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

[PATCH] ASoC: amd: acp: soc-acpi: add is_device_rt712_vb() helper
by Vijendar Mukunda 13 Jan '26

13 Jan '26

Add a filter to skip the RT172 VB configuration if a SmartMic Function is not found in the SDCA descriptors. If the ACPI information is incorrect this can only be quirked further with DMI information. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/Kconfig | 6 +++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 50 ++++++++++++++++++++ sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c | 42 ++++++++++++++++ sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h | 14 ++++++ 5 files changed, 114 insertions(+) create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index c2a60bc80ee6..ed2fa055f7f6 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -15,8 +15,14 @@ config SND_SOC_AMD_ACP_COMMON config SND_SOC_ACPI_AMD_MATCH tristate + select SND_SOC_ACPI_AMD_SDCA_QUIRKS select SND_SOC_ACPI if ACPI +config SND_SOC_ACPI_AMD_SDCA_QUIRKS + tristate + depends on ACPI + depends on SND_SOC_SDCA + if SND_SOC_AMD_ACP_COMMON config SND_SOC_AMD_ACP_PDM diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 08220b9a3802..81d23aded348 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -27,6 +27,7 @@ snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o amd-acp70-acpi-match.o snd-acp-sdw-mach-y := acp-sdw-mach-common.o snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o snd-acp-sdw-legacy-mach-y += acp-sdw-legacy-mach.o +snd-soc-acpi-amd-sdca-quirks-y += soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o +obj-$(CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS) += snd-soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_AMD_SOUNDWIRE_ACPI) += snd-amd-sdw-acpi.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 871b4f054a84..fa39f18578ca 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -7,6 +7,7 @@ */ #include <sound/soc-acpi.h> +#include "soc-acpi-amd-sdca-quirks.h" #include "../mach-config.h" static const struct snd_soc_acpi_endpoint single_endpoint = { @@ -44,6 +45,39 @@ static const struct snd_soc_acpi_endpoint spk_3_endpoint = { .group_id = 1 }; +static const struct snd_soc_acpi_endpoint jack_amp_g1_dmic_endpoints[] = { + /* Jack Endpoint */ + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + /* Amp Endpoint, work as spk_l_endpoint */ + { + .num = 1, + .aggregated = 1, + .group_position = 0, + .group_id = 1, + }, + /* DMIC Endpoint */ + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device rt712_vb_1_group1_adr[] = { + { + .adr = 0x000130025D071201ull, + .num_endpoints = ARRAY_SIZE(jack_amp_g1_dmic_endpoints), + .endpoints = jack_amp_g1_dmic_endpoints, + .name_prefix = "rt712" + } +}; + static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = { { .adr = 0x000030025D071101ull, @@ -254,6 +288,15 @@ static const struct snd_soc_acpi_link_adr acp70_cs35l56x4_l1[] = { {} }; +static const struct snd_soc_acpi_link_adr acp70_alc712_vb_l1[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt712_vb_1_group1_adr), + .adr_d = rt712_vb_1_group1_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr acp70_rt722_only[] = { { .mask = BIT(0), @@ -308,6 +351,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { .links = acp70_cs35l56x4_l1, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(1), + .links = acp70_alc712_vb_l1, + .machine_check = snd_soc_acpi_amd_sdca_is_device_rt712_vb, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sdw_machines); @@ -327,3 +376,4 @@ EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sof_sdw_machines); MODULE_DESCRIPTION("AMD ACP7.0 & ACP7.1 tables and support for ACPI enumeration"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_IMPORT_NS("SND_SOC_ACPI_AMD_SDCA_QUIRKS"); diff --git a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c new file mode 100644 index 000000000000..63bf9e3c0ae1 --- /dev/null +++ b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-acpi-amd-sdca-quirks.c - tables and support for SDCA quirks + * + * Copyright(c) 2025 Advanced Micro Devices, Inc. All rights reserved. + * + */ + +#include <linux/soundwire/sdw_amd.h> +#include <sound/sdca.h> +#include <sound/soc-acpi.h> +#include "soc-acpi-amd-sdca-quirks.h" + +/* + * Pretend machine quirk. The argument type is not the traditional + * 'struct snd_soc_acpi_mach' pointer but instead the sdw_amd_ctx + * which contains the peripheral information required for the + * SoundWire/SDCA filter on the SMART_MIC setup and interface + * revision. When the return value is false, the entry in the + * 'snd_soc_acpi_mach' table needs to be skipped. + */ +bool snd_soc_acpi_amd_sdca_is_device_rt712_vb(void *arg) +{ + struct sdw_amd_ctx *ctx = arg; + int i; + + if (!ctx) + return false; + + for (i = 0; i < ctx->peripherals->num_peripherals; i++) { + if (sdca_device_quirk_match(ctx->peripherals->array[i], + SDCA_QUIRKS_RT712_VB)) + return true; + } + + return false; +} +EXPORT_SYMBOL_NS(snd_soc_acpi_amd_sdca_is_device_rt712_vb, "SND_SOC_ACPI_AMD_SDCA_QUIRKS"); + +MODULE_DESCRIPTION("ASoC ACPI AMD SDCA quirks"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS("SND_SOC_SDCA"); diff --git a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h new file mode 100644 index 000000000000..7e345a236da1 --- /dev/null +++ b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * soc-acpi-amd-sdca-quirks.h - tables and support for SDCA quirks + * + * Copyright(c) 2025 Advanced Micro Devices, Inc. All rights reserved. + * + */ + +#ifndef _SND_SOC_ACPI_AMD_SDCA_QUIRKS +#define _SND_SOC_ACPI_AMD_SDCA_QUIRKS + +bool snd_soc_acpi_amd_sdca_is_device_rt712_vb(void *arg); + +#endif -- 2.45.2

