = FL | FR | LFE | FC | RL | RR }, { .ca_id = 0x0a, .n_ch = 6, .mask = FL | FR | FC | RL | RR }, - /* 6.1 */ - { .ca_id = 0x0f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | RC }, - /* surround71 */ + { .ca_id = 0x09, .n_ch = 6, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 6, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 6, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 6, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 6, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 6, + .mask = FL | FR | RC }, { .ca_id = 0x13, .n_ch = 8, .mask = FL | FR | LFE | FC | RL | RR | RLC | RRC }, - /* others */ - { .ca_id = 0x03, .n_ch = 8, - .mask = FL | FR | LFE | FC }, - { .ca_id = 0x04, .n_ch = 8, - .mask = FL | FR | RC}, - { .ca_id = 0x05, .n_ch = 8, - .mask = FL | FR | LFE | RC }, - { .ca_id = 0x06, .n_ch = 8, - .mask = FL | FR | FC | RC }, - { .ca_id = 0x07, .n_ch = 8, - .mask = FL | FR | LFE | FC | RC }, - { .ca_id = 0x0c, .n_ch = 8, - .mask = FL | FR | RC | RL | RR }, - { .ca_id = 0x0d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RC }, - { .ca_id = 0x0e, .n_ch = 8, - .mask = FL | FR | FC | RL | RR | RC }, - { .ca_id = 0x10, .n_ch = 8, - .mask = FL | FR | RL | RR | RLC | RRC }, - { .ca_id = 0x11, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, { .ca_id = 0x12, .n_ch = 8, .mask = FL | FR | FC | RL | RR | RLC | RRC }, - { .ca_id = 0x14, .n_ch = 8, - .mask = FL | FR | FLC | FRC }, - { .ca_id = 0x15, .n_ch = 8, - .mask = FL | FR | LFE | FLC | FRC }, - { .ca_id = 0x16, .n_ch = 8, - .mask = FL | FR | FC | FLC | FRC }, - { .ca_id = 0x17, .n_ch = 8, - .mask = FL | FR | LFE | FC | FLC | FRC }, - { .ca_id = 0x18, .n_ch = 8, - .mask = FL | FR | RC | FLC | FRC }, - { .ca_id = 0x19, .n_ch = 8, - .mask = FL | FR | LFE | RC | FLC | FRC }, - { .ca_id = 0x1a, .n_ch = 8, - .mask = FL | FR | RC | FC | FLC | FRC }, - { .ca_id = 0x1b, .n_ch = 8, - .mask = FL | FR | LFE | RC | FC | FLC | FRC }, - { .ca_id = 0x1c, .n_ch = 8, - .mask = FL | FR | RL | RR | FLC | FRC }, - { .ca_id = 0x1d, .n_ch = 8, - .mask = FL | FR | LFE | RL | RR | FLC | FRC }, { .ca_id = 0x1e, .n_ch = 8, .mask = FL | FR | FC | RL | RR | FLC | FRC }, - { .ca_id = 0x1f, .n_ch = 8, - .mask = FL | FR | LFE | FC | RL | RR | FLC | FRC }, + { .ca_id = 0x11, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RLC | RRC }, + { .ca_id = 0x1d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | FLC | FRC }, + { .ca_id = 0x10, .n_ch = 8, + .mask = FL | FR | RL | RR | RLC | RRC }, + { .ca_id = 0x1c, .n_ch = 8, + .mask = FL | FR | RL | RR | FLC | FRC }, + { .ca_id = 0x0f, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR | RC }, + { .ca_id = 0x1b, .n_ch = 8, + .mask = FL | FR | LFE | RC | FC | FLC | FRC }, + { .ca_id = 0x0e, .n_ch = 8, + .mask = FL | FR | FC | RL | RR | RC }, + { .ca_id = 0x1a, .n_ch = 8, + .mask = FL | FR | RC | FC | FLC | FRC }, + { .ca_id = 0x0d, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR | RC }, + { .ca_id = 0x19, .n_ch = 8, + .mask = FL | FR | LFE | RC | FLC | FRC }, + { .ca_id = 0x0c, .n_ch = 8, + .mask = FL | FR | RC | RL | RR }, + { .ca_id = 0x18, .n_ch = 8, + .mask = FL | FR | RC | FLC | FRC }, + { .ca_id = 0x17, .n_ch = 8, + .mask = FL | FR | LFE | FC | FLC | FRC }, + { .ca_id = 0x16, .n_ch = 8, + .mask = FL | FR | FC | FLC | FRC }, + { .ca_id = 0x15, .n_ch = 8, + .mask = FL | FR | LFE | FLC | FRC }, + { .ca_id = 0x14, .n_ch = 8, + .mask = FL | FR | FLC | FRC }, + { .ca_id = 0x0b, .n_ch = 8, + .mask = FL | FR | LFE | FC | RL | RR }, + { .ca_id = 0x0a, .n_ch = 8, + .mask = FL | FR | FC | RL | RR }, + { .ca_id = 0x09, .n_ch = 8, + .mask = FL | FR | LFE | RL | RR }, + { .ca_id = 0x08, .n_ch = 8, + .mask = FL | FR | RL | RR }, + { .ca_id = 0x07, .n_ch = 8, + .mask = FL | FR | LFE | FC | RC }, + { .ca_id = 0x06, .n_ch = 8, + .mask = FL | FR | FC | RC }, + { .ca_id = 0x05, .n_ch = 8, + .mask = FL | FR | LFE | RC }, + { .ca_id = 0x04, .n_ch = 8, + .mask = FL | FR | RC }, + { .ca_id = 0x03, .n_ch = 8, + .mask = FL | FR | LFE | FC }, + { .ca_id = 0x02, .n_ch = 8, + .mask = FL | FR | FC }, + { .ca_id = 0x01, .n_ch = 8, + .mask = FL | FR | LFE }, }; struct hdmi_codec_priv { @@ -371,7 +384,8 @@ static int hdmi_codec_chmap_ctl_get(struct snd_kcontrol *kcontrol, struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol); struct hdmi_codec_priv *hcp = info->private_data; - map = info->chmap[hcp->chmap_idx].map; + if (hcp->chmap_idx != HDMI_CODEC_CHMAP_IDX_UNKNOWN) + map = info->chmap[hcp->chmap_idx].map; for (i = 0; i < info->max_channels; i++) { if (hcp->chmap_idx == HDMI_CODEC_CHMAP_IDX_UNKNOWN) -- 2.34.1

1 0

Fix JackControl name
by GitHub pull_request - opened 15 Nov '24

15 Nov '24

alsa-project/alsa-ucm-conf pull request #464 was opened from lbilli: Not sure whether it makes a difference but that is the name reported by `amixer events`. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/464 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/464.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/6] Add generic AMD Soundwire machine driver for Legacy(No
by Vijendar Mukunda 14 Nov '24

14 Nov '24

This patch series add SoundWire machines for RT711, RT714, RT1316 and RT722 codecs for ACP 6.3 platform. Also, it adds a generic SoundWire machine driver code for legacy(No DSP) stack. Vijendar Mukunda (6): ASoC: amd: acp: add rt722 based soundwire machines ASoC: amd: acp: add RT711, RT714 & RT1316 support for acp 6.3 platform ASoC: amd: ps: add soundwire machines for acp6.3 platform ASoC: amd: acp: move get_acp63_cpu_pin_id() to common file ASoC: amd: acp: add soundwire machine driver for legacy stack ASoC: amd: ps: fix the pcm device numbering for acp 6.3 platform sound/soc/amd/acp/Kconfig | 29 ++ sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 486 +++++++++++++++++++++++ sound/soc/amd/acp/acp-sdw-mach-common.c | 64 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 49 +-- sound/soc/amd/acp/amd-acp63-acpi-match.c | 54 +++ sound/soc/amd/acp/soc_amd_sdw_common.h | 4 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/pci-ps.c | 1 + sound/soc/amd/ps/ps-sdw-dma.c | 2 + 10 files changed, 646 insertions(+), 48 deletions(-) create mode 100644 sound/soc/amd/acp/acp-sdw-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-sdw-mach-common.c -- 2.34.1

2 7

[GIT PULL] ASoC fixes for v6.12-rc7
by Mark Brown 14 Nov '24

14 Nov '24

The following changes since commit 48b86532c10128cf50c854a90c2d5b1410f4012d: ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits (2024-11-07 15:35:52 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc7 for you to fetch changes up to 2ae6da569e34e1d26c5275442d17ffd75fd343b3: ASoC: max9768: Fix event generation for playback mute (2024-11-12 13:16:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 Some last updates for v6.12, one quirk plus a couple of fixes. One is a minor fix for a relatively obscure driver and the other is a relatively important fix for boot hangs with some audio graph based cards. ---------------------------------------------------------------- Deep Harsora (1): ASoC: intel: sof_sdw: add quirk for Dell SKU John Watts (1): ASoC: audio-graph-card2: Purge absent supplies for device tree nodes Mark Brown (1): ASoC: max9768: Fix event generation for playback mute sound/soc/codecs/max9768.c | 11 +++++++++-- sound/soc/generic/audio-graph-card2.c | 3 +++ sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 3 files changed, 20 insertions(+), 2 deletions(-)

2 1

src/Versions.in.in: Update *_tempo_base name
by GitHub pull_request - edited 14 Nov '24

14 Nov '24

alsa-project/alsa-lib pull request #421 was edited from nvinson: Change @SYMBOL_PREFIX@snd_has_tempo_base to @SYMBOL_PREFIX@snd_has_queue_tempo_base. Starting with version 1.2.13, alsa-lib fails to link with ld.lld-19 due to "version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined". Per commit 769d1db1b0a213a39c7e59c0d1d724e7f45b1ac3 the correct name for the symbol is @SYMBOL_PREFIX@snd_has_queue_tempo_base; therefore, update src/Vesions.in.in to match. Fixes bug #420 Fixes Gentoo bug 943399 (https://bugs.gentoo.org/943399) Request URL : https://github.com/alsa-project/alsa-lib/pull/421 Patch URL : https://github.com/alsa-project/alsa-lib/pull/421.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

src/Versions.in.in: Update *_tempo_base name
by GitHub pull_request - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-lib pull request #421 was opened from nvinson: Change @SYMBOL_PREFIX@snd_has_tempo_base to @SYMBOL_PREFIX@snd_has_queue_tempo_base. Starting with version 1.2.13, alsa-lib fails to link with ld.lld-19 due to "version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined". Per commit 769d1db1b0a213a39c7e59c0d1d724e7f45b1ac3 the correct name for the symbol is @SYMBOL_PREFIX@snd_has_queue_tempo_base; therefore, update src/Vesions.in.in to match. Fixes bug 420 Fixes Gentoo bug 943399 (https://bugs.gentoo.org/943399) Request URL : https://github.com/alsa-project/alsa-lib/pull/421 Patch URL : https://github.com/alsa-project/alsa-lib/pull/421.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib-1.2.13: ld.lld: error: version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined
by GitHub issues - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-lib issue #420 was opened from nvinson: libasound.so fails to link with ld.lld with error ```ld.lld: error: version script assignment of 'ALSA_1.2.13' to symbol 'snd_seq_has_tempo_base' failed: symbol not defined``` Bug: [Gentoo bug 943399](https://bugs.gentoo.org/9433399) Issue URL : https://github.com/alsa-project/alsa-lib/issues/420 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

udevd[142]: GOTO 'alsa_restore_std' has no matching label in: '/usr/lib/udev/rules.d/90-alsa-restore.rules'
by GitHub issues - opened 14 Nov '24

14 Nov '24

alsa-project/alsa-utils issue #280 was opened from sxzzsf: There is no 'alsa_restore_std' label in the udev rule file, is it misspelled alsa_restore_go? Issue URL : https://github.com/alsa-project/alsa-utils/issues/280 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

UCM2: Intel: sof-hda-dsp: Fix handling of empty sys_vendor string
by GitHub pull_request - opened 13 Nov '24

13 Nov '24

alsa-project/alsa-ucm-conf pull request #463 was opened from singalsu: The mistake in UCM syntax caused in alsaucm start error: ALSA lib ucm_cond.c:367:(if_eval) unknown If.Condition.Type Fixes: 13022a97711d ("sof-hda-dsp: Fix the case where sysfs dmi sys_vendor attribute is not set") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/463 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/463.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

memory leakage detected in alsa-lib-1.2.9 in Linux-5.10.24
by GitHub issues - edited 13 Nov '24

13 Nov '24

alsa-project/alsa-lib issue #419 was edited from wangtao13: I wrote an ALSA application using alsa-lib-1.2.9, the main logic of the application is ``` static int init_pcm_play(snd_pcm_t **playback_handle...) { snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format; snd_pcm_uframes_t frames, periodsize; int ret; //1. openPCM， if (0 > snd_pcm_open(playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) { return -1; } //2. snd_pcm_hw_params_t if (0 > snd_pcm_hw_params_malloc (&hw_params)) { return -1; } //3. hw_params if (0 > snd_pcm_hw_params_any (*playback_handle, hw_params)) { return -1; } //4. if (0 > snd_pcm_hw_params_set_access (*playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) { return -1; } //5. SND_PCM_FORMAT_U8,8 if (8 == bits_per_sample) { format = SND_PCM_FORMAT_U8; } if (16 == bits_per_sample) { format = SND_PCM_FORMAT_S16_LE; } if (0 > snd_pcm_hw_params_set_format (*playback_handle, hw_params, format)) { return -1; } //6. if (0 > snd_pcm_hw_params_set_rate_near (*playback_handle, hw_params, &rate, 0)) { return -1; } //7. if (0 > snd_pcm_hw_params_set_channels(*playback_handle, hw_params, channels)) { return -1; } frames = 64; periodsize = frames; ret = snd_pcm_hw_params_set_buffer_size_near(*playback_handle, hw_params, &periodsize); if (ret < 0) { printf("Unable to set buffer size %li : %s\n", frames * 2, snd_strerror(ret)); } periodsize /= channels; ret = snd_pcm_hw_params_set_period_size_near(*playback_handle, hw_params, &periodsize, 0); if (ret < 0) { printf("Unable to set period size %li : %s\n", periodsize, snd_strerror(ret)); } //8. set hw_params if (0 > snd_pcm_hw_params (*playback_handle, hw_params)) { return -1; } snd_pcm_uframes_t temp; snd_pcm_hw_params_get_buffer_size(hw_params, &temp); printf("Got buffer size: %ld\n", temp); snd_pcm_hw_params_get_period_size(hw_params, &temp, 0); printf("Got period size: %ld\n", temp); snd_pcm_hw_params_free(hw_params); ret = snd_pcm_prepare(playback_handle); if (ret < 0) { printf("snd_pcm_prepare err\n"); } return ret; } int sound_play(playback_handle, ...) { rc = snd_pcm_writei(playback_handle, buf, buflen); if (rc < 0) { if (rc == -EAGAIN) { usleep(1000); return -1; } snd_pcm_prepare(playback_handle); return -1; } else if (rc == 0) { usleep(10 * 1000); } return 0; } int app_exit(....) { snd_pcm_close(*playback_handle); snd_config_update_free_global(); } #define TEST_LOOPS 2 /* > 1, can trigger memory leakage */ int main_test() { init_pcm_play(&playback_handle, ...., ..); for (int i=0; i<2; i++) { pcm_exit(); init_pcm_play(&playback_handle); sound_play(playback_handle) } return 0; } ``` ``` ==5176==ERROR: LeakSanitizer: detected memory leaks Direct leak of 12792 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Direct leak of 12 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) Indirect leak of 151552 byte(s) in 37 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x76281230 in snd_pcm_mmap (/usr/lib/libasound.so.2+0x61230) Indirect leak of 37392 byte(s) in 114 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Indirect leak of 31200 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b91b8 in snd_pcm_softvol_open (/usr/lib/libasound.so.2+0x991b8) Indirect leak of 15808 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d1c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d1c) Indirect leak of 10032 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b4e28 (/usr/lib/libasound.so.2+0x94e28) Indirect leak of 5168 byte(s) in 38 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x7627e644 (/usr/lib/libasound.so.2+0x5e644) Indirect leak of 3344 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7627e4e0 (/usr/lib/libasound.so.2+0x5e4e0) Indirect leak of 1976 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762939ac in snd_pcm_plug_open (/usr/lib/libasound.so.2+0x739ac) Indirect leak of 1824 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d3c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d3c) Indirect leak of 1596 byte(s) in 38 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762c09b8 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa09b8) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b7c58 (/usr/lib/libasound.so.2+0x97c58) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762512d4 (/usr/lib/libasound.so.2+0x312d4) Indirect leak of 1520 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762c0978 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa0978) Indirect leak of 1419 byte(s) in 153 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x7626c47c (/usr/lib/libasound.so.2+0x4c47c) Indirect leak of 468 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762572e8 in snd_ctl_hw_open (/usr/lib/libasound.so.2+0x372e8) Indirect leak of 304 byte(s) in 76 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x762690c4 (/usr/lib/libasound.so.2+0x490c4) Indirect leak of 273 byte(s) in 1 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x77569f3c in av_buffer_realloc (/usr/lib/libavutil.so.56+0x19f3c) Indirect leak of 195 byte(s) in 39 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762512fc (/usr/lib/libasound.so.2+0x312fc) Indirect leak of 28 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) SUMMARY: AddressSanitizer: 280023 byte(s) leaked in 997 allocation(s). ``` If the `TEST_LOOPS` is defined as 1, there is NO memory leakage reported by ASAN. Issue URL : https://github.com/alsa-project/alsa-lib/issues/419 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

memory leakage detected in alsa-lib-1.2.9 in Linux-5.10.24
by GitHub issues - opened 13 Nov '24

13 Nov '24

alsa-project/alsa-lib issue #419 was opened from wangtao13: I wrote an ALSA application using alsa-lib-1.2.9, the main logic of the application is ``` static int init_pcm_play(snd_pcm_t **playback_handle...) { snd_pcm_hw_params_t *hw_params; snd_pcm_format_t format; snd_pcm_uframes_t frames, periodsize; int ret; //1. openPCM， if (0 > snd_pcm_open(playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0)) { return -1; } //2. snd_pcm_hw_params_t if (0 > snd_pcm_hw_params_malloc (&hw_params)) { return -1; } //3. hw_params if (0 > snd_pcm_hw_params_any (*playback_handle, hw_params)) { return -1; } //4. if (0 > snd_pcm_hw_params_set_access (*playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) { return -1; } //5. SND_PCM_FORMAT_U8,8 if (8 == bits_per_sample) { format = SND_PCM_FORMAT_U8; } if (16 == bits_per_sample) { format = SND_PCM_FORMAT_S16_LE; } if (0 > snd_pcm_hw_params_set_format (*playback_handle, hw_params, format)) { return -1; } //6. if (0 > snd_pcm_hw_params_set_rate_near (*playback_handle, hw_params, &rate, 0)) { return -1; } //7. if (0 > snd_pcm_hw_params_set_channels(*playback_handle, hw_params, channels)) { return -1; } frames = 64; periodsize = frames; ret = snd_pcm_hw_params_set_buffer_size_near(*playback_handle, hw_params, &periodsize); if (ret < 0) { printf("Unable to set buffer size %li : %s\n", frames * 2, snd_strerror(ret)); } periodsize /= channels; ret = snd_pcm_hw_params_set_period_size_near(*playback_handle, hw_params, &periodsize, 0); if (ret < 0) { printf("Unable to set period size %li : %s\n", periodsize, snd_strerror(ret)); } //8. set hw_params if (0 > snd_pcm_hw_params (*playback_handle, hw_params)) { return -1; } snd_pcm_uframes_t temp; snd_pcm_hw_params_get_buffer_size(hw_params, &temp); printf("Got buffer size: %ld\n", temp); snd_pcm_hw_params_get_period_size(hw_params, &temp, 0); printf("Got period size: %ld\n", temp); snd_pcm_hw_params_free(hw_params); ret = snd_pcm_prepare(playback_handle); if (ret < 0) { printf("snd_pcm_prepare err\n"); } return ret; } int sound_play(playback_handle, ...) { rc = snd_pcm_writei(playback_handle, buf, buflen); if (rc < 0) { if (rc == -EAGAIN) { usleep(1000); return -1; } snd_pcm_prepare(playback_handle); return -1; } else if (rc == 0) { usleep(10 * 1000); } return 0; } int app_exit(....) { snd_pcm_close(*playback_handle); snd_config_update_free_global(); } int main_test() { init_pcm_play(&playback_handle, ...., ..); for (int i=0; i<2; i++) { pcm_exit(); init_pcm_play(&playback_handle); sound_play(playback_handle) } return 0; } ``` ``` ==5176==ERROR: LeakSanitizer: detected memory leaks Direct leak of 12792 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Direct leak of 12 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) Indirect leak of 151552 byte(s) in 37 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x76281230 in snd_pcm_mmap (/usr/lib/libasound.so.2+0x61230) Indirect leak of 37392 byte(s) in 114 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7626c458 (/usr/lib/libasound.so.2+0x4c458) Indirect leak of 31200 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b91b8 in snd_pcm_softvol_open (/usr/lib/libasound.so.2+0x991b8) Indirect leak of 15808 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d1c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d1c) Indirect leak of 10032 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b4e28 (/usr/lib/libasound.so.2+0x94e28) Indirect leak of 5168 byte(s) in 38 object(s) allocated from: #0 0x77900acc in __interceptor_malloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:86 #1 0x7627e644 (/usr/lib/libasound.so.2+0x5e644) Indirect leak of 3344 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x7627e4e0 (/usr/lib/libasound.so.2+0x5e4e0) Indirect leak of 1976 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762939ac in snd_pcm_plug_open (/usr/lib/libasound.so.2+0x739ac) Indirect leak of 1824 byte(s) in 76 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x76280d3c in snd_pcm_mmap (/usr/lib/libasound.so.2+0x60d3c) Indirect leak of 1596 byte(s) in 38 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762c09b8 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa09b8) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762b7c58 (/usr/lib/libasound.so.2+0x97c58) Indirect leak of 1560 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762512d4 (/usr/lib/libasound.so.2+0x312d4) Indirect leak of 1520 byte(s) in 38 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762c0978 in snd_timer_hw_open (/usr/lib/libasound.so.2+0xa0978) Indirect leak of 1419 byte(s) in 153 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x7626c47c (/usr/lib/libasound.so.2+0x4c47c) Indirect leak of 468 byte(s) in 39 object(s) allocated from: #0 0x77900d24 in __interceptor_calloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:95 #1 0x762572e8 in snd_ctl_hw_open (/usr/lib/libasound.so.2+0x372e8) Indirect leak of 304 byte(s) in 76 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x762690c4 (/usr/lib/libasound.so.2+0x490c4) Indirect leak of 273 byte(s) in 1 object(s) allocated from: #0 0x779011a4 in __interceptor_realloc /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:105 #1 0x77569f3c in av_buffer_realloc (/usr/lib/libavutil.so.56+0x19f3c) Indirect leak of 195 byte(s) in 39 object(s) allocated from: #0 0x7786d584 in __interceptor_strdup /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_interceptors.cc:405 #1 0x762512fc (/usr/lib/libasound.so.2+0x312fc) Indirect leak of 28 byte(s) in 1 object(s) allocated from: #0 0x77901ad8 in __interceptor_posix_memalign /home/users/at/work/arm-gnu-toolchain/arm-gcc/libsanitizer/asan/asan_malloc_linux.cc:155 #1 0x7758b3f4 in av_malloc (/usr/lib/libavutil.so.56+0x3b3f4) SUMMARY: AddressSanitizer: 280023 byte(s) leaked in 997 allocation(s). ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/419 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] Revert "aplay: fix S24_LE wav header"
by Shengjiu Wang 13 Nov '24

13 Nov '24

S24_LE means the lower 24bit in 32bit space has valid data. So previous code is correct. The commit causes regression, then revert it. This reverts commit 9cdcbe0e932b97fac88a6195f87647159a6c7194. Fixes: 9cdcbe0e932b ("aplay: fix S24_LE wav header") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- aplay/aplay.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index c58c2e9..d11e63a 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2719,11 +2719,11 @@ static void begin_wave(int fd, size_t cnt) case SND_PCM_FORMAT_S16_LE: bits = 16; break; - case SND_PCM_FORMAT_S24_LE: /* S24_LE is 24 bits stored in 32 bit width with 8 bit padding */ case SND_PCM_FORMAT_S32_LE: - case SND_PCM_FORMAT_FLOAT_LE: + case SND_PCM_FORMAT_FLOAT_LE: bits = 32; break; + case SND_PCM_FORMAT_S24_LE: case SND_PCM_FORMAT_S24_3LE: bits = 24; break; -- 2.34.1

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | ### Problem statement The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ### Relevant files & the such * [alsa-info info dump](https://github.com/user-attachments/files/17721762/alsa-info.txt) * Screenshot of the other USB device * ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) ### Relevant files & the such [alsa-info.txt](https://github.com/user-attachments/files/17721762/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up on | Relevant PRs | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | Related issues | | ------------------| ---------------- | ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | #434 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs (edit: *for the ALC408x*) are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Clevo Platfrom with ALC255
by Kailang 12 Nov '24

12 Nov '24

Hi Takashi, Attach was let Clevo platform with ALC255 enable headset Mic. BR, Kailang

2 1

[PATCH AUTOSEL 5.4 4/5] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index b7dc9d3192597..e8cd58c0838ee 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -318,7 +318,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.4 3/5] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 7e965848796c3..b7dc9d3192597 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -379,8 +379,8 @@ 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

1 0

[PATCH AUTOSEL 5.10 5/6] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3a7f0102b4c5c..90e4757f76b0f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -319,7 +319,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 7/8] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index aa9cdd93b5778..d71b4aecd269f 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -319,7 +319,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 6/8] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 9c3b8e2096565..aa9cdd93b5778 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

1 0

[PATCH AUTOSEL 6.1 10/12] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3d237f75e81f5..0629aa5f2fe4b 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/12] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 eb31b49e65978..3d237f75e81f5 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.6 13/15] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 1b61110cb9174..dcbcd1a59a3aa 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 12/15] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 0acc848c1f004..1b61110cb9174 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.6 06/15] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Sasha Levin 12 Nov '24

12 Nov '24

From: Shenghao Ding <shenghao-ding(a)ti.com> [ Upstream commit fe09de2db2365eed8b44b572cff7d421eaf1754a ] Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 629e2195a890b..1cc64ed8de6da 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2022,6 +2022,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 14/16] ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div()
by Sasha Levin 12 Nov '24

12 Nov '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 23569c8b314925bdb70dd1a7b63cfe6100868315 ] 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 is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Reviewed-by: Olivier Moysan <olivier.moysan(a)foss.st.com> Link: https://patch.msgid.link/20241107015936.211902-1-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 | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 19307812ec765..64f52c75e2aa8 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -317,7 +317,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); - if (div > SAI_XCR1_MCKDIV_MAX(version)) { + if (div > SAI_XCR1_MCKDIV_MAX(version) || div <= 0) { dev_err(&sai->pdev->dev, "Divider %d out of range\n", div); return -EINVAL; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 13/16] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 12 Nov '24

12 Nov '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 ad2492efb1cdc..19307812ec765 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -378,8 +378,8 @@ 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

1 0

[PATCH AUTOSEL 6.11 07/16] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Sasha Levin 12 Nov '24

12 Nov '24

From: Shenghao Ding <shenghao-ding(a)ti.com> [ Upstream commit fe09de2db2365eed8b44b572cff7d421eaf1754a ] Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> Link: https://patch.msgid.link/20241104100055.48-1-shenghao-ding@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f07104..6474cc551d551 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.43.0

1 0

[RESEND PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 12 Nov '24

12 Nov '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

1 6

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The "USB" audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another USB device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: | Following up | Relevant PR | | ------------------| ---------------- | | https://github.com/alsa-project/alsa-ucm-conf/issues/334 | https://github.com/alsa-project/alsa-ucm-conf/pull/461 | The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - edited 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was edited from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 | Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082 - ASRock X870 Steel Legend
by GitHub issues - opened 12 Nov '24

12 Nov '24

alsa-project/alsa-ucm-conf issue #462 was opened from cybik: Following up on https://github.com/alsa-project/alsa-ucm-conf/issues/334 The card "works", but there is one feature missing - though I hesitate to call it "critical": S/PDIF / TOSLINK support is incomplete. While sound works "fine" (ish), it's limited to 2 channels. * Plugging in another device (CM106-type, for example) gives out a plethora of configuration options (per the screenshot), though this also comes with having to select the right output * The USB audio outs are output-specific, which is VERY welcome, but the S/PDIF out does not allow for any advanced selection like "Supported Codecs", and the HiFi 5.1 and 7.1 do not appear to enable any IEC958/AC3 support in the background Is there something missing in the config files to allow for such support? ![image](https://github.com/user-attachments/assets/906ea829-0210-49a8-ba16-ad1b5b1285ae) Relevant PR: https://github.com/alsa-project/alsa-ucm-conf/pull/461 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/462 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

`amixer sset Master` doesnt increase and decrease both sides equally
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #279 was opened from lquenti: I look through the issues, man pages and internet and hope I didn't miss anything. Please tell me if its a user error. Whenever I use `amixer sset Master` to increase volume, it will not increase them evenly. Here is a before ![before](https://github.com/user-attachments/assets/4ccfa220-608c-4407-9f6e-fc1d27f1a91b) and after ![after](https://github.com/user-attachments/assets/250ab254-d46a-44e4-a201-58f9224b6293) Interestingly amixer seem to think that it is still evenly balanced, although alsamixer shows that it is unbalanced afterwards. --- Interestingly, when decreasing, the unevenness flips to the other side. [Other people also seem to struggle with it](https://stackoverflow.com/questions/69826139/using-amixer-to-set-audio-… Thank you so much. If it is something somewhat simple to fix, I can also write a fix myself, but for now I am unsure what the problem is. Issue URL : https://github.com/alsa-project/alsa-utils/issues/279 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Change 26ce:0a08 to list multiple motherboards
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #461 was opened from cybik: The hardware id `26ce:0a08` seems to be re-used by multiple motherboards at ASRock. Identify them within `[brackets]` to point out the specific boards in question. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ThinkPad P14s Gen 5 AMD micmute LED is not working sometimes.
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #278 was opened from pma1: The mic and sound output works fine. The mute LED works fine. But when press Fn+F4, the mic-mute led is not working sometimes after bootup. The failrate is like 1/10. Tested with latest alsa-ucm from git, it's reproduced. /sys/class/sound/ctl-led/mic/card1# cat list 7 # amixer -c 1 controls numid=12,iface=CARD,name='Headphone Jack' numid=11,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Phantom Jack' numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=2,iface=MIXER,name='Headphone Playback Switch' numid=1,iface=MIXER,name='Headphone Playback Volume' numid=16,iface=MIXER,name='PCM Playback Volume' numid=17,iface=MIXER,name='Mic ACP LED Capture Switch' numid=8,iface=MIXER,name='Mic Boost Volume' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=5,iface=MIXER,name='Auto-Mute Mode' numid=4,iface=MIXER,name='Speaker Playback Switch' numid=3,iface=MIXER,name='Speaker Playback Volume' numid=15,iface=PCM,name='Capture Channel Map' numid=14,iface=PCM,name='Playback Channel Map' # echo 'Capture Switch' > detach # echo 'Mic ACP LED Capture Switch' > attach /sys/class/sound/ctl-led/mic/card1# cat list 17 Then the micmute LED will work. [alsa-info-amd-micmutled.txt](https://github.com/user-attachments/files/1627… Issue URL : https://github.com/alsa-project/alsa-utils/issues/278 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

USB-Audio: ALC4080: Add support for MSI MPG X870E CARBON WIFI (0db0:0b58)
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #460 was opened from xatier: This PR updates the regex for MSI MPG X870E CARBON WIFI motherboard. ``` $ lsusb | ag audio Bus 003 Device 007: ID 0db0:0b58 Micro Star International USB Audio ``` ``` Audio Realtek® ALC4080 Codec ``` https://www.msi.com/Motherboard/MPG-X870E-CARBON-WIFI/Specification Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 6.1.y] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3
by Werner Sembach 09 Nov '24

09 Nov '24

From: Christoffer Sandberg <cs(a)tuxedo.de> Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 a8bc95ffa41a3..da14ab97783f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10001,6 +10001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.43.0

2 3

Odd behavior with Tempotec Serenade PCIe (CM8888)
by GitHub issues - opened 09 Nov '24

09 Nov '24

alsa-project/alsa-lib issue #418 was opened from lss4: This card is powered by CM8888, using `snd-hda-intel` driver, and it does work in Linux, but its behavior is quite odd. 1. Both the RCA connection and 6.35 headphone jack on the card are connected at the moment. Preferably I'd like to be able to switch to use either RCA (to speakers) or headphone, but I can't -- Audio always goes through both outputs, and there's only one analog output from this card. 2. This is the odd behavior I'm talking about -- the card's outputs are muted by default. Playing an audio file will trigger the card to unmute the output after 2-3 seconds followed by an audible click from the relay on the sound card. When the output is active, if no audio is being actively played for at least 10 seconds, the output will be muted again on its own, and this is again followed by an audible click from the relay. Honestly I've never seen an audio card capable of doing something that "intelligent" before. `alsa-info.sh` output attached here. The sound card in question is `card1` (`hwC1D1`). [alsa-info-20241109.txt](https://github.com/user-attachments/files/17687665/… I'm currently using Garuda Linux on this system. Previously I was using Manjaro. Both are Arch-based, and the behaviors are the same. I'm not sure about the support of this chip (CM8888) in general, as the information I could find online about CM8888 on Linux is limited and mostly outdated (several years ago). Issue URL : https://github.com/alsa-project/alsa-lib/issues/418 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[Question] About Roland Rubix series, alsa-ucm-conf and snd_usb_audio
by GitHub issues - opened 08 Nov '24

08 Nov '24

alsa-project/alsa-ucm-conf issue #459 was opened from sim590: I want to understand what `alsa_ucm-conf` is about exactly regarding the support for specific devices. I'm thinking to buy a Roland Rubix series audio interface (either 22 or 24) and I'm wondering if both my XLR inputs are gonna appear in alsa/pulseaudio/pipewire as separate input lines. Does that require a potential change in `snd_usb_audio` kernel module or only a configuration file on this repository? I've looked at the configuration files on this repository and I have found a Roland directory, but nothing about the Rubix series explicitly. Presently, I have a Focusrite Scarlett 2i2 *fake clone* (unfortunately) which I'm gonna return shortly because it's counterfeit. As of right now, I don't see the two separate outputs, but only the following input: ![image](https://github.com/user-attachments/assets/7bf58e54-67d1-4e20-989c-5e1ec0828850) which takes audio from my first XLR input port. I guess that's the class compliant interface behaviour. Correct me if I'm wrong, I'm trying to learn about all of this. So, I just wanna be sure that if I go and buy my Roland Rubix, that the device is actually gonna show up the two input lines, and if not that I will know what I will have to do to make it so. Is there gonna be a configuration file required in `alsa-ucm-conf`? Any potentiel change in `snd_usb_audio` kernel module? I'm a bit confused about the details regarding the support of the audio interface devices. Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/459 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASoC Kernel 6.1: Query regarding dynamic dai link/dapm registration during bootup
by Shalini Manjunatha (Consultant) (QUIC) 08 Nov '24

