February 2026 - Alsa-devel - mailman.alsa-project.org

USB Audio: AlphaTheta / Pioneer DJM-S11 (2b73:0037) fails UAC2 clock enumeration
by Federico Beffa 05 Mar '26

05 Mar '26

Hello ALSA developers, I am investigating Linux support for the AlphaTheta / Pioneer DJM-S11 mixer and would like to report the current behavior and ask for guidance on possible driver or quirk development. Device: Vendor ID: 2b73 (AlphaTheta Corporation) Product ID: 0037 (DJM-S11) Firmware: bcdDevice 1.05 USB speed: High Speed (480 Mbps) Environment: NixOS 25.11 Kernel: Linux 6.12.62 Driver: snd-usb-audio Summary of the issue: The DJM-S11 enumerates correctly on USB and exposes AudioControl, AudioStreaming, MIDI, and HID interfaces. However, ALSA fails to create any PCM devices, and the mixer does not appear in aplay -l. Kernel log shows the following errors during enumeration: parse_audio_format_rates_v2v3(): unable to retrieve number of sample rates (clock 1) Quirk or no altset; falling back to MIDI 1.0 As a result, only MIDI and HID interfaces are exposed; audio streaming is disabled. Descriptor observations: - The device advertises itself as USB Audio Class 2.0 (bcdADC 2.00, InterfaceProtocol 32). - A CLOCK_SOURCE descriptor is present (ClockID 1) with Clock Frequency Control marked as read/write. - No valid clock rate list can be retrieved by the driver (GET_RANGE appears to fail). - Audio streaming endpoints do exist: - Playback: 14 channels, 24-bit PCM, isochronous async OUT - Capture: 10 channels, 24-bit PCM, isochronous async IN (implicit feedback) From inspection of lsusb -vv, this appears to be a case where the device advertises UAC2 but requires vendor-specific clock initialization before audio streaming becomes usable. Without that, snd-usb-audio correctly refuses to expose PCM devices. I can provide the full lsusb -vv output and additional logs if helpful. Questions: 1. Is this a known pattern for newer Pioneer/AlphaTheta mixers, and has similar hardware been handled via ALSA quirks in the past? 2. Would it be acceptable to experiment with a quirk that forces a fixed sample rate (e.g. 48 kHz) and bypasses clock queries, for development/testing purposes? 3. Or does this class of device realistically require vendor-specific control handling beyond what snd-usb-audio quirks can support? I am interested in starting development work on this (initially out-of-tree) and would appreciate any guidance on the most appropriate direction. Thank you for your time and for ALSA’s continued work. Best regards, Federico

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USB-Audio: Motu: M6-HiFi.conf - m6 has only 4 outputs, not 6
by GitHub pull_request - opened 05 Mar '26

05 Mar '26

alsa-project/alsa-ucm-conf pull request #701 was opened from damex: hi, here is the fix for https://github.com/alsa-project/alsa-ucm-conf/issues/700 motu m6 has 6 input and 4 outputs. https://motu.com/products/m-series/m6 'The M6 is a 6-in/4-out USB-C audio interface...' in config it is listed as 6 outputs and that throws ALSA UCM error. adjusting configuration on working system makes error go away. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/701 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/701.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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MOTU M6 reports [ALSA UCM error] but seems to be working fine.
by GitHub issues - opened 02 Mar '26

02 Mar '26

alsa-project/alsa-ucm-conf issue #700 was opened from damex: hi, i did fresh fedora 43 install and noticed that my MOTU m6 reports ALSA UCM error on all inputs and outputs but those inputs and outputs work just fine. <img width="659" height="381" alt="Image" src="https://github.com/user-attachments/assets/cdce7782-7581-4c05-b501-2e22d592…" /> <img width="819" height="206" alt="Image" src="https://github.com/user-attachments/assets/2290dd4b-f0af-4f75-a58b-6b5a056e…" /> in some software it is difficult to choose correct input due to presence of this error in device name. i did find https://github.com/alsa-project/alsa-ucm-conf/issues/552 this report to be similar to mine. attaching full alsa info output here (it was requested in that report) [alsa-info.txt](https://github.com/user-attachments/files/25109878/alsa-info… please let me know if i you need any more information from me. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/700 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v7.0-rc1
by Mark Brown 26 Feb '26