5 5

UCM HiFi profile fails on hardware with Master volume instead of Headphone volume controls (ASUS ZenBook UX425EA)
by GitHub issues - opened 13 Jan '26

13 Jan '26

alsa-project/alsa-ucm-conf issue #687 was opened from akincano: ### Hardware Information - Laptop: ASUS ZenBook UX425EA_UX425EA - Audio Controller: Intel Tiger Lake-LP Smart Sound Technology - Audio Controller (PCI ID: 0x8086:0xa0c8) - Codec: Realtek ALC294 (Vendor ID: 0x10ec0294, Subsystem ID: 0x10431cee) - Kernel: 6.18.5.arch1-1 - Distribution: Arch Linux ### Software Versions ```bash alsa-ucm-conf 1.2.15.2-1 sof-firmware 2025.12-1 linux 6.18.5.arch1-1 ``` ### Problem Description The SOF HiFi profile fails to create speaker/headphone audio devices on this hardware. WirePlumber reports: ```bash spa.alsa: No sink and source at HiFi: Headphones ``` The HiFi profile shows as available: no and only HDMI outputs are exposed, with no analog audio devices. ### Root Cause **This is a UCM configuration issue**, not a kernel driver issue. The problem is that /usr/share/alsa/ucm2/HDA/HiFi-analog.conf checks for ALSA controls that don't exist on this hardware. **ALSA controls exposed by this hardware (identical for both SOF and HDA):** ```bash numid=3,iface=CARD,name='Speaker Phantom Jack' numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' [HDMI/IEC958 controls omitted for brevity] ``` **Controls the UCM config expects but don't exist:** - Headphone Playback Volume ❌ (only Master Playback Volume exists) - Headphone Playback Switch ❌ (only Master Playback Switch exists) - Headphone Jack ❌ (only Speaker Phantom Jack exists) **UCM config logic that fails:** In /usr/share/alsa/ucm2/HDA/HiFi-analog.conf: ```bash If.hpvol { Condition { Type ControlExists Control "name='Headphone Playback Volume'" } False.Define.hpvol "" } If.hp { Condition { Type String Empty "${var:hpvol}" } False.SectionDevice."Headphones" { # ... creates headphone device } } ``` Because Headphone Playback Volume doesn't exist, hpvol is set to empty string, and the If.hp condition prevents the headphone device from being created. Similarly, the jack detection logic checks for: ```bash Define.hpjack "Headphone Jack" If.hpjack { Condition { Type ControlExists Control "iface=CARD,name='Headphone Mic Jack'" } # ... falls back to checking 'Front Headphone Jack' } ``` But never checks for Speaker Phantom Jack, which is what this hardware exposes. ### Working Configuration (Legacy HDA Driver) With the legacy snd_hda_intel driver (forced via blacklisting SOF), PulseAudio's built-in profile logic correctly handles this hardware: **Working profile:** output:analog-stereo+input:analog-stereo **Exposed ports:** - analog-output-speaker: Speakers (uses Master Playback Volume/Switch) - analog-output-headphones: Headphones (uses Master Playback Volume/Switch) - Automatic jack detection via Speaker Phantom Jack The legacy HDA driver works because PulseAudio has fallback logic that handles hardware with only Master volume controls, but SOF's UCM-based configuration doesn't have this fallback. ### Proposed Solution The UCM configuration needs to be updated to handle hardware that only exposes Master Playback Volume/Switch instead of dedicated Headphone controls. **Option 1: Add fallback logic to HiFi-analog.conf** Check for Master Playback Volume when Headphone Playback Volume doesn't exist: ```bash If.hpvol { Condition { Type ControlExists Control "name='Headphone Playback Volume'" } False.If.master { Condition { Type ControlExists Control "name='Master Playback Volume'" } True.Define.hpvol "Master" } } ``` Similarly for jack detection, check Speaker Phantom Jack: ```bash If.hpjack { Condition { Type ControlExists Control "iface=CARD,name='Headphone Mic Jack'" } True.Define.hpjack "Headphone Mic Jack" False.If.front { Condition { Type ControlExists Control "iface=CARD,name='Front Headphone Jack'" } True.Define.hpjack "Front Headphone Jack" False.If.speaker { Condition { Type ControlExists Control "iface=CARD,name='Speaker Phantom Jack'" } True.Define.hpjack "Speaker Phantom Jack" } } } ``` **Option 2: Create a specific profile for this hardware** Create a UCM profile specifically for Tiger Lake systems with Realtek ALC294 that only expose Master controls. **Option 3: Update SOF topology** If this is a SOF topology issue, update the topology to expose proper Headphone controls instead of just Master. ### Additional Debug Information **Pin 0x21 configuration (headphone jack):** ```bash Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Pincap 0x0001001c: OUT HP EAPD Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 ``` The pin is capable of jack detection (EAPD Detect in capabilities) but is misconfigured as "N/A" with "NO_PRESENCE". However, the jack detection still works via the Speaker Phantom Jack ALSA control. **Full codec information:** Available upon request **Full WirePlumber logs:** Available upon request ### Impact This affects all ASUS ZenBook UX425EA users (and potentially other laptops with similar hardware configurations) running SOF. Users must either: 1. Use pro-audio profile (no automatic headphone/speaker switching) 2. Blacklist SOF and force legacy HDA driver (not a long-term solution) ### Related Issues This was initially reported in #686 which was closed as a kernel issue, but further investigation revealed it's actually a UCM configuration problem. The kernel driver is exposing the ALSA controls correctly; the UCM config just doesn't know how to handle this hardware configuration. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/687 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA 1.2.8 release
by Jaroslav Kysela 13 Jan '26

13 Jan '26

Hello all, new ALSA userspace packages were released. You may download them from the ALSA website http://www.alsa-project.org or directly: HTTP: https://www.alsa-project.org/files/pub FTP: ftp://ftp.alsa-project.org/pub Released packages: alsa-lib alsa-utils alsa-ucm-conf tinycompress Full list of changes: https://www.alsa-project.org/wiki/Changes_v1.2.7.2_v1.2.8 The fingerprint of the public signing key is: F04D F507 37AC 1A88 4C4B 3D71 8380 596D A6E5 9C91 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 12

snd_seq_drain_output change breaks existing code.
by GitHub issues - edited 13 Jan '26