08 Nov '24

Hi All, On kernel6.1 based Qualcomm target for automotive platform based on Android U, we have a feature early services to launch certain early apps at early init of android bootup cycle. Hence in order to launch early chime sound of audio at early init of android boot up cycle, we need to load all required audio kernel modules at early init stage in order to get sound card registered. We have certain KPI goals for every early apps to get completed within lesser duration so that we can achieve final KPI goal of android boot complete as per the target specification. In this journey, while optimizing the sound card registration through machine driver probe to achieve lower KPI for audio early chime sound, we find that for early chime launch we required only selected pcm front end dai links only. Hence if we keep FE dai link structure in our machine driver with only required items for early chime, overall sound card registrations gets completed within 200msec from the time it starts, which is a very good KPI number to achieve. So we were trying out code change to split overall FE dai link structure into two, one for early chime and another for all other required dai link's, then some how we need to register early chime related dai link during early init so that sound card registration process gets completed quickly, later stage we want to dynamically load all other FE dai links. Could you please help us to suggest on how can we achieve this?. Is there any possible way out to avoid static FE dai link registration for all required ones from machine driver during machine driver probe, instead register only limited list of dai links. Later can we dynamically register the dai links in order to get PCM devices entries created based on use case. Currently we have this API devm_snd_soc_register_card() being called with snd_soc_card parameter which has all possible FE dai links populated at once. Queries: 1) can we register sound card twice during bootup, once with limited dai links and next with all required ones? 2) can we register dai links/dapm dynamically based on use case requriement? 3) If it is possible, could you share any reference from any other vendor who is able to achieve the same?. Regards, Shalini M

3 2

[PATCH RESEND v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 08 Nov '24

08 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

3 5

[GIT PULL] ASoC fixes for v6.12-rc6
by Mark Brown 08 Nov '24

08 Nov '24

The following changes since commit 2db63e92186d7201ee1cb2f5af11757c5e5a1020: wcd937x codec fixes (2024-10-29 19:18:48 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc6 for you to fetch changes up to 48b86532c10128cf50c854a90c2d5b1410f4012d: ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits (2024-11-07 15:35:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 A moderately large pile of small changes here, split fairly evenly between fixes and ID additions/quirks and all of it driver specific. ---------------------------------------------------------------- Amelie Delaunay (1): ASoC: stm32: spdifrx: fix dma channel release in stm32_spdifrx_remove Jyri Sarha (1): ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits Kuninori Morimoto (1): MAINTAINERS: Generic Sound Card section Luo Yifan (2): ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate() ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div() Markus Petri (1): ASoC: amd: yc: Support dmic on another model of Lenovo Thinkpad E14 Gen 6 Mingcong Bai (1): ASoC: amd: yc: fix internal mic on Xiaomi Book Pro 14 2022 Shenghao Ding (1): ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip Venkata Prasad Potturu (1): ASoC: SOF: amd: Fix for incorrect DMA ch status register offset MAINTAINERS | 9 +++++++++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/codecs/tas2781-fmwlib.c | 1 + sound/soc/sof/amd/acp.c | 10 +++++++++- sound/soc/sof/sof-client-probes-ipc4.c | 1 + sound/soc/stm/stm32_sai_sub.c | 6 +++--- sound/soc/stm/stm32_spdifrx.c | 2 +- 7 files changed, 38 insertions(+), 5 deletions(-)

2 1

[PATCH RESEND v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 07 Nov '24

07 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

4 7

[PATCH v3 1/2] ASoC: dt-bindings: irondevice,sma1307: Add initial DT binding
by Ki-Seok Jo 07 Nov '24

07 Nov '24

This adds the schema binding for the Iron Device SMA1307 Amp Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- .../bindings/sound/irondevice,sma1307.yaml | 54 ++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml new file mode 100644 index 000000000..0bb4ee664 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) %YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/irondevice,sma1307.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Iron Device SMA1307 Audio Amplifier + +maintainers: + - Kiseok Jo <kiseok.jo(a)irondevice.com> + +description: + SMA1307 boosted digital speaker amplifier + with feedback-loop. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + enum: + - irondevice,sma1307a + - irondevice,sma1307aq + description: + If a 'q' is added, it indicated the product is AEC-Q100 + qualified for automotive applications. SMA1307A supports + both WLCSP and QFN packages. However, SMA1307AQ only + supports the QFN package. + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + amplifier@1e { + compatible = "irondevice,sma1307a"; + reg = <0x1e>; + #sound-dai-cells = <1>; + }; + }; -- 2.39.2

3 4

[PATCH v2] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset
by Venkata Prasad Potturu 07 Nov '24

07 Nov '24

DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: 490be7ba2a01 ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index de3001f5b9bb..95d4762c9d93 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) -- 2.39.2

2 1

[PATCH] ASoC: SOF: amd: Fix for incorrect DMA ch status register offset
by Venkata Prasad Potturu 07 Nov '24

07 Nov '24

DMA ch status register offset change in acp7.0 platform Incorrect DMA channel status register offset check lead to firmware boot failure. [ 14.432497] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump start ]------------ [ 14.432533] snd_sof_amd_acp70 0000:c4:00.5: Firmware boot failure due to timeout [ 14.432549] snd_sof_amd_acp70 0000:c4:00.5: fw_state: SOF_FW_BOOT_IN_PROGRESS (3) [ 14.432610] snd_sof_amd_acp70 0000:c4:00.5: invalid header size 0x71c41000. FW oops is bogus [ 14.432626] snd_sof_amd_acp70 0000:c4:00.5: unexpected fault 0x71c40000 trace 0x71c40000 [ 14.432642] snd_sof_amd_acp70 0000:c4:00.5: ------------[ DSP dump end ]------------ [ 14.432657] snd_sof_amd_acp70 0000:c4:00.5: error: failed to boot DSP firmware -5 [ 14.432672] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 3 -> 4 [ 14.433260] dmic-codec dmic-codec: ASoC: Unregistered DAI 'dmic-hifi' [ 14.433319] snd_sof_amd_acp70 0000:c4:00.5: fw_state change: 4 -> 0 [ 14.433358] snd_sof_amd_acp70 0000:c4:00.5: error: sof_probe_work failed err: -5 Update correct register offset for DMA ch status register. Fixes: bff3c2a4062d ("ASoC: SOF: amd: add support for acp7.0 based platform") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/acp.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index ce10d128de25..f7814dadf3ba 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -342,11 +342,19 @@ int acp_dma_status(struct acp_dev_data *adata, unsigned char ch) { struct snd_sof_dev *sdev = adata->dev; unsigned int val; + unsigned int acp_dma_ch_sts; int ret = 0; + switch (adata->pci_rev) { + case ACP70_PCI_ID: + acp_dma_ch_sts = ACP70_DMA_CH_STS; + break; + default: + acp_dma_ch_sts = ACP_DMA_CH_STS; + } val = snd_sof_dsp_read(sdev, ACP_DSP_BAR, ACP_DMA_CNTL_0 + ch * sizeof(u32)); if (val & ACP_DMA_CH_RUN) { - ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, ACP_DMA_CH_STS, val, !val, + ret = snd_sof_dsp_read_poll_timeout(sdev, ACP_DSP_BAR, acp_dma_ch_sts, val, !val, ACP_REG_POLL_INTERVAL, ACP_DMA_COMPLETE_TIMEOUT_US); if (ret < 0) -- 2.39.2

3 3

topology: Fix strcat() to uninitialized memory in add_define()
by GitHub pull_request - opened 07 Nov '24

07 Nov '24

alsa-project/alsa-utils pull request #277 was opened from rfvirgil: If *defs == NULL the realloc() will be a simple malloc() and return a pointer to uninitialized memory. Doing a strcat() to this is bad, so in that case strcpy() the string into the newly allocated buffer. This fixes the buffer overflow bug, such as: *** buffer overflow detected ***: terminated Program received signal SIGABRT, Aborted. __pthread_kill_implementation (no_tid=0, signo=6, threadid=<optimized out>) at ./nptl/pthread_kill.c:44 warning: 44 ./nptl/pthread_kill.c: No such file or directory (gdb) bt at ../sysdeps/posix/libc_fatal.c:132 at ./debug/fortify_fail.c:24 dest=dest@entry=0x5555555682c0 "hUUU\005PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passth"..., src=src@entry=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., destlen=<optimized out>) at ./debug/strcat_chk.c:34 __src=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., __dest=<optimized out>) at /usr/include/x86_64-linux-gnu/bits/string_fortified.h:130 d=0x7fffffffe0fa "PLATFORM=mtl,PREPROCESS_PLUGINS=nhlt,NHLT_BIN=nhlt-sof-mtl-sdw-cs42l42-l0-max98363-l2-4ch-48k-96k.bin,NUM_DMICS=4,PDM1_MIC_A_ENABLE=1,PDM1_MIC_B_ENABLE=1,DMIC0_ID=3,DMIC1_ID=4,DMIC1_ENABLE=passthrough"..., defs=<synthetic pointer>) at topology.c:400 Request URL : https://github.com/alsa-project/alsa-utils/pull/277 Patch URL : https://github.com/alsa-project/alsa-utils/pull/277.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR
by schorpp 07 Nov '24

07 Nov '24

[BUG] HDA Intel PCH, HDMI , HDA Intel PCH, ALC886 Digital, no IEC958 / HDMI output/passthrough to AVR ALSA versions 1.0.25...current Kernel Driver versions 3.12.40...6.x Hardware: DFI Mainboard MB330-CRM https://www.dfi.com/de/product/index/273#specification Reproducible Always Steps to reproduce and full report: https://www-vdr--portal-de.translate.goog/forum/index.php?thread/136519-gel… Software crosscheck with CM106 USB Soundbox iec958: works OK. Where to start in the drivers source code to fix it? y tom

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is over. The BridgeCast seems to be assigned to the new conf files with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! > _Edits : Spelling/Grammar mistakes_ Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is over. The BridgeCast seems to be assigned to the new conf files with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! > _Edits : Spellings/Grammar mistakes_ Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working properly despite V2 configuration files
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was edited from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is done. The BridgeCast seems to be assigned to the correct conf file with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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BridgeCast v2 not working despite V2 module
by GitHub issues - opened 06 Nov '24

06 Nov '24

alsa-project/alsa-ucm-conf issue #458 was opened from acerammos: Hi everybody, I (sadly) made the firmware update for the Roland BridgeCast and it came out broken. It was working flawlessly before the update, thanks to you i had all the Output/Input channels available and i was able to assign sound sources on them individually through the system settings gui (Kmix - Plasma). Today, as a now "V2", it is done. The BridgeCast seems to be assigned to the correct conf file with its new usb tag, to `BridgeCastV2-Hifi.conf` and not former the `BridgeCast-Hifi.conf`. It seems to be a good start, thanks to Fulmer for the merge and first testing Roland's update. But now, i have a weird mix of "auxiliary channels" instead of proper audio outputs like before : ![Copie d'écran_20241106_154428](https://github.com/user-attachments/assets/4240924b-… > _(Tell me if you need Fr>En translation on that screenshot)_ Note that "chromium" (wich is in fact a client like spotify, playing music) cannot be assigned to a specific output, because BridgeCast is now detected to be "only one channel" so the option is missing. An exemple after also enabling my speakers : ![Copie d'écran_20241106_155745](https://github.com/user-attachments/assets/d01d1a67-… > _Old behaviour allowed me to choose between all the output channels of the BridgeCast, like "Games", "Aux", "Music", Chat" or "System"_ Something else i've (maybe) noticed is the differences for the channel maps, between the `cat /proc/asound/cardX/stream0` channels output and what is written in the `Roland/BridgeCastV2-Hifi.conf`. `cat /proc/asound/cardX/stream0 :` ``` Roland BRIDGE CAST V2 at usb-0000:00:14.0-1, high speed : USB Audio Playback: Status: Running Interface = 1 Altset = 1 Packet Size = 392 Momentary freq = 48000 Hz (0x6.0000) Interface 1 Altset 1 Format: S32_LE Channels: 14 Endpoint: 0x0d (13 OUT) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR FLC FRC RC SL SR TC -- -- Sync Endpoint: 0x8e (14 IN) Sync EP Interface: 2 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 2 Altset 1 Format: S32_LE Channels: 6 Endpoint: 0x8e (14 IN) (ASYNC) Rates: 48000 Data packet interval: 125 us Bits: 24 Channel map: FL FR FC LFE RL RR Sync Endpoint: 0x0d (13 OUT) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` `BridgeCastV2-Hifi.conf :` ``` Macro [ { SplitPCM { Name "bc_stereo_out" Direction Playback Channels 2 HWChannels 14 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR } } { SplitPCM { Name "bc_stereo_in" Direction Capture Channels 2 HWChannels 6 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR } } ] ``` I'm completely new to this kind of code/linux driver wizardry so i haven't been able to gather more data, but i'm not afraid to be messing around, so if i can be of any help, screenshot, command outputs, random file testing, i'll gladly help. Regards, and thank you all ! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/458 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 06 Nov '24

06 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): This adds the schema binding for the Iron Device SMA1307 Amp The Iron Device SMA1307 is a boosted digital speaker amplifier .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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Help to solve pop when sound loops
by GitHub issues - edited 06 Nov '24

06 Nov '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ```c #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/8] ASoC: cleasnup rtd and its ID
by Kuninori Morimoto 05 Nov '24

05 Nov '24

Hi Mark rtd has own ID, but it is naming "num" for it. The naming is confusable. This patch rename it to "id". And many functions request both "rtd" and its "id" as function parameter, but rtd itself has rtd->id. This patch cleanup it. And, Qcom driver want to use irregular rtd ID because of its topology, and thus, soc-core need irregular calculation. I'm not sure why only Qcom needs such calculation, but this patch also cleanup it. But I guess we want to cleanup is not soc-core but Qcom side (?) Kuninori Morimoto (8): ASoC: rename rtd->num to rtd->id ASoC: fsl: switch to use rtd->id from rtd->num ASoC: meson: switch to use rtd->id from rtd->num ASoC: sh: switch to use rtd->id from rtd->num ASoC: generic: switch to use rtd->id from rtd->num ASoC: remove rtd->num ASoC: soc-core: do rtd->id trick at snd_soc_add_pcm_runtime() ASoC: cleanup function parameter for rtd and its id include/sound/soc-dai.h | 5 ++- include/sound/soc.h | 8 ++--- sound/soc/fsl/imx-card.c | 6 ++-- sound/soc/generic/simple-card-utils.c | 10 +++--- sound/soc/meson/axg-card.c | 6 ++-- sound/soc/meson/gx-card.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 9 +++-- sound/soc/soc-core.c | 50 +++++++++++++++------------ sound/soc/soc-dai.c | 4 +-- sound/soc/soc-pcm.c | 16 ++++----- 11 files changed, 60 insertions(+), 58 deletions(-) -- 2.43.0

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[PATCH 0/2] Soundwire dai link init logic refactor
by Vijendar Mukunda 05 Nov '24

05 Nov '24

This patch series refactors the SoundWire dai link init logic for Intel and AMD generic SoundWire machine driver and also updates the stream_name in dai_links structure. Link: https://github.com/thesofproject/linux/pull/5218 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Vijendar Mukunda (2): ASoC: sdw_utils/intel/amd: refactor dai link init logic ASoC: sdw_utils: Update stream_name in dai_links structure include/sound/soc_sdw_utils.h | 5 +++-- sound/soc/amd/acp/acp-sdw-sof-mach.c | 8 ++++---- sound/soc/intel/boards/sof_sdw.c | 12 ++++++------ sound/soc/sdw_utils/soc_sdw_utils.c | 10 ++++++---- 4 files changed, 19 insertions(+), 16 deletions(-) -- 2.34.1

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[PATCH] ASoC: rt721-sdca: change interrupt mask from XU to GE
by Jack Yu 05 Nov '24

05 Nov '24

Change interrupt mask from XU to GE to fix jack detection interrupt issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt721-sdca-sdw.c | 13 ++++--------- sound/soc/codecs/rt721-sdca.c | 6 ++++-- sound/soc/codecs/rt721-sdca.h | 1 + 3 files changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c index c0f8cccae3b2..c71453da088a 100644 --- a/sound/soc/codecs/rt721-sdca-sdw.c +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -203,7 +203,7 @@ static int rt721_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt721_sdca_jack_init set. */ sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } @@ -280,7 +280,7 @@ static int rt721_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 900; + prop->clk_stop_timeout = 1380; /* wake-up event */ prop->wake_capable = 1; @@ -337,11 +337,6 @@ static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); if (ret < 0) goto io_error; - } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { - ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, - SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); - if (ret < 0) - goto io_error; } ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); if (ret < 0) @@ -475,7 +470,7 @@ static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) mutex_lock(&rt721_sdca->disable_irq_lock); rt721_sdca->disable_irq = true; ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + SDW_SCP_SDCA_INTMASK_SDCA_0, 0); ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8, 0); mutex_unlock(&rt721_sdca->disable_irq_lock); @@ -502,7 +497,7 @@ static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) if (!slave->unattach_request) { mutex_lock(&rt721->disable_irq_lock); if (rt721->disable_irq == true) { - sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); rt721->disable_irq = false; } diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index bdd160b80b64..1c9f32e405cf 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -39,7 +39,7 @@ static void rt721_sdca_jack_detect_handler(struct work_struct *work) return; /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { rt721->jack_type = rt_sdca_headset_detect(rt721->regmap, RT721_SDCA_ENT_GE49); if (rt721->jack_type < 0) @@ -286,7 +286,7 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) mutex_lock(&rt721->calibrate_mutex); if (rt721->hs_jack) { sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_0); sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); @@ -298,6 +298,8 @@ static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) regmap_write(rt721->regmap, SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + rt_sdca_index_write(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, + RT721_XU_REL_CTRL, 0x0000); rt_sdca_index_update_bits(rt721->mbq_regmap, RT721_HDA_SDCA_FLOAT, RT721_GE_REL_CTRL1, 0x4000, 0x4000); } diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h index e2f071909da8..0a82c107b19a 100644 --- a/sound/soc/codecs/rt721-sdca.h +++ b/sound/soc/codecs/rt721-sdca.h @@ -133,6 +133,7 @@ struct rt721_sdca_dmic_kctrl_priv { #define RT721_HDA_LEGACY_UAJ_CTL 0x02 #define RT721_HDA_LEGACY_CTL1 0x05 #define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_XU_REL_CTRL 0x0c #define RT721_GE_REL_CTRL1 0x0d #define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e #define RT721_GE_SDCA_RST_CTRL 0x10 -- 2.34.1

2 1

[PATCH] ASoC: amd: acp: Fix for ACP SOF dmic tplg component load failure
by Venkata Prasad Potturu 05 Nov '24

05 Nov '24

Stream name mismatch with topology file causes tplg load failure. As SOF framework assigns dailink->stream name, overriding stream name other than link name causes SOF dmic component load failure. [ 35.474995] snd_sof_amd_acp70 0000:c4:00.5: error: can't connect DAI ACPDMIC0.IN stream acp-dmic-codec [ 35.475001] snd_sof_amd_acp70 0000:c4:00.5: failed to add widget type 28 name : ACPDMIC0.IN stream acp-dmic-codec [ 35.475013] sof_mach acp70-dsp: ASoC: failed to load widget ACPDMIC0.IN [ 35.475018] sof_mach acp70-dsp: ASoC: topology: could not load header: -22 [ 35.475072] snd_sof_amd_acp70 0000:c4:00.5: error: tplg component load failed -22 [ 35.475083] snd_sof_amd_acp70 0000:c4:00.5: error: failed to load DSP topology -22 [ 35.475090] snd_sof_amd_acp70 0000:c4:00.5: ASoC: error at snd_soc_component_probe on 0000:c4:00.5: -22 [ 35.475117] sof_mach acp70-dsp: ASoC: failed to instantiate card -22 [ 35.475254] sof_mach acp70-dsp: error -EINVAL: Failed to register card(sof-acp70-dsp) [ 35.475261] sof_mach acp70-dsp: probe with driver sof_mach failed with error -22 Fixes: b2385de2ae11 ("ASoC: amd: acp: Add stream name to ACP PDM DMIC devices") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/acp-mach-common.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index 67aa0ad83486..d314253207d5 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -1561,7 +1561,6 @@ int acp_sofdsp_dai_links_create(struct snd_soc_card *card) if (drv_data->dmic_cpu_id == DMIC) { links[i].name = "acp-dmic-codec"; - links[i].stream_name = "DMIC capture"; links[i].id = DMIC_BE_ID; links[i].codecs = dmic_codec; links[i].num_codecs = ARRAY_SIZE(dmic_codec); -- 2.39.2

5 6

[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 05 Nov '24

05 Nov '24

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

2 1

[PATCH] ASoC: codecs: ES8326: Reduce pop noise
by Zhang Yi 05 Nov '24

05 Nov '24

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index aa3e364827c8..a5603b617688 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); } } } else { @@ -1082,7 +1082,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

1 0

[PATCH v3 2/2] ASoC: sma1307: Add driver for Iron Device SMA1307
by Ki-Seok Jo 05 Nov '24

