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

sof-soundwire: missing codecs/rt721+rt1320/init.conf makes UCM fail to load on cards reporting "spk:rt721+rt1320"
by GitHub issues - opened 06 Sep '26

06 Sep '26

alsa-project/alsa-ucm-conf issue #851 was opened from edrdmolina: ### Summary On a machine whose card components report `spk:rt721+rt1320`, UCM fails to open the card entirely because `sof-soundwire.conf` includes `/codecs/${var:SpeakerCodecFile}/init.conf` with `SpeakerCodecFile` still set to the *combined* string `rt721+rt1320`, and no such directory exists. `ucm2/sof-soundwire/rt721+rt1320.conf` was added by #744, but no matching `ucm2/codecs/rt721+rt1320/` was added alongside it. `codecs/rt721/` and `codecs/rt1320/` both exist; only the combined directory is missing. This is still the case on master. The result is not a degraded profile but a hard failure: PipeWire/WirePlumber falls back to a generic ACP profile that opens the wrong PCM, so the machine has no working speakers and no microphones. ### Error ``` $ alsaucm -c hw:1 list _verbs ALSA lib utils.c:373:(uc_mgr_config_load_into) [error.ucm] could not open configuration file /usr/share/alsa/ucm2/codecs/rt721+rt1320/init.conf ALSA lib parser.c:80:(uc_mgr_config_load_file) [error.ucm] failed to open file /usr/share/alsa/ucm2/codecs/rt721+rt1320/init.conf: -2 ALSA lib main.c:1805:(snd_use_case_mgr_open) [error.ucm] failed to import hw:1 use case configuration -2 alsaucm: error failed to open sound card hw:1: No such file or directory ``` ### Environment - alsa-ucm-conf 1.2.16.1 - alsa-lib 1.2.16.1 - kernel 7.1.9 (Arch), Intel Panther Lake, `sof-audio-pci-intel-ptl` - Card longname: `LENOVO-83SN-YogaPro715IPH11-LNVNB161216` - Components: `HDA:80862822,80860101,00100000 cfg-amp:2 mic:dmic cfg-mics:4 iec61937-pcm:7,6,5 hs:rt721 spk:rt721+rt1320` The kernel has no RT721 machine-driver entry for this platform, so `sof_sdw` uses the generic fallback ("Use SoundWire default machine driver with function topologies"). That path emits `spk:rt721+rt1320` as a single combined token. ### Root cause In `ucm2/conf.d/sof-soundwire/sof-soundwire.conf`, `SpeakerCodecFile` is initialised to `SpeakerCodec1` (`rt721+rt1320`), then `MultiCodecInit` is supposed to narrow it to a single codec id. For this components string the macro is: - `Haystack` = `=rt721+rt1320=rt721=dmic` - For `CodecId=rt721`: the outer `Needle "=rt721"` matches, but the inner speaker test uses `Regex "=(rt721(-sdca)?)