13 Jan '26

alsa-project/alsa-lib issue #493 was edited from Hoosier-Beagler: The little gem shown below has broken a lot of code. Previously, no positive results were returned and software relied on -EAGAIN to detect remaining data. That result is no longer returned if some data was processed. This has broken the old and stable version of my application written in C(++) which is no longer being maintained. I have ported my app to python but this change has also broken public python wrappers for the alsa library. In [alsa-python / pyalsa / alsaseq.c](https://github.com/alsa-project/alsa-python/blob/master/pyalsa/a… below line 2800 and also [python-alsa-midi / alsa_midi / client.py](https://github.com/Jajcus/python-alsa-midi/blob/main/alsa_midi/c… at line 626 and possibly the sub-classes as well (lines 1679, 1871). None of these wrappers expected a valid return value and do not pass along said value returned from the c library call. Consequently, there is no longer an indication that the output has not been fully drained. ``` --- alsa-lib-1.2.14/src/seq/seq.c 2025-04-14 11:42:04.000000000 -0500 +++ alsa-lib-1.2.15.2/src/seq/seq.c 2026-01-08 08:27:57.000000000 -0600 @@ -4445,21 +4445,22 @@ int snd_seq_drain_output(snd_seq_t *seq) { ssize_t result, processed = 0; assert(seq); while (seq->obufused > 0) { result = seq->ops->write(seq, seq->obuf, seq->obufused); if (result < 0) { - if (result == -EAGAIN && processed) + if (result == -EAGAIN && processed > 0) return seq->obufused; return result; } if ((size_t)result < seq->obufused) memmove(seq->obuf, seq->obuf + result, seq->obufused - result); seq->obufused -= result; + processed += result; } return 0; } /** * \brief extract the first event in output buffer * \param seq sequencer handle ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/493 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_seq_drain_output change breaks existing code.
by GitHub issues - opened 13 Jan '26

13 Jan '26

alsa-project/alsa-lib issue #493 was opened from Hoosier-Beagler: The little gem shown below has broken a lot of code. Previously, no positive results were returned and software relied on -EAGAIN to detect remaining data. That result is no longer returned if some data was processed. This has broken the old and stable version of my application written in C(++) which is no longer being maintained. I have ported my app to python but this change has also broken public python wrappers for the alsa library. In [alsa-python / pyalsa / alsaseq.c](https://github.com/alsa-project/alsa-python/blob/master/pyalsa/a… below line 2800 and also [python-alsa-midi / alsa_midi / client.py](https://github.com/Jajcus/python-alsa-midi/blob/main/alsa_midi/c… at line 626 and possibly the sub-classes as well (lines 1679, 1871). None of these wrappers expected a valid return value and do not pass along said value returned from the c library call. ``` --- alsa-lib-1.2.14/src/seq/seq.c 2025-04-14 11:42:04.000000000 -0500 +++ alsa-lib-1.2.15.2/src/seq/seq.c 2026-01-08 08:27:57.000000000 -0600 @@ -4445,21 +4445,22 @@ int snd_seq_drain_output(snd_seq_t *seq) { ssize_t result, processed = 0; assert(seq); while (seq->obufused > 0) { result = seq->ops->write(seq, seq->obuf, seq->obufused); if (result < 0) { - if (result == -EAGAIN && processed) + if (result == -EAGAIN && processed > 0) return seq->obufused; return result; } if ((size_t)result < seq->obufused) memmove(seq->obuf, seq->obuf + result, seq->obufused - result); seq->obufused -= result; + processed += result; } return 0; } /** * \brief extract the first event in output buffer * \param seq sequencer handle ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/493 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SOF HiFi profile fails to detect headphone jack on ASUS ZenBook UX425EA (ALC294)
by GitHub issues - opened 12 Jan '26