26 Feb '26

The following changes since commit 6de23f81a5e08be8fbf5e8d7e9febc72a5b5f27f: Linux 7.0-rc1 (2026-02-22 13:18:59 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v7.0-rc1 for you to fetch changes up to 32fc4168fa56f6301d858c778a3d712774e9657e: ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table (2026-02-23 14:52:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v7.0 One quirk and a fix for handling of exotic peripherals on cs42l43. ---------------------------------------------------------------- Azamat Almazbek uulu (1): ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table Charles Keepax (1): ASoC: cs42l43: Report insert for exotic peripherals sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 1 + 2 files changed, 8 insertions(+)

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[PATCH] ALSA: hda/ca0132: Set HP/Speaker auto-detect default from headphone pin verb
by Matt DeVillier 26 Feb '26

26 Feb '26

From 79d54b6b2b2410ac3c169329b93607042b137693 Mon Sep 17 00:00:00 2001 From: Matt DeVillier <matt.devillier(a)gmail.com> Date: Wed, 25 Feb 2026 11:25:18 -0600 Subject: [PATCH] ALSA: hda/ca0132: Set HP/Speaker auto-detect default from headphone pin verb HP/Speaker auto-detect (VNID_HP_ASEL) has been off by default for every CA0132 device since the driver was added in 2012. vnode_lswitch is always initialized to 0 in ca0132_init_chip(), and no quirk or other code path enables it. As a result, headphone jack detection works only after the user manually turns on "HP/Speaker Auto Detect" in alsamixer, which is not obvious on laptops with combo jacks (e.g. Google Link, Alienware). Change the default to follow the headphone pin config: if the pin verb has presence detect enabled (no AC_DEFCFG_MISC_NO_PRESENCE) and the codec supports it (AC_PINCAP_PRES_DETECT), enable HP_ASEL by default. This lets firmware (coreboot, UEFI, etc.) express whether the headphone jack supports insertion detection. Devices with combo jacks can default to auto-detect; devices with fixed/no jack leave it off. Signed-off-by: Matt DeVillier <matt.devillier(a)gmail.com> --- sound/hda/codecs/ca0132.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/codecs/ca0132.c b/sound/hda/codecs/ca0132.c index dd054aedd501..ca6cb57347ea 100644 --- a/sound/hda/codecs/ca0132.c +++ b/sound/hda/codecs/ca0132.c @@ -9816,6 +9816,15 @@ static void ca0132_config(struct hda_codec *codec) spec->dig_in = 0x09; break; } + + /* Default HP/Speaker auto-detect from headphone pin verb: enable if the + * pin config indicates presence detect (not AC_DEFCFG_MISC_NO_PRESENCE). + */ + if (spec->unsol_tag_hp && + (snd_hda_query_pin_caps(codec, spec->unsol_tag_hp) & AC_PINCAP_PRES_DETECT) && + !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, spec->unsol_tag_hp)) & + AC_DEFCFG_MISC_NO_PRESENCE)) + spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID] = 1; } static int ca0132_prepare_verbs(struct hda_codec *codec) -- 2.51.0

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Why does MMAP access ignore start_threshold?
by Mark Hills 25 Feb '26