="` against `String "=rt721+rt1320="`, which does **not** match (the `+rt1320` sits between `rt721` and the closing `=`). So `SpeakerCodecFile` is not narrowed. It *does* match the headset branch, so `HeadsetCodecFile` correctly becomes `rt721`. - For `CodecId=rt1320`: the outer `Needle "=rt1320"` does not match the haystack at all, because `rt1320` is preceded by `+`, not `=`. `SpeakerCodecFile` therefore stays `rt721+rt1320`. The subsequent include filter regex contains the alternative `rt721`, which matches as a substring search against `rt721+rt1320`, so the guard passes and: ``` True.Include.a.File "/codecs/${var:SpeakerCodecFile}/init.conf" ``` resolves to the nonexistent `/codecs/rt721+rt1320/init.conf` and aborts the whole import. For comparison, the Cirrus combination is handled by shipping the combined directory (`codecs/cs42l43-spk+cs35l56/`) rather than by narrowing, so the same code path succeeds there. ### Fix Adding an (effectively empty) `ucm2/codecs/rt721+rt1320/init.conf` is sufficient. I verified this against an unmodified tree using `ALSA_CONFIG_UCM2`, and the verb then loads correctly: ``` $ alsaucm -c hw:1 set _verb HiFi list _devices 0: HDMI3 ... 3: Headphones 4: Headset Headset Microphone 5: Mic Digital Microphone 6: Speaker Speaker $ alsaucm -c hw:1 set _verb HiFi get PlaybackPCM/Speaker PlaybackPCM/Speaker=_ucm0001.hw:sofsoundwire,2 ``` Nothing from `codecs/rt1320/init.conf` is actually required for this combination: - `codecs/rt721/init.conf` is already pulled in via the headset branch (`HeadsetCodecFile` = `rt721`), so the RT721 `BootSequence` still runs. - `sof-soundwire/rt721+rt1320.conf` drives `rt1320-1 OT23 L/R Switch` directly in its Enable/DisableSequence rather than through `${var:SpeakerMixerElem}`, so the mixer-element definitions in `codecs/rt1320/init.conf` are unused here. Including `codecs/rt1320/init.conf` from the combined file would in fact be actively wrong on this machine: the components report `cfg-amp:2`, which selects the `twoAmps` branch and remaps controls for a second amp (`rt1320-2 ...`) that does not exist — there is one RT1320 driving both channels, and `rt721+rt1320.conf` already handles that correctly via its `ControlExists` test on `rt1320-2 OT23 L Switch`. If you would prefer the narrowing approach over shipping a combined directory, the alternative is to make `MultiCodecInit` split `A+B` speaker tokens (and set `SpeakerCodecFile` to the amp), but that would change behaviour for the existing Cirrus combination too, so the combined directory looks like the lower-risk fix. ### Possibly also affected `ucm2/sof-soundwire/rt712+rt1320.conf` exists with no corresponding `ucm2/codecs/rt712+rt1320/`. If any machine reports `spk:rt712+rt1320`, it should fail in exactly the same way. I do not have that hardware to confirm. I'm happy to open a PR with the added file if that's useful. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/851 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Common/HeadphonesOnly PlaybackMixerElem "PCM" does not match Topping D10s mixer (152a:8750)
by GitHub issues - opened 05 Sep '26

