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

ALSA 1.2.8 release
by Jaroslav Kysela 14 Jun '26

14 Jun '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.

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[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 13 Mar '26

13 Mar '26

I own a MOTU Midi Express XT 8x8 interface which is connected through the parallel port. Recently I upgraded my system and the new motherboard does not have an ISA parallel port anymore... So I bought a PCI parallel port extension card - hoping that it will work with the ALSA driver (mtpav). So I changed the interrupt an port address in modules.conf but the driver is no longer recognized as an ALSA driver even though it loads without errors. As the device still perfectly fits my needs I would like to modify the driver to get it working again. Can anyone give me advice what should be changed in the drivers code to make it work on a pci port? Thanks Holger

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[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 13 Mar '26

13 Mar '26

The patch ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fd2df3aeafa4b4cc468d58e147e0822967034b71 Mon Sep 17 00:00:00 2001 From: Annaliese McDermond <nh6z(a)nh6z.net> Date: Thu, 21 Mar 2019 17:58:46 -0700 Subject: [PATCH] ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF Model and manage codec clock input as a component in the Core Clock Framework. This should allow us to do some more complex clock management and power control. Also, some of the on-board chip clocks can be exposed to the outside, and this change will make those clocks easier to consume by other parts of the kernel. Signed-off-by: Annaliese McDermond <nh6z(a)nh6z.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4-clk.c | 34 ++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 18 +++++++++++---- 2 files changed, 47 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4-clk.c b/sound/soc/codecs/tlv320aic32x4-clk.c index 5e495fc8d931..cded85009c8c 100644 --- a/sound/soc/codecs/tlv320aic32x4-clk.c +++ b/sound/soc/codecs/tlv320aic32x4-clk.c @@ -265,6 +265,30 @@ static const struct clk_ops aic32x4_pll_ops = { .get_parent = clk_aic32x4_pll_get_parent, }; +static int clk_aic32x4_codec_clkin_set_parent(struct clk_hw *hw, u8 index) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + + return regmap_update_bits(mux->regmap, + AIC32X4_CLKMUX, + AIC32X4_CODEC_CLKIN_MASK, index << AIC32X4_CODEC_CLKIN_SHIFT); +} + +static u8 clk_aic32x4_codec_clkin_get_parent(struct clk_hw *hw) +{ + struct clk_aic32x4 *mux = to_clk_aic32x4(hw); + unsigned int val; + + regmap_read(mux->regmap, AIC32X4_CLKMUX, &val); + + return (val & AIC32X4_CODEC_CLKIN_MASK) >> AIC32X4_CODEC_CLKIN_SHIFT; +} + +static const struct clk_ops aic32x4_codec_clkin_ops = { + .set_parent = clk_aic32x4_codec_clkin_set_parent, + .get_parent = clk_aic32x4_codec_clkin_get_parent, +}; + static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { { .name = "pll", @@ -274,6 +298,14 @@ static struct aic32x4_clkdesc aic32x4_clkdesc_array[] = { .ops = &aic32x4_pll_ops, .reg = 0, }, + { + .name = "codec_clkin", + .parent_names = + (const char *[]) { "mclk", "bclk", "gpio", "pll" }, + .num_parents = 4, + .ops = &aic32x4_codec_clkin_ops, + .reg = 0, + }, }; static struct clk *aic32x4_register_clk(struct device *dev, @@ -314,6 +346,8 @@ int aic32x4_register_clocks(struct device *dev, const char *mclk_name) */ aic32x4_clkdesc_array[0].parent_names = (const char* []) { mclk_name, "bclk", "gpio", "din" }; + aic32x4_clkdesc_array[1].parent_names = + (const char *[]) { mclk_name, "bclk", "gpio", "pll" }; for (i = 0; i < ARRAY_SIZE(aic32x4_clkdesc_array); ++i) aic32x4_register_clk(dev, &aic32x4_clkdesc_array[i]); diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 7cf8c7cedfe1..5496e4e080f4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -735,12 +735,9 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, aic32x4_set_processing_blocks(component, aic32x4_divs[i].r_block, aic32x4_divs[i].p_block); - /* PLL as CODEC_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_CLKMUX, - AIC32X4_CODEC_CLKIN_MASK, - AIC32X4_CODEC_CLKIN_PLL << AIC32X4_CODEC_CLKIN_SHIFT); /* DAC_MOD_CLK as BDIV_CLKIN */ - snd_soc_component_update_bits(component, AIC32X4_IFACE3, AIC32X4_BDIVCLK_MASK, + snd_soc_component_update_bits(component, AIC32X4_IFACE3, + AIC32X4_BDIVCLK_MASK, AIC32X4_DACMOD2BCLK << AIC32X4_BDIVCLK_SHIFT); /* NDAC divider value */ @@ -987,6 +984,15 @@ static int aic32x4_component_probe(struct snd_soc_component *component) { struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u32 tmp_reg; + int ret; + + struct clk_bulk_data clocks[] = { + { .id = "codec_clkin" }, + }; + + ret = devm_clk_bulk_get(component->dev, ARRAY_SIZE(clocks), clocks); + if (ret) + return ret; if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); @@ -999,6 +1005,8 @@ static int aic32x4_component_probe(struct snd_soc_component *component) if (aic32x4->setup) aic32x4_setup_gpios(component); + clk_set_parent(clocks[0].clk, clocks[1].clk); + /* Power platform configuration */ if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) { snd_soc_component_write(component, AIC32X4_MICBIAS, -- 2.20.1