25 Feb '26

It seems that start_threshold (swparams) is not honoured if MMAP access is used. Whereas I expect that once a total of enough bytes have been committed the device automatically starts the consuming of the buffer, as it does with snd_pcm_writei(). Is this a bug, or intentional design? It does not seem to be the subject of any documentation -- yet for many years I seem to be be working around it. Perhaps I am consistently doing something wrong, here's a test case below. -- Mark #include <alsa/asoundlib.h> #define always(r) _always(r, __LINE__) #define FORMAT SND_PCM_FORMAT_S16 #define ACCESS SND_PCM_ACCESS_MMAP_INTERLEAVED static void _always(int r, unsigned line) { if (r < 0) { fprintf(stderr, "%d: %s\n", line, snd_strerror(r)); abort(); } } static void set_hw(snd_pcm_t *pcm, unsigned rate, unsigned channels, snd_pcm_uframes_t buffer_size, snd_pcm_uframes_t period_size) { int r; snd_pcm_hw_params_t *hw; snd_pcm_hw_params_alloca(&hw); r = snd_pcm_hw_params_any(pcm, hw); always(r); r = snd_pcm_hw_params_set_access(pcm, hw, ACCESS); always(r); r = snd_pcm_hw_params_set_format(pcm, hw, FORMAT); always(r); r = snd_pcm_hw_params_set_rate(pcm, hw, rate, 0); always(r); r = snd_pcm_hw_params_set_channels(pcm, hw, channels); always(r); r = snd_pcm_hw_params_set_buffer_size(pcm, hw, buffer_size); always(r); r = snd_pcm_hw_params_set_period_size(pcm, hw, period_size, 0); always(r); r = snd_pcm_hw_params(pcm, hw); always(r); } static void set_sw(snd_pcm_t *pcm, snd_pcm_uframes_t start_threshold, snd_pcm_uframes_t avail_min) { int r; snd_pcm_sw_params_t *sw; snd_pcm_sw_params_alloca(&sw); r = snd_pcm_sw_params_current(pcm, sw); always(r); r = snd_pcm_sw_params_set_start_threshold(pcm, sw, start_threshold); always(r); r = snd_pcm_sw_params_set_avail_min(pcm, sw, avail_min); always(r); r = snd_pcm_sw_params(pcm, sw); always(r); } snd_pcm_uframes_t do_mmap(snd_pcm_t *pcm) { int r; snd_pcm_uframes_t frames, offset; const snd_pcm_channel_area_t *area; r = snd_pcm_wait(pcm, 1000); always(r); r = snd_pcm_mmap_begin(pcm, &area, &offset, &frames); always(r); r = snd_pcm_mmap_commit(pcm, offset, frames); always(r); return frames; } snd_pcm_uframes_t do_write(snd_pcm_t *pcm, unsigned channels, snd_pcm_uframes_t frames) { int r; void *buf; size_t len = channels * snd_pcm_format_size(FORMAT, frames); buf = alloca(len); bzero(buf, len); r = snd_pcm_writei(pcm, buf, frames); always(r); return frames; } int main(int argc, char *argv[]) { int r; snd_pcm_t *pcm; r = snd_pcm_open(&pcm, argv[1], SND_PCM_STREAM_PLAYBACK, 0); always(r); set_hw(pcm, 48000, 2, 2048, 512); set_sw(pcm, 1024, 512); r = snd_pcm_prepare(pcm); always(r); for (;;) { snd_pcm_uframes_t frames; if (ACCESS == SND_PCM_ACCESS_MMAP_INTERLEAVED) frames = do_mmap(pcm); else frames = do_write(pcm, 2, 512); fprintf(stderr, "written %lu\n", frames); } r = snd_pcm_close(pcm); always(r); return 0; }

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ALC255 internal microphone not working on Acer Aspire VN7-592G (subsystem 1025:1039)
by Valerio e. 24 Feb '26

24 Feb '26

Hello, I am reporting an issue with the internal microphone on my Acer Aspire V15 Nitro Black Edition (VN7-592G) running Linux Mint with kernel 6.17.0-14-generic. Hardware details: - Laptop: Acer Aspire VN7-592G - Codec: Realtek ALC255 - Vendor ID: 0x10ec0255 - Subsystem ID: 0x10251039 - Kernel: 6.17.0-14-generic - Audio server: PipeWire 1.0.5 Issue: The internal microphone is not working. The device is recognized by ALSA (arecord -l shows ALC255 Analog), and recording produces a file with non-zero amplitude (maximum ~0.17), but the signal is only static noise regardless of any input. The microphone works correctly under Windows. All input pin defaults in /proc/asound/card0/codec#0 show 0x411111f0 ([N/A]), suggesting the BIOS does not correctly declare the internal microphone pin. Specifically, pin 0x19 (labeled "Internal Mic Boost Volume" by the driver) appears to be the internal microphone but its Pin Default is 0x411111f0 instead of a valid configuration. I have tried the following without success: - Various model= parameters (alc255-acer, headset-mic, laptop-amic, dell-headset-multi) - Manual hda-verb commands to activate pin 0x19 - Firmware patch via /lib/firmware/hda-jack-retask.fw with pincfg 0x19 0x01a1913c - hdajackretask GUI tool The alsa-info output is attached. Thank you for your time and help. Best regards, Valerio