05 Sep '26

alsa-project/alsa-ucm-conf issue #850 was opened from z23: USB ID `152a:8750` is mapped to `Common/HeadphonesOnly`, which hard-codes `PlaybackMixerElem "PCM"`. A Topping D10s using that same Thesycon ID has no `PCM` mixer control. PipeWire/WirePlumber then falls back to software volume and leaves the DAC Feature Unit volume untouched, so desktop volume (e.g. 70%) stacks on top of a quiet hardware fader (here 84/127 = -43 dB). ## Hardware - Device: Topping D10s (USB DAC) - `lsusb`: `152a:8750 Thesycon Systemsoftware & Consulting GmbH D10s` - iManufacturer: Topping; iProduct: D10s - Distro: Arch Linux (`alsa-ucm-conf 1.2.16.1-1`, `alsa-lib 1.2.16.1-1`) - Kernel: `7.2.3-arch1-2` - Session: PipeWire 1.6.8 + WirePlumber 0.5.17 ## Actual mixer controls There is no control named `PCM`. The playback volume is named after the product string (note the trailing space): ``` numid=2,iface=MIXER,name='D10s Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=3,iface=MIXER,name='D10s Playback Volume' ; type=INTEGER,access=rw---R--,values=1,min=0,max=127,step=0 : values=84 | dBminmax-min=-127.00dB,max=0.00dB ``` (`values=84` was the live hardware volume when the mismatch was noticed: 66% / -43.00 dB.) ## What UCM applies `alsaucm -c D10s dump text`: ``` Verb.HiFi { Comment Default Device.Headphones { Comment Headphones Values { CaptureCTL "_ucm0001.hw:D10s" PlaybackCTL "_ucm0001.hw:D10s" PlaybackMixerElem PCM PlaybackPCM "_ucm0001.hw:D10s,0" } } } ``` This comes from: - `ucm2/USB-Audio/USB-Audio.conf`: `Macro.hp-only.StringMatch "Id='152a:8750' Profile='Common/HeadphonesOnly'"` - `ucm2/USB-Audio/Common/HeadphonesOnly-HiFi.conf`: `PlaybackMixerElem "PCM"` ## Session manager effect WirePlumber logs this on every start: ``` spa.alsa: Path PCM is not a volume or mute control ``` The PipeWire sink is created (`HiFi: Headphones`) but **without** `HW_VOLUME_CTRL`. Flags were `HARDWARE DECIBEL_VOLUME LATENCY`. Desktop volume therefore only applies software gain. The USB Feature Unit stays at whatever the DAC last stored (class-compliant volume is in the device), which made output far quieter than the UI suggested. ## History `152a:8750` was originally added for a Topping DX3 Pro+ ([03fed99](https://github.com/alsa-project/alsa-ucm-conf/commit/03fed99a96da3…) That profile only set `PlaybackPCM` and did **not** name a mixer element. When it was generalized to `Common/HeadphonesOnly` ([93d015d](https://github.com/alsa-project/alsa-ucm-conf/commit/93d015d12c68a…) `PlaybackMixerElem "PCM"` was added. Several Thesycon-based Topping devices share `152a:8750` (DX3 Pro+ and D10s at least), and the D10s mixer is `D10s `, not `PCM`. ## Suggested fix `Common/HeadphonesOnly` should not assume a mixer named `PCM`. Preferred: drop `PlaybackMixerElem "PCM"` (back to the original DX3 Pro+ snippet) so ACP/PipeWire can bind the card's real playback volume control. Alternative: only set it when the control exists: ``` If.pcm { Condition { Type ControlExists Control "name='PCM Playback Volume'" } True.Value.PlaybackMixerElem "PCM" } ``` Hard-coding `PlaybackMixerElem "D10s "` would fix this one unit but is fragile (trailing space, and the USB ID is shared). ## Workaround ``` amixer -c D10s sset 'D10s ' 100% alsactl store ``` That puts the Feature Unit at 0 dB so the desktop software volume is the only attenuation. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/850 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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sys/endian.h: No such file or directory
by GitHub issues - opened 05 Sep '26

05 Sep '26

alsa-project/alsa-lib issue #522 was opened from FuwwyCodingCOwO: It seems that GNU/Hurd does not have `sys/endian.h`, this could probably be fixed by instead using `endian.h` provided by [POSIX](https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.h…. The [Hurd porting guidelines](https://www.gnu.org/software/hurd/hurd/porting/guidelines.html) seem to agree with this. I will make a pull request to go along with this. configured with: ``` configure --prefix=/usr \ --build=x86_64-pc-linux-gnu \ --host=i686-gnu \ --mandir=/usr/share/man \ --infodir=/usr/share/info \ --datadir=/usr/share \ --sysconfdir=/etc \ --localstatedir=/var/lib \ --datarootdir=/usr/share \ --disable-dependency-tracking \ --disable-silent-rules \ --disable-static \ --docdir=/usr/share/doc/alsa-lib-1.2.16.1 \ --htmldir=/usr/share/doc/alsa-lib-1.2.16.1/html \ --with-sysroot=/usr/i686-gnu \ --libdir=/usr/lib \ --disable-maintainer-mode \ --disable-resmgr \ --enable-aload \ --enable-rawmidi \ --enable-seq \ --enable-shared \ --enable-thread-safety \ --disable-python \ --without-debug ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/522 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2 0/2] ASoC: codecs: nau8360: Add DSP fallback and PEQ band initialization
by Neo Chang 04 Sep '26