12 Jan '26

alsa-project/alsa-ucm-conf issue #686 was opened from akincano: Component: Sound(ALSA) → PCI Description: Hardware: Laptop: ASUS ZenBook UX425EA_UX425EA Audio Controller: Intel Tiger Lake-LP Smart Sound Technology Audio Controller (PCI ID: 0x8086:0xa0c8) Codec: Realtek ALC294 (Vendor ID: 0x10ec0294, Subsystem ID: 0x10431cee) Kernel: 6.18.4-arch1-1 Distribution: Arch Linux ALSA components: HDA:80862812,80860101,00100000 HDA:10ec0294,10431cee,00100004 cfg-dmics:2 Problem: When using the SOF (Sound Open Firmware) driver, the HiFi profile fails to create audio sinks/sources for the headphone jack. WirePlumber logs show: spa.alsa: No sink and source at HiFi: Headphones The HiFi profile shows as available: no and the active profile defaults to off. Attempting to switch to HiFi results in only HDMI outputs being available, with no analog speaker or headphone outputs. The headphone jack (pin 0x21) is misconfigured: Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x02 0x03* The key issues are: Pin Default 0x411111f0 - configured as "N/A" instead of headphone jack Misc = NO_PRESENCE - no jack detection enabled Pin-ctls: 0x00 - pin not configured for output Unsolicited: tag=00, enabled=0 - jack detection events disabled This causes PipeWire/WirePlumber to only create HDMI outputs in the HiFi profile, with no analog audio devices. Expected Behavior: The HiFi profile should detect and expose: Internal speakers Headphone jack with jack detection Automatic switching between speakers and headphones when plugging/unplugging Digital microphone Current Workaround: Blacklisting SOF and forcing legacy HDA driver works perfectly: ```bash # /etc/modprobe.d/alsa-base.conf blacklist snd_sof_pci_intel_tgl blacklist snd_sof_intel_hda_common blacklist snd_sof options snd-intel-dspcfg dsp_driver=1 ``` With legacy HDA driver, the same pin 0x21 is properly configured: ```bash Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Control: name="Headphone Playback Switch", index=0, device=0 ``` The `output:analog-stereo` profile works perfectly with automatic jack detection and switching. **Reproduction:** 1. Boot ASUS ZenBook UX425EA with default SOF driver (no blacklisting) 2. Check `pactl list cards` - shows: - HiFi profile: `available: no` - Active profile: `off` - Only HDMI ports listed under HiFi profile, no analog outputs 3. Check `journalctl --user -u wireplumber` - see error: `spa.alsa: No sink and source at HiFi: Headphones` 4. Attempting to use HiFi or even pro-audio profile results in no working analog audio **SOF Firmware/Topology Information:** ```bash sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware file: intel/sof/sof-tgl.ri sof-audio-pci-intel-tgl 0000:00:1f.3: Topology file: intel/sof-tplg/sof-hda-generic-2ch.tplg sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:1 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:1 sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 ``` **Card Configuration (SOF):** ``` Name: alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_generic api.alsa.card.name = "sof-hda-dsp" alsa.long_card_name = "ASUSTeKCOMPUTERINC.