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Re: [RFC PATCH] ASoC: Intel: sof_es8336: Add DMI quirk for Huawei BOD-WXX9
by Honey Tydik 23 Feb '26

23 Feb '26

Gentle ping on this RFC. I'm still investigating why GPIO control from DAPM event doesn't work on Huawei BOD-WXX9, while the same code works from probe context. Any feedback or suggestions would be appreciated. Thanks, Tagir вс, 1 февр. 2026 г. в 15:19, Tagir Garaev <tgaraev653(a)gmail.com>: > Add DMI entry for Huawei Matebook D (BOD-WXX9) with HEADPHONE_GPIO > and DMIC quirks. > > This device has ES8336 codec with: > - GPIO 16 (headphone-enable) for headphone amplifier control > - GPIO 17 (speakers-enable) for speaker amplifier control > - GPIO 269 for jack detection IRQ > - 2-channel DMIC > > Hardware investigation shows that both GPIO 16 and 17 are required > for proper audio routing, as headphones and speakers share the same > physical output (HPOL/HPOR) and are separated only via amplifier > enable signals. > > RFC: Seeking advice on GPIO control issue: > > GPIO values change in driver (gpiod_get_value() shows logical value > changes) but not physically (debugfs gpio shows no change). The same > gpiod_set_value_cansleep() calls work correctly in probe context with > msleep(), but fail when called from DAPM event callbacks. > > Context information from diagnostics: > - in_atomic=0, in_interrupt=0, irqs_disabled=0 > - Process context: pipewire > - GPIO 17 (speakers): changes in driver, no physical change > - GPIO 16 (headphone): changes in driver, no physical change > > In Windows, audio switching works without visible GPIO changes, > suggesting possible ACPI/firmware involvement. > > Any suggestions on how to properly control these GPIOs from DAPM > events would be appreciated. > Signed-off-by: Tagir Garaev <tgaraev653(a)gmail.com> > --- > sound/soc/intel/boards/sof_es8336.c | 9 +++++++++ > 1 file changed, 9 insertions(+) > > diff --git a/sound/soc/intel/boards/sof_es8336.c > b/sound/soc/intel/boards/sof_es8336.c > index 7dd02d700d7e..8cedfce99edc 100644 > --- a/sound/soc/intel/boards/sof_es8336.c > +++ b/sound/soc/intel/boards/sof_es8336.c > @@ -363,6 +363,15 @@ static int sof_es8336_quirk_cb(const struct > dmi_system_id *id) > * if the topology file is modified as well. > */ > static const struct dmi_system_id sof_es8336_quirk_table[] = { > + { > + .callback = sof_es8336_quirk_cb, > + .matches = { > + DMI_MATCH(DMI_SYS_VENDOR, "HUAWEI"), > + DMI_MATCH(DMI_PRODUCT_NAME, "BOD-WXX9"), > + }, > + .driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO | > + SOF_ES8336_ENABLE_DMIC) > + }, > { > .callback = sof_es8336_quirk_cb, > .matches = { > -- > 2.52.0 > >

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ucm2: sof-soundwire: Add support for Cirrus Logic CS47L47
by GitHub pull_request - opened 23 Feb '26

23 Feb '26

alsa-project/alsa-ucm-conf pull request #709 was opened from rfvirgil: Add the sof-soundwire and codec files for the CS47L47. This is a SDCA codec with headphone/headset and digital mic. The configuration is based on the CS42L45 files. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/709 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/709.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add Audient EVO4
by GitHub pull_request - edited 23 Feb '26

23 Feb '26

alsa-project/alsa-ucm-conf pull request #708 was edited from motolav: This is a 4in/4out device with 1 analog stereo out and 2 analog mono in. The hardware mixed loopback interface takes up the remaining 4 channels. Without this UCM the device is configured as 4.0 surround for in and out. I copied the way Windows configures the device except for the stereo mic mix as I couldn't get it to work somehow(might have been a wiremix issue). The loopback interface requires software to configure what channels are mixed into its recording interface that's only available on Windows and macOS but there is an incomplete script to control the device's hardware registers at github.com/vijay-prema/audient-evo-linux-tools. Closes: #303 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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