04 Sep '26

This patch series adds a fallback mechanism for the NAU8360 DSP during firmware load failure and ensures proper initialization for PEQ bands. It includes the following changes: - Fallback to bypass mode automatically when DSP firmware fails to load, ensuring basic audio playback remains functional. - Set a default PEQ band number during DSP initialization to prevent invalid state or uninitialized parameters before user configuration is applied. Changes in v2: - Use snd_soc_component_update_bits() to prevent register overwriting. (Reported by Sashiko AI) - Fix missing bit shift for default PEQ band. (Reported by Sashiko AI) Neo Chang (2): ASoC: codecs: nau8360: Set DAC path to bypass mode when DSP firmware load fails ASoC: codecs: nau8360: Add default PEQ band number for DSP init sound/soc/codecs/nau8360.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.25.1

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[PATCH 5.15.y] ALSA: seq: Fix port lock leak in deliver_to_subscribers()
by Tzung-Bi Shih 04 Sep '26

04 Sep '26

Commit 46397622a3fa ("ALSA: seq: Add UMP support") mistakenly removed the `snd_seq_port_unlock(src_port)` call in deliver_to_subscribers(). As a result, every time an event is delivered to a subscriber, the source port's `use_lock` atomic counter is increased but never decreased. The leak prevents ALSA sequencer ports from closing cleanly. When the client process exits or destroys the port, port_delete() hangs forever at snd_use_lock_sync(). The process becomes stuck in an unkillable D state, triggering the following warning: ALSA: seq_lock: waiting [1 left] in sound/core/seq/seq_ports.c:250 Restore the missing `snd_seq_port_unlock(src_port)`. Note that commit 177ccf811df4 ("ALSA: seq: Support MIDI 2.0 UMP Endpoint port") incidentally resolved this issue upstream when it refactored the locking logic into a wrapper function. However, since the commit introduces a new feature and might not be suitable for stable release branches, this targeted fix is necessary. Fixes: 6389f2c13531 ("ALSA: seq: Add UMP support") Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- sound/core/seq/seq_clientmgr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index f2feeb97765c..bd2d7a9fab5b 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -733,6 +733,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, read_unlock(&grp->list_lock); else up_read(&grp->list_mutex); + snd_seq_port_unlock(src_port); memcpy(event, &event_saved, saved_size); return (result < 0) ? result : num_ev; } -- 2.55.0.979.g7e5102b832-goog

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[PATCH 6.1.y] ALSA: seq: Fix port lock leak in deliver_to_subscribers()
by Tzung-Bi Shih 04 Sep '26

04 Sep '26

Commit 46397622a3fa ("ALSA: seq: Add UMP support") mistakenly removed the `snd_seq_port_unlock(src_port)` call in deliver_to_subscribers(). As a result, every time an event is delivered to a subscriber, the source port's `use_lock` atomic counter is increased but never decreased. The leak prevents ALSA sequencer ports from closing cleanly. When the client process exits or destroys the port, port_delete() hangs forever at snd_use_lock_sync(). The process becomes stuck in an unkillable D state, triggering the following warning: ALSA: seq_lock: waiting [1 left] in sound/core/seq/seq_ports.c:250 Restore the missing `snd_seq_port_unlock(src_port)`. Note that commit 177ccf811df4 ("ALSA: seq: Support MIDI 2.0 UMP Endpoint port") incidentally resolved this issue upstream when it refactored the locking logic into a wrapper function. However, since the commit introduces a new feature and might not be suitable for stable release branches, this targeted fix is necessary. Fixes: d7e2ce72833b ("ALSA: seq: Add UMP support") Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- sound/core/seq/seq_clientmgr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 9b82fccd959c..37930d74e394 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -733,6 +733,7 @@ static int deliver_to_subscribers(struct snd_seq_client *client, read_unlock(&grp->list_lock); else up_read(&grp->list_mutex); + snd_seq_port_unlock(src_port); memcpy(event, &event_saved, saved_size); return (result < 0) ? result : num_ev; } -- 2.55.0.979.g7e5102b832-goog

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ucm2: sof-soundwire: add Senary SN624x SDCA support
by GitHub pull_request - opened 04 Sep '26