-ZenBookUX425EA_UX425EA-1.0-UX425EA" alsa.driver_name = "snd_soc_skl_hda_dsp" alsa.mixer_name = "Realtek ALC294" Profiles: off: Off (sinks: 0, sources: 0, priority: 0, available: yes) HiFi: Play HiFi quality Music (sinks: 3, sources: 1, priority: 9900, available: no) pro-audio: Pro Audio (sinks: 5, sources: 3, priority: 1, available: yes) Active Profile: off HiFi profile ports (no analog outputs): [Out] HDMI3: HDMI / DisplayPort 3 Output [Out] HDMI2: HDMI / DisplayPort 2 Output [Out] HDMI1: HDMI / DisplayPort 1 Output [In] Mic1: Digital Microphone Request: The ASUS ZenBook UX425EA needs either: A UCM configuration that properly maps pin 0x21 as a headphone jack with presence detection, OR A kernel quirk to set the correct pin configuration for this laptop model, OR An updated SOF topology file that includes the analog outputs The hardware clearly supports headphone jack detection (the legacy HDA driver proves this), but SOF is not utilizing it correctly. Impact: This affects all ASUS ZenBook UX425EA users running modern kernels with SOF as the default driver. Users either have no audio or must manually blacklist SOF to use the legacy HDA driver. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/686 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Can't resample into ALSA plugin
by Mark Hills 11 Jan '26

11 Jan '26

I've written an ALSA plugin (ioplug) of the form below. My plugin offers only S16, 48000Hz. I expected a "plug" device would take the application's request and resample to 48000Hz, but instead it fails: $ aplay -D plug:mytest cd-audio.wav Playing WAVE 'cd-audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:1456: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: 125000 PERIOD_SIZE: (5512 5513) PERIOD_BYTES: (22048 22052) PERIODS: (3 5) BUFFER_TIME: 500000 BUFFER_SIZE: 22050 BUFFER_BYTES: 88200 TICK_TIME: 0 It doesn't call .hw_params, so the negotiation has failed based on snd_pcm_ioplug_set_param_*. Without the plug: $ aplay -D mytest cd-audio.wav Playing WAVE 'cd-audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Warning: rate is not accurate (requested = 44100Hz, got = 48000Hz) please, try the plug plugin (-Dplug:mytest) hw_params: format=S16_LE channels=2 rate=48000 period_size=6000 buffer_size=2400 Modify the plugin 48000Hz to be 44100Hz and negotiation passes, so it suggests the rate is the problem (not some other parameter like period size) I tried a variety of things including directing aplay to use specific period and buffer size. Thanks --- /* * Code extracts, error checks removed for brevity */ static snd_pcm_ioplug_callback_t callback = { .close = _close, .hw_params = hw_params, /* prints them */ .start = start, .stop = stop, .pointer = pointer, .transfer = transfer, .delay = delay, .pause = _pause, .resume = resume, .prepare = prepare, }; static snd_pcm_ioplug_t initialise = { .version = SND_PCM_IOPLUG_VERSION, .name = "My test", .callback = &callback, }; SND_PCM_PLUGIN_DEFINE_FUNC(mytest) { struct state *s; snd_pcm_ioplug_t *io; unsigned access = SND_PCM_ACCESS_RW_INTERLEAVED; unsigned format = SND_PCM_FORMAT_S16; unsigned rate = 48000; s = malloc(sizeof(*s)); ... s->io = initialise; s->io.private_data = s; io = &s->io; snd_pcm_ioplug_create(io, name, stream, mode); snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_ACCESS, 1, &access); snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_FORMAT, 1, &format); snd_pcm_ioplug_set_param_list(io, SND_PCM_IOPLUG_HW_RATE, 1, &rate); *pcmp = io->pcm; return 0; } -- Mark