05 Nov '24

The Iron Device SMA1307 is a boosted digital speaker amplifier v2 -> v3 - Use BIT() macro - Changed the value in sizeof - Designated as const - Removed the retry functionality from the regmap functions - Removed the remove callback function from the component driver - Added match data and I2C ID table - Applied the value when setting the binary mode - Removed unused variable - Removed some error log - Removed cache_type member - Changed the configuration for one shot control - Modified the 'Amp Enable Switch' and 'Binary Mode Control' - Removed the reutrn value for regmap - Change 'check fault' from sysfs to ALSA control - Fix for the error in workqueue polling operation - Uevent configuration in fault situation - Fixed typos Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- To | 0 sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++++++++ 5 files changed, 2508 insertions(+) diff --git a/To b/To new file mode 100644 index 000000000..e69de29bb diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c6c4c7481..3484a3f68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -236,6 +236,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SMA1303 + imply SND_SOC_SMA1307 imply SND_SOC_SPDIF imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 @@ -1836,6 +1837,15 @@ config SND_SOC_SMA1303 help Enable support for Iron Device SMA1303 Boosted Class-D amplifier +config SND_SOC_SMA1307 + tristate "Iron Device SMA1307 Audio Amplifier" + depends on I2C + help + Enable support for Iron Device SMA1307 boosted digital speaker + amplifier with feedback-loop. + If you are using a system with an SMA1307 amplifier connected + via I2C, enable this option. + config SND_SOC_SPDIF tristate "S/PDIF CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 850c6249e..4b29fd4bd 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -274,6 +274,7 @@ snd-soc-sigmadsp-i2c-y := sigmadsp-i2c.o snd-soc-sigmadsp-regmap-y := sigmadsp-regmap.o snd-soc-si476x-y := si476x.o snd-soc-sma1303-y := sma1303.o +snd-soc-sma1307-y := sma1307.o snd-soc-spdif-tx-y := spdif_transmitter.o snd-soc-spdif-rx-y := spdif_receiver.o snd-soc-src4xxx-y := src4xxx.o @@ -678,6 +679,7 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SMA1303) += snd-soc-sma1303.o +obj-$(CONFIG_SND_SOC_SMA1307) += snd-soc-sma1307.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o diff --git a/sound/soc/codecs/sma1307.c b/sound/soc/codecs/sma1307.c new file mode 100644 index 000000000..985a247b3 --- /dev/null +++ b/sound/soc/codecs/sma1307.c @@ -0,0 +1,2052 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// sma1307.c -- sma1307 ALSA SoC Audio driver +// +// Copyright 2024 Iron Device Corporation +// +// Auther: Gyuhwa Park <gyuwha.park(a)irondevice.com> +// Auther: Kiseok Jo <kiseok.jo(a)irondevice.com> + +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include "sma1307.h" + +#define CHECK_PERIOD_TIME 1 /* sec per HZ */ +#define PLL_MATCH(_input_clk_name, _output_clk_name, _input_clk,\ + _post_n, _n, _vco, _p_cp)\ +{\ + .input_clk_name = _input_clk_name,\ + .output_clk_name = _output_clk_name,\ + .input_clk = _input_clk,\ + .post_n = _post_n,\ + .n = _n,\ + .vco = _vco,\ + .p_cp = _p_cp,\ +} + +static const char *setting_file = "sma1307_setting.bin"; +#define SMA1307_SETTING_CHECKSUM 0x100000 + +/* PLL clock setting Table */ +struct sma1307_pll_match { + char *input_clk_name; + char *output_clk_name; + unsigned int input_clk; + unsigned int post_n; + unsigned int n; + unsigned int vco; + unsigned int p_cp; +}; + +struct sma1307_data { + char *name; + void (*init)(struct regmap *regmap); +}; + +struct sma1307_priv { + bool check_fault_status; + bool force_mute_status; + bool sw_ot1_prot; + char *name; + enum sma1307_mode amp_mode; + int binary_mode; + int dapm_aif_in; + int dapm_aif_out0; + int dapm_aif_out1; + int dapm_sdo_en; + int dapm_sdo_setting; + int num_of_pll_matches; + int check_fault_period; + struct delayed_work check_fault_work; + struct device *dev; + struct kobject *kobj; + struct mutex default_lock; + struct regmap *regmap; + struct sma1307_setting_file set; + const struct sma1307_pll_match *pll_matches; + const struct sma1307_data *data; + unsigned int cur_vol; + unsigned int format; + unsigned int frame_size; + unsigned int init_vol; + unsigned int last_bclk; + unsigned int otp_trm2; + unsigned int otp_trm3; + unsigned int rev_num; + unsigned int sys_clk_id; + unsigned int tdm_slot0_rx; + unsigned int tdm_slot1_rx; + unsigned int tdm_slot0_tx; + unsigned int tdm_slot1_tx; + unsigned int tsdw_cnt; +}; + +static const struct sma1307_pll_match sma1307_pll_matches[] = { + /* in_clk_name, out_clk_name, input_clk post_n, n, vco, p_cp */ + PLL_MATCH("1.411MHz", "24.554MHz", + 1411200, 0x06, 0xD1, 0x88, 0x00), + PLL_MATCH("1.536MHz", "24.576MHz", + 1536000, 0x06, 0xC0, 0x88, 0x00), + PLL_MATCH("2.822MHz", "24.554MHz", + 2822400, 0x06, 0xD1, 0x88, 0x04), + PLL_MATCH("3.072MHz", "24.576MHz", + 3072000, 0x06, 0x60, 0x88, 0x00), + PLL_MATCH("6.144MHz", "24.576MHz", + 6144000, 0x06, 0x60, 0x88, 0x04), + PLL_MATCH("12.288MHz", "24.576MHz", + 12288000, 0x06, 0x60, 0x88, 0x08), + PLL_MATCH("19.2MHz", "24.48MHz", + 19200000, 0x06, 0x7B, 0x88, 0x0C), + PLL_MATCH("24.576MHz", "24.576MHz", + 24576000, 0x06, 0x60, 0x88, 0x0C), +}; + +static struct snd_soc_component *sma1307_amp_component; + +static void sma1307_startup(struct snd_soc_component *); +static void sma1307_shutdown(struct snd_soc_component *); +static void sma1307_reset(struct snd_soc_component *); +static void sma1307_set_binary(struct snd_soc_component *); +static void sma1307_set_default(struct snd_soc_component *); + +/* Initial register value - 6.0W SPK (8ohm load) */ +static const struct reg_default sma1307_reg_def[] = { + { 0x00, 0x80 }, + { 0x01, 0x00 }, + { 0x02, 0x52 }, + { 0x03, 0x4C }, + { 0x04, 0x47 }, + { 0x05, 0x42 }, + { 0x06, 0x40 }, + { 0x07, 0x40 }, + { 0x08, 0x3C }, + { 0x09, 0x2F }, + { 0x0A, 0x32 }, + { 0x0B, 0x50 }, + { 0x0C, 0x8C }, + { 0x0D, 0x00 }, + { 0x0E, 0x3F }, + { 0x0F, 0x00 }, + { 0x10, 0x00 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x09 }, + { 0x14, 0x12 }, + { 0x1C, 0x00 }, + { 0x1D, 0x85 }, + { 0x1E, 0xA1 }, + { 0x1F, 0x67 }, + { 0x22, 0x00 }, + { 0x23, 0x1F }, + { 0x24, 0x7A }, + { 0x25, 0x00 }, + { 0x26, 0xFF }, + { 0x27, 0x39 }, + { 0x28, 0x54 }, + { 0x29, 0x92 }, + { 0x2A, 0xB0 }, + { 0x2B, 0xED }, + { 0x2C, 0xED }, + { 0x2D, 0xFF }, + { 0x2E, 0xFF }, + { 0x2F, 0xFF }, + { 0x30, 0xFF }, + { 0x31, 0xFF }, + { 0x32, 0xFF }, + { 0x34, 0x01 }, + { 0x35, 0x17 }, + { 0x36, 0x92 }, + { 0x37, 0x00 }, + { 0x38, 0x01 }, + { 0x39, 0x10 }, + { 0x3E, 0x01 }, + { 0x3F, 0x08 }, + { 0x8B, 0x05 }, + { 0x8C, 0x50 }, + { 0x8D, 0x80 }, + { 0x8E, 0x10 }, + { 0x8F, 0x02 }, + { 0x90, 0x02 }, + { 0x91, 0x83 }, + { 0x92, 0xC0 }, + { 0x93, 0x00 }, + { 0x94, 0xA4 }, + { 0x95, 0x74 }, + { 0x96, 0x57 }, + { 0xA2, 0xCC }, + { 0xA3, 0x28 }, + { 0xA4, 0x40 }, + { 0xA5, 0x01 }, + { 0xA6, 0x41 }, + { 0xA7, 0x08 }, + { 0xA8, 0x04 }, + { 0xA9, 0x27 }, + { 0xAA, 0x10 }, + { 0xAB, 0x10 }, + { 0xAC, 0x10 }, + { 0xAD, 0x0F }, + { 0xAE, 0xCD }, + { 0xAF, 0x70 }, + { 0xB0, 0x03 }, + { 0xB1, 0xEF }, + { 0xB2, 0x03 }, + { 0xB3, 0xEF }, + { 0xB4, 0xF3 }, + { 0xB5, 0x3D }, +}; + +static bool sma1307_readable_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_00_SYSTEM_CTRL ... SMA1307_1F_TONE_FINE_VOLUME: + case SMA1307_22_COMP_HYS_SEL ... SMA1307_32_BROWN_OUT_PROT19: + case SMA1307_34_OCP_SPK ... SMA1307_39_PMT_NZ_VAL: + case SMA1307_3B_TEST1 ... SMA1307_3F_ATEST2: + case SMA1307_8B_PLL_POST_N ... SMA1307_9A_OTP_TRM3: + case SMA1307_A0_PAD_CTRL0 ... SMA1307_BE_MCBS_CTRL2: + case SMA1307_F5_READY_FOR_V_SAR: + case SMA1307_F7_READY_FOR_T_SAR ... SMA1307_FF_DEVICE_INDEX: + break; + default: + return false; + } + return true; +} + +static bool sma1307_writeable_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_00_SYSTEM_CTRL ... SMA1307_1F_TONE_FINE_VOLUME: + case SMA1307_22_COMP_HYS_SEL ... SMA1307_32_BROWN_OUT_PROT19: + case SMA1307_34_OCP_SPK ... SMA1307_39_PMT_NZ_VAL: + case SMA1307_3B_TEST1 ... SMA1307_3F_ATEST2: + case SMA1307_8B_PLL_POST_N ... SMA1307_9A_OTP_TRM3: + case SMA1307_A0_PAD_CTRL0 ... SMA1307_BE_MCBS_CTRL2: + break; + default: + return false; + } + return true; +} + +static bool sma1307_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg > SMA1307_FF_DEVICE_INDEX) + return false; + + switch (reg) { + case SMA1307_F8_STATUS_T1 ... SMA1307_FF_DEVICE_INDEX: + break; + default: + return false; + } + return true; +} + +/* DB scale conversion of speaker volume */ +static const DECLARE_TLV_DB_SCALE(sma1307_spk_tlv, -6000, 50, 0); + +static const char *const sma1307_aif_in_source_text[] = { + "Mono", "Left", "Right" +}; + +static const char *const sma1307_sdo_setting_text[] = { + "Data_One_48k", "Data_Two_48k", "Data_Two_24k", + "Clk_PLL", "Clk_OSC" +}; + +static const char *const sma1307_aif_out_source_text[] = { + "Disable", "After_FmtC", "After_Mixer", "After_DSP", + "Vrms2_Avg", "Battery", "Temperature", "After_Delay" +}; + +static const char *const sma1307_tdm_slot_text[] = { + "Slot0", "Slot1", "Slot2", "Slot3", + "Slot4", "Slot5", "Slot6", "Slot7" +}; + +static const char *const sma1307_binary_mode_text[] = { + "Mode0", "Mode1", "Mode2", "Mode3", "Mode4" +}; + +static const char *const sma1307_reset_text[] = { + "Reset" +}; + +static const struct soc_enum sma1307_aif_in_source_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_aif_in_source_text), + sma1307_aif_in_source_text); +static const struct soc_enum sma1307_sdo_setting_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_sdo_setting_text), + sma1307_sdo_setting_text); +static const struct soc_enum sma1307_aif_out_source_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_aif_out_source_text), + sma1307_aif_out_source_text); +static const struct soc_enum sma1307_tdm_slot_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_tdm_slot_text), + sma1307_tdm_slot_text); +static const struct soc_enum sma1307_binary_mode_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_binary_mode_text), + sma1307_binary_mode_text); +static const struct soc_enum sma1307_reset_enum = +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1307_reset_text), + sma1307_reset_text); + +static int sma1307_force_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->force_mute_status; + + return 0; +} + +static int sma1307_force_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->force_mute_status == val) { + change = false; + } else { + change = true; + sma1307->force_mute_status = val; + } + + return change; +} + +static int sma1307_tdm_slot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int val1, val2; + + regmap_read(sma1307->regmap, SMA1307_A5_TDM1, &val1); + regmap_read(sma1307->regmap, SMA1307_A6_TDM2, &val2); + + if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX0_POS_NAME)) { + ucontrol->value.integer.value[0] + = (val1 & SMA1307_TDM_SLOT0_RX_POS_MASK) >> 3; + sma1307->tdm_slot0_rx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX1_POS_NAME)) { + ucontrol->value.integer.value[0] + = val1 & SMA1307_TDM_SLOT1_RX_POS_MASK; + sma1307->tdm_slot1_rx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX0_POS_NAME)) { + ucontrol->value.integer.value[0] + = (val2 & SMA1307_TDM_SLOT0_TX_POS_MASK) >> 3; + sma1307->tdm_slot0_tx = ucontrol->value.integer.value[0]; + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX1_POS_NAME)) { + ucontrol->value.integer.value[0] + = val2 & SMA1307_TDM_SLOT1_TX_POS_MASK; + sma1307->tdm_slot1_tx = ucontrol->value.integer.value[0]; + } else { + return -EINVAL; + } + + return 0; +} + +static int sma1307_tdm_slot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int val = (int)ucontrol->value.integer.value[0]; + bool change; + + if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX0_POS_NAME)) { + if (sma1307->tdm_slot0_rx == val) + change = false; + else { + change = true; + sma1307->tdm_slot0_rx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A5_TDM1, + SMA1307_TDM_SLOT0_RX_POS_MASK, val << 3); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_RX1_POS_NAME)) { + if (sma1307->tdm_slot1_rx == val) + change = false; + else { + change = true; + sma1307->tdm_slot1_rx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A5_TDM1, + SMA1307_TDM_SLOT1_RX_POS_MASK, val); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX0_POS_NAME)) { + if (sma1307->tdm_slot0_tx == val) + change = false; + else { + change = true; + sma1307->tdm_slot0_tx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A6_TDM2, + SMA1307_TDM_SLOT0_TX_POS_MASK, val << 3); + } + } else if (!strcmp(kcontrol->id.name, SMA1307_TDM_TX1_POS_NAME)) { + if (sma1307->tdm_slot1_tx == val) + change = false; + else { + change = true; + sma1307->tdm_slot1_tx = val; + regmap_update_bits(sma1307->regmap, SMA1307_A6_TDM2, + SMA1307_TDM_SLOT1_TX_POS_MASK, val); + } + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + + return change; +} + +static int sma1307_sw_ot1_prot_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->sw_ot1_prot; + + return 0; +} + +static int sma1307_sw_ot1_prot_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->sw_ot1_prot == val) + change = false; + else { + change = true; + sma1307->sw_ot1_prot = val; + } + + return change; +} + +static int sma1307_check_fault_status_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = (int)sma1307->check_fault_status; + + return 0; +} + +static int sma1307_check_fault_status_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + bool change = false, val = (bool)ucontrol->value.integer.value[0]; + + if (sma1307->check_fault_status == val) { + change = false; + } else { + change = true; + sma1307->check_fault_status = val; + } + + return change; +} + +static int sma1307_check_fault_period_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = sma1307->check_fault_period; + + return 0; +} + +static int sma1307_check_fault_period_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + bool change = false; + int val = ucontrol->value.integer.value[0]; + + if (val < mc->min || val > mc->max) + return -EINVAL; + if (sma1307->check_fault_period == val) { + change = false; + } else { + change = true; + sma1307->check_fault_period = val; + } + + return change; +} + +static int sma1307_reset_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_RESET_MASK, SMA1307_RESET_ON); + sma1307_reset(component); + + snd_ctl_notify(component->card->snd_card, SNDRV_CTL_EVENT_MASK_VALUE, + &kcontrol->id); + + return true; +} + +static int sma1307_binary_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct sma1307_priv *sma1307 = snd_kcontrol_chip(kcontrol); + + sma1307->binary_mode = (int)ucontrol->value.enumerated.item[0]; + if (sma1307->set.status) + sma1307_set_binary(component); + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + +static void sma1307_startup(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + regmap_update_bits(sma1307->regmap, SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_ON); + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_POWER_MASK, SMA1307_POWER_ON); + + if (sma1307->amp_mode == SMA1307_MONO_MODE) { + regmap_update_bits(sma1307->regmap, + SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, + SMA1307_SPK_MONO); + } else { + regmap_update_bits(sma1307->regmap, + SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, + SMA1307_SPK_STEREO); + } + + if (sma1307->check_fault_status) { + if (sma1307->check_fault_period > 0) + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + sma1307->check_fault_period * HZ); + else + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + CHECK_PERIOD_TIME * HZ); + } +} + +static void sma1307_shutdown(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + /* for SMA1307A */ + cancel_delayed_work_sync(&sma1307->check_fault_work); + + regmap_update_bits(sma1307->regmap, SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, SMA1307_SPK_MUTE); + /* Need to wait time for mute slope */ + msleep(55); + + regmap_update_bits(sma1307->regmap, SMA1307_10_SYSTEM_CTRL1, + SMA1307_SPK_MODE_MASK, SMA1307_SPK_OFF); + regmap_update_bits(sma1307->regmap, SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_OFF); + regmap_update_bits(sma1307->regmap, SMA1307_00_SYSTEM_CTRL, + SMA1307_POWER_MASK, SMA1307_POWER_OFF); +} + +static int sma1307_aif_in_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = sma1307->dapm_aif_in; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (mux) { + case SMA1307_MONO_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_ON); + break; + case SMA1307_LEFT_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_OFF); + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_LR_DATA_SW_MASK, + SMA1307_LR_DATA_SW_NORMAL); + break; + case SMA1307_RIGHT_MODE: + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_MONOMIX_MASK, + SMA1307_MONOMIX_OFF); + regmap_update_bits(sma1307->regmap, + SMA1307_11_SYSTEM_CTRL2, + SMA1307_LR_DATA_SW_MASK, + SMA1307_LR_DATA_SW_SWAP); + break; + default: + + dev_err(sma1307->dev, "%s: Invalid value (%d)\n", + __func__, mux); + return -EINVAL; + } + sma1307->amp_mode = mux; + break; + } + return 0; +} + +static int sma1307_sdo_setting_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = sma1307->dapm_sdo_setting; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (mux) { + case SMA1307_OUT_DATA_ONE_48K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_ONE_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_OUTPUT3_DIS + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_DATA_TWO_48K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_TWO_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_OUTPUT3_DIS + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_DATA_TWO_24K: + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_SDO_OUTPUT2_MASK, + SMA1307_TWO_SDO_PER_CH); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT3_MASK + | + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_TWO_SDO_PER_CH_24K + | SMA1307_SDO_DATA); + break; + case SMA1307_OUT_CLK_PLL: + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_CLK_PLL); + + break; + case SMA1307_OUT_CLK_OSC: + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_DATA_CLK_SEL_MASK, + SMA1307_SDO_CLK_OSC); + + break; + default: + dev_err(sma1307->dev, "%s: Invalid value (%d)\n", + __func__, mux); + return -EINVAL; + } + break; + } + return 0; +} + +static int sma1307_aif_out_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int mux = 0, val = 0, mask = 0; + + if (!strcmp(w->name, SMA1307_AIF_OUT0_NAME)) { + mux = sma1307->dapm_aif_out0; + val = mux; + mask = SMA1307_SDO_OUT0_SEL_MASK; + } else if (!strcmp(w->name, SMA1307_AIF_OUT1_NAME)) { + mux = sma1307->dapm_aif_out1; + val = mux << 3; + mask = SMA1307_SDO_OUT1_SEL_MASK; + } else { + dev_err(sma1307->dev, "%s: Invalid widget - %s\n", + __func__, w->name); + return -EINVAL; + } + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(sma1307->regmap, SMA1307_09_OUTPUT_CTRL, + mask, val); + break; + } + return 0; +} + +static int sma1307_sdo_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(sma1307->regmap, + SMA1307_09_OUTPUT_CTRL, + SMA1307_PORT_CONFIG_MASK, + SMA1307_OUTPUT_PORT_ENABLE); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT_MASK, + SMA1307_LOGIC_OUTPUT); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sma1307->regmap, + SMA1307_09_OUTPUT_CTRL, + SMA1307_PORT_CONFIG_MASK, + SMA1307_INPUT_PORT_ONLY); + regmap_update_bits(sma1307->regmap, + SMA1307_A3_TOP_MAN2, + SMA1307_SDO_OUTPUT_MASK, + SMA1307_HIGH_Z_OUTPUT); + break; + } + return 0; +} + +static int sma1307_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + sma1307_startup(component); + break; + case SND_SOC_DAPM_PRE_PMD: + sma1307_shutdown(component); + break; + } + return 0; +} + +static int sma1307_dapm_aif_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.enumerated.item[0] = (unsigned int)sma1307->dapm_aif_in; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_aif_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_aif_in_source_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_aif_in != val) { + change = true; + sma1307->dapm_aif_in = val; + } else + change = false; + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_sdo_setting_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.enumerated.item[0] = + (unsigned int)sma1307->dapm_sdo_setting; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_sdo_setting_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_sdo_setting_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_sdo_setting != val) { + change = true; + sma1307->dapm_sdo_setting = val; + } else + change = false; + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_aif_out_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + unsigned int val = 0; + + if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT0_NAME)) { + val = (unsigned int)sma1307->dapm_aif_out0; + } else if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT1_NAME)) { + val = (unsigned int)sma1307->dapm_aif_out1; + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + ucontrol->value.enumerated.item[0] = val; + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_aif_out_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.enumerated.item[0]; + bool change; + + if ((val < 0) || (val >= ARRAY_SIZE(sma1307_aif_out_source_text))) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT0_NAME)) { + if (sma1307->dapm_aif_out0 != val) { + change = true; + sma1307->dapm_aif_out0 = val; + } else + change = false; + } else if (!strcmp(kcontrol->id.name, SMA1307_AIF_OUT1_NAME)) { + if (sma1307->dapm_aif_out1 != val) { + change = true; + sma1307->dapm_aif_out1 = val; + } else + change = false; + } else { + dev_err(sma1307->dev, "%s: Invalid Control ID - %s\n", + __func__, kcontrol->id.name); + return -EINVAL; + } + + snd_soc_dapm_put_enum_double(kcontrol, ucontrol); + + return change; +} + +static int sma1307_dapm_sdo_enable_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + + ucontrol->value.integer.value[0] = (long)sma1307->dapm_sdo_en; + snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + return 0; +} + +static int sma1307_dapm_sdo_enable_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct sma1307_priv *sma1307 = + snd_soc_component_get_drvdata(dapm->component); + int val = (int)ucontrol->value.integer.value[0]; + bool change; + + if ((val < 0) || (val > 1)) { + dev_err(sma1307->dev, "%s: Out of range\n", __func__); + return -EINVAL; + } + + if (sma1307->dapm_sdo_en != val) { + change = true; + sma1307->dapm_sdo_en = val; + } else + change = false; + + snd_soc_dapm_put_volsw(kcontrol, ucontrol); + + return change; +} + +static const struct snd_kcontrol_new sma1307_aif_in_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_IN_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_in_get, + .put = sma1307_dapm_aif_in_put, + .private_value = (unsigned long)&sma1307_aif_in_source_enum +}; + +static const struct snd_kcontrol_new sma1307_sdo_setting_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SDO Setting", + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_sdo_setting_get, + .put = sma1307_dapm_sdo_setting_put, + .private_value = (unsigned long)&sma1307_sdo_setting_enum +}; + +static const struct snd_kcontrol_new sma1307_aif_out0_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_OUT0_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_out_get, + .put = sma1307_dapm_aif_out_put, + .private_value = (unsigned long)&sma1307_aif_out_source_enum +}; + +static const struct snd_kcontrol_new sma1307_aif_out1_source_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SMA1307_AIF_OUT1_NAME, + .info = snd_soc_info_enum_double, + .get = sma1307_dapm_aif_out_get, + .put = sma1307_dapm_aif_out_put, + .private_value = (unsigned long)&sma1307_aif_out_source_enum +}; + +static const struct snd_kcontrol_new sma1307_sdo_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Switch", + .info = snd_soc_info_volsw, + .get = sma1307_dapm_sdo_enable_get, + .put = sma1307_dapm_sdo_enable_put, + .private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, 0, 1, 0, 0) +}; + +static const struct snd_kcontrol_new sma1307_enable_control = + SOC_DAPM_SINGLE("Switch", SMA1307_00_SYSTEM_CTRL, 0, 1, 0); + +static const struct snd_kcontrol_new sma1307_binary_mode_control[] = { + SOC_ENUM_EXT("Binary Mode", sma1307_binary_mode_enum, + snd_soc_get_enum_double, sma1307_binary_mode_put), +}; + +static const struct snd_kcontrol_new sma1307_snd_controls[] = { + SOC_SINGLE_TLV(SMA1307_VOL_CTRL_NAME, SMA1307_0A_SPK_VOL, + 0, 167, 1, sma1307_spk_tlv), + SOC_ENUM_EXT(SMA1307_TDM_RX0_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_RX1_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_TX0_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_TDM_TX1_POS_NAME, sma1307_tdm_slot_enum, + sma1307_tdm_slot_get, sma1307_tdm_slot_put), + SOC_ENUM_EXT(SMA1307_RESET_CTRL_NAME, sma1307_reset_enum, + snd_soc_get_enum_double, sma1307_reset_put), + SOC_SINGLE_BOOL_EXT(SMA1307_FORCE_MUTE_CTRL_NAME, 0, + sma1307_force_mute_get, sma1307_force_mute_put), + SOC_SINGLE_BOOL_EXT(SMA1307_OT1_SW_PROT_CTRL_NAME, 0, + sma1307_sw_ot1_prot_get, sma1307_sw_ot1_prot_put), + SOC_SINGLE_BOOL_EXT(SMA1307_CHECK_FAULT_STATUS_NAME, 0, + sma1307_check_fault_status_get, + sma1307_check_fault_status_put), + SOC_SINGLE_EXT(SMA1307_CHECK_FAULT_PERIOD_NAME, SND_SOC_NOPM, 0, 600, 0, + sma1307_check_fault_period_get, + sma1307_check_fault_period_put), +}; + +static const struct snd_soc_dapm_widget sma1307_dapm_widgets[] = { + /* platform domain */ + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("SDO"), + + /* path domain */ + SND_SOC_DAPM_MUX_E(SMA1307_AIF_IN_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_in_source_control, + sma1307_aif_in_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("SDO Setting", SND_SOC_NOPM, 0, 0, + &sma1307_sdo_setting_control, + sma1307_sdo_setting_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MUX_E(SMA1307_AIF_OUT0_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_out0_source_control, + sma1307_aif_out_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MUX_E(SMA1307_AIF_OUT1_NAME, SND_SOC_NOPM, 0, 0, + &sma1307_aif_out1_source_control, + sma1307_aif_out_event, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SWITCH_E("SDO Enable", SND_SOC_NOPM, 0, 0, + &sma1307_sdo_control, + sma1307_sdo_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("Entry", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("AMP Power", SND_SOC_NOPM, 0, 0, NULL, 0, + sma1307_power_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("AMP Enable", SND_SOC_NOPM, 0, 0, + &sma1307_enable_control), + + /* stream domain */ + SND_SOC_DAPM_AIF_IN("AIF IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF OUT", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route sma1307_audio_map[] = { + /* Playback */ + { "AIF IN Source", "Mono", "AIF IN" }, + { "AIF IN Source", "Left", "AIF IN" }, + { "AIF IN Source", "Right", "AIF IN" }, + + { "SDO Enable", "Switch", "AIF IN" }, + + { "SDO Setting", "Data_One_48k", "SDO Enable" }, + { "SDO Setting", "Data_Two_48k", "SDO Enable" }, + { "SDO Setting", "Data_Two_24k", "SDO Enable" }, + { "SDO Setting", "Clk_PLL", "SDO Enable" }, + { "SDO Setting", "Clk_OSC", "SDO Enable" }, + + { "AIF OUT0 Source", "Disable", "SDO Setting" }, + { "AIF OUT0 Source", "After_FmtC", "SDO Setting" }, + { "AIF OUT0 Source", "After_Mixer", "SDO Setting" }, + { "AIF OUT0 Source", "After_DSP", "SDO Setting" }, + { "AIF OUT0 Source", "Vrms2_Avg", "SDO Setting" }, + { "AIF OUT0 Source", "Battery", "SDO Setting" }, + { "AIF OUT0 Source", "Temperature", "SDO Setting" }, + { "AIF OUT0 Source", "After_Delay", "SDO Setting" }, + + { "AIF OUT1 Source", "Disable", "SDO Setting" }, + { "AIF OUT1 Source", "After_FmtC", "SDO Setting" }, + { "AIF OUT1 Source", "After_Mixer", "SDO Setting" }, + { "AIF OUT1 Source", "After_DSP", "SDO Setting" }, + { "AIF OUT1 Source", "Vrms2_Avg", "SDO Setting" }, + { "AIF OUT1 Source", "Battery", "SDO Setting" }, + { "AIF OUT1 Source", "Temperature", "SDO Setting" }, + { "AIF OUT1 Source", "After_Delay", "SDO Setting" }, + + { "Entry", NULL, "AIF OUT0 Source" }, + { "Entry", NULL, "AIF OUT1 Source" }, + { "Entry", NULL, "AIF IN Source" }, + + { "AMP Power", NULL, "Entry" }, + + { "AMP Enable", "Switch", "AMP Power" }, + { "SPK", NULL, "AMP Enable" }, + + /* Capture */ + { "AIF OUT", NULL, "AMP Enable" }, +}; + +static void sma1307_setup_pll(struct snd_soc_component *component, + unsigned int bclk) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + int i = 0; + + dev_dbg(component->dev, "%s: BCLK = %dHz\n", __func__, bclk); + + if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_MCLK) { + dev_warn(component->dev, "%s: MCLK is not supported\n", + __func__); + } else if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_BCLK) { + for (i = 0; i < sma1307->num_of_pll_matches; i++) { + if (sma1307->pll_matches[i].input_clk == bclk) + break; + } + if (i == sma1307->num_of_pll_matches) { + dev_warn(component->dev, + "%s: No matching value between pll table and SCK\n", + __func__); + return; + } + + regmap_update_bits(sma1307->regmap, + SMA1307_A2_TOP_MAN1, + SMA1307_PLL_MASK, SMA1307_PLL_ON); + } + + regmap_write(sma1307->regmap, SMA1307_8B_PLL_POST_N, + sma1307->pll_matches[i].post_n); + regmap_write(sma1307->regmap, SMA1307_8C_PLL_N, + sma1307->pll_matches[i].n); + regmap_write(sma1307->regmap, SMA1307_8D_PLL_A_SETTING, + sma1307->pll_matches[i].vco); + regmap_write(sma1307->regmap, SMA1307_8E_PLL_P_CP, + sma1307->pll_matches[i].p_cp); +} + +static int sma1307_dai_hw_params_amp(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int bclk = 0; + int ret = 0; + + if (sma1307->format == SND_SOC_DAIFMT_DSP_A) + bclk = params_rate(params) * sma1307->frame_size; + else + bclk = params_rate(params) * params_physical_width(params) + * params_channels(params); + + dev_dbg(component->dev, + "%s: rate = %d : bit size = %d : channel = %d\n", + __func__, params_rate(params), params_width(params), + params_channels(params)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (sma1307->sys_clk_id == SMA1307_PLL_CLKIN_BCLK) { + if (sma1307->last_bclk != bclk) { + sma1307_setup_pll(component, bclk); + sma1307->last_bclk = bclk; + } + } + + switch (params_rate(params)) { + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 44100: + case 48000: + break; + + case 96000: + dev_warn(component->dev, + "%s: %d rate not support SDO\n", __func__, + params_rate(params)); + break; + + default: + dev_err(component->dev, "%s: not support rate : %d\n", + __func__, params_rate(params)); + + return -EINVAL; + } + + /* substream->stream is SNDRV_PCM_STREAM_CAPTURE */ + } else { + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_32FS | + SMA1307_DATA_16BIT); + break; + + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_64FS | + SMA1307_DATA_24BIT); + break; + + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_SCK_RATE_MASK + | + SMA1307_DATA_WIDTH_MASK, + SMA1307_SCK_64FS | + SMA1307_DATA_24BIT); + break; + default: + dev_err(component->dev, + "%s: not support data bit : %d\n", __func__, + params_format(params)); + return -EINVAL; + } + } + + switch (sma1307->format) { + case SND_SOC_DAIFMT_I2S: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_STANDARD_I2S); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_I2S_FORMAT); + break; + case SND_SOC_DAIFMT_LEFT_J: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, SMA1307_LJ); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_LJ_FORMAT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + switch (params_width(params)) { + case 16: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_RJ_16BIT); + break; + case 24: + case 32: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_RJ_24BIT); + break; + } + break; + case SND_SOC_DAIFMT_DSP_A: + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_I2S_MODE_MASK, + SMA1307_STANDARD_I2S); + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, + SMA1307_TDM_FORMAT); + break; + } + + switch (params_width(params)) { + case 16: + case 24: + case 32: + break; + default: + dev_err(component->dev, + "%s: not support data bit : %d\n", __func__, + params_format(params)); + return -EINVAL; + } + if (ret < 0) + return -EINVAL; + + return 0; +} + +static int sma1307_dai_set_sysclk_amp(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (clk_id) { + case SMA1307_EXTERNAL_CLOCK_19_2: + case SMA1307_EXTERNAL_CLOCK_24_576: + case SMA1307_PLL_CLKIN_MCLK: + case SMA1307_PLL_CLKIN_BCLK: + break; + default: + dev_err(component->dev, "%s: Invalid clk id: %d\n", + __func__, clk_id); + return -EINVAL; + } + sma1307->sys_clk_id = clk_id; + + return 0; +} + +static int sma1307_dai_set_fmt_amp(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + + case SND_SOC_DAIFMT_CBC_CFC: + dev_dbg(component->dev, + "%s: %s\n", __func__, "I2S/TDM Device mode"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_CONTROLLER_DEVICE_MASK, + SMA1307_DEVICE_MODE); + break; + + case SND_SOC_DAIFMT_CBP_CFP: + dev_dbg(component->dev, + "%s: %s\n", __func__, "I2S/TDM Controller mode"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_CONTROLLER_DEVICE_MASK, + SMA1307_CONTROLLER_MODE); + break; + + default: + dev_err(component->dev, + "%s: Unsupported Controller/Device : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + sma1307->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + dev_err(component->dev, + "%s: Unsupported Audio Interface Format : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + + case SND_SOC_DAIFMT_IB_NF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Invert BCLK + Normal Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_SCK_RISING_MASK, + SMA1307_SCK_RISING_EDGE); + break; + case SND_SOC_DAIFMT_IB_IF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Invert BCLK + Invert Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_LEFTPOL_MASK + | SMA1307_SCK_RISING_MASK, + SMA1307_HIGH_FIRST_CH + | SMA1307_SCK_RISING_EDGE); + break; + case SND_SOC_DAIFMT_NB_IF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Normal BCLK + Invert Frame"); + regmap_update_bits(sma1307->regmap, + SMA1307_01_INPUT_CTRL1, + SMA1307_LEFTPOL_MASK, + SMA1307_HIGH_FIRST_CH); + break; + case SND_SOC_DAIFMT_NB_NF: + dev_dbg(component->dev, "%s: %s\n", + __func__, "Normal BCLK + Normal Frame"); + break; + default: + dev_err(component->dev, + "%s: Unsupported Bit & Frameclock : 0x%x\n", + __func__, fmt); + return -EINVAL; + } + + return 0; +} + +static int sma1307_dai_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s: slots = %d, slot_width - %d\n", + __func__, slots, slot_width); + + sma1307->frame_size = slot_width * slots; + + regmap_update_bits(sma1307->regmap, + SMA1307_A4_TOP_MAN3, + SMA1307_INTERFACE_MASK, SMA1307_TDM_FORMAT); + + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_TX_MODE_MASK, + SMA1307_TDM_TX_MONO); + + switch (slot_width) { + case 16: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_DL_MASK, + SMA1307_TDM_DL_16); + break; + case 32: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_DL_MASK, + SMA1307_TDM_DL_32); + break; + default: + dev_err(component->dev, "%s: not support TDM %d slot_width\n", + __func__, slot_width); + return -EINVAL; + } + + switch (slots) { + case 4: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_N_SLOT_MASK, + SMA1307_TDM_N_SLOT_4); + break; + case 8: + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_N_SLOT_MASK, + SMA1307_TDM_N_SLOT_8); + break; + default: + dev_err(component->dev, "%s: not support TDM %d slots\n", + __func__, slots); + return -EINVAL; + } + + if (sma1307->tdm_slot0_rx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_SLOT0_RX_POS_MASK, + sma1307->tdm_slot0_rx << 3); + else + dev_err(component->dev, "%s: Incorrect tdm-slot0-rx %d set\n", + __func__, sma1307->tdm_slot0_rx); + + if (sma1307->tdm_slot1_rx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A5_TDM1, + SMA1307_TDM_SLOT1_RX_POS_MASK, + sma1307->tdm_slot1_rx); + else + dev_err(component->dev, "%s: Incorrect tdm-slot1-rx %d set\n", + __func__, sma1307->tdm_slot1_rx); + + if (sma1307->tdm_slot0_tx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_SLOT0_TX_POS_MASK, + sma1307->tdm_slot0_tx << 3); + else + dev_err(component->dev, "%s: Incorrect tdm-slot0-tx %d set\n", + __func__, sma1307->tdm_slot0_tx); + + if (sma1307->tdm_slot1_tx < slots) + regmap_update_bits(sma1307->regmap, + SMA1307_A6_TDM2, + SMA1307_TDM_SLOT1_TX_POS_MASK, + sma1307->tdm_slot1_tx); + else + dev_err(component->dev, "%s: Incorrect tdm-slot1-tx %d set\n", + __func__, sma1307->tdm_slot1_tx); + + return 0; +} + +static int sma1307_dai_mute_stream(struct snd_soc_dai *dai, int mute, + int stream) +{ + struct snd_soc_component *component = dai->component; + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + + if (stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + if (mute) { + dev_dbg(component->dev, "%s: %s\n", __func__, "MUTE"); + regmap_update_bits(sma1307->regmap, + SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, + SMA1307_SPK_MUTE); + } else { + if (!sma1307->force_mute_status) { + dev_dbg(component->dev, "%s: %s\n", __func__, + "UNMUTE"); + regmap_update_bits(sma1307->regmap, + SMA1307_0E_MUTE_VOL_CTRL, + SMA1307_SPK_MUTE_MASK, + SMA1307_SPK_UNMUTE); + } else { + dev_dbg(sma1307->dev, "%s: FORCE MUTE!!!\n", __func__); + } + } + + return 0; +} + +static const struct snd_soc_dai_ops sma1307_dai_ops_amp = { + .hw_params = sma1307_dai_hw_params_amp, + .set_fmt = sma1307_dai_set_fmt_amp, + .set_sysclk = sma1307_dai_set_sysclk_amp, + .set_tdm_slot = sma1307_dai_set_tdm_slot, + .mute_stream = sma1307_dai_mute_stream, +}; + +#define SMA1307_RATES_PLAYBACK SNDRV_PCM_RATE_8000_96000 +#define SMA1307_RATES_CAPTURE SNDRV_PCM_RATE_8000_48000 +#define SMA1307_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver sma1307_dai[] = { + { + .name = "sma1307-amplifier", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SMA1307_RATES_PLAYBACK, + .formats = SMA1307_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SMA1307_RATES_CAPTURE, + .formats = SMA1307_FORMATS, + }, + .ops = &sma1307_dai_ops_amp, + }, +}; + +static void sma1307_check_fault_worker(struct work_struct *work) +{ + struct sma1307_priv *sma1307 = + container_of(work, struct sma1307_priv, check_fault_work.work); + unsigned int status1_val, status2_val; + char *envp[3] = { NULL, NULL, NULL }; + + if (sma1307->tsdw_cnt) + regmap_read(sma1307->regmap, + SMA1307_0A_SPK_VOL, &sma1307->cur_vol); + else + regmap_read(sma1307->regmap, + SMA1307_0A_SPK_VOL, &sma1307->init_vol); + + regmap_read(sma1307->regmap, SMA1307_FA_STATUS1, &status1_val); + regmap_read(sma1307->regmap, SMA1307_FB_STATUS2, &status2_val); + + if (~status1_val & SMA1307_OT1_OK_STATUS) { + dev_crit(sma1307->dev, + "%s: OT1(Over Temperature Level 1)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT1"); + if (sma1307->sw_ot1_prot) { + /* Volume control (Current Volume -3dB) */ + if ((sma1307->cur_vol + 6) <= 0xFA) { + sma1307->cur_vol += 6; + regmap_write(sma1307->regmap, + SMA1307_0A_SPK_VOL, + sma1307->cur_vol); + envp[1] = kasprintf(GFP_KERNEL, + "VOLUME=0x%02X", sma1307->cur_vol); + } + } + sma1307->tsdw_cnt++; + } else if (sma1307->tsdw_cnt) { + regmap_write(sma1307->regmap, + SMA1307_0A_SPK_VOL, sma1307->init_vol); + sma1307->tsdw_cnt = 0; + sma1307->cur_vol = sma1307->init_vol; + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT1_CLEAR"); + envp[1] = kasprintf(GFP_KERNEL, + "VOLUME=0x%02X", sma1307->cur_vol); + } + + if (~status1_val & SMA1307_OT2_OK_STATUS) { + dev_crit(sma1307->dev, + "%s: OT2(Over Temperature Level 2)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OT2"); + } + if (status1_val & SMA1307_UVLO_STATUS) { + dev_crit(sma1307->dev, + "%s: UVLO(Under Voltage Lock Out)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=UVLO"); + } + if (status1_val & SMA1307_OVP_BST_STATUS) { + dev_crit(sma1307->dev, + "%s: OVP_BST(Over Voltage Protection)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OVP_BST"); + } + if (status2_val & SMA1307_OCP_SPK_STATUS) { + dev_crit(sma1307->dev, + "%s: OCP_SPK(Over Current Protect SPK)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OCP_SPK"); + } + if (status2_val & SMA1307_OCP_BST_STATUS) { + dev_crit(sma1307->dev, + "%s: OCP_BST(Over Current Protect Boost)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=OCP_BST"); + } + if (status2_val & SMA1307_CLK_MON_STATUS) { + dev_crit(sma1307->dev, + "%s: CLK_FAULT(No clock input)\n", __func__); + envp[0] = kasprintf(GFP_KERNEL, "STATUS=CLK_FAULT"); + } + + if (envp[0] != NULL) { + if (kobject_uevent_env(sma1307->kobj, KOBJ_CHANGE, envp)) + dev_err(sma1307->dev, + "%s: Error sending uevent\n", __func__); + kfree(envp[0]); + kfree(envp[1]); + } + + if (sma1307->check_fault_status) { + if (sma1307->check_fault_period > 0) + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + sma1307->check_fault_period * HZ); + else + queue_delayed_work(system_freezable_wq, + &sma1307->check_fault_work, + CHECK_PERIOD_TIME * HZ); + } +} + +static void sma1307_setting_loaded(struct sma1307_priv *sma1307, const char *file) +{ + const struct firmware *fw; + int *data, size, offset, num_mode; + + request_firmware(&fw, file, sma1307->dev); + + if (!fw) { + dev_err(sma1307->dev, "%s: failed to read \"%s\"\n", + __func__, setting_file); + release_firmware(fw); + sma1307->set.status = false; + return; + } else if ((fw->size) < SMA1307_SETTING_HEADER_SIZE) { + dev_err(sma1307->dev, "%s: Invalid file\n", __func__); + release_firmware(fw); + sma1307->set.status = false; + return; + } + + data = kzalloc(fw->size, GFP_KERNEL); + size = fw->size >> 2; + memcpy(data, fw->data, fw->size); + + release_firmware(fw); + + /* HEADER */ + sma1307->set.header_size = SMA1307_SETTING_HEADER_SIZE; + sma1307->set.checksum = data[sma1307->set.header_size - 2]; + sma1307->set.num_mode = data[sma1307->set.header_size - 1]; + num_mode = sma1307->set.num_mode; + sma1307->set.header = devm_kzalloc(sma1307->dev, + sma1307->set.header_size, + GFP_KERNEL); + memcpy(sma1307->set.header, data, + sma1307->set.header_size * sizeof(int)); + + if ((sma1307->set.checksum >> 8) != SMA1307_SETTING_CHECKSUM) { + dev_err(sma1307->dev, "%s: failed by dismatch \"%s\"\n", + __func__, setting_file); + sma1307->set.status = false; + return; + } + + /* DEFAULT */ + sma1307->set.def_size = SMA1307_SETTING_DEFAULT_SIZE; + sma1307->set.def + = devm_kzalloc(sma1307->dev, + sma1307->set.def_size * sizeof(int), GFP_KERNEL); + memcpy(sma1307->set.def, + &data[sma1307->set.header_size], + sma1307->set.def_size * sizeof(int)); + + /* MODE */ + offset = sma1307->set.header_size + sma1307->set.def_size; + sma1307->set.mode_size = DIV_ROUND_CLOSEST(size - offset, num_mode + 1); + for (int i = 0; i < num_mode; i++) { + sma1307->set.mode_set[i] + = devm_kzalloc(sma1307->dev, + sma1307->set.mode_size * 2 * sizeof(int), + GFP_KERNEL); + for (int j = 0; j < sma1307->set.mode_size; j++) { + sma1307->set.mode_set[i][2 * j] + = data[offset + ((num_mode + 1) * j)]; + sma1307->set.mode_set[i][2 * j + 1] + = data[offset + ((num_mode + 1) * j + i + 1)]; + } + } + + kfree(data); + sma1307->set.status = true; + +} + +static void sma1307_reset(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + unsigned int status = 0; + + regmap_read(sma1307->regmap, SMA1307_FF_DEVICE_INDEX, &status); + + sma1307->rev_num = status & SMA1307_REV_NUM_STATUS; + dev_dbg(component->dev, "%s: SMA1307 Revision %d\n", + __func__, sma1307->rev_num); + regmap_read(sma1307->regmap, SMA1307_99_OTP_TRM2, &sma1307->otp_trm2); + regmap_read(sma1307->regmap, SMA1307_9A_OTP_TRM3, &sma1307->otp_trm3); + + if ((sma1307->otp_trm2 & SMA1307_OTP_STAT_MASK) != SMA1307_OTP_STAT_1) + dev_warn(component->dev, "%s: SMA1307 OTP Status Fail\n", + __func__); + + /* Register Initial Value Setting */ + sma1307_setting_loaded(sma1307, setting_file); + if (sma1307->set.status) + sma1307_set_binary(component); + else + sma1307_set_default(component); + + regmap_update_bits(sma1307->regmap, + SMA1307_93_INT_CTRL, + SMA1307_DIS_INT_MASK, SMA1307_HIGH_Z_INT); + regmap_write(sma1307->regmap, SMA1307_0A_SPK_VOL, sma1307->init_vol); +} + +static void sma1307_set_binary(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int i = 0, mode = 0; + + for (i = 0; i < (sma1307->set.def_size); i++) { + if (sma1307_writeable_register(sma1307->dev, i) + && ((i < SMA1307_97_OTP_TRM0) + || (i > SMA1307_9A_OTP_TRM3))) { + regmap_write(sma1307->regmap, i, sma1307->set.def[i]); + + } + } + for (i = 0; i < (sma1307->set.mode_size); i++) { + if (sma1307_writeable_register(sma1307->dev, i) + && ((i < SMA1307_97_OTP_TRM0) + || (i > SMA1307_9A_OTP_TRM3))) { + mode = sma1307->binary_mode; + regmap_write(sma1307->regmap, + sma1307->set.mode_set[mode][2 * i], + sma1307->set.mode_set[mode][2 * i + + 1]); + } + } +} + +static void sma1307_set_default(struct snd_soc_component *component) +{ + struct sma1307_priv *sma1307 = snd_soc_component_get_drvdata(component); + int i = 0; + + for (i = 0; i < (unsigned int)ARRAY_SIZE(sma1307_reg_def); i++) + regmap_write(sma1307->regmap, + sma1307_reg_def[i].reg, + sma1307_reg_def[i].def); + + if (!strcmp(sma1307->name, DEVICE_NAME_SMA1307AQ)) + sma1307->data->init(sma1307->regmap); +} + +static int sma1307_probe(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + snd_soc_dapm_sync(dapm); + + sma1307_amp_component = component; + + snd_soc_add_component_controls(component, sma1307_binary_mode_control, + ARRAY_SIZE(sma1307_binary_mode_control)); + sma1307_reset(component); + + return 0; +} + +static const struct snd_soc_component_driver sma1307_component = { + .probe = sma1307_probe, + .controls = sma1307_snd_controls, + .num_controls = ARRAY_SIZE(sma1307_snd_controls), + .dapm_widgets = sma1307_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sma1307_dapm_widgets), + .dapm_routes = sma1307_audio_map, + .num_dapm_routes = ARRAY_SIZE(sma1307_audio_map), +}; + +static const struct regmap_config sma_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = SMA1307_FF_DEVICE_INDEX, + .readable_reg = sma1307_readable_register, + .writeable_reg = sma1307_writeable_register, + .volatile_reg = sma1307_volatile_register, + + .reg_defaults = sma1307_reg_def, + .num_reg_defaults = ARRAY_SIZE(sma1307_reg_def), +}; + +static void sma1307aq_init(struct regmap *regmap) +{ + /* Guidelines for driving 4ohm load */ + /* Brown Out Protection */ + regmap_write(regmap, SMA1307_02_BROWN_OUT_PROT1, 0x62); + regmap_write(regmap, SMA1307_03_BROWN_OUT_PROT2, 0x5D); + regmap_write(regmap, SMA1307_04_BROWN_OUT_PROT3, 0x57); + regmap_write(regmap, SMA1307_05_BROWN_OUT_PROT8, 0x54); + regmap_write(regmap, SMA1307_06_BROWN_OUT_PROT9, 0x51); + regmap_write(regmap, + SMA1307_07_BROWN_OUT_PROT10, 0x4D); + regmap_write(regmap, + SMA1307_08_BROWN_OUT_PROT11, 0x4B); + regmap_write(regmap, SMA1307_27_BROWN_OUT_PROT4, 0x3C); + regmap_write(regmap, SMA1307_28_BROWN_OUT_PROT5, 0x5B); + regmap_write(regmap, + SMA1307_29_BROWN_OUT_PROT12, 0x78); + regmap_write(regmap, + SMA1307_2A_BROWN_OUT_PROT13, 0x96); + regmap_write(regmap, + SMA1307_2B_BROWN_OUT_PROT14, 0xB4); + regmap_write(regmap, + SMA1307_2C_BROWN_OUT_PROT15, 0xD3); + /* FDPEC Gain */ + regmap_write(regmap, SMA1307_35_FDPEC_CTRL0, 0x16); + /* FLT Vdd */ + regmap_write(regmap, SMA1307_92_FDPEC_CTRL1, 0xA0); + /* Boost Max */ + regmap_write(regmap, SMA1307_AB_BOOST_CTRL4, 0x0F); +} + +static const struct sma1307_data sma1307aq_data = { + .name = DEVICE_NAME_SMA1307AQ, + .init = sma1307aq_init, +}; + +static int sma1307_i2c_probe(struct i2c_client *client) +{ + struct sma1307_priv *sma1307; + const struct sma1307_data *data; + int ret = 0; + unsigned int device_info; + + sma1307 = devm_kzalloc(&client->dev, + sizeof(*sma1307), GFP_KERNEL); + if (!sma1307) + return -ENOMEM; + + sma1307->regmap = devm_regmap_init_i2c(client, &sma_i2c_regmap); + if (IS_ERR(sma1307->regmap)) { + return dev_err_probe(&client->dev, PTR_ERR(sma1307->regmap), + "%s: failed to allocate register map\n", __func__); + } + + data = device_get_match_data(&client->dev); + if (!data) + return -ENODEV; + + sma1307->data = data; + + /* set initial value as normal AMP IC status */ + sma1307->name = client->name; + sma1307->format = SND_SOC_DAIFMT_I2S; + sma1307->sys_clk_id = SMA1307_PLL_CLKIN_BCLK; + sma1307->num_of_pll_matches = ARRAY_SIZE(sma1307_pll_matches); + + sma1307->check_fault_period = CHECK_PERIOD_TIME; + sma1307->check_fault_status = true; + sma1307->init_vol = 0x32; + sma1307->cur_vol = sma1307->init_vol; + sma1307->sw_ot1_prot = true; + + mutex_init(&sma1307->default_lock); + + INIT_DELAYED_WORK(&sma1307->check_fault_work, + sma1307_check_fault_worker); + + sma1307->dev = &client->dev; + sma1307->kobj = &client->dev.kobj; + + i2c_set_clientdata(client, sma1307); + + sma1307->pll_matches = sma1307_pll_matches; + + regmap_read(sma1307->regmap, + SMA1307_FF_DEVICE_INDEX, &device_info); + + if ((device_info & 0xF8) != SMA1307_DEVICE_ID) { + dev_err(&client->dev, + "%s: device initialization error (0x%02X)", + __func__, device_info); + return -ENODEV; + } + dev_dbg(&client->dev, "%s: chip version 0x%02X\n", + __func__, device_info); + + i2c_set_clientdata(client, sma1307); + + ret = devm_snd_soc_register_component(&client->dev, + &sma1307_component, sma1307_dai, + 1); + + if (ret) { + dev_err(&client->dev, "%s: failed to register component\n", + __func__); + + return ret; + } + + return ret; +} + +static void sma1307_i2c_remove(struct i2c_client *client) +{ + struct sma1307_priv *sma1307 = + (struct sma1307_priv *)i2c_get_clientdata(client); + + cancel_delayed_work_sync(&sma1307->check_fault_work); +} + +static const struct i2c_device_id sma1307_i2c_id[] = { + { "sma1307a", 0 }, + { "sma1307aq", 0 }, + { } +}; + +MODULE_DEVICE_TABLE(i2c, sma1307_i2c_id); + +static const struct of_device_id sma1307_of_match[] = { + { + .compatible = "irondevice,sma1307a", + }, + { + .compatible = "irondevice,sma1307aq", + .data = &sma1307aq_data //AEC-Q100 Qualificated + }, + { } +}; + +MODULE_DEVICE_TABLE(of, sma1307_of_match); + +static struct i2c_driver sma1307_i2c_driver = { + .driver = { + .name = "sma1307", + .of_match_table = sma1307_of_match, + }, + .probe = sma1307_i2c_probe, + .remove = sma1307_i2c_remove, + .id_table = sma1307_i2c_id, +}; + +module_i2c_driver(sma1307_i2c_driver); + +MODULE_DESCRIPTION("ALSA SoC SMA1307 driver"); +MODULE_AUTHOR("Gyuhwa Park, <gyuhwa.park(a)irondevice.com>"); +MODULE_AUTHOR("KS Jo, <kiseok.jo(a)irondevice.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sma1307.h b/sound/soc/codecs/sma1307.h new file mode 100644 index 000000000..44aab52a3 --- /dev/null +++ b/sound/soc/codecs/sma1307.h @@ -0,0 +1,444 @@ +/* SPDX-License-Identifier: GPL-2.0-or-later + * sma1307.h -- sma1307 ALSA SoC Audio driver + * + * Copyright 2024 Iron Device Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _SMA1307_H +#define _SMA1307_H + +#include <sound/soc.h> + +enum sma1307_fault { + SMA1307_FAULT_OT1, + SMA1307_FAULT_OT2, + SMA1307_FAULT_UVLO, + SMA1307_FAULT_OVP_BST, + SMA1307_FAULT_OCP_SPK, + SMA1307_FAULT_OCP_BST, + SMA1307_FAULT_CLK +}; + +enum sma1307_mode { + SMA1307_MONO_MODE, + SMA1307_LEFT_MODE, + SMA1307_RIGHT_MODE, +}; + +enum sma1307_sdo_mode { + SMA1307_OUT_DATA_ONE_48K, + SMA1307_OUT_DATA_TWO_48K, + SMA1307_OUT_DATA_TWO_24K, + SMA1307_OUT_CLK_PLL, + SMA1307_OUT_CLK_OSC +}; + +enum sma1307_sdo_source { + SMA1307_OUT_DISABLE, + SMA1307_OUT_FORMAT_C, + SMA1307_OUT_MIXER_OUT, + SMA1307_OUT_AFTER_DSP, + SMA1307_OUT_VRMS2_AVG, + SMA1307_OUT_BATTERY, + SMA1307_OUT_TEMP, + SMA1307_OUT_AFTER_DELAY +}; + +struct sma1307_setting_file { + bool status; + char *header; + int *def; + int *mode_set[5]; + int checksum; + int num_mode; + size_t header_size; + size_t def_size; + size_t mode_size; +}; + +#define SMA1307_I2C_ADDR_00 0x1e +#define SMA1307_I2C_ADDR_01 0x3e +#define SMA1307_I2C_ADDR_10 0x5e +#define SMA1307_I2C_ADDR_11 0x7e + +#define DEVICE_NAME_SMA1307A "sma1307a" +#define DEVICE_NAME_SMA1307AQ "sma1307aq" + +#define SMA1307_EXTERNAL_CLOCK_19_2 0x00 +#define SMA1307_EXTERNAL_CLOCK_24_576 0x01 +#define SMA1307_PLL_CLKIN_MCLK 0x02 +#define SMA1307_PLL_CLKIN_BCLK 0x03 + +#define SMA1307_OFFSET_DEFAULT_MODE 0x00 +#define SMA1307_OFFSET_BURNING_MODE 0x01 + +#define SMA1307_SETTING_HEADER_SIZE 0x08 +#define SMA1307_SETTING_DEFAULT_SIZE 0xC0 + +#define SMA1307_DEFAULT_SET 0x00 +#define SMA1307_BINARY_FILE_SET 0x01 + +/* Controls Name */ +#define SMA1307_REG_CTRL_NAME "Register Byte Control" +#define SMA1307_VOL_CTRL_NAME "Speaker Volume" +#define SMA1307_FORCE_MUTE_CTRL_NAME "Force Mute Switch" +#define SMA1307_TDM_RX0_POS_NAME "TDM RX Slot0 Position" +#define SMA1307_TDM_RX1_POS_NAME "TDM RX Slot1 Position" +#define SMA1307_TDM_TX0_POS_NAME "TDM TX Slot0 Position" +#define SMA1307_TDM_TX1_POS_NAME "TDM TX Slot1 Position" +#define SMA1307_OT1_SW_PROT_CTRL_NAME "OT1 SW Protection Switch" +#define SMA1307_RESET_CTRL_NAME "Reset Switch" +#define SMA1307_CHECK_FAULT_STATUS_NAME "Check Fault Status" +#define SMA1307_CHECK_FAULT_PERIOD_NAME "Check Fault Period" + +/* DAPM Name */ +#define SMA1307_AIF_IN_NAME "AIF IN Source" +#define SMA1307_AIF_OUT0_NAME "AIF OUT0 Source" +#define SMA1307_AIF_OUT1_NAME "AIF OUT1 Source" + +/* + * SMA1307 Register Definition + */ + +/* SMA1307 Register Addresses */ +#define SMA1307_00_SYSTEM_CTRL 0x00 +#define SMA1307_01_INPUT_CTRL1 0x01 +#define SMA1307_02_BROWN_OUT_PROT1 0x02 +#define SMA1307_03_BROWN_OUT_PROT2 0x03 +#define SMA1307_04_BROWN_OUT_PROT3 0x04 +#define SMA1307_05_BROWN_OUT_PROT8 0x05 +#define SMA1307_06_BROWN_OUT_PROT9 0x06 +#define SMA1307_07_BROWN_OUT_PROT10 0x07 +#define SMA1307_08_BROWN_OUT_PROT11 0x08 +#define SMA1307_09_OUTPUT_CTRL 0x09 +#define SMA1307_0A_SPK_VOL 0x0A +#define SMA1307_0B_BST_TEST 0x0B +#define SMA1307_0C_BOOST_CTRL8 0x0C +#define SMA1307_0D_SPK_TEST 0x0D +#define SMA1307_0E_MUTE_VOL_CTRL 0x0E +#define SMA1307_0F_VBAT_TEMP_SENSING 0x0F + +#define SMA1307_10_SYSTEM_CTRL1 0x10 +#define SMA1307_11_SYSTEM_CTRL2 0x11 +#define SMA1307_12_SYSTEM_CTRL3 0x12 +#define SMA1307_13_DELAY 0x13 +#define SMA1307_14_MODULATOR 0x14 +#define SMA1307_15_BASS_SPK1 0x15 +#define SMA1307_16_BASS_SPK2 0x16 +#define SMA1307_17_BASS_SPK3 0x17 +#define SMA1307_18_BASS_SPK4 0x18 +#define SMA1307_19_BASS_SPK5 0x19 +#define SMA1307_1A_BASS_SPK6 0x1A +#define SMA1307_1B_BASS_SPK7 0x1B +#define SMA1307_1C_BROWN_OUT_PROT20 0x1C +#define SMA1307_1D_BROWN_OUT_PROT0 0x1D +#define SMA1307_1E_TONE_GENERATOR 0x1E +#define SMA1307_1F_TONE_FINE_VOLUME 0x1F + +#define SMA1307_22_COMP_HYS_SEL 0x22 +#define SMA1307_23_COMPLIM1 0x23 +#define SMA1307_24_COMPLIM2 0x24 +#define SMA1307_25_COMPLIM3 0x25 +#define SMA1307_26_COMPLIM4 0x26 +#define SMA1307_27_BROWN_OUT_PROT4 0x27 +#define SMA1307_28_BROWN_OUT_PROT5 0x28 +#define SMA1307_29_BROWN_OUT_PROT12 0x29 +#define SMA1307_2A_BROWN_OUT_PROT13 0x2A +#define SMA1307_2B_BROWN_OUT_PROT14 0x2B +#define SMA1307_2C_BROWN_OUT_PROT15 0x2C +#define SMA1307_2D_BROWN_OUT_PROT6 0x2D +#define SMA1307_2E_BROWN_OUT_PROT7 0x2E +#define SMA1307_2F_BROWN_OUT_PROT16 0x2F + +#define SMA1307_30_BROWN_OUT_PROT17 0x30 +#define SMA1307_31_BROWN_OUT_PROT18 0x31 +#define SMA1307_32_BROWN_OUT_PROT19 0x32 +#define SMA1307_34_OCP_SPK 0x34 +#define SMA1307_35_FDPEC_CTRL0 0x35 +#define SMA1307_36_PROTECTION 0x36 +#define SMA1307_37_SLOPECTRL 0x37 +#define SMA1307_38_POWER_METER 0x38 +#define SMA1307_39_PMT_NZ_VAL 0x39 +#define SMA1307_3B_TEST1 0x3B +#define SMA1307_3C_TEST2 0x3C +#define SMA1307_3D_TEST3 0x3D +#define SMA1307_3E_IDLE_MODE_CTRL 0x3E +#define SMA1307_3F_ATEST2 0x3F +#define SMA1307_8B_PLL_POST_N 0x8B +#define SMA1307_8C_PLL_N 0x8C +#define SMA1307_8D_PLL_A_SETTING 0x8D +#define SMA1307_8E_PLL_P_CP 0x8E +#define SMA1307_8F_ANALOG_TEST 0x8F + +#define SMA1307_90_CRESTLIM1 0x90 +#define SMA1307_91_CRESTLIM2 0x91 +#define SMA1307_92_FDPEC_CTRL1 0x92 +#define SMA1307_93_INT_CTRL 0x93 +#define SMA1307_94_BOOST_CTRL9 0x94 +#define SMA1307_95_BOOST_CTRL10 0x95 +#define SMA1307_96_BOOST_CTRL11 0x96 +#define SMA1307_97_OTP_TRM0 0x97 +#define SMA1307_98_OTP_TRM1 0x98 +#define SMA1307_99_OTP_TRM2 0x99 +#define SMA1307_9A_OTP_TRM3 0x9A + +#define SMA1307_A0_PAD_CTRL0 0xA0 +#define SMA1307_A1_PAD_CTRL1 0xA1 +#define SMA1307_A2_TOP_MAN1 0xA2 +#define SMA1307_A3_TOP_MAN2 0xA3 +#define SMA1307_A4_TOP_MAN3 0xA4 +#define SMA1307_A5_TDM1 0xA5 +#define SMA1307_A6_TDM2 0xA6 +#define SMA1307_A7_CLK_MON 0xA7 +#define SMA1307_A8_BOOST_CTRL1 0xA8 +#define SMA1307_A9_BOOST_CTRL2 0xA9 +#define SMA1307_AA_BOOST_CTRL3 0xAA +#define SMA1307_AB_BOOST_CTRL4 0xAB +#define SMA1307_AC_BOOST_CTRL5 0xAC +#define SMA1307_AD_BOOST_CTRL6 0xAD +#define SMA1307_AE_BOOST_CTRL7 0xAE +#define SMA1307_AF_LPF 0xAF + +#define SMA1307_B0_RMS_TC1 0xB0 +#define SMA1307_B1_RMS_TC2 0xB1 +#define SMA1307_B2_AVG_TC1 0xB2 +#define SMA1307_B3_AVG_TC2 0xB3 +#define SMA1307_B4_PRVALUE1 0xB4 +#define SMA1307_B5_PRVALUE2 0xB5 +#define SMA1307_B8_SPK_NG_CTRL1 0xB8 +#define SMA1307_B9_SPK_NG_CTRL2 0xB9 +#define SMA1307_BA_DGC1 0xBA +#define SMA1307_BB_DGC2 0xBB +#define SMA1307_BC_DGC3 0xBC +#define SMA1307_BD_MCBS_CTRL1 0xBD +#define SMA1307_BE_MCBS_CTRL2 0xBE + +/* Status Register Read Only */ +#define SMA1307_F5_READY_FOR_V_SAR 0xF5 +#define SMA1307_F7_READY_FOR_T_SAR 0xF7 +#define SMA1307_F8_STATUS_T1 0xF8 +#define SMA1307_F9_STATUS_T2 0xF9 +#define SMA1307_FA_STATUS1 0xFA +#define SMA1307_FB_STATUS2 0xFB +#define SMA1307_FC_STATUS3 0xFC +#define SMA1307_FD_STATUS4 0xFD +#define SMA1307_FE_STATUS5 0xFE +#define SMA1307_FF_DEVICE_INDEX 0xFF + +/* SMA1307 Registers Bit Fields */ +/* Power On/Off */ +#define SMA1307_POWER_MASK BIT(0) +#define SMA1307_POWER_OFF 0 +#define SMA1307_POWER_ON BIT(0) + +/* Reset */ +#define SMA1307_RESET_MASK BIT(1) +#define SMA1307_RESET_ON BIT(1) + +/* Left Polarity */ +#define SMA1307_LEFTPOL_MASK BIT(3) +#define SMA1307_LOW_FIRST_CH 0 +#define SMA1307_HIGH_FIRST_CH BIT(3) + +/* SCK Falling/Rising */ +#define SMA1307_SCK_RISING_MASK BIT(2) +#define SMA1307_SCK_FALLING_EDGE 0 +#define SMA1307_SCK_RISING_EDGE BIT(2) + +/* SPK Mute */ +#define SMA1307_SPK_MUTE_MASK BIT(0) +#define SMA1307_SPK_UNMUTE 0 +#define SMA1307_SPK_MUTE BIT(0) + +/* SPK Mode */ +#define SMA1307_SPK_MODE_MASK (BIT(2)|BIT(3)|BIT(4)) +#define SMA1307_SPK_OFF 0 +#define SMA1307_SPK_MONO BIT(2) +#define SMA1307_SPK_STEREO BIT(4) + +/* Mono Mix */ +#define SMA1307_MONOMIX_MASK BIT(0) +#define SMA1307_MONOMIX_OFF 0 +#define SMA1307_MONOMIX_ON BIT(0) + +/* LR Data Swap */ +#define SMA1307_LR_DATA_SW_MASK BIT(4) +#define SMA1307_LR_DATA_SW_NORMAL 0 +#define SMA1307_LR_DATA_SW_SWAP BIT(4) + +/* PLL On/Off */ +#define SMA1307_PLL_MASK BIT(6) +#define SMA1307_PLL_ON 0 +#define SMA1307_PLL_OFF BIT(6) + +/* Input Format */ +#define SMA1307_I2S_MODE_MASK (BIT(4)|BIT(5)|BIT(6)) +#define SMA1307_STANDARD_I2S 0 +#define SMA1307_LJ BIT(4) +#define SMA1307_RJ_16BIT BIT(6) +#define SMA1307_RJ_18BIT (BIT(4)|BIT(6)) +#define SMA1307_RJ_20BIT (BIT(5)|BIT(6)) +#define SMA1307_RJ_24BIT (BIT(4)|BIT(5)|BIT(6)) + +/* Controller / Device Setting */ +#define SMA1307_CONTROLLER_DEVICE_MASK BIT(7) +#define SMA1307_DEVICE_MODE 0 +#define SMA1307_CONTROLLER_MODE BIT(7) + +/* Port Config */ +#define SMA1307_PORT_CONFIG_MASK (BIT(6)|BIT(7)) +#define SMA1307_INPUT_PORT_ONLY 0 +#define SMA1307_OUTPUT_PORT_ENABLE BIT(7) + +/* SDO Output */ +#define SMA1307_SDO_OUTPUT_MASK BIT(3) +#define SMA1307_LOGIC_OUTPUT 0 +#define SMA1307_HIGH_Z_OUTPUT BIT(3) + +#define SMA1307_DATA_CLK_SEL_MASK (BIT(6)|BIT(7)) +#define SMA1307_SDO_DATA 0 +#define SMA1307_SDO_CLK_PLL BIT(6) +#define SMA1307_SDO_CLK_OSC (BIT(6)|BIT(7)) + +/* SDO Output2 */ +#define SMA1307_SDO_OUTPUT2_MASK BIT(0) +#define SMA1307_ONE_SDO_PER_CH 0 +#define SMA1307_TWO_SDO_PER_CH BIT(0) + +/* SDO Output3 */ +#define SMA1307_SDO_OUTPUT3_MASK BIT(2) +#define SMA1307_SDO_OUTPUT3_DIS 0 +#define SMA1307_TWO_SDO_PER_CH_24K BIT(2) + +/* SDO OUT1 Select*/ +#define SMA1307_SDO_OUT1_SEL_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_SDO1_DISABLE 0 +#define SMA1307_SDO1_FORMAT_C BIT(3) +#define SMA1307_SDO1_MONO_MIX BIT(4) +#define SMA1307_SDO1_AFTER_DSP (BIT(3)|BIT(4)) +#define SMA1307_SDO1_VRMS2_AVG BIT(5) +#define SMA1307_SDO1_VBAT_MON (BIT(3)|BIT(5)) +#define SMA1307_SDO1_TEMP_MON (BIT(4)|BIT(5)) +#define SMA1307_SDO1_AFTER_DELAY (BIT(3)|BIT(4)|BIT(5)) + +/* SDO OUT0 Select*/ +#define SMA1307_SDO_OUT0_SEL_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_SDO0_DISABLE 0 +#define SMA1307_SDO0_FORMAT_C BIT(0) +#define SMA1307_SDO0_MONO_MIX BIT(1) +#define SMA1307_SDO0_AFTER_DSP (BIT(0)|BIT(1)) +#define SMA1307_SDO0_VRMS2_AVG BIT(2) +#define SMA1307_SDO0_VBAT_MON (BIT(0)|BIT(2)) +#define SMA1307_SDO0_TEMP_MON (BIT(1)|BIT(2)) +#define SMA1307_SDO0_AFTER_DELAY (BIT(0)|BIT(1)|BIT(2)) + +/* INTERRUPT Operation */ +#define SMA1307_SEL_INT_MASK BIT(2) +#define SMA1307_INT_CLEAR_AUTO 0 +#define SMA1307_INT_CLEAR_MANUAL BIT(2) + +/* INTERRUPT CLEAR */ +#define SMA1307_CLR_INT_MASK BIT(1) +#define SMA1307_INT_READY 0 +#define SMA1307_INT_CLEAR BIT(1) + +/* INTERRUPT Disable */ +#define SMA1307_DIS_INT_MASK BIT(0) +#define SMA1307_NORMAL_INT 0 +#define SMA1307_HIGH_Z_INT BIT(0) + +/* Interface Control */ +#define SMA1307_INTERFACE_MASK (BIT(5)|BIT(6)|BIT(7)) +#define SMA1307_LJ_FORMAT BIT(5) +#define SMA1307_I2S_FORMAT (BIT(5)|BIT(6)) +#define SMA1307_TDM_FORMAT BIT(7) + +#define SMA1307_SCK_RATE_MASK (BIT(3)|BIT(4)) +#define SMA1307_SCK_64FS 0 +#define SMA1307_SCK_32FS BIT(4) + +#define SMA1307_DATA_WIDTH_MASK (BIT(1)|BIT(2)) +#define SMA1307_DATA_24BIT 0 +#define SMA1307_DATA_16BIT (BIT(1)|BIT(2)) + +#define SMA1307_TDM_TX_MODE_MASK BIT(6) +#define SMA1307_TDM_TX_MONO 0 +#define SMA1307_TDM_TX_STEREO BIT(6) + +#define SMA1307_TDM_SLOT0_RX_POS_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_0 0 +#define SMA1307_TDM_SLOT0_RX_POS_1 BIT(3) +#define SMA1307_TDM_SLOT0_RX_POS_2 BIT(4) +#define SMA1307_TDM_SLOT0_RX_POS_3 (BIT(3)|BIT(4)) +#define SMA1307_TDM_SLOT0_RX_POS_4 BIT(5) +#define SMA1307_TDM_SLOT0_RX_POS_5 (BIT(3)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_6 (BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_RX_POS_7 (BIT(3)|BIT(4)|BIT(5)) + +#define SMA1307_TDM_SLOT1_RX_POS_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_0 0 +#define SMA1307_TDM_SLOT1_RX_POS_1 BIT(0) +#define SMA1307_TDM_SLOT1_RX_POS_2 BIT(1) +#define SMA1307_TDM_SLOT1_RX_POS_3 (BIT(0)|BIT(1)) +#define SMA1307_TDM_SLOT1_RX_POS_4 BIT(2) +#define SMA1307_TDM_SLOT1_RX_POS_5 (BIT(0)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_6 (BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_RX_POS_7 (BIT(0)|BIT(1)|BIT(2)) + +/* TDM2 FORMAT : 0xA6 */ +#define SMA1307_TDM_DL_MASK BIT(7) +#define SMA1307_TDM_DL_16 0 +#define SMA1307_TDM_DL_32 BIT(7) + +#define SMA1307_TDM_N_SLOT_MASK BIT(6) +#define SMA1307_TDM_N_SLOT_4 0 +#define SMA1307_TDM_N_SLOT_8 BIT(6) + +#define SMA1307_TDM_SLOT0_TX_POS_MASK (BIT(3)|BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_0 0 +#define SMA1307_TDM_SLOT0_TX_POS_1 BIT(3) +#define SMA1307_TDM_SLOT0_TX_POS_2 BIT(4) +#define SMA1307_TDM_SLOT0_TX_POS_3 (BIT(3)|BIT(4)) +#define SMA1307_TDM_SLOT0_TX_POS_4 BIT(5) +#define SMA1307_TDM_SLOT0_TX_POS_5 (BIT(3)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_6 (BIT(4)|BIT(5)) +#define SMA1307_TDM_SLOT0_TX_POS_7 (BIT(3)|BIT(4)|BIT(5)) + +#define SMA1307_TDM_SLOT1_TX_POS_MASK (BIT(0)|BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_0 0 +#define SMA1307_TDM_SLOT1_TX_POS_1 BIT(0) +#define SMA1307_TDM_SLOT1_TX_POS_2 BIT(1) +#define SMA1307_TDM_SLOT1_TX_POS_3 (BIT(0)|BIT(1)) +#define SMA1307_TDM_SLOT1_TX_POS_4 BIT(2) +#define SMA1307_TDM_SLOT1_TX_POS_5 (BIT(0)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_6 (BIT(1)|BIT(2)) +#define SMA1307_TDM_SLOT1_TX_POS_7 (BIT(0)|BIT(1)|BIT(2)) + +/* OTP STATUS */ +#define SMA1307_OTP_STAT_MASK BIT(6) +#define SMA1307_OTP_STAT_0 0 +#define SMA1307_OTP_STAT_1 BIT(6) + +/* STATUS */ +#define SMA1307_OT1_OK_STATUS BIT(7) +#define SMA1307_OT2_OK_STATUS BIT(6) +#define SMA1307_UVLO_STATUS BIT(5) +#define SMA1307_OVP_BST_STATUS BIT(4) +#define SMA1307_POWER_FLAG BIT(3) + +#define SMA1307_SCAN_CHK BIT(7) +#define SMA1307_OCP_SPK_STATUS BIT(5) +#define SMA1307_OCP_BST_STATUS BIT(4) +#define SMA1307_BOP_STATE (BIT(1)|BIT(2)|BIT(3)) +#define SMA1307_CLK_MON_STATUS BIT(0) + +#define SMA1307_DEVICE_ID (BIT(3)|BIT(4)) +#define SMA1307_REV_NUM_STATUS (BIT(0)|BIT(1)) +#define SMA1307_REV_NUM_REV0 0 +#define SMA1307_REV_NUM_REV1 BIT(0) + +#endif -- 2.39.2