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Lenovo Yoga 7 2-in-1 16AKP10 volume control bug
by Sandro Gähler 08 Mar '26

08 Mar '26

Hello I would like to report a bug for the brand new Lenovo Yoga 7 2-in-1 16AKP10. It has the same sound chip and also the same volume control bug as several older models. I thus assume that the same workaround, written by Philipp Jungkamp and later by Pascal Gross, will also work for this model. There are several lines with laptops that apply this workaround, for example this line in linux/sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN) Could you please add this line for my model? SND_PCI_QUIRK(0x????, 0x????, "Lenovo Yoga 7 2-in-1 16AKP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), Please advise on how to determine the first two parameters. Thanks a lot for your great work and support. Best regards, --Sandro Gähler

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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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ucm2: sof-soundwire: fix a case MultiCodec1 string is in HeadsetCodec1 string
by GitHub pull_request - opened 10 Feb '26

10 Feb '26

alsa-project/alsa-ucm-conf pull request #653 was opened from shumingfan: ucm2: sof-soundwire: fix a case MultiCodec1 string is in HeadsetCodec1 string The component string is Components : 'HDA:80862822,80860101,00100000 cfg-amp:1 iec61937-pcm:7,6,5 hs:rt713-sdca mic:rt713 spk:rt1320' Therefore, HeadsetCodec1 string is 'rt713-sdca' and MultiCodec1 string is 'rt713'. The Regex and String should be swapped and check again. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/653 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/653.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Can't resample into ALSA plugin
by Mark Hills 05 Feb '26

05 Feb '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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[PATCH] MAINTAINERS: update Tzung-Bi's email address
by Tzung-Bi Shih 05 Feb '26

05 Feb '26

Use kernel.org account instead. Signed-off-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- .../devicetree/bindings/sound/google,cros-ec-codec.yaml | 2 +- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 2 +- MAINTAINERS | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml index c3e9f3485449..dea293f403d9 100644 --- a/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml +++ b/Documentation/devicetree/bindings/sound/google,cros-ec-codec.yaml @@ -8,7 +8,7 @@ title: Audio codec controlled by ChromeOS EC maintainers: - Cheng-Yi Chiang <cychiang(a)chromium.org> - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Google's ChromeOS EC codec is a digital mic codec provided by the diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 1d73204451b1..ea7d4900ee4a 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Realtek rt1015p codec devicetree bindings maintainers: - - Tzung-Bi Shih <tzungbi(a)google.com> + - Tzung-Bi Shih <tzungbi(a)kernel.org> description: | Rt1015p is a rt1015 variant which does not support I2C and diff --git a/MAINTAINERS b/MAINTAINERS index cf0f18502372..f9749afc0b9d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4906,7 +4906,7 @@ F: drivers/platform/chrome/ CHROMEOS EC CODEC DRIVER M: Cheng-Yi Chiang <cychiang(a)chromium.org> -M: Tzung-Bi Shih <tzungbi(a)google.com> +M: Tzung-Bi Shih <tzungbi(a)kernel.org> R: Guenter Roeck <groeck(a)chromium.org> L: chrome-platform(a)lists.linux.dev S: Maintained -- 2.38.0.135.g90850a2211-goog

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[GIT PULL] ASoC fixes for v6.19-rc3
by Mark Brown 05 Feb '26