04 Sep '26

alsa-project/alsa-ucm-conf pull request #849 was opened from wangxiaohu2026: ## Summary Add sof-soundwire UCM for Senary SN624x SDCA (jack / speaker / dmic). Kernel `CardComponents` example: `hs:sn624x-sdca spk:sn624x mic:sn624x-dmic` Files added under `ucm2/sof-soundwire/`: - `sn624x.conf` — Speaker - `sn624x-sdca.conf` — Headphones / Headset Mic - `sn624x-dmic.conf` — DMIC Note: existing `sn6242*` UCM entries cannot substitute these names. ## Hardware - Platform: Intel Panther Lake / Wildcat Lake (`sof-soundwire`) - Codec: Senary SN624x SDCA SoundWire (e.g. part_id `0x6244`) - Kernel: ASoC SN624x series in broonie/sound `for-7.4` ## Test - HiFi profile exposes Speaker / Dmic / Headphones / Microphones - `alsaucm -c sof-soundwire dump text` shows the corresponding devices ## alsa-info See attached `alsa-info-sn624x-20260825.txt`. [alsa-info-sn624x-20260825.txt](https://github.com/user-attachments/files/31… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/849 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/849.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/2] ASoC: codecs: nau8360: Add DSP fallback and PEQ band initialization
by Neo Chang 04 Sep '26

04 Sep '26

This patch series adds a fallback mechanism for the NAU8360 DSP during firmware load failure and ensures proper initialization for PEQ bands. It includes the following changes: - Fallback to bypass mode automatically when DSP firmware fails to load, ensuring basic audio playback remains functional. - Set a default PEQ band number during DSP initialization to prevent invalid state or uninitialized parameters before user configuration is applied. Neo Chang (2): ASoC: codecs: nau8360: Set DAC path to bypass mode when DSP firmware load fails ASoC: codecs: nau8360: Add default PEQ band number for DSP init sound/soc/codecs/nau8360.c | 3 +++ 1 file changed, 3 insertions(+) -- 2.25.1

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conf: USB-Audio: Add four more devices to the IEC958 blacklist
by GitHub pull_request - edited 04 Sep '26

04 Sep '26

alsa-project/alsa-lib pull request #518 was edited from tonymitchell: None of these devices has an S/PDIF connector, but each shows up in GNOME Settings with a non-existent digital profile. Add them to the list to prevent them from being opened. Razer BlackShark V3 Xbox (USB ID 1532:0a58) Logitech BRIO (USB ID 046d:085e, capture only) Plantronics Blackwire 5220 Series (USB ID 047f:c053) Poly Voyager Focus 2 UC / BT700 (USB ID 047f:02e6) Signed-off-by: Tony Mitchell <tony.mitchell(a)live.ca> Request URL : https://github.com/alsa-project/alsa-lib/pull/518 Patch URL : https://github.com/alsa-project/alsa-lib/pull/518.patch Repository URL: https://github.com/alsa-project/alsa-lib

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conf: USB-Audio: Add three more devices to the IEC958 blacklist
by GitHub pull_request - edited 04 Sep '26

04 Sep '26

alsa-project/alsa-lib pull request #518 was edited from tonymitchell: None of these devices has an S/PDIF connector, but each shows up in GNOME Settings with a non-existent digital profile. Add them to the list to prevent them from being opened. Razer BlackShark V3 Xbox (USB ID 1532:0a58) Logitech BRIO (USB ID 046d:085e, capture only) Plantronics Blackwire 5220 Series (USB ID 047f:c053) Poly Voyager Focus 2 UC / BT700 (USB ID 047f:02e6) Signed-off-by: Tony Mitchell <tony.mitchell(a)live.ca> Request URL : https://github.com/alsa-project/alsa-lib/pull/518 Patch URL : https://github.com/alsa-project/alsa-lib/pull/518.patch Repository URL: https://github.com/alsa-project/alsa-lib

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