2 2

[PATCH v1] ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip
by Shenghao Ding 04 Nov '24

04 Nov '24

Add new driver version to support tas2563 & tas2781 qfn chip Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-fmwlib.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index ae360c97fe1e..0aeb88abbf52 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -1992,6 +1992,7 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, break; case 0x202: case 0x400: + case 0x401: tas_priv->fw_parse_variable_header = fw_parse_variable_header_git; tas_priv->fw_parse_program_data = -- 2.34.1

2 1

[tiwai-sound:for-next 8/9] sound/pci/hda/hda_intel.c:1035:26: warning: variable 'bus' set but not used
by kernel test robot 04 Nov '24

04 Nov '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 149cb7d9537e241b43056fb4133f56832ac51b7a commit: cac99f73f0752e1c83674e12fb2c605dca9ce474 [8/9] ALSA: hda: intel: Don't free interrupt when suspending config: x86_64-defconfig (https://download.01.org/0day-ci/archive/20241105/202411050247.3esQz7Am-lkp@…) compiler: gcc-11 (Debian 11.3.0-12) 11.3.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241105/202411050247.3esQz7Am-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202411050247.3esQz7Am-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/pci/hda/hda_intel.c: In function 'azx_suspend': >> sound/pci/hda/hda_intel.c:1035:26: warning: variable 'bus' set but not used [-Wunused-but-set-variable] 1035 | struct hdac_bus *bus; | ^~~ vim +/bus +1035 sound/pci/hda/hda_intel.c f5dac54d9d9382 Kai-Heng Feng 2020-10-27 1030 3baffc4a84d759 Takashi Iwai 2018-12-07 1031 static int azx_suspend(struct device *dev) 3baffc4a84d759 Takashi Iwai 2018-12-07 1032 { 3baffc4a84d759 Takashi Iwai 2018-12-07 1033 struct snd_card *card = dev_get_drvdata(dev); 3baffc4a84d759 Takashi Iwai 2018-12-07 1034 struct azx *chip; 3baffc4a84d759 Takashi Iwai 2018-12-07 @1035 struct hdac_bus *bus; 3baffc4a84d759 Takashi Iwai 2018-12-07 1036 3baffc4a84d759 Takashi Iwai 2018-12-07 1037 if (!azx_is_pm_ready(card)) c5c215232dfb71 Takashi Iwai 2012-12-04 1038 return 0; c5c215232dfb71 Takashi Iwai 2012-12-04 1039 3baffc4a84d759 Takashi Iwai 2018-12-07 1040 chip = card->private_data; a41d122449bea3 Takashi Iwai 2015-04-14 1041 bus = azx_bus(chip); 472e18f63c425d Imre Deak 2021-06-23 1042 azx_shutdown_chip(chip); 785d8c4be82b45 Libin Yang 2015-05-12 1043 785d8c4be82b45 Libin Yang 2015-05-12 1044 trace_azx_suspend(chip); ^1da177e4c3f41 Linus Torvalds 2005-04-16 1045 return 0; ^1da177e4c3f41 Linus Torvalds 2005-04-16 1046 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 1047 :::::: The code at line 1035 was first introduced by commit :::::: 3baffc4a84d759ba54c461e8e4583bd8890c749a ALSA: hda/intel: Refactoring PM code :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 0/2] Add a driver for the Iron Device SMA1307 Amp
by Ki-Seok Jo 04 Nov '24

04 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> --- Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT binding ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 54 + To | 0 sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 ++++++++++++++++++++ sound/soc/codecs/sma1307.h | 444 +++++ 6 files changed, 2562 insertions(+) --- base-commit: 06df5968fd2f418b5729c43fffd65388ef0448f4 change-id: 20241007-irondevice-sma1307-ffc42b218574 Best regards, -- Kiseok Jo <kiseok.jo(a)irondevice.com>

1 0

Add support for ASRock X870E Taichi
by GitHub pull_request - opened 01 Nov '24

01 Nov '24

alsa-project/alsa-ucm-conf pull request #457 was opened from julmajustus: Included id 26ce:0a0b to realtek-alc4080 list. USB Audio S/PDIF Output, USB Audio Rear input and USB Audio Front Headphones seems to work now without issues with HiFi 2.0 channels. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/457 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/457.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Direct MIDI event delivery to ALSA sequencer's hardware port without any connections/subscriptions fails with -ENODEV.
by giedrius＠blokas.io 31 Oct '24

31 Oct '24

Hello, I had written an `aseqsend` utility to send MIDI events from hex strings to ALSA sequencer ports, similar to the -S arg of `amidi`, but I just noticed that while sending events directly to application ALSA Sequencer ports works just fine, trying to send to a hardware MIDI port fails with -19 (-ENODEV). Here's the aseqsend utility code, the highlighted line is the one returning -ENODEV: https://github.com/BlokasLabs/aseqsend/blob/main/aseqsend.c#L173 Here's some sample commands showcasing the problem (I've added a printf with drain result just after the indicated line above locally): https://paste.debian.net/hidden/9690d660/ It shows that in case there's no active subscriptions to the port, the event delivery fails, with no data ever reaching the hardware device. As long as there's at least one subscription to the port, it works ok. I've reproduced this on 5.15.36-rt41-v7+ and 6.6.58-v8-16k (running on Raspberry Pi, based on https://github.com/raspberrypi/linux) I've traced the -ENODEV to be returned at this location: https://elixir.bootlin.com/linux/v6.6.58/source/sound/core/seq/seq_midi.c#L… Whenever there's a subscription, the 'substream' is not NULL. But what I would naturally expect is for the events to get delivered to the hardware port when using direct addressing, regardless of its subscription status, the same way how it's possible to send events to application aseq ports: -- Terminal 1 -- patch@patchbox:~ $ aseqdump Waiting for data at port 128:0. Press Ctrl+C to end. Source Event Ch Data 129:0 Note on 0, note 64, velocity 48 -- Terminal 2 -- patch@patchbox:~/work/aseqsend $ ./aseqsend 128:0 90 40 30 drain: 0 Is this a bug in ALSA Sequencer, or is my expectation of being able to write directly to a hardware ALSA sequencer port incorrect?