05 Feb '26

The following changes since commit 56d953a8d0da5e53c2594edde23465ec49385b1c: ASoC: Intel: mtl-match: Add 6-amp matches for CS35L56 (2025-12-19 13:25:16 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.19-rc3 for you to fetch changes up to 3572ad82201baeed840f1959787e0e1281e7f937: ASoC: drop bogus container_of() error handling (2025-12-22 22:13:03 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.19 A small collection of driver speciifc fixes, mostly relatively minor, plus an adjustment to the topology file naming for Intel systems with Bluetooth required for disambiguation. ---------------------------------------------------------------- Bard Liao (1): ASoC: SOF: Intel: add -bt tplg suffix if BT is present Chen-Yu Tsai (1): ASoC: sun4i-spdif: Add missing kerneldoc fields for sun4i_spdif_quirks Johan Hovold (4): ASoC: codecs: pm4125: drop bogus container_of() error handling ASoC: codecs: wcd937x: drop bogus container_of() error handling ASoC: soc_sdw_utils: drop bogus container_of() error handling ASoC: codecs: pm4125: clean up bind() device reference handling Mac Chiang (1): ASoC: Intel: sof_sdw: shift SSP BT mask bits. Mark Brown (1): ASoC: drop bogus container_of() error handling sound/soc/codecs/pm4125.c | 17 ++--------------- sound/soc/codecs/wcd937x.c | 5 ----- sound/soc/intel/boards/sof_sdw_common.h | 6 +++--- sound/soc/sdw_utils/soc_sdw_utils.c | 4 ---- sound/soc/sof/intel/hda.c | 14 +++++++++++++- sound/soc/sunxi/sun4i-spdif.c | 2 ++ 6 files changed, 20 insertions(+), 28 deletions(-)

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Persistent bug
by 58 6 24 Jan '26

24 Jan '26

Hello ALSA developers, I would like to report an issue with internal speakers not working on a MSI Thin A15 laptop using Fedora Workstation. After extensive debugging, this appears to be related to a missing or incorrect quirk for the Realtek ALC256 codec on this machine. Hardware - Laptop: MSI Thin A15 - Audio codec: Realtek ALC256 - Subsystem ID: 0x14621437 - Other audio devices: - NVIDIA HDA (HDMI/DP) - ATI R6xx HDMI Software - Distribution: Fedora Workstation - Kernel: Fedora stock kernel - Audio stack: ALSA + PipeWire - Desktop: GNOME Incident description The internal speakers stopped working after the following event: - I was running *Arknights: Endfield* via Heroic Games Launcher - At the same time, I launched *Counter-Strike 2* from Steam - The system froze completely - I performed a hard power-off After rebooting, the internal speakers never worked again. Current behavior - Headphones output works correctly - HDMI / DP audio works correctly - External speakers connected to the headphone jack work, but only in *mono* - Internal speakers are listed in GNOME and pavucontrol as "Internal Speaker", but: - Connectivity field is empty - Jack detection is not present - Volume slider is present but audio is always silent Debugging performed - Confirmed ALSA, PipeWire, and GNOME audio stack are working correctly - Tested multiple snd_hda_intel model= parameters (no effect) - Inspected /proc/asound/card*/codec#*: - Node 0x14 is declared as *Internal Speaker*, with EAPD present - Node 0x21 (Headphones) works correctly - Tried hda-verb commands: - SET_EAPD_BTLENABLE - SET_AMP_GAIN_MUTE - SET_COEF_INDEX / SET_PROC_COEF - All commands return value=0, but the internal speaker never activates - Node 0x14 does not respond to amplifier or routing commands - HDMI/DP digital pins function normally Observations - The hardware declares the internal speaker pin, but there appears to be no functional routing - No existing model= option activates the speaker - Behavior matches other reports of MSI Thin laptops with ALC256 - This suggests a missing or incorrect ALSA quirk for this subsystem ID Expected behavior Internal speakers should function as a standard output device, as they do under other operating systems. Conclusion This does not appear to be a user-space or configuration issue. The problem is likely: - Incorrect or incomplete ACPI/DSDT information from MSI, and/or - Missing ALSA quirk for Realtek ALC256 on subsystem ID 0x14621437 If needed, I can provide full codec dumps, alsa-info.sh output, or test patches. Thank you for your time and for maintaining ALSA. Best regards, O. Leonardo Planes

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