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Laptop built-in speakers not detected after c8b5541
by GitHub issues - opened 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf issue #685 was opened from Xeltalliv: Updating from v1.2.15.1 to v1.2.15.2 broke sound on ASUS Zenbook 14 UM425UA laptop. It has no external headphone jacks. I narrowed it down the cause to the commit https://github.com/alsa-project/alsa-ucm-conf/commit/c8b55418ff36036fc96073… With v1.2.15.1 `wpctl status` shows: ``` Audio ├─ Devices: │ 48. Renoir/Cezanne HDMI/DP Audio Controller [alsa] │ 49. Ryzen HD Audio Controller [alsa] │ ├─ Sinks: │ * 56. Ryzen HD Audio Controller Headphones [vol: 0.40] │ ├─ Sources: │ * 57. Ryzen HD Audio Controller Stereo Microphone [vol: 1.00] │ 58. Ryzen HD Audio Controller Digital Microphone [vol: 1.00] │ ├─ Filters: │ └─ Streams: ``` With v1.2.15.2 `wpctl status` shows: ``` Audio ├─ Devices: │ 48. Renoir/Cezanne HDMI/DP Audio Controller [alsa] │ 49. Ryzen HD Audio Controller [alsa] │ ├─ Sinks: │ * 34. Dummy Output [vol: 1.00] │ ├─ Sources: │ * 56. Ryzen HD Audio Controller Digital Microphone [vol: 0.97] │ ├─ Filters: │ └─ Streams: ``` Not sure if any of the following helps: In both cases `dmesg | grep -iE 'snd|alsa|pipewire'` shows: ``` [ 8.235764] snd_rn_pci_acp3x 0000:04:00.5: enabling device (0000 -> 0002) [ 8.383393] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client [ 8.404286] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 8.434270] snd_hda_codec_alc269 hdaudioC1D0: autoconfig for ALC294: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 8.434280] snd_hda_codec_alc269 hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.434284] snd_hda_codec_alc269 hdaudioC1D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.434288] snd_hda_codec_alc269 hdaudioC1D0: mono: mono_out=0x0 [ 8.434290] snd_hda_codec_alc269 hdaudioC1D0: inputs: ``` `aplay -l` shows: ``` **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic_1 [HD-Audio Generic], device 0: ALC294 Analog [ALC294 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `amixer -c 1 scontrols` shows: ``` Simple mixer control 'Master',0 Simple mixer control 'Mic ACP LED',0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/685 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for UR24C firmware version channel differences, fix UR22C
by GitHub pull_request - edited 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf pull request #684 was edited from jwlademann: Steinberg UR24C has a similar change to the number of capture channels for version 3 firmware that UR22C has. During implementation I found an issue with the condition used for UR22C which I fixed also. See: https://github.com/alsa-project/alsa-ucm-conf/pull/554 and https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add support for UR24C firmware version channel differences, fix UR22C
by GitHub pull_request - edited 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf pull request #684 was edited from jwlademann: Steinberg UR24C has a similar change to the number of capture channels for version 3 firmware that UR22C. During implementation I found an issue with the condition used for UR22C which I fixed also. See: https://github.com/alsa-project/alsa-ucm-conf/pull/554 and https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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