3 3

[PATCH v29 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 31 Oct '24

31 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v29: - Fixed some phrases/wording within the SOC USB documentation, and also added an output with aplay -l for the example output. - Fixed allocated string buffer for creating the USB SND offload mixer, and added a PCM index check to ensure that the pcm index is less than the expected number. - Added a complement enable jack call if USB backend DAI link drivers need access to it. Changes in v28: - Updated comments and commit log in the stop endpoint sync patch. Clarified that the default stop endpoint completion routine won't fully run as expected since it has a completion associated w/ the command. - Added a null check for sb->xhci within xhci_sideband_create_interrupter(). This is to just ensure that caller has registered sideband before calling create interrupter. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (3): xhci: support setting interrupt moderation IMOD for secondary interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 491 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 51 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 147 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 369 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1974 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 +++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5957 insertions(+), 57 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[GIT PULL] ASoC fixes for v6.12-rc5
by Mark Brown 30 Oct '24

30 Oct '24

The following changes since commit db7e59e6a39a4d3d54ca8197c796557e6d480b0d: ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc (2024-10-22 21:00:38 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc5 for you to fetch changes up to 2db63e92186d7201ee1cb2f5af11757c5e5a1020: wcd937x codec fixes (2024-10-29 19:18:48 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 The biggest set of changes here is Hans' fixes and quirks for various Baytrail based platforms with RT5640 CODECs, and there's one core fix for a missed length assignment for __counted_by() checking. Otherwise it's small device specific fixes, several of them in the DT bindings. ---------------------------------------------------------------- Aleksei Vetrov (1): ASoC: dapm: fix bounds checker error in dapm_widget_list_create Alexey Klimov (2): ASoC: codecs: wcd937x: add missing LO Switch control ASoC: codecs: wcd937x: relax the AUX PDM watchdog Christophe JAILLET (1): ASoC: cs42l51: Fix some error handling paths in cs42l51_probe() Dmitry Yashin (1): ASoC: dt-bindings: rockchip,rk3308-codec: add port property Hans de Goede (5): ASoC: codecs: rt5640: Always disable IRQs from rt5640_cancel_work() ASoC: Intel: bytcr_rt5640: Add support for non ACPI instantiated codec ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet ASoC: Intel: sst: Support LPE0F28 ACPI HID ASoC: Intel: sst: Fix used of uninitialized ctx to log an error Mark Brown (1): wcd937x codec fixes .../bindings/sound/rockchip,rk3308-codec.yaml | 4 ++ sound/hda/intel-dsp-config.c | 4 ++ sound/soc/codecs/cs42l51.c | 7 ++- sound/soc/codecs/rt5640.c | 27 +++++---- sound/soc/codecs/wcd937x.c | 12 +++- sound/soc/codecs/wcd937x.h | 4 ++ sound/soc/intel/atom/sst/sst_acpi.c | 64 +++++++++++++++++++--- sound/soc/intel/boards/bytcr_rt5640.c | 48 +++++++++++++++- sound/soc/soc-dapm.c | 2 + 9 files changed, 144 insertions(+), 28 deletions(-)

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Why Choose a Dedicated Server for Your Business?
by David Menk 29 Oct '24

29 Oct '24

Hello everyone, I wanted to share some insights on why dedicated server hosting can be a game-changer for businesses, especially those with high-traffic websites or specific performance needs. What is a Dedicated Server? A dedicated server is a physical server that's exclusively used by one client. Unlike shared hosting, where resources are split between multiple users, a dedicated server gives you complete control over CPU, RAM, storage, and bandwidth, making it ideal for websites or applications that need consistent performance, security, and reliability. Benefits of Using a Dedicated Server: Performance and Speed: With dedicated resources, your website or application can handle high volumes of traffic without slowing down. This is especially important for e-commerce sites or any business where speed impacts the user experience. Enhanced Security: Dedicated servers offer stronger security, as you're not sharing the server with other users. You can implement custom security measures tailored to your business, helping to protect against potential breaches and data loss. Full Control and Customization: Since the server is dedicated to you, you have complete control over the operating system, software, and applications you want to run. This level of customization can be essential for companies with specific software requirements. Scalability: As your business grows, dedicated servers can be upgraded with additional resources to meet your increasing demands, providing flexibility without the need to migrate to another server. Better Reliability: Dedicated servers are less prone to outages or downtime compared to shared hosting, providing a stable environment that’s crucial for businesses that rely on 24/7 availability. Is a Dedicated Server Right for You? If your website or application has high traffic, complex applications, or sensitive data, a dedicated server could be the right choice. It's a bit more of an investment, but the control, performance, and reliability often pay off in the long run. If anyone has experience with dedicated servers or is considering making the switch, I’d love to hear your thoughts and any advice you might have! Looking forward to the discussion!

1 1

[PATCH] ALSA: usb-audio: Add Pioneer DJ/AlphaTheta DJM-A9 Mixer
by Sarah Grant 29 Oct '24

29 Oct '24

This also includes a change to the capture level as they differ with this device, as well as additional values for prefader channels. Acked-by: Olivia Mackintosh <livvy(a)base.nu> Signed-off-by: Sarah Grant <s(a)srd.tw> --- sound/usb/mixer_quirks.c | 68 +++++++++++++++++++++++++++++++++++++--- sound/usb/quirks-table.h | 57 +++++++++++++++++++++++++++++++++ 2 files changed, 121 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 2a9594f34dac..15c300eddf53 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3498,7 +3498,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) } /* - * Pioneer DJ DJM Mixers + * Pioneer DJ / AlphaTheta DJM Mixers * * These devices generally have options for soft-switching the playback and * capture sources in addition to the recording level. Although different @@ -3515,17 +3515,23 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_CAP_CDLINE 0x01 #define SND_DJM_CAP_DIGITAL 0x02 #define SND_DJM_CAP_PHONO 0x03 +#define SND_DJM_CAP_PREFADER 0x05 #define SND_DJM_CAP_PFADER 0x06 #define SND_DJM_CAP_XFADERA 0x07 #define SND_DJM_CAP_XFADERB 0x08 #define SND_DJM_CAP_MIC 0x09 #define SND_DJM_CAP_AUX 0x0d #define SND_DJM_CAP_RECOUT 0x0a +#define SND_DJM_CAP_RECOUT_NOMIC 0x0e #define SND_DJM_CAP_NONE 0x0f #define SND_DJM_CAP_CH1PFADER 0x11 #define SND_DJM_CAP_CH2PFADER 0x12 #define SND_DJM_CAP_CH3PFADER 0x13 #define SND_DJM_CAP_CH4PFADER 0x14 +#define SND_DJM_CAP_CH1PREFADER 0x31 +#define SND_DJM_CAP_CH2PREFADER 0x32 +#define SND_DJM_CAP_CH3PREFADER 0x33 +#define SND_DJM_CAP_CH4PREFADER 0x34 // Playback types #define SND_DJM_PB_CH1 0x00 @@ -3551,6 +3557,7 @@ static int snd_rme_digiface_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_900NXS2_IDX 0x3 #define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_450_IDX 0x5 +#define SND_DJM_A9_IDX 0x6 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -3579,7 +3586,7 @@ struct snd_djm_ctl { u16 wIndex; }; -static const char *snd_djm_get_label_caplevel(u16 wvalue) +static const char *snd_djm_get_label_caplevel_common(u16 wvalue) { switch (wvalue) { case 0x0000: return "-19dB"; @@ -3590,6 +3597,20 @@ static const char *snd_djm_get_label_caplevel(u16 wvalue) } }; +// The DJM-A9 has different capture levels than other, older models +static const char *snd_djm_get_label_caplevel_a9(u16 wvalue) +{ + switch (wvalue) { + case 0x0000: return "+15dB"; + case 0x0100: return "+12dB"; + case 0x0200: return "+9dB"; + case 0x0300: return "+6dB"; + case 0x0400: return "+3dB"; + case 0x0500: return "0dB"; + default: return NULL; + } +}; + static const char *snd_djm_get_label_cap_common(u16 wvalue) { switch (wvalue & 0x00ff) { @@ -3602,8 +3623,13 @@ static const char *snd_djm_get_label_cap_common(u16 wvalue) case SND_DJM_CAP_XFADERB: return "Cross Fader B"; case SND_DJM_CAP_MIC: return "Mic"; case SND_DJM_CAP_RECOUT: return "Rec Out"; + case SND_DJM_CAP_RECOUT_NOMIC: return "Rec Out without Mic"; case SND_DJM_CAP_AUX: return "Aux"; case SND_DJM_CAP_NONE: return "None"; + case SND_DJM_CAP_CH1PREFADER: return "Pre Fader Ch1"; + case SND_DJM_CAP_CH2PREFADER: return "Pre Fader Ch2"; + case SND_DJM_CAP_CH3PREFADER: return "Pre Fader Ch3"; + case SND_DJM_CAP_CH4PREFADER: return "Pre Fader Ch4"; case SND_DJM_CAP_CH1PFADER: return "Post Fader Ch1"; case SND_DJM_CAP_CH2PFADER: return "Post Fader Ch2"; case SND_DJM_CAP_CH3PFADER: return "Post Fader Ch3"; @@ -3623,6 +3649,14 @@ static const char *snd_djm_get_label_cap_850(u16 wvalue) } }; +static const char *snd_djm_get_label_caplevel(u8 device_idx, u16 wvalue) +{ + switch (device_idx) { + case SND_DJM_A9_IDX: return snd_djm_get_label_caplevel_a9(wvalue); + default: return snd_djm_get_label_caplevel_common(wvalue); + } +}; + static const char *snd_djm_get_label_cap(u8 device_idx, u16 wvalue) { switch (device_idx) { @@ -3644,7 +3678,7 @@ static const char *snd_djm_get_label_pb(u16 wvalue) static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) { switch (windex) { - case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(wvalue); + case SND_DJM_WINDEX_CAPLVL: return snd_djm_get_label_caplevel(device_idx, wvalue); case SND_DJM_WINDEX_CAP: return snd_djm_get_label_cap(device_idx, wvalue); case SND_DJM_WINDEX_PB: return snd_djm_get_label_pb(wvalue); default: return NULL; @@ -3653,7 +3687,7 @@ static const char *snd_djm_get_label(u8 device_idx, u16 wvalue, u16 windex) // common DJM capture level option values static const u16 snd_djm_opts_cap_level[] = { - 0x0000, 0x0100, 0x0200, 0x0300 }; + 0x0000, 0x0100, 0x0200, 0x0300, 0x400, 0x500 }; // DJM-250MK2 @@ -3795,6 +3829,28 @@ static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { }; +// DJM-A9 +static const u16 snd_djm_opts_a9_cap1[] = { + 0x0107, 0x0108, 0x0109, 0x010a, 0x010e, + 0x111, 0x112, 0x113, 0x114, 0x0131, 0x132, 0x133, 0x134 }; +static const u16 snd_djm_opts_a9_cap2[] = { + 0x0201, 0x0202, 0x0203, 0x0205, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a, 0x020e }; +static const u16 snd_djm_opts_a9_cap3[] = { + 0x0301, 0x0302, 0x0303, 0x0305, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a, 0x030e }; +static const u16 snd_djm_opts_a9_cap4[] = { + 0x0401, 0x0402, 0x0403, 0x0405, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a, 0x040e }; +static const u16 snd_djm_opts_a9_cap5[] = { + 0x0501, 0x0502, 0x0503, 0x0505, 0x0506, 0x0507, 0x0508, 0x0509, 0x050a, 0x050e }; + +static const struct snd_djm_ctl snd_djm_ctls_a9[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Master Input", a9_cap1, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Input", a9_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", a9_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", a9_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", a9_cap5, 2, SND_DJM_WINDEX_CAP) +}; + static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_250MK2_IDX] = SND_DJM_DEVICE(250mk2), [SND_DJM_750_IDX] = SND_DJM_DEVICE(750), @@ -3802,6 +3858,7 @@ static const struct snd_djm_device snd_djm_devices[] = { [SND_DJM_900NXS2_IDX] = SND_DJM_DEVICE(900nxs2), [SND_DJM_750MK2_IDX] = SND_DJM_DEVICE(750mk2), [SND_DJM_450_IDX] = SND_DJM_DEVICE(450), + [SND_DJM_A9_IDX] = SND_DJM_DEVICE(a9), }; @@ -4076,6 +4133,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ err = snd_djm_controls_create(mixer, SND_DJM_900NXS2_IDX); break; + case USB_ID(0x2b73, 0x003c): /* Pioneer DJ / AlphaTheta DJM-A9 */ + err = snd_djm_controls_create(mixer, SND_DJM_A9_IDX); + break; } return err; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 24c981c9b240..b758655b948c 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3119,6 +3119,63 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +{ + /* + * Pioneer DJ / AlphaTheta DJM-A9 + * 10 channels playback & 12 channels capture @ 44.1/48/96kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x003c), + QUIRK_DRIVER_INFO { + QUIRK_DATA_COMPOSITE { + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + { + QUIRK_DATA_AUDIOFORMAT(0) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100| + SNDRV_PCM_RATE_48000| + SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 3, + .rate_table = (unsigned int[]) { + 44100, 48000, 96000 + } + } + }, + QUIRK_COMPOSITE_END + } + } +}, + /* * MacroSilicon MS2100/MS2106 based AV capture cards * -- 2.40.1

2 1

[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 29 Oct '24

29 Oct '24

Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

2 1

[PATCH v5 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 29 Oct '24

29 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

1 6

[PATCH] ASoC: codecs: ES8326: Modify the configuration of and micbias
by Zhang Yi 28 Oct '24

28 Oct '24

Because we designed a new version of ES8326, the configuration of micbias needed to be modified.We tested the new driver, on both the new version and the old one. It works well. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index d5362b3be484..aa3e364827c8 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -614,6 +614,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x0F); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x00); + } } } else { if (!es8326->calibrated) { @@ -640,6 +644,10 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) ES8326_MUTE_MASK, ~(ES8326_MUTE)); } else { msleep(300); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); + } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, 0x0F, 0x00); } @@ -821,7 +829,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); - if ((es8326->jack_remove_retry == 1) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 1) && (es8326->version < ES8326_VERSION_B)) { if (iface & ES8326_HPINSERT_FLAG) es8326->jack_remove_retry = 2; else @@ -859,7 +867,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) /* * Inverted HPJACK_POL bit to trigger one IRQ to double check HP Removal event */ - if ((es8326->jack_remove_retry == 0) && (es8326->version != ES8326_VERSION_B)) { + if ((es8326->jack_remove_retry == 0) && (es8326->version < ES8326_VERSION_B)) { es8326->jack_remove_retry = 1; dev_dbg(comp->dev, "remove event check, invert HPJACK_POL, cnt = %d\n", es8326->jack_remove_retry); @@ -954,7 +962,7 @@ static int es8326_calibrate(struct snd_soc_component *component) regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); es8326->version = reg; - if ((es8326->version == ES8326_VERSION_B) && (es8326->calibrated == false)) { + if ((es8326->version >= ES8326_VERSION_B) && (es8326->calibrated == false)) { dev_dbg(component->dev, "ES8326_VERSION_B, calibrating\n"); regmap_write(es8326->regmap, ES8326_CLK_INV, 0xc0); regmap_write(es8326->regmap, ES8326_CLK_DIV1, 0x03); @@ -1047,7 +1055,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15); regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x80 | - ((es8326->version == ES8326_VERSION_B) ? + ((es8326->version >= ES8326_VERSION_B) ? (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol) : (ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol | 0x04))); usleep_range(5000, 10000); @@ -1073,6 +1081,10 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44); regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x03); + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + } msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

1 0

[PATCH] ALSA: hda/realtek: Add subwoofer quirk for Infinix ZERO BOOK 13
by Piyush Raj Chouhan 28 Oct '24

28 Oct '24

From: pc1598 <piyushchouhan1598(a)gmail.com> Infinix ZERO BOOK 13 has a 2+2 speaker system which isn't probed correctly. This patch adds a quirk with the proper pin connections. Also The mic in this laptop suffers too high gain resulting in mostly fan noise being recorded, This patch Also limit mic boost. HW Probe for device; https://linux-hardware.org/?probe=a2e892c47b Test: All 4 speaker works, Mic has low noise. Signed-off-by: Piyush Raj Chouhan <piyushchouhan1598(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3567b14b52b7..075935707f8a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7551,6 +7551,7 @@ enum { ALC290_FIXUP_SUBWOOFER_HSJACK, ALC269_FIXUP_THINKPAD_ACPI, ALC269_FIXUP_DMIC_THINKPAD_ACPI, + ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13, ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO, ALC255_FIXUP_ACER_MIC_NO_PRESENCE, ALC255_FIXUP_ASUS_MIC_NO_PRESENCE, @@ -7996,6 +7997,16 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_pincfg_U7x7_headset_mic, }, + [ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170151 }, /* use as internal speaker (LFE) */ + { 0x1b, 0x90170152 }, /* use as internal speaker (back) */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, [ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10989,6 +11000,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.46.2

1 0

[tiwai-sound:topic/compress-accel 1/1] sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used
by kernel test robot 25 Oct '24

25 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/compress-accel head: 04177158cf98a79744937893b100020d77e6f9ac commit: 04177158cf98a79744937893b100020d77e6f9ac [1/1] ALSA: compress_offload: introduce accel operation mode config: arm-randconfig-001-20241026 (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) compiler: arm-linux-gnueabi-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241026/202410260257.PNGCF9yq-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410260257.PNGCF9yq-lkp@intel.com/ All warnings (new ones prefixed by >>): sound/core/compress_offload.c: In function 'snd_compr_poll': >> sound/core/compress_offload.c:409:35: warning: variable 'runtime' set but not used [-Wunused-but-set-variable] 409 | struct snd_compr_runtime *runtime; | ^~~~~~~ vim +/runtime +409 sound/core/compress_offload.c 404 405 static __poll_t snd_compr_poll(struct file *f, poll_table *wait) 406 { 407 struct snd_compr_file *data = f->private_data; 408 struct snd_compr_stream *stream; > 409 struct snd_compr_runtime *runtime; 410 size_t avail; 411 __poll_t retval = 0; 412 413 if (snd_BUG_ON(!data)) 414 return EPOLLERR; 415 416 stream = &data->stream; 417 runtime = stream->runtime; 418 419 guard(mutex)(&stream->device->lock); 420 421 switch (stream->runtime->state) { 422 case SNDRV_PCM_STATE_OPEN: 423 case SNDRV_PCM_STATE_XRUN: 424 return snd_compr_get_poll(stream) | EPOLLERR; 425 default: 426 break; 427 } 428 429 poll_wait(f, &stream->runtime->sleep, wait); 430 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Help to solve pop when sound loops
by Charles Mulder 25 Oct '24

25 Oct '24

Hi. I hope this is the correct place to ask for help relating to ALSA's C lib? I'm working on a wavetable project for college. Please see attached. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on buffer > period > frames, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C -- Sent with Tuta; enjoy secure & ad-free emails: https://tuta.com

1 0

Help to solve pop when sound loops
by GitHub issues - edited 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was edited from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

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Help to address pop at end of buffer
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #417 was opened from charlesmulder: Hi. Apologies for asking for help here. I don't know where else to go. I'm working on a wavetable project for college. I've managed to create a wavetable containing a sine wave. The pitch can be altered via a frequency variable (f). There is an audible popping sound when the sound loops. I have noticed that increasing the buffer size, postpones the pop. I've been reading up on `buffer > period > frames`, but I don't know how to go about removing the pop. Would really appreciate some guidance. Thanks C ``` #include <stdio.h> #include <alsa/asoundlib.h> #include <math.h> unsigned char sinuc( float ); #define TWOPI 2*M_PI #define ALSA_INFO 1 static char *device = "default"; /* playback device */ unsigned char buffer[24*1024]; int main(void) { // ALSA playback related int err; snd_pcm_t *playback_handle; // pcm snd_pcm_hw_params_t *hw_params; unsigned int val, val2; int dir; snd_pcm_uframes_t period_size; snd_pcm_format_t format; // Wavetable related unsigned int N = 1024; float f = 440; // A4 note //float f = 261.626; // C4 note unsigned int fs = 44100; // playback if ((err = snd_pcm_open(&playback_handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } // ALSA configuration /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&hw_params); /* Fill it in with default values. */ snd_pcm_hw_params_any(playback_handle, hw_params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Unsigned 8-bit little-endian format */ snd_pcm_hw_params_set_format(playback_handle, hw_params, SND_PCM_FORMAT_U8); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(playback_handle, hw_params, 1); /* 44100 bits/second sampling rate (CD quality) */ snd_pcm_hw_params_set_rate_near(playback_handle, hw_params, &fs, &dir); /* Set period size to 32 frames. */ period_size = 32; snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params, &period_size, &dir); /* Write the parameters to the driver */ err = snd_pcm_hw_params(playback_handle, hw_params); if (err < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(err)); exit(1); } // Wavetable init unsigned char wavetable[N]; // wavetable buffer float angle_inc = TWOPI/(float)N; // sine wave angle increment float index_inc = N*f/(float)fs; // wavetable index increment // Populate wavetable with a sine wave for( int n = 0; n < N; n++ ) { wavetable[n] = sinuc( angle_inc * n ); // 0 - 255 range } // ALSA Sample Buffer // period = 940 frames // buffer = 15052 frames float n = 0; for (int i = 0; i < sizeof(buffer)/sizeof(char); i++) { buffer[i] = wavetable[(int)n]; //printf("%d\n",buffer[i]); n = n+index_inc; if( (int)n >= N ) { n = 0; } } if ((err = snd_pcm_prepare (playback_handle)) < 0) { fprintf (stderr, "cannot prepare audio interface for use (%s)\n", snd_strerror (err)); exit (1); } //for (int i = 0; i < 16; i++) { while(1) { period_size = snd_pcm_writei(playback_handle, buffer, sizeof(buffer)); if (period_size < 0) period_size = snd_pcm_recover(playback_handle, period_size, 0); if (period_size < 0) { printf("snd_pcm_writei failed: %s\n", snd_strerror(period_size)); break; } if (period_size > 0 && period_size < (long)sizeof(buffer)) printf("Short write (expected %li, wrote %li)\n", (long)sizeof(buffer), period_size); } // pass the remaining samples, otherwise they're dropped in close err = snd_pcm_drain(playback_handle); if (err < 0) printf("snd_pcm_drain failed: %s\n", snd_strerror(err)); //snd_pcm_hw_params_free(hw_params); snd_pcm_close(playback_handle); return 0; } /** * Sine unsigned char. * Scales sine output to a char * Original range -1 to 1. * New range 0 - 255. */ unsigned char sinuc( float angle ) { return (sinf( angle ) * 255 + 255) / 2; } ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/417 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALC225 the microphone does not work in jack3.5
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #416 was opened from PavelGu: I have notebook clevo NL55AU with Ubuntu24.04 installed. The microphone does not work on a headset connected to a combined audio output jack 3.5. Logs: https://alsa-project.org/db/?f=89d2b631cb1635dd48357610d54e062bddef81b4 Can you help me? Issue URL : https://github.com/alsa-project/alsa-lib/issues/416 Repository URL: https://github.com/alsa-project/alsa-lib

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[Discussion] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by GitHub issues - opened 25 Oct '24

25 Oct '24

alsa-project/alsa-lib issue #415 was opened from xeroxslayer: I raised this discussion a few months ago on the ALSA mailing list, but there was no response regarding this. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). Now, this profile will be autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software, I wouldn't know, I haven't actually tried adding a mono profile manually), but it seemed odd that one has to go through such trouble just to listen to mono sound... something that should be available out of the box IMO. Issue URL : https://github.com/alsa-project/alsa-lib/issues/415 Repository URL: https://github.com/alsa-project/alsa-lib

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Limit internal Mic boost on Dell platform
by Kailang 25 Oct '24

25 Oct '24

Hi Takashi, Attach patch will limit internal Mic boost on all Dell platform. BR, Kailang

2 1

Update ucm conf for MediaTek Genio platforms
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/alsa-ucm-conf pull request #456 was opened from zoran055: 1. Add dynamic configuration to detect HDMI/DP devices for mt8370-evk, mt8390-evk and mt8395-evk. 2. Add headset jack detection support for mt8370-evk, mt8390-evk and mt8395-evk. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/456.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.12-rc4
by Mark Brown 24 Oct '24

24 Oct '24

The following changes since commit 8cf0b93919e13d1e8d4466eb4080a4c4d9d66d7b: Linux 6.12-rc2 (2024-10-06 15:32:27 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc4 for you to fetch changes up to db7e59e6a39a4d3d54ca8197c796557e6d480b0d: ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc (2024-10-22 21:00:38 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.12 An uncomfortably large set of fixes due to me not getting round to sending them for longer than I should due to travel and illness. This is mostly smaller driver specific changes, but there are a couple of generic changes: - Bumping the minimal topology ABI we check for during validation, the code had support for v4 removed previously but the update of the define used for initial validation was missed. - Fix the assumption that DAPM structs will be embedded in a component which isn't true for card widgets when doing name comparisons, though fortunately this is rarely triggered. We've pulled in one Soundwire fix which was part of a larger series fixing cleanup issues in on Intel Soundwire systems. ---------------------------------------------------------------- Alexey Klimov (2): ASoC: qcom: sdm845: add missing soundwire runtime stream alloc ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs Amadeusz Sławiński (2): ASoC: Intel: avs: Update stream status in a separate thread ASoC: topology: Bump minimal topology ABI version Andrei Simion (1): MAINTAINERS: Update maintainer list for MICROCHIP ASOC, SSC and MCP16502 drivers Benjamin Bara (1): ASoC: dapm: avoid container_of() to get component Binbin Zhou (1): ASoC: loongson: Fix component check failed on FDT systems Chancel Liu (1): ASoC: fsl_micfil: Add a flag to distinguish with different volume control types Christian Heusel (1): ASoC: amd: yc: Add quirk for ASUS Vivobook S15 M3502RA Colin Ian King (1): ASoC: max98388: Fix missing increment of variable slot_found Derek Fang (1): ASoC: Intel: soc-acpi: lnl: Add match entry for TM2 laptops Ilya Dudikov (1): ASoC: amd: yc: Fix non-functional mic on ASUS E1404FA Jack Yu (1): ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue Kai Vehmanen (1): ASoC: SOF: Intel: hda-loader: do not wait for HDaudio IOC Kirill Marinushkin (1): ASoC: Change my e-mail to gmail Krzysztof Kozlowski (2): ASoC: qcom: Select missing common Soundwire module code on SDM845 ASoC: qcom: sc7280: Fix missing Soundwire runtime stream alloc Lad Prabhakar (1): ASoC: rsnd: Fix probe failure on HiHope boards due to endpoint parsing Mark Brown (1): ASoC/SoundWire: clean up link DMA during stop for IPC4 Miquel Raynal (1): ASoC: dt-bindings: davinci-mcasp: Fix interrupt properties Ranjani Sridharan (4): ASoC: SOF: ipc4-topology: Do not set ALH node_id for aggregated DAIs ASoC: SOF: Intel: hda: Handle prepare without close for non-HDA DAI's soundwire: intel_ace2x: Send PDI stream number during prepare ASoC: SOF: Intel: hda: Always clean up link DMA during stop Shengjiu Wang (2): ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler ASoC: fsl_micfil: Add sample rate constraint Venkata Prasad Potturu (2): ASoC: SOF: amd: Add error log for DSP firmware validation failure ASoC: SOF: amd: Fix for ACP SRAM addr for acp7.0 platform Zhu Jun (1): ASoC: codecs: Fix error handling in aw_dev_get_dsp_status function Zichen Xie (1): ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe() .../bindings/sound/davinci-mcasp-audio.yaml | 18 ++--- MAINTAINERS | 5 +- drivers/soundwire/intel_ace2x.c | 19 ++--- include/uapi/sound/asoc.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 ++++ sound/soc/codecs/aw88399.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 15 ++-- sound/soc/codecs/max98388.c | 1 + sound/soc/codecs/pcm3060-i2c.c | 4 +- sound/soc/codecs/pcm3060-spi.c | 4 +- sound/soc/codecs/pcm3060.c | 4 +- sound/soc/codecs/pcm3060.h | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/fsl/fsl_esai.c | 4 +- sound/soc/fsl/fsl_micfil.c | 81 +++++++++++++++++++++- sound/soc/intel/avs/core.c | 3 +- sound/soc/intel/avs/pcm.c | 19 +++++ sound/soc/intel/avs/pcm.h | 16 +++++ sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 38 ++++++++++ sound/soc/loongson/loongson_card.c | 1 + sound/soc/qcom/Kconfig | 2 + sound/soc/qcom/lpass-cpu.c | 2 + sound/soc/qcom/sc7280.c | 10 ++- sound/soc/qcom/sdm845.c | 7 +- sound/soc/sh/rcar/core.c | 7 +- sound/soc/soc-dapm.c | 4 +- sound/soc/sof/amd/acp-loader.c | 5 +- sound/soc/sof/amd/acp.c | 4 +- sound/soc/sof/intel/hda-dai-ops.c | 23 +++--- sound/soc/sof/intel/hda-dai.c | 37 ++++++++-- sound/soc/sof/intel/hda-loader.c | 17 ----- sound/soc/sof/ipc4-topology.c | 15 +++- 32 files changed, 298 insertions(+), 89 deletions(-) create mode 100644 sound/soc/intel/avs/pcm.h

2 1

[PATCH v4 0/7] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 24 Oct '24

24 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… Add it to this patch set. Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Jaroslav Kysela (1): ALSA: compress_offload: introduce accel operation mode Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device .../sound/designs/compress-accel.rst | 136 ++++ include/sound/compress_driver.h | 46 ++ include/uapi/sound/compress_offload.h | 63 +- include/uapi/sound/compress_params.h | 23 +- sound/core/Kconfig | 3 + sound/core/compress_offload.c | 351 ++++++++- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 ++ sound/soc/fsl/fsl_asrc_m2m.c | 727 ++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 ++++++- sound/soc/fsl/fsl_easrc.h | 4 + 14 files changed, 1851 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/designs/compress-accel.rst create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

4 10

Demo application for accel operation.
by GitHub pull_request - opened 24 Oct '24

24 Oct '24

alsa-project/tinycompress pull request #26 was opened from TE-N-ShengjiuWang: This 'caccel' applicaton is a demo to show how to call the new introduced accel interface. The kernel patch is: ALSA: compress_offload: introduce accel operation mode As there is conflict when using the API of alsa-lib, add alsa_wrap file to separate the header files: alsa/asoundlib.h sound/asound.h command for conversion to 48kHz: caccel -i in.wav -o out.wav -r 48000 Request URL : https://github.com/alsa-project/tinycompress/pull/26 Patch URL : https://github.com/alsa-project/tinycompress/pull/26.patch Repository URL: https://github.com/alsa-project/tinycompress

1 0

[PATCH v5 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark I got kernel-test report that v4 had compile warning. This is v5 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v4 -> v5 - fixup sumsung odroid compile warning - rebased on latest asoc/for-6.13 v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o73mqj5b.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 11 +-- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 584 deletions(-) -- 2.43.0

2 14

[PATCH v4 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 23 Oct '24

23 Oct '24

Hi Mark 2 weeks past. This is v4 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v3 -> v4 - adjust latest amd - rebased on asoc/for-6.13 v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87y138vbv7.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

2 14

Update default depop procedure
by Kailang 23 Oct '24

23 Oct '24

Hi Takashi, Attach patch will solve issue which Headphone has a chance no sound. Many Thanks. BR, Kailang

2 1

ALC4080 onboard audio support on the MSI X870 Tomahawk motherboard
by GitHub issues - opened 23 Oct '24

23 Oct '24

alsa-project/alsa-ucm-conf issue #455 was opened from nfp0: @perexg As mentioned [here](https://github.com/alsa-project/alsa-ucm-conf/issues/447#issuecomment…, it seems the ALC4080 audio on the new MSI X870 Tomahawk motherboard is not supported. USB ID is `0db0:cd0e` I have tested adding `cd0e` on the `USB-Audio.conf` file like so: ``` Regex "USB((0414:a0(0e|1[0124]))|(0b05:(19(84|9[69])|1a(16|2[07]|5[23c]|97|f1)))|(0db0:(cd0e|005a|124b|151f|1feb|3130|36e7|4(19c|22d|240|88c)|62a4|6c[0c]9|70d3|7696|82c7|8af7|961e|a(073|228|47c|74b)|b202|d1d7|d6e7))|(26ce:0a0[68]))" ``` With this, the microphone now seems to be correctly mapped and works. But the problem is that it's volume is way too low. It is set to 100% in both KDE and alsamixer (it's the Line one). ![image](https://github.com/user-attachments/assets/034e8f45-9e17-4338-9c1b-fda0b56dae29) How can I help solve this problem? Should I open a separate issue? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/455 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [RFC PATCH 00/10] Add support for DMA and audio codec of F1C100s
by Mesih Kilinc 23 Oct '24

23 Oct '24

This is RFC patchset for Allwinner suniv F1C100s to support DMA and audio codec. Allwinner F1C100s has a audio codec that has necessary digital and analog parts. It has r-l headphone output and microphone, line, r-l FM inputs. ADC can capture any inputs and also output channels via mux. Any input channels or DAC samples can feed output channels. Add support for this audio codec. F1C100s utilizes DMA channels to send and receive ADC-DAC samples. So DMA support needed. Patch 1~5 adds support for DMA. Suniv F1C100s has very similar DMA to sun4i. But there is some dissimilarities also. Suniv features a DMA reset bit in clock control unit. It has smaller number of DMA channels. Several registers has different addresses. It's max burst size is 4 instead of 8. Also DMA endpoint numbers are different. Patch 6 adds DMA max burst option to sun4i-codec. Patch 7~8 Add support for suniv F1C100s audio codec. Patch 9 adds audio codec to suniv-f1c100s.dtsi Patch 10 adds audio codec support to Lichee Pi Nano board. Thanks! Mesih Kilinc (10): dma-engine: sun4i: Add a quirk to support different chips dma-engine: sun4i: Add has_reset option to quirk dt-bindings: dmaengine: Add Allwinner suniv F1C100s DMA dma-engine: sun4i: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for DMA ASoC: sun4i-codec: Add DMA Max Burst field dt-bindigs: sound: Add Allwinner suniv F1C100s Audio Codec ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s ARM: dts: suniv: f1c100s: Add support for Audio Codec ARM: dts: suniv: f1c100s: Activate Audio Codec for Lichee Pi Nano .../devicetree/bindings/dma/sun4i-dma.txt | 4 +- .../devicetree/bindings/sound/sun4i-codec.txt | 5 + arch/arm/boot/dts/suniv-f1c100s-licheepi-nano.dts | 8 + arch/arm/boot/dts/suniv-f1c100s.dtsi | 25 ++ drivers/dma/Kconfig | 4 +- drivers/dma/sun4i-dma.c | 221 ++++++++++-- sound/soc/sunxi/sun4i-codec.c | 371 ++++++++++++++++++++- 7 files changed, 601 insertions(+), 37 deletions(-) -- 2.7.4

5 17

[PATCH 0/2] ASoC: fsl_xcvr: enable some interrupts
by Shengjiu Wang 23 Oct '24

23 Oct '24

Enable interrupt of cmdc status update and the interrupts for wrong preamble received. Shengjiu Wang (2): ASoC: fsl_xcvr: enable interrupt of cmdc status update ASoC: fsl_xcvr: reset RX dpath after wrong preamble sound/soc/fsl/fsl_xcvr.c | 94 ++++++++++++++++++++++++++++++++++------ sound/soc/fsl/fsl_xcvr.h | 5 +++ 2 files changed, 86 insertions(+), 13 deletions(-) -- 2.34.1

2 3

ALSA problem , audio doesn't work
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubuntu releases
by GitHub issues - edited 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was edited from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

AlSA problem , audio on/off across multiple ubutnu
by GitHub issues - opened 22 Oct '24

22 Oct '24

alsa-project/alsa-lib issue #414 was opened from Kermilli: Hi, I've been having problems with audio on ubuntu on and off. I'm not a professional but everything seems to point at alsa . The pulse troubleshooting page at some point recommends to run alsa directly/bypass pulse with the command `mplayer -ao alsa:device=hw=0 yourchosentrack.mp3` which gives `[AO_ALSA] alsa-lib: pcm_hw.c:1715:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-2): No such file or directory [AO_ALSA] Playback open error: No such file or directory Failed to initialize audio driver 'alsa:device=hw=0' Could not open/initialize audio device -> no sound. Audio: no sound Video: no video ` sudo pulseaudio -vvv gives `E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="1" name="pci-0000_00_1b.0" card_name="alsa_card.pci-0000_00_1b.0" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed.` I've tried everything . The only time it worked was when I added options snd-hda-intel model=auto to /etc/modprobe.d/alsa-base.conf. , but even then it stopped working after some time when I was listening to audio. Any help would be greatly appreciated! Issue URL : https://github.com/alsa-project/alsa-lib/issues/414 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: rt722-sdca: increase clk_stop_timeout to fix clock stop issue
by Jack Yu 22 Oct '24

22 Oct '24

clk_stop_timeout should be increased to 900ms to fix clock stop issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index 87354bb1564e..d5c985ff5ac5 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -253,7 +253,7 @@ static int rt722_sdca_read_prop(struct sdw_slave *slave) } /* set the timeout values */ - prop->clk_stop_timeout = 200; + prop->clk_stop_timeout = 900; /* wake-up event */ prop->wake_capable = 1; -- 2.34.1

2 1

ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #413 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. kernel: linux-6.8.1-gentoo the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 On the new system the 'old' kernel linux-6.8.1-gentoo is also available. Rolled back alsa-lib from alsa-lib-1.2.12 to alsa-lib-1.2.10 and booted kernel kernel linux-6.8.1-gentoo Still same problem. Issue URL : https://github.com/alsa-project/alsa-lib/issues/413 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: firewire-lib: Avoid division by zero in apply_constraint_to_size()
by Andrey Shumilin 21 Oct '24

21 Oct '24

The step variable is initialized to zero. It is changed in the loop, but if it's not changed it will remain zero. Add a variable check before the division. The observed behavior was introduced by commit 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size"), and it is difficult to show that any of the interval parameters will satisfy the snd_interval_test() condition with data from the amdtp_rate_table[] table. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 826b5de90c0b ("ALSA: firewire-lib: fix insufficient PCM rule for period/buffer size") Signed-off-by: Andrey Shumilin <shum.sdl(a)nppct.ru> --- sound/firewire/amdtp-stream.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4e2f2bb7879f..6c45ee3545f9 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -163,6 +163,9 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, step = max(step, amdtp_syt_intervals[i]); } + if (step == 0) + return -EINVAL; + t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; -- 2.30.2

1 0

snd_pcm_hw_open open '/dev/snd/pcmC0D0p' failed (-16)
by GitHub issues - edited 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #412 was edited from SVENsuzhou: Hi, I am using pjsip to make a sip call function. But I got the following error in alsa-lib. ``` Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 pjsua_call.c !Answering call 0: code=200 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 inv0x7f989c003fc8 ..SDP negotiation done: Success Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c ...Call 0: updating media.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c .....Media stream call00:0 is destroyed Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 ....UDP media transport started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_aud.c ....Audio channel update.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 .....UDP media transport attached Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Encoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Decoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_media.c ....Audio updated, stream #0: PCMA (sendrecv) Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ...Conf connect: 1 --> 0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ....Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@44100/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: start media transmit failed: pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@48000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@32000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@8000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c .....Unable to open sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c ....Error opening sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 media.cpp ...pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.321 pjsua_core.c ....TX 966 bytes Response msg 200/INVITE/cseq=30946 (tdta0x5559ce2e7378) to UDP 10.57.0.200:5060: Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: SIP/2.0 200 OK Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Via: SIP/2.0/UDP 10.57.0.200:5060;rport=5060;received=10.57.0.200;branch=z9hG4bKPje5a12777-db72-4db0-a5a5-a72743589a27 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Call-ID: 0bc91889-099c-42de-beb2-4fedefac6e7f Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: answer incoming call, remote uri: <sip:passenger_pei@10.57.0.200>, state: CONNECTING Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: From: <sip:passenger_pei@10.57.0.200>;tag=9efc50b5-c57a-4aa7-823e-2c3df064a27c ``` ``` root@tos-hmi-02:~# dmesg | grep snd [ 16.251073] snd_hda_intel 0000:00:0e.0: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 16.261081] snd_hda_intel 0000:00:0e.0: bound 0000:00:02.0 (ops 0xffffffff8df312a0) [ 16.351558] snd_hda_codec_idt hdaudioC0D0: autoconfig for 92HD89D3: line_outs=3 (0xd/0xf/0x10/0x0/0x0) type:line [ 16.362460] snd_hda_codec_idt hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 16.362470] snd_hda_codec_idt hdaudioC0D0: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) [ 16.362476] snd_hda_codec_idt hdaudioC0D0: mono: mono_out=0x0 [ 16.383430] snd_hda_codec_idt hdaudioC0D0: dig-out=0x22/0x23 [ 16.383447] snd_hda_codec_idt hdaudioC0D0: inputs: [ 16.383452] snd_hda_codec_idt hdaudioC0D0: Internal Mic=0x14 [ 16.383455] snd_hda_codec_idt hdaudioC0D0: Rear Mic=0xe [ 16.383459] snd_hda_codec_idt hdaudioC0D0: Front Mic=0xb [ 16.383461] snd_hda_codec_idt hdaudioC0D0: Internal Mic=0x13 [ 16.383464] snd_hda_codec_idt hdaudioC0D0: Line=0xc [ 16.383466] snd_hda_codec_idt hdaudioC0D0: CD=0x12 [ 16.383469] snd_hda_codec_idt hdaudioC0D0: dig-in=0x24 ``` I try to use `lsof /dev/snd/*` to check which process hold the file handle. And I got nothing. Can you give me some leads on troubleshooting? Issue URL : https://github.com/alsa-project/alsa-lib/issues/412 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_hw_open open '/dev/snd/pcmC0D0p' failed (-16)
by GitHub issues - opened 21 Oct '24

21 Oct '24

alsa-project/alsa-lib issue #412 was opened from SVENsuzhou: Hi, I am using pjsip to make a sip call function. But I got the following error in alsa-lib. ``` Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 pjsua_call.c !Answering call 0: code=200 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.315 inv0x7f989c003fc8 ..SDP negotiation done: Success Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c ...Call 0: updating media.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_media.c .....Media stream call00:0 is destroyed Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 ....UDP media transport started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 pjsua_aud.c ....Audio channel update.. Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.316 udp0x7f989c010710 .....UDP media transport attached Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Encoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 strm0x5559ce2eb538 .....Decoder stream started Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_media.c ....Audio updated, stream #0: PCMA (sendrecv) Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ...Conf connect: 1 --> 0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c ....Set sound device: capture=-1, playback=-2, mode=0, use_default_settings=0 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.317 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@44100/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: start media transmit failed: pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.318 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@48000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@32000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.319 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@8000/1/20ms Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 alsa_dev.c ......ALSA lib /usr/src/debug/alsa-lib/1.2.11/src/pcm/pcm_hw.c:1785:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-16): Device o Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c .....Unable to open sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 pjsua_aud.c ....Error opening sound device: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) [status=420002] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.320 media.cpp ...pjsua_conf_connect(id, sink.id) error: Unknown error from audio driver (PJMEDIA_EAUD_SYSERR) (status=420002) [/usr/src/debug/pjproject/2.14.1/pjsip/src/pjsua2/media.cpp:218] Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: 08:14:36.321 pjsua_core.c ....TX 966 bytes Response msg 200/INVITE/cseq=30946 (tdta0x5559ce2e7378) to UDP 10.57.0.200:5060: Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: SIP/2.0 200 OK Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Via: SIP/2.0/UDP 10.57.0.200:5060;rport=5060;received=10.57.0.200;branch=z9hG4bKPje5a12777-db72-4db0-a5a5-a72743589a27 Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: Call-ID: 0bc91889-099c-42de-beb2-4fedefac6e7f Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: answer incoming call, remote uri: <sip:passenger_pei@10.57.0.200>, state: CONNECTING Oct 21 08:14:36 tos-hmi-02 ads_uartboardd[1404]: From: <sip:passenger_pei@10.57.0.200>;tag=9efc50b5-c57a-4aa7-823e-2c3df064a27c ``` I try to use `lsof /dev/snd/*` to check which process hold the file handle. And I got nothing. Can you give me some leads on troubleshooting? Issue URL : https://github.com/alsa-project/alsa-lib/issues/412 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v6] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 20 Oct '24

20 Oct '24

Due to internal re-org, Kevin has no longer maintained audio driver. Due to job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v6: - fixed spelling mistakes v5: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410021557.FByBO9Dp-lkp@intel.com/ v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 87198c5e589e..d1484b5658e3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22910,25 +22910,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22938,9 +22955,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

[PATCH v2] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs
by Alexey Klimov 19 Oct '24

19 Oct '24

Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add rxn_reg_stride2 on top. While at this, also remove specific if-check for INTERP_AUX from rx_macro_digital_mute() and rx_macro_enable_interp_clk(). These if-check was used to handle such special offset for AUX interpolator but since CDC_RX_RXn_RX_PATH_SEC7(rx, n) and CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macros will generate the correst addresses of dsm register, they are no longer needed. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> --- Changes in v2: - updated macros as suggested by Mark and Dmitry; - removed INTERP_AUX if-check in two functions; Link to previous patch: https://lore.kernel.org/lkml/20240925043823.520218-1-alexey.klimov@linaro.o… sound/soc/codecs/lpass-rx-macro.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ef7a70fa6966..febbbe073962 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + (rx->rxn_reg_stride * n) + ((n > 1) ? rx->rxn_reg_stride2 : 0)) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -1929,9 +1932,6 @@ static int rx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) CDC_RX_PATH_PGA_MUTE_MASK, 0x0); } - if (j == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - int_mux_cfg0 = CDC_RX_INP_MUX_RX_INT0_CFG0 + j * 8; int_mux_cfg1 = int_mux_cfg0 + 4; int_mux_cfg0_val = snd_soc_component_read(component, int_mux_cfg0); @@ -2702,9 +2702,6 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, main_reg = CDC_RX_RXn_RX_PATH_CTL(rx, interp_idx); dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, interp_idx); - if (interp_idx == INTERP_AUX) - dsm_reg = CDC_RX_RXn_RX_PATH_DSM_CTL(rx, 2); - rx_cfg2_reg = CDC_RX_RXn_RX_PATH_CFG2(rx, interp_idx); if (SND_SOC_DAPM_EVENT_ON(event)) { @@ -3821,6 +3818,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0xc; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3832,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) -- 2.45.2

1 0

[PATCH] ASoC: fsl_micfil: Add a flag to distinguish with different volume control types
by Chancel Liu 19 Oct '24

19 Oct '24

On i.MX8MM the register of volume control has positive and negative values. It is different from other platforms like i.MX8MP and i.MX93 which only have positive values. Add a volume_sx flag to use SX_TLV volume control for this kind of platform. Use common TLV volume control for other platforms. Fixes: cdfa92eb90f5 ("ASoC: fsl_micfil: Correct the number of steps on SX controls") Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_micfil.c | 43 +++++++++++++++++++++++++++++++++++++- 1 file changed, 42 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 4161b674745b..0679d762f08a 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -78,6 +78,7 @@ struct fsl_micfil_soc_data { bool imx; bool use_edma; bool use_verid; + bool volume_sx; u64 formats; }; @@ -87,6 +88,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mm = { .fifo_depth = 8, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S16_LE, + .volume_sx = true, }; static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { @@ -95,6 +97,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx8mp = { .fifo_depth = 32, .dataline = 0xf, .formats = SNDRV_PCM_FMTBIT_S32_LE, + .volume_sx = false, }; static struct fsl_micfil_soc_data fsl_micfil_imx93 = { @@ -105,6 +108,7 @@ static struct fsl_micfil_soc_data fsl_micfil_imx93 = { .formats = SNDRV_PCM_FMTBIT_S32_LE, .use_edma = true, .use_verid = true, + .volume_sx = false, }; static const struct of_device_id fsl_micfil_dt_ids[] = { @@ -328,7 +332,26 @@ static int hwvad_detected(struct snd_kcontrol *kcontrol, return 0; } -static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { +static const struct snd_kcontrol_new fsl_micfil_volume_controls[] = { + SOC_SINGLE_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(0), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(1), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH2 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(2), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH3 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(3), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH4 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(4), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH5 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(5), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH6 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(6), 0xF, 0, gain_tlv), + SOC_SINGLE_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, + MICFIL_OUTGAIN_CHX_SHIFT(7), 0xF, 0, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_volume_sx_controls[] = { SOC_SINGLE_SX_TLV("CH0 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(0), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH1 Volume", REG_MICFIL_OUT_CTRL, @@ -345,6 +368,9 @@ static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { MICFIL_OUTGAIN_CHX_SHIFT(6), 0x8, 0xF, gain_tlv), SOC_SINGLE_SX_TLV("CH7 Volume", REG_MICFIL_OUT_CTRL, MICFIL_OUTGAIN_CHX_SHIFT(7), 0x8, 0xF, gain_tlv), +}; + +static const struct snd_kcontrol_new fsl_micfil_snd_controls[] = { SOC_ENUM_EXT("MICFIL Quality Select", fsl_micfil_quality_enum, micfil_quality_get, micfil_quality_set), @@ -844,6 +870,20 @@ static int fsl_micfil_dai_probe(struct snd_soc_dai *cpu_dai) return 0; } +static int fsl_micfil_component_probe(struct snd_soc_component *component) +{ + struct fsl_micfil *micfil = snd_soc_component_get_drvdata(component); + + if (micfil->soc->volume_sx) + snd_soc_add_component_controls(component, fsl_micfil_volume_sx_controls, + ARRAY_SIZE(fsl_micfil_volume_sx_controls)); + else + snd_soc_add_component_controls(component, fsl_micfil_volume_controls, + ARRAY_SIZE(fsl_micfil_volume_controls)); + + return 0; +} + static const struct snd_soc_dai_ops fsl_micfil_dai_ops = { .probe = fsl_micfil_dai_probe, .startup = fsl_micfil_startup, @@ -865,6 +905,7 @@ static struct snd_soc_dai_driver fsl_micfil_dai = { static const struct snd_soc_component_driver fsl_micfil_component = { .name = "fsl-micfil-dai", + .probe = fsl_micfil_component_probe, .controls = fsl_micfil_snd_controls, .num_controls = ARRAY_SIZE(fsl_micfil_snd_controls), .legacy_dai_naming = 1, -- 2.46.2

3 2

[PATCH] Use card->dev in replace of the &pdev->dev argument in the dev_err function
by Liu Jing 18 Oct '24

18 Oct '24

Because card->dev = &pdev->dev is already defined in the rx51_soc_probe function, and then &pdev->dev is still used. Signed-off-by: Liu Jing <liujing(a)cmss.chinamobile.com> --- sound/soc/ti/rx51.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index 77296237575a..d9900c69e536 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -371,7 +371,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0); if (!dai_node) { - dev_err(&pdev->dev, "McBSP node is not provided\n"); + dev_err(card->dev, "McBSP node is not provided\n"); return -EINVAL; } rx51_dai[0].cpus->dai_name = NULL; @@ -381,7 +381,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 0); if (!dai_node) { - dev_err(&pdev->dev, "Codec node is not provided\n"); + dev_err(card->dev, "Codec node is not provided\n"); return -EINVAL; } rx51_dai[0].codecs->name = NULL; @@ -389,7 +389,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,audio-codec", 1); if (!dai_node) { - dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n"); + dev_err(card->dev, "Auxiliary Codec node is not provided\n"); return -EINVAL; } rx51_aux_dev[0].dlc.name = NULL; @@ -399,7 +399,7 @@ static int rx51_soc_probe(struct platform_device *pdev) dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0); if (!dai_node) { - dev_err(&pdev->dev, "Headphone amplifier node is not provided\n"); + dev_err(card->dev, "Headphone amplifier node is not provided\n"); return -EINVAL; } rx51_aux_dev[1].dlc.name = NULL; @@ -408,7 +408,7 @@ static int rx51_soc_probe(struct platform_device *pdev) rx51_codec_conf[1].dlc.of_node = dai_node; } - pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL); + pdata = devm_kzalloc(card->dev, sizeof(*pdata), GFP_KERNEL); if (pdata == NULL) return -ENOMEM; @@ -439,7 +439,7 @@ static int rx51_soc_probe(struct platform_device *pdev) err = devm_snd_soc_register_card(card->dev, card); if (err) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err); + dev_err(card->dev, "snd_soc_register_card failed (%d)\n", err); return err; } -- 2.27.0

1 0

[PATCH 5.10 0/1] soundwire: cadence: Fix error check in cdns_xfer_msg()
by Daniil Dulov 18 Oct '24

18 Oct '24

Svacer reports redundant comparison in cdns_xfer_msg(). The problem is present in 5.10 stable release and can be fixed by the following upstream patch that can be cleanly applied to 5.10 stable branch.

1 1

[PATCH alsa-ucm-conf v5 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 18 Oct '24

18 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v5: Change PlaybackMixerElem to just "Speaker" v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..176f880 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 3

Qualcomm: fix codec initialisation
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #454 was opened from jhovold: Fix the Qualcomm configurations that are using the same identifier for lazy includes of codec initialisation files which means that only the last one is actually included. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/454.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Revert "ucm2: Qualcomm: x1e80100: add USB DisplayPort playback"
by GitHub pull_request - opened 18 Oct '24

18 Oct '24

alsa-project/alsa-ucm-conf pull request #453 was opened from jhovold: This reverts commit 1dcb58e4fac989bafe0705e49a7f029727207fb3. The recently added DisplayPort support depends on kernel changes that never made it upstream and has some usability issues that remain to be resolved. Revert for now to avoid breaking existing setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/453.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument.
by GitHub issues - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-plugins issue #62 was opened from linuxandos2: Playing a flac with a samplerate higer than 96000 with audacious gives: > _ALSA error: snd_pcm_hw_params_set_format failed: Invalid argument._ Output device for audacious is hw:1.0 - USB Audio the device is a soudbaster blasterx G6. output of _cat /proc/asound/card1/stream0_ > Creative Technology Ltd Sound BlasterX G6 at usb-0000:00:14.0-4.1, high speed : > USB Audio > > Playback: > Status: Stop > Interface 1 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 1 > Implicit Feedback Mode: No > Interface 1 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x01 (1 OUT) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > Sync Endpoint: 0x81 (1 IN) > Sync EP Interface: 1 > Sync EP Altset: 2 > Implicit Feedback Mode: No > > Capture: > Status: Stop > Interface 2 > Altset 1 > Format: S24_3LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 24 > Channel map: FL FR > Interface 2 > Altset 2 > Format: S32_LE > Channels: 2 > Endpoint: 0x82 (2 IN) (ASYNC) > Rates: 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 > Data packet interval: 125 us > Bits: 32 > Channel map: FL FR > linux: gentoo kernel: 6.9.9-gentoo-x86_64 alsa-lib-1.2.12 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.12 alsa-plugins-1.2.12 alsa-utils-1.2.12 selecting > pulse - PulseAudio Sound Server in audacious results in no sound and audacious hangs. Pipewire and wireplumber are used. On an old gentoo- system on another partition all samplerates play successfully. Using the same kernel: linux-6.8.1-gentoo as in the old system on the new one does not solve the problem. the versions of alsa are: alsa-lib-1.2.10-r2 alsa-topology-conf-1.2.5.1 alsa-ucm-conf-1.2.10-r1 alsa-plugins-1.2.7.1-r1 alsa-utils-1.2.10-r1 Issue URL : https://github.com/alsa-project/alsa-plugins/issues/62 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

acppdmmach: add support for ACP 7.0
by GitHub pull_request - opened 17 Oct '24

17 Oct '24

alsa-project/alsa-ucm-conf pull request #452 was opened from superm1: Starting with ACP 7.0 there is a generic 'acppdmmach' driver that will be utilized. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/452.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH alsa-ucm-conf v4 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 17 Oct '24

17 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v4: Add "CaptureMixer" to headset config v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 3 ++- 4 files changed, 14 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..ddca67a 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,8 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixer "default:${CardId}" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 2

MUSL static build - conf.c still insists on using dynamic link libraries
by GitHub issues - opened 16 Oct '24

16 Oct '24

alsa-project/alsa-lib issue #411 was opened from pietervandermeer: alsa-lib v 1.2.12 configured for musl static build like so: ```CC=arm-linux-musleabihf-gcc ./configure --enable-shared=no --enable-static=yes --with-pic --host=armv7-unknown-linux-musleabihf --prefix=/home/vagrant/alsa-lib-1.2.12/build --libdir=/usr/lib/arm-linux-musleabihf --includedir=/usr/include/arm-linux-musleabihf/``` This builds and links fine against my executable. But when I run it, there's an issue: ``` INFO opening alsa audio playback device [hw:CARD=sndrpihifiberry,DEV=0] ALSA lib conf.c:3558:(snd_config_hooks_call) Cannot open shared library (null) ((null): Dynamic loading not supported) ALSA lib conf.c:4026:(snd_config_update_r) hooks failed, removing configuration WARN Not found playback device, error ALSA function 'snd_pcm_open' failed with error 'No such file or directory (2)'. Trying next.. ``` Note the "cannot open shared library". Well, yes, because you should be configured as 100% static, as per configure arguments ```--enable-shared=no --enable-static=yes``` Upon inspecting the code in ```conf.c```, there are various functions like ```snd_config_hooks_call()``` that just call ```snd_dlopen()``` and ```snd_dlsym()```. Right now I'm just replacing these with ```strcmp``` on the function_name followed by a static function call to the respective function. But I don't know how much work I'll need, and of course it is really hacky. It would be great if a 100% static build was supported. Issue URL : https://github.com/alsa-project/alsa-lib/issues/411 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Working on audio for qcom apq8098 / msm8998
by Marc Gonzalez 15 Oct '24

15 Oct '24

Hello everyone, I've been trying to get audio working on a qcom apq8098 (cousin of msm8998) board. Code is mostly based on work from here: https://gitlab.com/msm8998-mainline/linux This is the diff I have on top of v6.12-rc1 https://paste.debian.net/1331666/ Kernel log after init starts: [ 13.311201] random: crng init done [ 14.724797] atl1c 0000:01:00.0 enp1s0: renamed from eth0 (while UP) [ 14.895978] Detected VIPT I-cache on CPU1 [ 14.896041] GICv3: CPU1: found redistributor 1 region 0:0x0000000017b20000 [ 14.896113] CPU1: Booted secondary processor 0x0000000001 [0x51af8014] [ 14.949128] Detected VIPT I-cache on CPU2 [ 14.949198] GICv3: CPU2: found redistributor 2 region 0:0x0000000017b40000 [ 14.949272] CPU2: Booted secondary processor 0x0000000002 [0x51af8014] [ 15.025271] Detected VIPT I-cache on CPU3 [ 15.025349] GICv3: CPU3: found redistributor 3 region 0:0x0000000017b60000 [ 15.025423] CPU3: Booted secondary processor 0x0000000003 [0x51af8014] [ 15.106412] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:3 [ 15.109935] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:4 [ 15.119041] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_probe:1759 [ 15.134672] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: q6afe_dai_dev_probe [ 15.137948] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: of_q6afe_parse_dai_data [ 15.146461] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:7 [ 15.153563] remoteproc remoteproc0: powering up 4080000.remoteproc [ 15.166371] remoteproc remoteproc0: Booting fw image mba.mbn, size 234152 [ 15.169747] q6asm-dai 17300000.remoteproc:glink-edge:apr:apr-service@7:dais: Adding to iommu group 5 [ 15.170559] qcom,apr 17300000.remoteproc:glink-edge.apr_audio_svc.-1.-1: Adding APR/GPR dev: aprsvc:apr-service:4:8 [ 15.179732] q6afe-dai 17300000.remoteproc:glink-edge:apr:apr-service@4:dais: YOA: msm_dai_q6_dai_probe [ 15.199345] qcom-q6afe aprsvc:apr-service:4:4: YO: q6afe_find_port:860 [ 15.208834] msm-snd-msm8998 sound: ASoC: registered pcm #0 MultiMedia1 (*) [ 15.216194] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia1 mapping ok [ 15.221994] msm-snd-msm8998 sound: ASoC: registered pcm #1 MultiMedia2 (*) [ 15.229582] msm-snd-msm8998 sound: snd-soc-dummy-dai <-> MultiMedia2 mapping ok [ 15.236045] msm-snd-msm8998 sound: ASoC: registered pcm #2 (HDMI) [ 15.243211] msm-snd-msm8998 sound: i2s-hifi <-> HDMI mapping ok [ 15.285770] input: batfish Headset Jack as /devices/platform/sound/sound/card0/input2 [ 15.298010] qcom-q6v5-mss 4080000.remoteproc: MBA booted without debug policy, loading mpss [ 15.542937] remoteproc remoteproc0: remote processor 4080000.remoteproc is now up [ 17.727724] ath10k_snoc 18800000.wifi: qmi chip_id 0x30214 chip_family 0x4001 board_id 0xff soc_id 0x40010002 [ 17.727923] ath10k_snoc 18800000.wifi: qmi fw_version 0x100204b2 fw_build_timestamp 2019-09-04 03:01 fw_build_id QC_IMAGE_VERSION_STRING=WLAN.HL.1.0-01202-QCAHLSWMTPLZ-1.221523.2 [ 17.736928] ath10k_snoc 18800000.wifi: qmi not waiting for msa_ready indicator [ 17.903361] platform a800000.usb: deferred probe pending: dwc3: failed to initialize core [ 21.512409] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.512619] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.522450] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.599761] ath10k_snoc 18800000.wifi: wcn3990 hw1.0 target 0x00000008 chip_id 0x00000000 sub 0000:0000 [ 21.599935] ath10k_snoc 18800000.wifi: kconfig debug 1 debugfs 1 tracing 0 dfs 0 testmode 0 [ 21.608221] ath10k_snoc 18800000.wifi: firmware ver api 5 features wowlan,mgmt-tx-by-reference,non-bmi crc32 3ec1ac4b [ 21.657531] ath10k_snoc 18800000.wifi: htt-ver 3.44 wmi-op 4 htt-op 3 cal file max-sta 32 raw 0 hwcrypto 1 [ 21.713363] arm-smmu 16c0000.iommu: Unhandled context fault: fsr=0x402, iova=0x00000000, fsynr=0x1, cbfrsynra=0x1900, cb=0 [ 21.713577] arm-smmu 16c0000.iommu: FSR = 00000402 [Format=2 TF], SID=0x1900 [ 21.723404] arm-smmu 16c0000.iommu: FSYNR0 = 00000001 [S1CBNDX=0 PLVL=1] [ 21.732374] ath10k_snoc 18800000.wifi: invalid MAC address; choosing random [ 21.737662] ath: EEPROM regdomain: 0x0 [ 21.744356] ath: EEPROM indicates default country code should be used [ 21.748148] ath: doing EEPROM country->regdmn map search [ 21.754648] ath: country maps to regdmn code: 0x3a [ 21.759986] ath: Country alpha2 being used: US [ 21.764605] ath: Regpair used: 0x3a Does anyone see something obvious I might have missed? Regards

3 6

ucm-validator verify fails when including codecs/hda/hdmi.conf
by GitHub issues - opened 15 Oct '24

15 Oct '24

alsa-project/alsa-ucm-conf issue #451 was opened from ChrisLane: After increasing the verbosity of the validator in the GitHub workflow, we can see that the "Validate Syntax" step is failing on the `codecs/hda/hdmi.conf` file. Here is the relevant output: https://github.com/alsa-project/alsa-ucm-conf/actions/runs/11346809053/job/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/451 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v28 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 15 Oct '24

15 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v28: - Fixed some phrases/wording within the SOC USB documentation, and also added an output with aplay -l for the example output. - Fixed allocated string buffer for creating the USB SND offload mixer, and added a PCM index check to ensure that the pcm index is less than the expected number. - Added a complement enable jack call if USB backend DAI link drivers need access to it. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (3): xhci: support setting interrupt moderation IMOD for secondary interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 491 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 51 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 147 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 369 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1974 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 +++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5957 insertions(+), 57 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 68

ucm2: wsa884x: fix typo in mixer names
by GitHub pull_request - opened 14 Oct '24

14 Oct '24

alsa-project/alsa-ucm-conf pull request #450 was opened from joske: DTS fix for x1e80100-CRD will be submitted upstream separately Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/450 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/450.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable
by Jack Yu 14 Oct '24

14 Oct '24

If det_mode is not 0, 3 or 5 then function will return jack_type with an uninitialzed value. Enhance switch case to prevent uninitialized variable issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt-sdw-common.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c index a422da6cf702..9f51fec383f9 100644 --- a/sound/soc/codecs/rt-sdw-common.c +++ b/sound/soc/codecs/rt-sdw-common.c @@ -150,15 +150,15 @@ int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) goto io_error; switch (det_mode) { - case 0x00: - jack_type = 0; - break; case 0x03: jack_type = SND_JACK_HEADPHONE; break; case 0x05: jack_type = SND_JACK_HEADSET; break; + default: + jack_type = 0; + break; } /* write selected_mode */ -- 2.34.1

2 1

Debian 12 : no audio
by GitHub issues - opened 14 Oct '24

14 Oct '24

alsa-project/alsa-ucm-conf issue #449 was opened from ezawadzki: After fresh debian 12 installation : - Speakers audio : ok - Headphones : no audio After installed `alsa-ucm-conf` and followed : ``` systemctl stop alsa-state rm /var/lib/alsa/asound.state systemctl start alsa-state ``` Now : - Speakers audio : no audio (speakers not recognized) - Headphones : no audio ``` cat /etc/os-release PRETTY_NAME="Debian GNU/Linux 12 (bookworm)" ``` ``` uname -a Linux debian 6.10.11+bpo-amd64 #1 SMP PREEMPT_DYNAMIC Debian 6.10.11-1~bpo12+1 (2024-10-03) x86_64 GNU/Linux ``` ``` alsamixer Card and Chipset : PipeWire ``` Any help ? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/449 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[tiwai-sound:for-next 10/14] sound/pci/hda/patch_realtek.c:10711:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 504f052aa3435ab2f15af8b20bc4f4de8ff259c7 [10/14] ALSA: hda/realtek: Use codec SSID matching for Lenovo devices config: i386-buildonly-randconfig-003-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141704.uIhTGCED-lkp@…) compiler: clang version 18.1.8 (https://github.com/llvm/llvm-project 3b5b5c1ec4a3095ab096dd780e84d7ab81f3d7ff) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241014/202410141704.uIhTGCED-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410141704.uIhTGCED-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pci/hda/patch_realtek.c:10711:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10711 | HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10716:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10716 | HDA_CODEC_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10731:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10731 | HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_realtek.c:10733:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 10733 | HDA_CODEC_QUIRK(0x17aa, 0x386f, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ 4 errors generated. vim +10711 sound/pci/hda/patch_realtek.c 9995 9996 static const struct hda_quirk alc269_fixup_tbl[] = { 9997 SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), 9998 SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), 9999 SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), 10000 SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), 10001 SND_PCI_QUIRK(0x1025, 0x072d, "Acer Aspire V5-571G", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10002 SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), 10003 SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), 10004 SND_PCI_QUIRK(0x1025, 0x0762, "Acer Aspire E1-472", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), 10005 SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), 10006 SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), 10007 SND_PCI_QUIRK(0x1025, 0x080d, "Acer Aspire V5-122P", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10008 SND_PCI_QUIRK(0x1025, 0x0840, "Acer Aspire E1", ALC269VB_FIXUP_ASPIRE_E1_COEF), 10009 SND_PCI_QUIRK(0x1025, 0x100c, "Acer Aspire E5-574G", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), 10010 SND_PCI_QUIRK(0x1025, 0x101c, "Acer Veriton N2510G", ALC269_FIXUP_LIFEBOOK), 10011 SND_PCI_QUIRK(0x1025, 0x102b, "Acer Aspire C24-860", ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE), 10012 SND_PCI_QUIRK(0x1025, 0x1065, "Acer Aspire C20-820", ALC269VC_FIXUP_ACER_HEADSET_MIC), 10013 SND_PCI_QUIRK(0x1025, 0x106d, "Acer Cloudbook 14", ALC283_FIXUP_CHROME_BOOK), 10014 SND_PCI_QUIRK(0x1025, 0x1094, "Acer Aspire E5-575T", ALC255_FIXUP_ACER_LIMIT_INT_MIC_BOOST), 10015 SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10016 SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10017 SND_PCI_QUIRK(0x1025, 0x1166, "Acer Veriton N4640G", ALC269_FIXUP_LIFEBOOK), 10018 SND_PCI_QUIRK(0x1025, 0x1167, "Acer Veriton N6640G", ALC269_FIXUP_LIFEBOOK), 10019 SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), 10020 SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), 10021 SND_PCI_QUIRK(0x1025, 0x1248, "Acer Veriton N4660G", ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE), 10022 SND_PCI_QUIRK(0x1025, 0x1269, "Acer SWIFT SF314-54", ALC256_FIXUP_ACER_HEADSET_MIC), 10023 SND_PCI_QUIRK(0x1025, 0x126a, "Acer Swift SF114-32", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10024 SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10025 SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10026 SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10027 SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), 10028 SND_PCI_QUIRK(0x1025, 0x129d, "Acer SWIFT SF313-51", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10029 SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10030 SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), 10031 SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), 10032 SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), 10033 SND_PCI_QUIRK(0x1025, 0x141f, "Acer Spin SP513-54N", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10034 SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10035 SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), 10036 SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), 10037 SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), 10038 SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), 10039 SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), 10040 SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), 10041 SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), 10042 SND_PCI_QUIRK(0x1028, 0x05bd, "Dell Latitude E6440", ALC292_FIXUP_DELL_E7X), 10043 SND_PCI_QUIRK(0x1028, 0x05be, "Dell Latitude E6540", ALC292_FIXUP_DELL_E7X), 10044 SND_PCI_QUIRK(0x1028, 0x05ca, "Dell Latitude E7240", ALC292_FIXUP_DELL_E7X), 10045 SND_PCI_QUIRK(0x1028, 0x05cb, "Dell Latitude E7440", ALC292_FIXUP_DELL_E7X), 10046 SND_PCI_QUIRK(0x1028, 0x05da, "Dell Vostro 5460", ALC290_FIXUP_SUBWOOFER), 10047 SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10048 SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10049 SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 10050 SND_PCI_QUIRK(0x1028, 0x0615, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), 10051 SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_SUBWOOFER_HSJACK), 10052 SND_PCI_QUIRK(0x1028, 0x062c, "Dell Latitude E5550", ALC292_FIXUP_DELL_E7X), 10053 SND_PCI_QUIRK(0x1028, 0x062e, "Dell Latitude E7450", ALC292_FIXUP_DELL_E7X), 10054 SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS_HSJACK), 10055 SND_PCI_QUIRK(0x1028, 0x064a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10056 SND_PCI_QUIRK(0x1028, 0x064b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10057 SND_PCI_QUIRK(0x1028, 0x0665, "Dell XPS 13", ALC288_FIXUP_DELL_XPS_13), 10058 SND_PCI_QUIRK(0x1028, 0x0669, "Dell Optiplex 9020m", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), 10059 SND_PCI_QUIRK(0x1028, 0x069a, "Dell Vostro 5480", ALC290_FIXUP_SUBWOOFER_HSJACK), 10060 SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), 10061 SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10062 SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10063 SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10064 SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10065 SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10066 SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10067 SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), 10068 SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), 10069 SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), 10070 SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), 10071 SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), 10072 SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), 10073 SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), 10074 SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), 10075 SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), 10076 SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10077 SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10078 SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), 10079 SND_PCI_QUIRK(0x1028, 0x0872, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), 10080 SND_PCI_QUIRK(0x1028, 0x0873, "Dell Precision 3930", ALC255_FIXUP_DUMMY_LINEOUT_VERB), 10081 SND_PCI_QUIRK(0x1028, 0x08ad, "Dell WYSE AIO", ALC225_FIXUP_DELL_WYSE_AIO_MIC_NO_PRESENCE), 10082 SND_PCI_QUIRK(0x1028, 0x08ae, "Dell WYSE NB", ALC225_FIXUP_DELL1_MIC_NO_PRESENCE), 10083 SND_PCI_QUIRK(0x1028, 0x0935, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), 10084 SND_PCI_QUIRK(0x1028, 0x097d, "Dell Precision", ALC289_FIXUP_DUAL_SPK), 10085 SND_PCI_QUIRK(0x1028, 0x097e, "Dell Precision", ALC289_FIXUP_DUAL_SPK), 10086 SND_PCI_QUIRK(0x1028, 0x098d, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), 10087 SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), 10088 SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), 10089 SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), 10090 SND_PCI_QUIRK(0x1028, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET), 10091 SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), 10092 SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), 10093 SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), 10094 SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10095 SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10096 SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), 10097 SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), 10098 SND_PCI_QUIRK(0x1028, 0x0b27, "Dell", ALC245_FIXUP_CS35L41_SPI_2), 10099 SND_PCI_QUIRK(0x1028, 0x0b28, "Dell", ALC245_FIXUP_CS35L41_SPI_2), 10100 SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10101 SND_PCI_QUIRK(0x1028, 0x0b71, "Dell Inspiron 16 Plus 7620", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10102 SND_PCI_QUIRK(0x1028, 0x0beb, "Dell XPS 15 9530 (2023)", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10103 SND_PCI_QUIRK(0x1028, 0x0c03, "Dell Precision 5340", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), 10104 SND_PCI_QUIRK(0x1028, 0x0c0b, "Dell Oasis 14 RPL-P", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10105 SND_PCI_QUIRK(0x1028, 0x0c0d, "Dell Oasis", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10106 SND_PCI_QUIRK(0x1028, 0x0c0e, "Dell Oasis 16", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10107 SND_PCI_QUIRK(0x1028, 0x0c19, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), 10108 SND_PCI_QUIRK(0x1028, 0x0c1a, "Dell Precision 3340", ALC236_FIXUP_DELL_DUAL_CODECS), 10109 SND_PCI_QUIRK(0x1028, 0x0c1b, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), 10110 SND_PCI_QUIRK(0x1028, 0x0c1c, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), 10111 SND_PCI_QUIRK(0x1028, 0x0c1d, "Dell Precision 3440", ALC236_FIXUP_DELL_DUAL_CODECS), 10112 SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), 10113 SND_PCI_QUIRK(0x1028, 0x0c28, "Dell Inspiron 16 Plus 7630", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), 10114 SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), 10115 SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10116 SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10117 SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10118 SND_PCI_QUIRK(0x1028, 0x0cc0, "Dell Oasis 13", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10119 SND_PCI_QUIRK(0x1028, 0x0cc1, "Dell Oasis 14 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10120 SND_PCI_QUIRK(0x1028, 0x0cc2, "Dell Oasis 14 2-in-1 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10121 SND_PCI_QUIRK(0x1028, 0x0cc3, "Dell Oasis 14 Low Weight MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10122 SND_PCI_QUIRK(0x1028, 0x0cc4, "Dell Oasis 16 MTL-H/U", ALC289_FIXUP_DELL_CS35L41_SPI_2), 10123 SND_PCI_QUIRK(0x1028, 0x0cc5, "Dell Oasis 14", ALC289_FIXUP_RTK_AMP_DUAL_SPK), 10124 SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10125 SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), 10126 SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), 10127 SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), 10128 SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), 10129 SND_PCI_QUIRK(0x103c, 0x21f9, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10130 SND_PCI_QUIRK(0x103c, 0x2210, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10131 SND_PCI_QUIRK(0x103c, 0x2214, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10132 SND_PCI_QUIRK(0x103c, 0x221b, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10133 SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC), 10134 SND_PCI_QUIRK(0x103c, 0x2221, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10135 SND_PCI_QUIRK(0x103c, 0x2225, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10136 SND_PCI_QUIRK(0x103c, 0x2236, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10137 SND_PCI_QUIRK(0x103c, 0x2237, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10138 SND_PCI_QUIRK(0x103c, 0x2238, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10139 SND_PCI_QUIRK(0x103c, 0x2239, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10140 SND_PCI_QUIRK(0x103c, 0x224b, "HP", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10141 SND_PCI_QUIRK(0x103c, 0x2253, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10142 SND_PCI_QUIRK(0x103c, 0x2254, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10143 SND_PCI_QUIRK(0x103c, 0x2255, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10144 SND_PCI_QUIRK(0x103c, 0x2256, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10145 SND_PCI_QUIRK(0x103c, 0x2257, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10146 SND_PCI_QUIRK(0x103c, 0x2259, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10147 SND_PCI_QUIRK(0x103c, 0x225a, "HP", ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED), 10148 SND_PCI_QUIRK(0x103c, 0x225f, "HP", ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY), 10149 SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10150 SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10151 SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10152 SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10153 SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10154 SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10155 SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10156 SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10157 SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), 10158 SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10159 SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), 10160 SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10161 SND_PCI_QUIRK(0x103c, 0x2273, "HP", ALC280_FIXUP_HP_DOCK_PINS), 10162 SND_PCI_QUIRK(0x103c, 0x2278, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10163 SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10164 SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10165 SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10166 SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10167 SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10168 SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10169 SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10170 SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10171 SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10172 SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10173 SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10174 SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10175 SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10176 SND_PCI_QUIRK(0x103c, 0x22db, "HP", ALC280_FIXUP_HP_9480M), 10177 SND_PCI_QUIRK(0x103c, 0x22dc, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10178 SND_PCI_QUIRK(0x103c, 0x22fb, "HP", ALC269_FIXUP_HP_GPIO_MIC1_LED), 10179 SND_PCI_QUIRK(0x103c, 0x2334, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10180 SND_PCI_QUIRK(0x103c, 0x2335, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10181 SND_PCI_QUIRK(0x103c, 0x2336, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10182 SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), 10183 SND_PCI_QUIRK(0x103c, 0x2b5e, "HP 288 Pro G2 MT", ALC221_FIXUP_HP_288PRO_MIC_NO_PRESENCE), 10184 SND_PCI_QUIRK(0x103c, 0x802e, "HP Z240 SFF", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10185 SND_PCI_QUIRK(0x103c, 0x802f, "HP Z240", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10186 SND_PCI_QUIRK(0x103c, 0x8077, "HP", ALC256_FIXUP_HP_HEADSET_MIC), 10187 SND_PCI_QUIRK(0x103c, 0x8158, "HP", ALC256_FIXUP_HP_HEADSET_MIC), 10188 SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC295_FIXUP_HP_X360), 10189 SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), 10190 SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), 10191 SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10192 SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10193 SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), 10194 SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10195 SND_PCI_QUIRK(0x103c, 0x841c, "HP Pavilion 15-CK0xx", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10196 SND_PCI_QUIRK(0x103c, 0x8497, "HP Envy x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10197 SND_PCI_QUIRK(0x103c, 0x84a6, "HP 250 G7 Notebook PC", ALC269_FIXUP_HP_LINE1_MIC1_LED), 10198 SND_PCI_QUIRK(0x103c, 0x84ae, "HP 15-db0403ng", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10199 SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), 10200 SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), 10201 SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), 10202 SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10203 SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360), 10204 SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10205 SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10206 SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), 10207 SND_PCI_QUIRK(0x103c, 0x86c1, "HP Laptop 15-da3001TU", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10208 SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), 10209 SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10210 SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10211 SND_PCI_QUIRK(0x103c, 0x86f9, "HP Spectre x360 13-aw0xxx", ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED), 10212 SND_PCI_QUIRK(0x103c, 0x8716, "HP Elite Dragonfly G2 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10213 SND_PCI_QUIRK(0x103c, 0x8720, "HP EliteBook x360 1040 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10214 SND_PCI_QUIRK(0x103c, 0x8724, "HP EliteBook 850 G7", ALC285_FIXUP_HP_GPIO_LED), 10215 SND_PCI_QUIRK(0x103c, 0x8728, "HP EliteBook 840 G7", ALC285_FIXUP_HP_GPIO_LED), 10216 SND_PCI_QUIRK(0x103c, 0x8729, "HP", ALC285_FIXUP_HP_GPIO_LED), 10217 SND_PCI_QUIRK(0x103c, 0x8730, "HP ProBook 445 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10218 SND_PCI_QUIRK(0x103c, 0x8735, "HP ProBook 435 G7", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10219 SND_PCI_QUIRK(0x103c, 0x8736, "HP", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10220 SND_PCI_QUIRK(0x103c, 0x8760, "HP", ALC285_FIXUP_HP_MUTE_LED), 10221 SND_PCI_QUIRK(0x103c, 0x876e, "HP ENVY x360 Convertible 13-ay0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10222 SND_PCI_QUIRK(0x103c, 0x877a, "HP", ALC285_FIXUP_HP_MUTE_LED), 10223 SND_PCI_QUIRK(0x103c, 0x877d, "HP", ALC236_FIXUP_HP_MUTE_LED), 10224 SND_PCI_QUIRK(0x103c, 0x8780, "HP ZBook Fury 17 G7 Mobile Workstation", 10225 ALC285_FIXUP_HP_GPIO_AMP_INIT), 10226 SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", 10227 ALC285_FIXUP_HP_GPIO_AMP_INIT), 10228 SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10229 SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10230 SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), 10231 SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10232 SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), 10233 SND_PCI_QUIRK(0x103c, 0x87d3, "HP Laptop 15-gw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10234 SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10235 SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10236 SND_PCI_QUIRK(0x103c, 0x87f1, "HP ProBook 630 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10237 SND_PCI_QUIRK(0x103c, 0x87f2, "HP ProBook 640 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10238 SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), 10239 SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), 10240 SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), 10241 SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), 10242 SND_PCI_QUIRK(0x103c, 0x87fd, "HP Laptop 14-dq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10243 SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10244 SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10245 SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10246 SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10247 SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), 10248 SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10249 SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10250 SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10251 SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10252 SND_PCI_QUIRK(0x103c, 0x884c, "HP EliteBook 840 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), 10253 SND_PCI_QUIRK(0x103c, 0x8862, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10254 SND_PCI_QUIRK(0x103c, 0x8863, "HP ProBook 445 G8 Notebook PC", ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10255 SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10256 SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10257 SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), 10258 SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10259 SND_PCI_QUIRK(0x103c, 0x888a, "HP ENVY x360 Convertible 15-eu0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10260 SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), 10261 SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), 10262 SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), 10263 SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10264 SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), 10265 SND_PCI_QUIRK(0x103c, 0x88dd, "HP Pavilion 15z-ec200", ALC285_FIXUP_HP_MUTE_LED), 10266 SND_PCI_QUIRK(0x103c, 0x8902, "HP OMEN 16", ALC285_FIXUP_HP_MUTE_LED), 10267 SND_PCI_QUIRK(0x103c, 0x890e, "HP 255 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10268 SND_PCI_QUIRK(0x103c, 0x8919, "HP Pavilion Aero Laptop 13-be0xxx", ALC287_FIXUP_HP_GPIO_LED), 10269 SND_PCI_QUIRK(0x103c, 0x896d, "HP ZBook Firefly 16 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10270 SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10271 SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10272 SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10273 SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10274 SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10275 SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10276 SND_PCI_QUIRK(0x103c, 0x897d, "HP mt440 Mobile Thin Client U74", ALC236_FIXUP_HP_GPIO_LED), 10277 SND_PCI_QUIRK(0x103c, 0x8981, "HP Elite Dragonfly G3", ALC245_FIXUP_CS35L41_SPI_4), 10278 SND_PCI_QUIRK(0x103c, 0x898e, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), 10279 SND_PCI_QUIRK(0x103c, 0x898f, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), 10280 SND_PCI_QUIRK(0x103c, 0x8991, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10281 SND_PCI_QUIRK(0x103c, 0x8992, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2), 10282 SND_PCI_QUIRK(0x103c, 0x8994, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10283 SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), 10284 SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), 10285 SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), 10286 SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), 10287 SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), 10288 SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), 10289 SND_PCI_QUIRK(0x103c, 0x89c0, "HP ZBook Power 15.6 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10290 SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10291 SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10292 SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10293 SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10294 SND_PCI_QUIRK(0x103c, 0x89e7, "HP Elite x2 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10295 SND_PCI_QUIRK(0x103c, 0x8a0f, "HP Pavilion 14-ec1xxx", ALC287_FIXUP_HP_GPIO_LED), 10296 SND_PCI_QUIRK(0x103c, 0x8a20, "HP Laptop 15s-fq5xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10297 SND_PCI_QUIRK(0x103c, 0x8a25, "HP Victus 16-d1xxx (MB 8A25)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), 10298 SND_PCI_QUIRK(0x103c, 0x8a28, "HP Envy 13", ALC287_FIXUP_CS35L41_I2C_2), 10299 SND_PCI_QUIRK(0x103c, 0x8a29, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10300 SND_PCI_QUIRK(0x103c, 0x8a2a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10301 SND_PCI_QUIRK(0x103c, 0x8a2b, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10302 SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10303 SND_PCI_QUIRK(0x103c, 0x8a2d, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10304 SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10305 SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10306 SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10307 SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4), 10308 SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), 10309 SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), 10310 SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), 10311 SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), 10312 SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), 10313 SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), 10314 SND_PCI_QUIRK(0x103c, 0x8ab9, "HP EliteBook 840 G8 (MB 8AB8)", ALC285_FIXUP_HP_GPIO_LED), 10315 SND_PCI_QUIRK(0x103c, 0x8abb, "HP ZBook Firefly 14 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10316 SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10317 SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10318 SND_PCI_QUIRK(0x103c, 0x8ad8, "HP 800 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10319 SND_PCI_QUIRK(0x103c, 0x8b0f, "HP Elite mt645 G7 Mobile Thin Client U81", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10320 SND_PCI_QUIRK(0x103c, 0x8b2f, "HP 255 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), 10321 SND_PCI_QUIRK(0x103c, 0x8b3a, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10322 SND_PCI_QUIRK(0x103c, 0x8b3f, "HP mt440 Mobile Thin Client U91", ALC236_FIXUP_HP_GPIO_LED), 10323 SND_PCI_QUIRK(0x103c, 0x8b42, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10324 SND_PCI_QUIRK(0x103c, 0x8b43, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10325 SND_PCI_QUIRK(0x103c, 0x8b44, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10326 SND_PCI_QUIRK(0x103c, 0x8b45, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10327 SND_PCI_QUIRK(0x103c, 0x8b46, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10328 SND_PCI_QUIRK(0x103c, 0x8b47, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10329 SND_PCI_QUIRK(0x103c, 0x8b59, "HP Elite mt645 G7 Mobile Thin Client U89", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10330 SND_PCI_QUIRK(0x103c, 0x8b5d, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10331 SND_PCI_QUIRK(0x103c, 0x8b5e, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10332 SND_PCI_QUIRK(0x103c, 0x8b63, "HP Elite Dragonfly 13.5 inch G4", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10333 SND_PCI_QUIRK(0x103c, 0x8b65, "HP ProBook 455 15.6 inch G10 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10334 SND_PCI_QUIRK(0x103c, 0x8b66, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10335 SND_PCI_QUIRK(0x103c, 0x8b70, "HP EliteBook 835 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10336 SND_PCI_QUIRK(0x103c, 0x8b72, "HP EliteBook 845 G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10337 SND_PCI_QUIRK(0x103c, 0x8b74, "HP EliteBook 845W G10", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10338 SND_PCI_QUIRK(0x103c, 0x8b77, "HP ElieBook 865 G10", ALC287_FIXUP_CS35L41_I2C_2), 10339 SND_PCI_QUIRK(0x103c, 0x8b7a, "HP", ALC236_FIXUP_HP_GPIO_LED), 10340 SND_PCI_QUIRK(0x103c, 0x8b7d, "HP", ALC236_FIXUP_HP_GPIO_LED), 10341 SND_PCI_QUIRK(0x103c, 0x8b87, "HP", ALC236_FIXUP_HP_GPIO_LED), 10342 SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), 10343 SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), 10344 SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), 10345 SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10346 SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10347 SND_PCI_QUIRK(0x103c, 0x8b96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10348 SND_PCI_QUIRK(0x103c, 0x8b97, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10349 SND_PCI_QUIRK(0x103c, 0x8bb3, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), 10350 SND_PCI_QUIRK(0x103c, 0x8bb4, "HP Slim OMEN", ALC287_FIXUP_CS35L41_I2C_2), 10351 SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10352 SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10353 SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10354 SND_PCI_QUIRK(0x103c, 0x8be0, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10355 SND_PCI_QUIRK(0x103c, 0x8be1, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10356 SND_PCI_QUIRK(0x103c, 0x8be2, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10357 SND_PCI_QUIRK(0x103c, 0x8be3, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10358 SND_PCI_QUIRK(0x103c, 0x8be5, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10359 SND_PCI_QUIRK(0x103c, 0x8be6, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10360 SND_PCI_QUIRK(0x103c, 0x8be7, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10361 SND_PCI_QUIRK(0x103c, 0x8be8, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10362 SND_PCI_QUIRK(0x103c, 0x8be9, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10363 SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), 10364 SND_PCI_QUIRK(0x103c, 0x8c15, "HP Spectre x360 2-in-1 Laptop 14-eu0xxx", ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX), 10365 SND_PCI_QUIRK(0x103c, 0x8c16, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), 10366 SND_PCI_QUIRK(0x103c, 0x8c17, "HP Spectre 16", ALC287_FIXUP_CS35L41_I2C_2), 10367 SND_PCI_QUIRK(0x103c, 0x8c21, "HP Pavilion Plus Laptop 14-ey0XXX", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10368 SND_PCI_QUIRK(0x103c, 0x8c30, "HP Victus 15-fb1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), 10369 SND_PCI_QUIRK(0x103c, 0x8c46, "HP EliteBook 830 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10370 SND_PCI_QUIRK(0x103c, 0x8c47, "HP EliteBook 840 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10371 SND_PCI_QUIRK(0x103c, 0x8c48, "HP EliteBook 860 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10372 SND_PCI_QUIRK(0x103c, 0x8c49, "HP Elite x360 830 2-in-1 G11", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10373 SND_PCI_QUIRK(0x103c, 0x8c4d, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), 10374 SND_PCI_QUIRK(0x103c, 0x8c4e, "HP Omen", ALC287_FIXUP_CS35L41_I2C_2), 10375 SND_PCI_QUIRK(0x103c, 0x8c4f, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), 10376 SND_PCI_QUIRK(0x103c, 0x8c50, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10377 SND_PCI_QUIRK(0x103c, 0x8c51, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10378 SND_PCI_QUIRK(0x103c, 0x8c52, "HP EliteBook 1040 G11", ALC285_FIXUP_HP_GPIO_LED), 10379 SND_PCI_QUIRK(0x103c, 0x8c53, "HP Elite x360 1040 2-in-1 G11", ALC285_FIXUP_HP_GPIO_LED), 10380 SND_PCI_QUIRK(0x103c, 0x8c66, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10381 SND_PCI_QUIRK(0x103c, 0x8c67, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10382 SND_PCI_QUIRK(0x103c, 0x8c68, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), 10383 SND_PCI_QUIRK(0x103c, 0x8c6a, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2), 10384 SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10385 SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10386 SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10387 SND_PCI_QUIRK(0x103c, 0x8c7b, "HP ProBook 445 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10388 SND_PCI_QUIRK(0x103c, 0x8c7c, "HP ProBook 445 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10389 SND_PCI_QUIRK(0x103c, 0x8c7d, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10390 SND_PCI_QUIRK(0x103c, 0x8c7e, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10391 SND_PCI_QUIRK(0x103c, 0x8c7f, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10392 SND_PCI_QUIRK(0x103c, 0x8c80, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10393 SND_PCI_QUIRK(0x103c, 0x8c81, "HP EliteBook 665 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10394 SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), 10395 SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED), 10396 SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), 10397 SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED), 10398 SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED), 10399 SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED), 10400 SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED), 10401 SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10402 SND_PCI_QUIRK(0x103c, 0x8c97, "HP ZBook", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10403 SND_PCI_QUIRK(0x103c, 0x8ca1, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), 10404 SND_PCI_QUIRK(0x103c, 0x8ca2, "HP ZBook Power", ALC236_FIXUP_HP_GPIO_LED), 10405 SND_PCI_QUIRK(0x103c, 0x8ca4, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10406 SND_PCI_QUIRK(0x103c, 0x8ca7, "HP ZBook Fury", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 10407 SND_PCI_QUIRK(0x103c, 0x8caf, "HP Elite mt645 G8 Mobile Thin Client", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), 10408 SND_PCI_QUIRK(0x103c, 0x8cbd, "HP Pavilion Aero Laptop 13-bg0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), 10409 SND_PCI_QUIRK(0x103c, 0x8cdd, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), 10410 SND_PCI_QUIRK(0x103c, 0x8cde, "HP Spectre", ALC287_FIXUP_CS35L41_I2C_2), 10411 SND_PCI_QUIRK(0x103c, 0x8cdf, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10412 SND_PCI_QUIRK(0x103c, 0x8ce0, "HP SnowWhite", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), 10413 SND_PCI_QUIRK(0x103c, 0x8cf5, "HP ZBook Studio 16", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), 10414 SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), 10415 SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), 10416 SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10417 SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), 10418 SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), 10419 SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10420 SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK), 10421 SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10422 SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10423 SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10424 SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), 10425 SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), 10426 SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), 10427 SND_PCI_QUIRK(0x1043, 0x12a3, "Asus N7691ZM", ALC269_FIXUP_ASUS_N7601ZM), 10428 SND_PCI_QUIRK(0x1043, 0x12af, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), 10429 SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), 10430 SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), 10431 SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), 10432 SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), 10433 SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), 10434 SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650PY/PZ/PV/PU/PYV/PZV/PIV/PVV", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), 10435 SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X/GA402N", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), 10436 SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604VI/VC/VE/VG/VJ/VQ/VU/VV/VY/VZ", ALC285_FIXUP_ASUS_HEADSET_MIC), 10437 SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603VQ/VU/VV/VJ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10438 SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601VV/VU/VJ/VQ/VI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10439 SND_PCI_QUIRK(0x1043, 0x14d3, "ASUS G614JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), 10440 SND_PCI_QUIRK(0x1043, 0x14e3, "ASUS G513PI/PU/PV", ALC287_FIXUP_CS35L41_I2C_2), 10441 SND_PCI_QUIRK(0x1043, 0x1503, "ASUS G733PY/PZ/PZV/PYV", ALC287_FIXUP_CS35L41_I2C_2), 10442 SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), 10443 SND_PCI_QUIRK(0x1043, 0x1533, "ASUS GV302XA/XJ/XQ/XU/XV/XI", ALC287_FIXUP_CS35L41_I2C_2), 10444 SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301VV/VQ/VU/VJ/VA/VC/VE/VVC/VQC/VUC/VJC/VEC/VCC", ALC285_FIXUP_ASUS_HEADSET_MIC), 10445 SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), 10446 SND_PCI_QUIRK(0x1043, 0x1663, "ASUS GU603ZI/ZJ/ZQ/ZU/ZV", ALC285_FIXUP_ASUS_HEADSET_MIC), 10447 SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), 10448 SND_PCI_QUIRK(0x1043, 0x16a3, "ASUS UX3402VA", ALC245_FIXUP_CS35L41_SPI_2), 10449 SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), 10450 SND_PCI_QUIRK(0x1043, 0x16d3, "ASUS UX5304VA", ALC245_FIXUP_CS35L41_SPI_2), 10451 SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), 10452 SND_PCI_QUIRK(0x1043, 0x16f3, "ASUS UX7602VI/BZ", ALC245_FIXUP_CS35L41_SPI_2), 10453 SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), 10454 SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), 10455 SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), 10456 SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), 10457 SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), 10458 SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), 10459 SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), 10460 SND_PCI_QUIRK(0x1043, 0x18d3, "ASUS UM3504DA", ALC294_FIXUP_CS35L41_I2C_2), 10461 SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), 10462 SND_PCI_QUIRK(0x1043, 0x194e, "ASUS UX563FD", ALC294_FIXUP_ASUS_HPE), 10463 SND_PCI_QUIRK(0x1043, 0x1970, "ASUS UX550VE", ALC289_FIXUP_ASUS_GA401), 10464 SND_PCI_QUIRK(0x1043, 0x1982, "ASUS B1400CEPE", ALC256_FIXUP_ASUS_HPE), 10465 SND_PCI_QUIRK(0x1043, 0x19ce, "ASUS B9450FA", ALC294_FIXUP_ASUS_HPE), 10466 SND_PCI_QUIRK(0x1043, 0x19e1, "ASUS UX581LV", ALC295_FIXUP_ASUS_MIC_NO_PRESENCE), 10467 SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), 10468 SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), 10469 SND_PCI_QUIRK(0x1043, 0x1a63, "ASUS UX3405MA", ALC245_FIXUP_CS35L41_SPI_2), 10470 SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), 10471 SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), 10472 SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), 10473 SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC), 10474 SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), 10475 SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), 10476 SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), 10477 SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10478 SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), 10479 SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), 10480 SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), 10481 SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1), 10482 SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), 10483 SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), 10484 SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10485 SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), 10486 SND_PCI_QUIRK(0x1043, 0x1ccf, "ASUS G814JU/JV/JI", ALC245_FIXUP_CS35L41_SPI_2), 10487 SND_PCI_QUIRK(0x1043, 0x1cdf, "ASUS G814JY/JZ/JG", ALC245_FIXUP_CS35L41_SPI_2), 10488 SND_PCI_QUIRK(0x1043, 0x1cef, "ASUS G834JY/JZ/JI/JG", ALC285_FIXUP_ASUS_HEADSET_MIC), 10489 SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS G713PI/PU/PV/PVN", ALC287_FIXUP_CS35L41_I2C_2), 10490 SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), 10491 SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), 10492 SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), 10493 SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), 10494 SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), 10495 SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), 10496 SND_PCI_QUIRK(0x1043, 0x1e1f, "ASUS Vivobook 15 X1504VAP", ALC2XX_FIXUP_HEADSET_MIC), 10497 SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), 10498 SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), 10499 SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), 10500 SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), 10501 SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), 10502 SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), 10503 SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), 10504 SND_PCI_QUIRK(0x1043, 0x1f12, "ASUS UM5302", ALC287_FIXUP_CS35L41_I2C_2), 10505 SND_PCI_QUIRK(0x1043, 0x1f1f, "ASUS H7604JI/JV/J3D", ALC245_FIXUP_CS35L41_SPI_2), 10506 SND_PCI_QUIRK(0x1043, 0x1f62, "ASUS UX7602ZM", ALC245_FIXUP_CS35L41_SPI_2), 10507 SND_PCI_QUIRK(0x1043, 0x1f92, "ASUS ROG Flow X16", ALC289_FIXUP_ASUS_GA401), 10508 SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), 10509 SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10510 SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10511 SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10512 SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10513 SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), 10514 SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), 10515 SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), 10516 SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), 10517 SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), 10518 SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), 10519 SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), 10520 SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), 10521 SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), 10522 SND_PCI_QUIRK(0x104d, 0x9099, "Sony VAIO S13", ALC275_FIXUP_SONY_DISABLE_AAMIX), 10523 SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10524 SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10525 SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), 10526 SND_PCI_QUIRK(0x10cf, 0x159f, "Lifebook E780", ALC269_FIXUP_LIFEBOOK_NO_HP_TO_LINEOUT), 10527 SND_PCI_QUIRK(0x10cf, 0x15dc, "Lifebook T731", ALC269_FIXUP_LIFEBOOK_HP_PIN), 10528 SND_PCI_QUIRK(0x10cf, 0x1629, "Lifebook U7x7", ALC255_FIXUP_LIFEBOOK_U7x7_HEADSET_MIC), 10529 SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), 10530 SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), 10531 SND_PCI_QUIRK(0x10ec, 0x10f2, "Intel Reference board", ALC700_FIXUP_INTEL_REFERENCE), 10532 SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), 10533 SND_PCI_QUIRK(0x10ec, 0x119e, "Positivo SU C1400", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10534 SND_PCI_QUIRK(0x10ec, 0x11bc, "VAIO VJFE-IL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10535 SND_PCI_QUIRK(0x10ec, 0x1230, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10536 SND_PCI_QUIRK(0x10ec, 0x124c, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10537 SND_PCI_QUIRK(0x10ec, 0x1252, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10538 SND_PCI_QUIRK(0x10ec, 0x1254, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10539 SND_PCI_QUIRK(0x10ec, 0x12cc, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10540 SND_PCI_QUIRK(0x10ec, 0x12f6, "Intel Reference board", ALC295_FIXUP_CHROME_BOOK), 10541 SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-SZ6", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10542 SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), 10543 SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_AMP), 10544 SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_AMP), 10545 SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_AMP), 10546 SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_AMP), 10547 SND_PCI_QUIRK(0x144d, 0xc1a3, "Samsung Galaxy Book Pro (NP935XDB-KC1SE)", ALC298_FIXUP_SAMSUNG_AMP), 10548 SND_PCI_QUIRK(0x144d, 0xc1a4, "Samsung Galaxy Book Pro 360 (NT935QBD)", ALC298_FIXUP_SAMSUNG_AMP), 10549 SND_PCI_QUIRK(0x144d, 0xc1a6, "Samsung Galaxy Book Pro 360 (NP930QBD)", ALC298_FIXUP_SAMSUNG_AMP), 10550 SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), 10551 SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), 10552 SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), 10553 SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), 10554 SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), 10555 SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), 10556 SND_PCI_QUIRK(0x144d, 0xc870, "Samsung Galaxy Book2 Pro (NP950XED)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), 10557 SND_PCI_QUIRK(0x144d, 0xc872, "Samsung Galaxy Book2 Pro (NP950XEE)", ALC298_FIXUP_SAMSUNG_AMP_V2_2_AMPS), 10558 SND_PCI_QUIRK(0x144d, 0xc886, "Samsung Galaxy Book3 Pro (NP964XFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10559 SND_PCI_QUIRK(0x144d, 0xc1ca, "Samsung Galaxy Book3 Pro 360 (NP960QFG)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10560 SND_PCI_QUIRK(0x144d, 0xc1cc, "Samsung Galaxy Book3 Ultra (NT960XFH)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10561 SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), 10562 SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), 10563 SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), 10564 SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), 10565 SND_PCI_QUIRK(0x152d, 0x1262, "Huawei NBLB-WAX9N", ALC2XX_FIXUP_HEADSET_MIC), 10566 SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10567 SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10568 SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10569 SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10570 SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10571 SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10572 SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10573 SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10574 SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10575 SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10576 SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10577 SND_PCI_QUIRK(0x1558, 0x4041, "Clevo NV4[15]PZ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10578 SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10579 SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10580 SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10581 SND_PCI_QUIRK(0x1558, 0x5015, "Clevo NH5[58]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10582 SND_PCI_QUIRK(0x1558, 0x5017, "Clevo NH7[79]H[HJK]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10583 SND_PCI_QUIRK(0x1558, 0x50a3, "Clevo NJ51GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10584 SND_PCI_QUIRK(0x1558, 0x50b3, "Clevo NK50S[BEZ]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10585 SND_PCI_QUIRK(0x1558, 0x50b6, "Clevo NK50S5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10586 SND_PCI_QUIRK(0x1558, 0x50b8, "Clevo NK50SZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10587 SND_PCI_QUIRK(0x1558, 0x50d5, "Clevo NP50D5", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10588 SND_PCI_QUIRK(0x1558, 0x50e1, "Clevo NH5[58]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10589 SND_PCI_QUIRK(0x1558, 0x50e2, "Clevo NH7[79]HPQ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10590 SND_PCI_QUIRK(0x1558, 0x50f0, "Clevo NH50A[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10591 SND_PCI_QUIRK(0x1558, 0x50f2, "Clevo NH50E[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10592 SND_PCI_QUIRK(0x1558, 0x50f3, "Clevo NH58DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10593 SND_PCI_QUIRK(0x1558, 0x50f5, "Clevo NH55EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10594 SND_PCI_QUIRK(0x1558, 0x50f6, "Clevo NH55DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10595 SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10596 SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10597 SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10598 SND_PCI_QUIRK(0x1558, 0x51b1, "Clevo NS50AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10599 SND_PCI_QUIRK(0x1558, 0x51b3, "Clevo NS70AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10600 SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10601 SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10602 SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10603 SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10604 SND_PCI_QUIRK(0x1558, 0x70f3, "Clevo NH77DPQ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10605 SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10606 SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10607 SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10608 SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10609 SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10610 SND_PCI_QUIRK(0x1558, 0x7724, "Clevo L140AU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10611 SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10612 SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10613 SND_PCI_QUIRK(0x1558, 0x8521, "Clevo NH77D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10614 SND_PCI_QUIRK(0x1558, 0x8535, "Clevo NH50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10615 SND_PCI_QUIRK(0x1558, 0x8536, "Clevo NH79D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10616 SND_PCI_QUIRK(0x1558, 0x8550, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10617 SND_PCI_QUIRK(0x1558, 0x8551, "Clevo NH[57][0-9][ER][ACDH]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10618 SND_PCI_QUIRK(0x1558, 0x8560, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), 10619 SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), 10620 SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), 10621 SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10622 SND_PCI_QUIRK(0x1558, 0x866d, "Clevo NP5[05]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10623 SND_PCI_QUIRK(0x1558, 0x867c, "Clevo NP7[01]PNP", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10624 SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10625 SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10626 SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), 10627 SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10628 SND_PCI_QUIRK(0x1558, 0x8a51, "Clevo NH70RCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10629 SND_PCI_QUIRK(0x1558, 0x8d50, "Clevo NH55RCQ-M", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10630 SND_PCI_QUIRK(0x1558, 0x951d, "Clevo N950T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10631 SND_PCI_QUIRK(0x1558, 0x9600, "Clevo N960K[PR]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10632 SND_PCI_QUIRK(0x1558, 0x961d, "Clevo N960S[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10633 SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10634 SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10635 SND_PCI_QUIRK(0x1558, 0xa554, "VAIO VJFH52", ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE), 10636 SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10637 SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10638 SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10639 SND_PCI_QUIRK(0x1558, 0xa741, "Clevo V54x_6x_TNE", ALC245_FIXUP_CLEVO_NOISY_MIC), 10640 SND_PCI_QUIRK(0x1558, 0xa763, "Clevo V54x_6x_TU", ALC245_FIXUP_CLEVO_NOISY_MIC), 10641 SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10642 SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10643 SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10644 SND_PCI_QUIRK(0x1558, 0xc018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10645 SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10646 SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), 10647 SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), 10648 SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), 10649 SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), 10650 SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), 10651 SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), 10652 SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), 10653 SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), 10654 SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), 10655 SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK_LIMIT_BOOST), 10656 SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), 10657 SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), 10658 SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), 10659 SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), 10660 SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad T440s", ALC292_FIXUP_TPT440), 10661 SND_PCI_QUIRK(0x17aa, 0x220e, "Thinkpad T440p", ALC292_FIXUP_TPT440_DOCK), 10662 SND_PCI_QUIRK(0x17aa, 0x2210, "Thinkpad T540p", ALC292_FIXUP_TPT440_DOCK), 10663 SND_PCI_QUIRK(0x17aa, 0x2211, "Thinkpad W541", ALC292_FIXUP_TPT440_DOCK), 10664 SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad T440", ALC292_FIXUP_TPT440_DOCK), 10665 SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad X240", ALC292_FIXUP_TPT440_DOCK), 10666 SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10667 SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), 10668 SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), 10669 SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), 10670 SND_PCI_QUIRK(0x17aa, 0x222d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10671 SND_PCI_QUIRK(0x17aa, 0x222e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10672 SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), 10673 SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), 10674 SND_PCI_QUIRK(0x17aa, 0x2234, "Thinkpad ICE-1", ALC287_FIXUP_TAS2781_I2C), 10675 SND_PCI_QUIRK(0x17aa, 0x2245, "Thinkpad T470", ALC298_FIXUP_TPT470_DOCK), 10676 SND_PCI_QUIRK(0x17aa, 0x2246, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10677 SND_PCI_QUIRK(0x17aa, 0x2247, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10678 SND_PCI_QUIRK(0x17aa, 0x2249, "Thinkpad", ALC292_FIXUP_TPT460), 10679 SND_PCI_QUIRK(0x17aa, 0x224b, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10680 SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10681 SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10682 SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10683 SND_PCI_QUIRK(0x17aa, 0x2292, "Thinkpad X1 Carbon 7th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), 10684 SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), 10685 SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), 10686 SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), 10687 SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10688 SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10689 SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10690 SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10691 SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10692 SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10693 SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10694 SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10695 SND_PCI_QUIRK(0x17aa, 0x231e, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), 10696 SND_PCI_QUIRK(0x17aa, 0x231f, "Thinkpad", ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318), 10697 SND_PCI_QUIRK(0x17aa, 0x2326, "Hera2", ALC287_FIXUP_TAS2781_I2C), 10698 SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), 10699 SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), 10700 SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10701 SND_PCI_QUIRK(0x17aa, 0x3111, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10702 SND_PCI_QUIRK(0x17aa, 0x312a, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10703 SND_PCI_QUIRK(0x17aa, 0x312f, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10704 SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), 10705 SND_PCI_QUIRK(0x17aa, 0x3151, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10706 SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10707 SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), 10708 SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), 10709 SND_PCI_QUIRK(0x17aa, 0x334b, "Lenovo ThinkCentre M70 Gen5", ALC283_FIXUP_HEADSET_MIC), 10710 SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10711 HDA_CODEC_QUIRK(0x17aa, 0x3802, "DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10712 SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga Pro 9 14IRP8", ALC287_FIXUP_TAS2781_I2C), 10713 SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), 10714 SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), 10715 SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), 10716 HDA_CODEC_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC), 10717 SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10718 SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), 10719 SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), 10720 SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10721 SND_PCI_QUIRK(0x17aa, 0x383d, "Legion Y9000X 2019", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), 10722 SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), 10723 SND_PCI_QUIRK(0x17aa, 0x3847, "Legion 7 16ACHG6", ALC287_FIXUP_LEGION_16ACHG6), 10724 SND_PCI_QUIRK(0x17aa, 0x384a, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10725 SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10726 SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), 10727 SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), 10728 SND_PCI_QUIRK(0x17aa, 0x3865, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), 10729 SND_PCI_QUIRK(0x17aa, 0x3866, "Lenovo 13X", ALC287_FIXUP_CS35L41_I2C_2), 10730 SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10731 HDA_CODEC_QUIRK(0x17aa, 0x386e, "Legion Y9000X 2022 IAH7", ALC287_FIXUP_CS35L41_I2C_2), 10732 SND_PCI_QUIRK(0x17aa, 0x386e, "Yoga Pro 7 14ARP8", ALC285_FIXUP_SPEAKER2_TO_DAC1), 10733 HDA_CODEC_QUIRK(0x17aa, 0x386f, "Legion Pro 7 16ARX8H", ALC287_FIXUP_TAS2781_I2C), 10734 SND_PCI_QUIRK(0x17aa, 0x386f, "Legion Pro 7i 16IAX7", ALC287_FIXUP_CS35L41_I2C_2), 10735 SND_PCI_QUIRK(0x17aa, 0x3870, "Lenovo Yoga 7 14ARB7", ALC287_FIXUP_YOGA7_14ARB7_I2C), 10736 SND_PCI_QUIRK(0x17aa, 0x3877, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 10737 SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), 10738 SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), 10739 SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), 10740 SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), 10741 SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10742 SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10743 SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), 10744 SND_PCI_QUIRK(0x17aa, 0x3891, "Lenovo Yoga Pro 7 14AHP9", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), 10745 SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), 10746 SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), 10747 SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10748 SND_PCI_QUIRK(0x17aa, 0x38ab, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), 10749 SND_PCI_QUIRK(0x17aa, 0x38b4, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), 10750 SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), 10751 SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), 10752 SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), 10753 SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), 10754 SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), 10755 SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), 10756 SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), 10757 SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), 10758 SND_PCI_QUIRK(0x17aa, 0x38c7, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), 10759 SND_PCI_QUIRK(0x17aa, 0x38c8, "Thinkbook 13x Gen 4", ALC287_FIXUP_CS35L41_I2C_4), 10760 SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10761 SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), 10762 SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), 10763 SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), 10764 SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), 10765 SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), 10766 SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), 10767 SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), 10768 SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), 10769 SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), 10770 SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), 10771 SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), 10772 SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10773 SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), 10774 SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), 10775 SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10776 SND_PCI_QUIRK(0x17aa, 0x5034, "Thinkpad T450", ALC292_FIXUP_TPT440_DOCK), 10777 SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), 10778 SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), 10779 SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), 10780 SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), 10781 SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460), 10782 SND_PCI_QUIRK(0x17aa, 0x5051, "Thinkpad L460", ALC292_FIXUP_TPT460), 10783 SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460), 10784 SND_PCI_QUIRK(0x17aa, 0x505d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10785 SND_PCI_QUIRK(0x17aa, 0x505f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10786 SND_PCI_QUIRK(0x17aa, 0x5062, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10787 SND_PCI_QUIRK(0x17aa, 0x508b, "Thinkpad X12 Gen 1", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), 10788 SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10789 SND_PCI_QUIRK(0x17aa, 0x511e, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10790 SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), 10791 SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), 10792 SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), 10793 SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), 10794 SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), 10795 SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), 10796 SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), 10797 SND_PCI_QUIRK(0x1854, 0x0488, "LG gram 16 (16Z90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10798 SND_PCI_QUIRK(0x1854, 0x048a, "LG gram 17 (17ZD90R)", ALC298_FIXUP_SAMSUNG_AMP_V2_4_AMPS), 10799 SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), 10800 SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), 10801 SND_PCI_QUIRK(0x19e5, 0x3212, "Huawei KLV-WX9 ", ALC256_FIXUP_ACER_HEADSET_MIC), 10802 SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), 10803 SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), 10804 SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), 10805 SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ 10806 SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), 10807 SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), 10808 SND_PCI_QUIRK(0x1c6c, 0x122a, "Positivo N14AP7", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10809 SND_PCI_QUIRK(0x1c6c, 0x1251, "Positivo N14KP6-TG", ALC288_FIXUP_DELL1_MIC_NO_PRESENCE), 10810 SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), 10811 SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), 10812 SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), 10813 SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10814 SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10815 SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), 10816 SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), 10817 SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), 10818 SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), 10819 SND_PCI_QUIRK(0x1d05, 0x1387, "TongFang GMxIXxx", ALC2XX_FIXUP_HEADSET_MIC), 10820 SND_PCI_QUIRK(0x1d17, 0x3288, "Haier Boyue G42", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), 10821 SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), 10822 SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), 10823 SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), 10824 SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC), 10825 SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC), 10826 SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), 10827 SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), 10828 SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), 10829 SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), 10830 SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), 10831 SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), 10832 SND_PCI_QUIRK(0x8086, 0x3038, "Intel NUC 13", ALC295_FIXUP_CHROME_BOOK), 10833 SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10834 SND_PCI_QUIRK(0xf111, 0x0006, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10835 SND_PCI_QUIRK(0xf111, 0x0009, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), 10836 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

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[PATCH v1] ALSA: hda/tas2781: Add new quirk for Lenovo, ASUS, Dell projects
by Baojun Xu 14 Oct '24

14 Oct '24

Add new vendor_id and subsystem_id in quirk for Lenovo, ASUS, and Dell projects. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/patch_realtek.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e2e927656cd..28d8e58e7fee 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10198,6 +10198,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0c1e, "Dell Precision 3540", ALC236_FIXUP_DELL_DUAL_CODECS), SND_PCI_QUIRK(0x1028, 0x0c28, "Dell Inspiron 16 Plus 7630", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x0c4d, "Dell", ALC287_FIXUP_CS35L41_I2C_4), + SND_PCI_QUIRK(0x1028, 0x0c94, "Dell Polaris 3 metal", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1028, 0x0c96, "Dell Polaris 2in1", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1028, 0x0cbd, "Dell Oasis 13 CS MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbe, "Dell Oasis 13 2-IN-1 MTL-U", ALC289_FIXUP_DELL_CS35L41_SPI_2), SND_PCI_QUIRK(0x1028, 0x0cbf, "Dell Oasis 13 Low Weight MTU-L", ALC289_FIXUP_DELL_CS35L41_SPI_2), @@ -10501,11 +10503,15 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x10a1, "ASUS UX391UA", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x10a4, "ASUS TP3407SA", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x10c0, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x10d0, "ASUS X540LA/X540LJ", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x10d3, "ASUS K6500ZC", ALC294_FIXUP_ASUS_SPK), + SND_PCI_QUIRK(0x1043, 0x1154, "ASUS TP3607SH", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x11c0, "ASUS X556UR", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1043, 0x1204, "ASUS Strix G615JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x1214, "ASUS Strix G615LH_LM_LP", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x125e, "ASUS Q524UQK", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1271, "ASUS X430UN", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1290, "ASUS X441SA", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), @@ -10583,6 +10589,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), @@ -10597,6 +10604,13 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), + SND_PCI_QUIRK(0x1043, 0x3e30, "ASUS TP3607SA", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3ee0, "ASUS Strix G815_JHR_JMR_JPR", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3ef0, "ASUS Strix G635LR_LW_LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f00, "ASUS Strix G815LH_LM_LP", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f10, "ASUS Strix G835LR_LW_LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f20, "ASUS Strix G615LR_LW", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x1043, 0x3f30, "ASUS Strix G815LR_LW", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), @@ -10819,11 +10833,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3878, "Lenovo Legion 7 Slim 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387f, "Yoga S780-16 pro dual LX", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3880, "Yoga S780-16 pro dual YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3882, "Lenovo Yoga Pro 7 14APH8", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3891, "Lenovo Yoga Pro 7 14AHP9", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38a5, "Y580P AMD dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a9, "Thinkbook 16P", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD), @@ -10832,6 +10849,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38b8, "Yoga S780-14.5 proX AMD YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38b9, "Yoga S780-14.5 proX AMD LX Dual", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), @@ -10842,12 +10861,22 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38d3, "Yoga S990-16 Pro IMH YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d4, "Yoga S990-16 Pro IMH VECO Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d5, "Yoga S990-16 Pro IMH YC Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38d6, "Yoga S990-16 Pro IMH VECO Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38df, "Yoga Y990 Intel YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38e0, "Yoga Y990 Intel VECO Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38f8, "Yoga Book 9i", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x38fd, "ThinkBook plus Gen5 Hybrid", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), + SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), -- 2.43.0

2 1

[PATCH] Support for Line 6 TonePort KB37
by Seth Poulsen 14 Oct '24

14 Oct '24

The Line 6 TonePort KB37 is a device with the same ports as the TonePort UX2, but with a MIDI keyboard onboard. It can use the same driver as the UX2, it just needed to be enabled. -Seth Poulsen

2 1

[PATCH alsa-ucm-conf v3 1/2] sof-soundwire: Change map control names to make them unique and user friendly
by Maciej Strozek 14 Oct '24

14 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- v3: Make Headset Microphone more user-friendly and contain the word "Capture" v2: Rename to "Speaker Playback" instead of "Speaker Combined" --- ucm2/codecs/cs42l43/init.conf | 10 ++++++++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- ucm2/sof-soundwire/cs42l43.conf | 2 +- 4 files changed, 13 insertions(+), 7 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..b2796e2 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,13 +2,19 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } + "name='cs42l43 Headset Microphone Capture Switch'" { + "name='cs42l43 Decimator 1 Switch'".vindex.0 0 + } + "name='cs42l43 Headset Microphone Capture Volume'" { + "name='cs42l43 Decimator 1 Volume'".vindex.0 0 + } } } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..a3cf6ad 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Playback Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Playback Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Playback" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 1702cf0..7e74fb8 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -40,7 +40,7 @@ SectionDevice."Headset" { Value { CapturePriority 200 CapturePCM "hw:${CardId},4" - CaptureMixerElem "cs42l43 Decimator 1" + CaptureMixerElem "cs42l43 Headset Microphone" JackControl "Headset Mic Jack" } } -- 2.39.5

2 2

[PATCH] ASoC: rt-sdw-common: Enhance switch case to prevent uninitialized variable
by Jack Yu 14 Oct '24

14 Oct '24

If det_mode is not 0, 3 or 5 then function will return jack_type with an uninitialzed value. Enhance switch case to prevent uninitialized variable issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt-sdw-common.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c index a422da6cf702..3ec576c576be 100644 --- a/sound/soc/codecs/rt-sdw-common.c +++ b/sound/soc/codecs/rt-sdw-common.c @@ -150,15 +150,14 @@ int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) goto io_error; switch (det_mode) { - case 0x00: - jack_type = 0; - break; case 0x03: jack_type = SND_JACK_HEADPHONE; break; case 0x05: jack_type = SND_JACK_HEADSET; break; + default: + jack_type = 0; } /* write selected_mode */ -- 2.34.1

2 1

[tiwai-sound:for-next 9/14] sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 1f55e3699fc9ced72400cdca39fe248bf2b288a2 [9/14] ALSA: hda/conexant: Use the new codec SSID matching config: i386-buildonly-randconfig-006-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141700.bF6I8r36-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241014/202410141700.bF6I8r36-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410141700.bF6I8r36-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All error/warnings (new ones prefixed by >>): In file included from sound/pci/hda/patch_conexant.c:18: >> sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^~~~ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:65: warning: excess elements in struct initializer 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:65: note: (near initialization for 'cxt5066_fixups[47]') 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1081:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:58: error: 'const struct hda_quirk' has no member named 'name' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^~~~ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ >> sound/pci/hda/hda_local.h:312:65: warning: excess elements in struct initializer 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:65: note: (near initialization for 'cxt5066_fixups[48]') 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ^ sound/pci/hda/patch_conexant.c:1082:9: note: in expansion of macro 'HDA_CODEC_QUIRK' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~ vim +312 sound/pci/hda/hda_local.h 5b1913a79c3e05 Takashi Iwai 2024-10-08 310 5b1913a79c3e05 Takashi Iwai 2024-10-08 311 #define HDA_CODEC_QUIRK(vend, dev, xname, val) \ 5b1913a79c3e05 Takashi Iwai 2024-10-08 @312 { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ 5b1913a79c3e05 Takashi Iwai 2024-10-08 313 .match_codec_ssid = true } 5b1913a79c3e05 Takashi Iwai 2024-10-08 314 :::::: The code at line 312 was first introduced by commit :::::: 5b1913a79c3e0518d9c5db343fa9fc4edcea041f ALSA: hda: Use own quirk lookup helper :::::: TO: Takashi Iwai <tiwai(a)suse.de> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 9/14] sound/pci/hda/patch_conexant.c:1081:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk'
by kernel test robot 14 Oct '24

14 Oct '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 commit: 1f55e3699fc9ced72400cdca39fe248bf2b288a2 [9/14] ALSA: hda/conexant: Use the new codec SSID matching config: i386-buildonly-randconfig-003-20241014 (https://download.01.org/0day-ci/archive/20241014/202410141648.JOPFB2du-lkp@…) compiler: clang version 18.1.8 (https://github.com/llvm/llvm-project 3b5b5c1ec4a3095ab096dd780e84d7ab81f3d7ff) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20241014/202410141648.JOPFB2du-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202410141648.JOPFB2du-lkp@intel.com/ Note: the tiwai-sound/for-next HEAD 0ddf2784d6c29e59409a62b8f32dc5abe56135a4 builds fine. It only hurts bisectability. All errors (new ones prefixed by >>): >> sound/pci/hda/patch_conexant.c:1081:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 1081 | HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ sound/pci/hda/patch_conexant.c:1082:2: error: field designator 'name' does not refer to any field in type 'const struct hda_quirk' 1082 | HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/hda_local.h:312:51: note: expanded from macro 'HDA_CODEC_QUIRK' 312 | { _SND_PCI_QUIRK_ID(vend, dev), .value = (val), .name = (xname),\ | ~^~~~~~~~~~~~~~ 2 errors generated. vim +1081 sound/pci/hda/patch_conexant.c 1029 1030 static const struct hda_quirk cxt5066_fixups[] = { 1031 SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), 1032 SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), 1033 SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), 1034 SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), 1035 SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), 1036 SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), 1037 SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), 1038 SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), 1039 SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), 1040 SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), 1041 SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), 1042 SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1043 SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1044 SND_PCI_QUIRK(0x103c, 0x82b4, "HP ProDesk 600 G3", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1045 SND_PCI_QUIRK(0x103c, 0x836e, "HP ProBook 455 G5", CXT_FIXUP_MUTE_LED_GPIO), 1046 SND_PCI_QUIRK(0x103c, 0x837f, "HP ProBook 470 G5", CXT_FIXUP_MUTE_LED_GPIO), 1047 SND_PCI_QUIRK(0x103c, 0x83b2, "HP EliteBook 840 G5", CXT_FIXUP_HP_DOCK), 1048 SND_PCI_QUIRK(0x103c, 0x83b3, "HP EliteBook 830 G5", CXT_FIXUP_HP_DOCK), 1049 SND_PCI_QUIRK(0x103c, 0x83d3, "HP ProBook 640 G4", CXT_FIXUP_HP_DOCK), 1050 SND_PCI_QUIRK(0x103c, 0x8402, "HP ProBook 645 G4", CXT_FIXUP_MUTE_LED_GPIO), 1051 SND_PCI_QUIRK(0x103c, 0x8427, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), 1052 SND_PCI_QUIRK(0x103c, 0x844f, "HP ZBook Studio G5", CXT_FIXUP_HP_ZBOOK_MUTE_LED), 1053 SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1054 SND_PCI_QUIRK(0x103c, 0x8456, "HP Z2 G4 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1055 SND_PCI_QUIRK(0x103c, 0x8457, "HP Z2 G4 mini", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1056 SND_PCI_QUIRK(0x103c, 0x8458, "HP Z2 G4 mini premium", CXT_FIXUP_HP_MIC_NO_PRESENCE), 1057 SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), 1058 SND_PCI_QUIRK(0x14f1, 0x0265, "SWS JS201D", CXT_PINCFG_SWS_JS201D), 1059 SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), 1060 SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), 1061 SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), 1062 SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), 1063 SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), 1064 SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), 1065 SND_PCI_QUIRK(0x17aa, 0x21d2, "Lenovo T420s", CXT_PINCFG_LENOVO_TP410), 1066 SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), 1067 SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), 1068 SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo IdeaPad Z560", CXT_FIXUP_MUTE_LED_EAPD), 1069 SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), 1070 SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), 1071 SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), 1072 /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same 1073 * PCI SSID is used on multiple Lenovo models 1074 */ 1075 SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), 1076 SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), 1077 SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), 1078 SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), 1079 SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), 1080 SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), > 1081 HDA_CODEC_QUIRK(0x2782, 0x12c3, "Sirius Gen1", CXT_PINCFG_TOP_SPEAKER), 1082 HDA_CODEC_QUIRK(0x2782, 0x12c5, "Sirius Gen2", CXT_PINCFG_TOP_SPEAKER), 1083 {} 1084 }; 1085 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH 6.1] ASoC: rt5682s: Return devm_of_clk_add_hw_provider to transfer the error
by Vitaliy Shevtsov 14 Oct '24

14 Oct '24

From: Ma Ke <make24(a)iscas.ac.cn> commit 3ff810b9bebe5578a245cfa97c252ab602e703f1 upstream. Return devm_of_clk_add_hw_provider() in order to transfer the error, if it fails due to resource allocation failure or device tree clock provider registration failure. Fixes: bdd229ab26be ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Ma Ke <make24(a)iscas.ac.cn> Link: https://patch.msgid.link/20240717115436.3449492-1-make24@iscas.ac.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Vitaliy Shevtsov <v.shevtsov(a)maxima.ru> --- sound/soc/codecs/rt5682s.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 80c673aa14db..07d514b4ce70 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2828,7 +2828,9 @@ static int rt5682s_register_dai_clks(struct snd_soc_component *component) } if (dev->of_node) { - devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + ret = devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, dai_clk_hw); + if (ret) + return ret; } else { ret = devm_clk_hw_register_clkdev(dev, dai_clk_hw, init.name, dev_name(dev)); -- 2.46.2

1 0

[PATCH v2] ASoC: rt-sdw-common: Common functions for Realtek soundwire driver
by Jack Yu 13 Oct '24

13 Oct '24

This is the first version of common functions for Realtek soundwire codec driver. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt-sdw-common.c | 238 +++++++++++++++++++++++++++++++ sound/soc/codecs/rt-sdw-common.h | 66 +++++++++ 4 files changed, 311 insertions(+) create mode 100644 sound/soc/codecs/rt-sdw-common.c create mode 100644 sound/soc/codecs/rt-sdw-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..b095eba9d634 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1545,6 +1545,11 @@ config SND_SOC_RL6231 default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m +config SND_SOC_RT_SDW_COMMON + tristate + default y if SND_SOC_RT721_SDCA_SDW=y + default m if SND_SOC_RT721_SDCA_SDW=m + config SND_SOC_RL6347A tristate default y if SND_SOC_RT274=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..4497b2e30cd3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -216,6 +216,7 @@ snd-soc-rk3308-y := rk3308_codec.o snd-soc-rk3328-y := rk3328_codec.o snd-soc-rk817-y := rk817_codec.o snd-soc-rl6231-y := rl6231.o +snd-soc-rt-sdw-common-y := rt-sdw-common.o snd-soc-rl6347a-y := rl6347a.o snd-soc-rt1011-y := rt1011.o snd-soc-rt1015-y := rt1015.o @@ -618,6 +619,7 @@ obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT_SDW_COMMON) += snd-soc-rt-sdw-common.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c new file mode 100644 index 000000000000..9ed0e9855699 --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.c @@ -0,0 +1,238 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt-sdw-common.c +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/bitops.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/jack.h> + +#include "rt-sdw-common.h" + +/** + * rt_sdca_index_write - Write a value to Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_write(map, addr, value); + if (ret < 0) + pr_err("Failed to set value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_write); + +/** + * rt_sdca_index_read - Read value from Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_read(map, addr, value); + if (ret < 0) + pr_err("Failed to get value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_read); + +/** + * rt_sdca_index_update_bits - Update value on Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @mask: Bitmask to change + * @value: New value for bitmask + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ + +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt_sdca_index_read(map, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt_sdca_index_write(map, nid, reg, tmp); +} +EXPORT_SYMBOL_GPL(rt_sdca_index_update_bits); + +/** + * rt_sdca_btn_type - Decision of button type. + * + * @buffer: UMP message buffer. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_btn_type(unsigned char *buffer) +{ + u8 btn_type = 0; + int ret; + + btn_type |= buffer[0] & 0xf; + btn_type |= (buffer[0] >> 4) & 0xf; + btn_type |= buffer[1] & 0xf; + btn_type |= (buffer[1] >> 4) & 0xf; + + if (btn_type & BIT(0)) + ret |= SND_JACK_BTN_2; + if (btn_type & BIT(1)) + ret |= SND_JACK_BTN_3; + if (btn_type & BIT(2)) + ret |= SND_JACK_BTN_0; + if (btn_type & BIT(3)) + ret |= SND_JACK_BTN_1; + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_btn_type); + +/** + * rt_sdca_headset_detect - Headset jack type detection. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * + * A headset jack type will be returned, a negative errno will + * be returned in error cases. + */ +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) +{ + unsigned int det_mode, jack_type; + int ret; + + /* get detected_mode */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + jack_type = 0; + break; + case 0x03: + jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_SELECTED_MODE, 0), det_mode); + if (ret < 0) + goto io_error; + } + + return jack_type; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_headset_detect); + +/** + * rt_sdca_button_detect - Read UMP message and decide button type. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * @hid_buf_addr: HID buffer address. + * @hid_id: Report ID for HID. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + unsigned char buf[3]; + int ret; + + /* get current UMP message owner */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(map, hid_buf_addr + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + /* Report ID for HID */ + if (buf[0] == hid_id) + btn_type = rt_sdca_btn_type(&buf[1]); + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(map, + SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01); + + return btn_type; +} +EXPORT_SYMBOL_GPL(rt_sdca_button_detect); + +MODULE_DESCRIPTION("Realtek soundwire common functions"); +MODULE_AUTHOR("jack yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt-sdw-common.h b/sound/soc/codecs/rt-sdw-common.h new file mode 100644 index 000000000000..4759516feb38 --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.h @@ -0,0 +1,66 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +// +// rt-sdw-common.h +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#ifndef __RT_SDW_COMMON_H__ +#define __RT_SDW_COMMON_H__ + +#define SDCA_NUM_JACK_CODEC 0x01 +#define SDCA_NUM_MIC_ARRAY 0x02 +#define SDCA_NUM_HID 0x03 +#define SDCA_NUM_AMP 0x04 +#define RT_SDCA_CTL_SELECTED_MODE 0x01 +#define RT_SDCA_CTL_DETECTED_MODE 0x02 +#define RT_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 + +struct rt_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +#define RT_SDCA_PR_VALUE(xreg_base, xcount, xmax, xinvert) \ + ((unsigned long)&(struct rt_sdca_dmic_kctrl_priv) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .invert = xinvert}) + +#define RT_SDCA_FU_CTRL(xname, reg_base, xmax, xinvert, xcount, \ + xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = xinfo, \ + .get = xget, \ + .put = xput, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, xinvert)} + +#define RT_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, xcount, xmax, tlv_array, xinfo) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = xinfo, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, 0) } + + +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value); +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value); +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val); +int rt_sdca_btn_type(unsigned char *buffer); +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id); +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id); + +#endif /* __RT_SDW_COMMON_H__ */ -- 2.34.1

3 5

[PATCH 0/2] ASoC: imx-card: add cs42888 codec support
by Shengjiu Wang 12 Oct '24

12 Oct '24

add cs42888 codec support Chancel Liu (2): ASoC: imx-card: Set mclk for codec ASoC: imx-card: Add CS42888 support sound/soc/fsl/imx-card.c | 59 +++++++++++++++++++++++++++++++++++----- 1 file changed, 52 insertions(+), 7 deletions(-) -- 2.34.1

3 4

[PATCH AUTOSEL 6.11 12/16] ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string
by Sasha Levin 12 Oct '24

12 Oct '24

From: Alexey Klimov <alexey.klimov(a)linaro.org> [ Upstream commit b97bc0656a66f89f78098d4d72dc04fa9518ab11 ] Add "qcom,qrb4210-rb2-sndcard" to the list of recognizable devices. Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> Link: https://patch.msgid.link/20241002022015.867031-3-alexey.klimov@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index a15dafb99b337..50e175fd521ce 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -166,6 +166,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) static const struct of_device_id snd_sm8250_dt_match[] = { {.compatible = "qcom,sm8250-sndcard"}, + {.compatible = "qcom,qrb4210-rb2-sndcard"}, {.compatible = "qcom,qrb5165-rb5-sndcard"}, {} }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 04/16] ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit
by Sasha Levin 12 Oct '24

12 Oct '24

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> [ Upstream commit 72455e33173c1a00c0ce93d2b0198eb45d5f4195 ] FCONT=1 means On FIFO error, the SAI will continue from the same word that caused the FIFO error to set after the FIFO warning flag has been cleared. Set FCONT bit in control register to avoid the channel swap issue after SAI xrun. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Link: https://patch.msgid.link/1727676508-22830-1-git-send-email-shengjiu.wang@nx… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_sai.c | 5 ++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad2..a1f03c97b7bb8 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -613,6 +613,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to avoid channel swap */ + val_cr4 |= FSL_SAI_CR4_FCONT; + /* Set to output mode to avoid tri-stated data pins */ if (tx) val_cr4 |= FSL_SAI_CR4_CHMOD; @@ -699,7 +702,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | - FSL_SAI_CR4_CHMOD_MASK, + FSL_SAI_CR4_CHMOD_MASK | FSL_SAI_CR4_FCONT_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index dadbd16ee3945..9c4d19fe22c65 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -137,6 +137,7 @@ /* SAI Transmit and Receive Configuration 4 Register */ +#define FSL_SAI_CR4_FCONT_MASK BIT(28) #define FSL_SAI_CR4_FCONT BIT(28) #define FSL_SAI_CR4_FCOMB_SHIFT BIT(26) #define FSL_SAI_CR4_FCOMB_SOFT BIT(27) -- 2.43.0

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[PATCH AUTOSEL 6.11 03/16] ASoC: codecs: lpass-rx-macro: add missing CDC_RX_BCL_VBAT_RF_PROC2 to default regs values
by Sasha Levin 12 Oct '24

12 Oct '24

From: Alexey Klimov <alexey.klimov(a)linaro.org> [ Upstream commit e249786b2188107a7c50e7174d35f955a60988a1 ] CDC_RX_BCL_VBAT_RF_PROC1 is listed twice and its default value is 0x2a which is overwriten by its next occurence in rx_defaults[]. The second one should be missing CDC_RX_BCL_VBAT_RF_PROC2 instead and its default value is expected 0x0. Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> Link: https://patch.msgid.link/20240925043823.520218-2-alexey.klimov@linaro.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ce42749660c87..ac759f4a880d0 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -958,7 +958,7 @@ static const struct reg_default rx_defaults[] = { { CDC_RX_BCL_VBAT_PK_EST2, 0x01 }, { CDC_RX_BCL_VBAT_PK_EST3, 0x40 }, { CDC_RX_BCL_VBAT_RF_PROC1, 0x2A }, - { CDC_RX_BCL_VBAT_RF_PROC1, 0x00 }, + { CDC_RX_BCL_VBAT_RF_PROC2, 0x00 }, { CDC_RX_BCL_VBAT_TAC1, 0x00 }, { CDC_RX_BCL_VBAT_TAC2, 0x18 }, { CDC_RX_BCL_VBAT_TAC3, 0x18 }, -- 2.43.0

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[PATCH] ASoC: fsl_esai: change dev_warn to dev_dbg in irq handler
by Shengjiu Wang 11 Oct '24

11 Oct '24

Irq handler need to be executed as fast as possible, so the log in irq handler is better to use dev_dbg which needs to be enabled when debugging. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index a65f5b9935a2..0b247f16a163 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -119,10 +119,10 @@ static irqreturn_t esai_isr(int irq, void *devid) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); if (esr & ESAI_ESR_RFF_MASK) - dev_warn(&pdev->dev, "isr: Receiving overrun\n"); + dev_dbg(&pdev->dev, "isr: Receiving overrun\n"); if (esr & ESAI_ESR_TFE_MASK) - dev_warn(&pdev->dev, "isr: Transmission underrun\n"); + dev_dbg(&pdev->dev, "isr: Transmission underrun\n"); if (esr & ESAI_ESR_TLS_MASK) dev_dbg(&pdev->dev, "isr: Just transmitted the last slot\n"); -- 2.34.1

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[PATCH v1] ALSA: hda/tas2781: Add speaker id check for ASUS projects
by Baojun Xu 11 Oct '24

11 Oct '24

Add speaker id check by gpio in ACPI for ASUS projects. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 3 +++ sound/pci/hda/tas2781_hda_i2c.c | 39 ++++++++++++++++++++++++++++++--- 2 files changed, 39 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 8cd6da0480b7..f8a3dc13cbbc 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -107,6 +107,8 @@ #define TASDEVICE_CMD_DELAY 0x3 #define TASDEVICE_CMD_FIELD_W 0x4 +#define TAS2781_ASUS_ID 0x10430000 + enum audio_device { TAS2563, TAS2781, @@ -156,6 +158,7 @@ struct tasdevice_priv { struct tasdevice_rca rcabin; struct calidata cali_data; struct tasdevice_fw *fmw; + struct gpio_desc *speaker_id; struct gpio_desc *reset; struct mutex codec_lock; struct regmap *regmap; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 370d847517f9..1f71927825b2 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -615,7 +615,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct hda_codec *codec = tas_priv->codec; - int i, ret; + int i, ret, spk_id; pm_runtime_get_sync(tas_priv->dev); mutex_lock(&tas_priv->codec_lock); @@ -648,8 +648,23 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; - scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", - codec->core.subsystem_id & 0xffff); + if ((codec->core.subsystem_id & 0xffff0000) == TAS2781_ASUS_ID) { + // Speaker id need to be checked. + if (tas_priv->speaker_id) + spk_id = gpiod_get_value(tas_priv->speaker_id); + else + spk_id = 0; + if (spk_id < 0 || spk_id > 1) { + // Speaker id is not valid, use default. + dev_dbg(tas_priv->dev, "Wrong spk_id = %d\n", spk_id); + spk_id = 0; + } + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X%01d.bin", + codec->core.subsystem_id & 0xffff, spk_id); + } else { + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + } ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", @@ -787,6 +802,15 @@ static void tas2781_hda_remove(struct device *dev) tasdevice_remove(tas_hda->priv); } +static const struct acpi_gpio_params speakerid_gpios = { 0, 0, false }; +static const struct acpi_gpio_params interrupt_gpios = { 1, 0, false }; + +static const struct acpi_gpio_mapping tas2781_speaker_id_gpios[] = { + { "speakerid-gpios", &speakerid_gpios, 1 }, + { "interrupt-gpios", &interrupt_gpios, 1 }, + { } +}; + static int tas2781_hda_i2c_probe(struct i2c_client *clt) { struct tas2781_hda *tas_hda; @@ -823,6 +847,15 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (ret) return dev_err_probe(tas_hda->dev, ret, "Platform not supported\n"); + ret = devm_acpi_dev_add_driver_gpios(tas_hda->dev, + tas2781_speaker_id_gpios); + if (ret) + dev_info(tas_hda->dev, "Unable to add GPIO mapping table\n"); + + tas_hda->priv->speaker_id = devm_gpiod_get(tas_hda->dev, "speakerid", + GPIOD_IN); + if (IS_ERR(tas_hda->priv->speaker_id)) + dev_info(tas_hda->dev, "Failed to get Speaker id gpio.\n"); ret = tasdevice_init(tas_hda->priv); if (ret) -- 2.43.0

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[PATCH] ALSA: hda/realtek: tas2781: Fix ROG ALLY X audio
by Jonathan LoBue 11 Oct '24

11 Oct '24

This patch enables the TI TAS2781 amplifier SoC for the ASUS ROG ALLY X. This is a design change from the original ASUS ROG ALLY, creating the need for this patch. All other Realtek Codec settings seem to be re-used from the original ROG ALLY design (on the ROG ALLY X). This patch maintains the previous settings for the Realtek codec portion, but enables the I2C binding for the TI TAS2781 amplifier (instead of the Cirrus CS35L41 amp used on the original ASUS ROG ALLY). One other requirement must be met for audio to work on the ASUS ROG ALLY X. A proper firmware file in the correct location with a proper symlink. We had reached out to TI engineers and confirmed that the firmware found in the Windows' driver package has a GPL license. Bazzite Github is hosting this firmware file for now until proper linux-firmware upstreaming can occur. https://github.com/ublue-os/bazzite This firmware file should be placed in /usr/lib/firmware/ti/tas2781/TAS2XXX1EB3.bin with a symlink to it from /usr/lib/firmware/TAS2XXX1EB3.bin Co-developed by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Signed-off-by: Kyle Gospodnetich <me(a)kylegospodneti.ch> Co-developed by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Jan Drogehoff <sentrycraft123(a)gmail.com> Signed-off-by: Antheas Kapenekakis <lkml(a)antheas.dev> Tested-by: Richard Alvarez <alvarez.richard(a)gmail.com> Tested-by: Miles Montierth <cyber_dopamine(a)intheblackmedia.com> Signed-off-by: Jonathan LoBue <jlobue10(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..9bc39c2ee6b9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7410,6 +7410,7 @@ enum { ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_X, ALC294_FIXUP_ASUS_ALLY_PINS, ALC294_FIXUP_ASUS_ALLY_VERBS, ALC294_FIXUP_ASUS_ALLY_SPEAKER, @@ -8877,6 +8878,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS }, + [ALC294_FIXUP_ASUS_ALLY_X] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, [ALC294_FIXUP_ASUS_ALLY_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -10313,6 +10320,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally NR2301L/X", ALC294_FIXUP_ASUS_ALLY), + SND_PCI_QUIRK(0x1043, 0x1eb3, "ROG Ally X RC72LA", ALC294_FIXUP_ASUS_ALLY_X), SND_PCI_QUIRK(0x1043, 0x1863, "ASUS UX6404VI/VV", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.46.0

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