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MOTU UltraLight mk5 USB mode switching
by GitHub issues - opened 05 Nov '25

05 Nov '25

alsa-project/alsa-ucm-conf issue #635 was opened from TeeTeufel: Hi, I am currently stuck getting higher sample rates work with the MOTU UltraLight mk5. Automatic switching between 44.1 and 48khz works fine, but anything higher not. For 88.2/96 khz and 176.4/192 khz the USB mode must be switched, but nothing happens. When setting the sample rate on the audio interface itself, audio output stays silent. **System**: - Fedora 43 - Pipewire - Latest firmware on the interface What do you need from me? Could this be an issue with pipewire rather than alsa or ucm? If yes, do you maybe have any resources that could give me some hints? Consulting existing issues, forums, chatgpt and perplexity did not result in any improvement. Thank you very much for anything that could help. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/635 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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SND_MIXER_SABSTRACT_BASIC Mixer Abstraction
by Shane Kirk 05 Nov '25

05 Nov '25

Does anyone know the state of the SND_MIXER_SABSTRACT_BASIC mixer abstraction? Has this been unofficially/officially deprecated? I've been curious about it for a while and have been wanting to experiment with it, but none of the distros I've used recently seem to install support for it. Also, the smixer.conf used by this mixer abstraction seems to lean on a library that's not built as part of alsa-lib. It's been many, many years since most of this code has been touched, so I'm wondering if I'm going down a rabbit hole that's been long abandoned. Thanks! -Shane Kirk

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[PATCH v2] ALSA: hda/conexant: Fix pop noise on conexant codec
by wangdich9700＠163.com 05 Nov '25

05 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback,mute speaker DAC(0x17)immediately and restore after 20ms delay to avoid audible popping. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 65 +++++++++++++++++++++++++++++++++++++ 1 file changed, 65 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..2c16d1930a75 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,9 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + struct hda_codec *conexant_codec; + struct delayed_work change_pinctl_work; }; @@ -216,6 +219,63 @@ static void cx_remove(struct hda_codec *codec) snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void change_pinctl_worker(struct work_struct *work) +{ + struct hda_codec *codec; + struct conexant_spec *spec; + + spec = container_of(work, struct conexant_spec, change_pinctl_work.work); + codec = spec->conexant_codec; + + return mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_mute_unmute_speaker(struct hda_codec *codec, struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + mute_unmute_speaker(codec, 0x17, true); + schedule_delayed_work(&spec->change_pinctl_work, 20); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1238,10 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + INIT_DELAYED_WORK(&spec->change_pinctl_work, change_pinctl_worker); + spec->conexant_codec = codec; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1251,7 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_mute_unmute_speaker); break; } -- 2.25.1

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[PATCH] ALSA: hda/conexant: Fix pop noise on conexant codec
by wangdich9700＠163.com 05 Nov '25

05 Nov '25

From: wangdicheng <wangdich9700(a)163.com> Pop noise mitigation: When headphones are unplugged during playback,mute speaker DAC(0x17)immediately and restore after 20ms delay to avoid audible popping. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/hda/codecs/conexant.c | 64 +++++++++++++++++++++++++++++++++++++ 1 file changed, 64 insertions(+) diff --git a/sound/hda/codecs/conexant.c b/sound/hda/codecs/conexant.c index 5fcbc1312c69..641cb738901d 100644 --- a/sound/hda/codecs/conexant.c +++ b/sound/hda/codecs/conexant.c @@ -43,6 +43,8 @@ struct conexant_spec { unsigned int gpio_mute_led_mask; unsigned int gpio_mic_led_mask; bool is_cx11880_sn6140; + + struct delayed_work change_pinctl_work; }; @@ -216,6 +218,63 @@ static void cx_remove(struct hda_codec *codec) snd_hda_gen_remove(codec); } +static void mute_unmute_speaker(struct hda_codec *codec, hda_nid_t nid, bool mute) +{ + unsigned int conn_sel, dac, conn_list, gain_left, gain_right; + + conn_sel = snd_hda_codec_read(codec, nid, 0, 0xf01, 0x0); + conn_list = snd_hda_codec_read(codec, nid, 0, 0xf02, 0x0); + + dac = ((conn_list >> (conn_sel * 8)) & 0xff); + if (dac == 0) + return; + + gain_left = snd_hda_codec_read(codec, dac, 0, 0xba0, 0x0); + gain_right = snd_hda_codec_read(codec, dac, 0, 0xb80, 0x0); + + if (mute) { + gain_left |= 0x80; + gain_right |= 0x80; + } else { + gain_left &= (~(0x80)); + gain_right &= (~(0x80)); + } + + snd_hda_codec_write(codec, dac, 0, 0x3a0, gain_left); + snd_hda_codec_write(codec, dac, 0, 0x390, gain_right); + + if (mute) { + snd_hda_codec_write(codec, nid, 0, 0x707, 0); + codec_dbg(codec, "mute_speaker, set 0x%x PinCtrl to 0.\n", nid); + } else { + snd_hda_codec_write(codec, nid, 0, 0x707, 0x40); + codec_dbg(codec, "unmute_speaker, set 0x%x PinCtrl to 0x40.\n", nid); + } +} + +static void change_pinctl_worker(struct work_struct *work) +{ + struct hda_codec *codec; + struct conexant_spec *spec; + + spec = container_of(work, struct conexant_spec, change_pinctl_work.work); + codec = spec->conexant_codec; + + return mute_unmute_speaker(codec, 0x17, false); +} + +static void cx_auto_mute_unmute_speaker(struct hda_codec *codec, struct hda_jack_callback *event) +{ + struct conexant_spec *spec = codec->spec; + int phone_present; + + phone_present = snd_hda_codec_read(codec, 0x16, 0, 0xf09, 0x0); + if (!(phone_present & 0x80000000)) { + mute_unmute_speaker(codec, 0x17, true); + schedule_delayed_work(&spec->change_pinctl_work, 20); + } +} + static void cx_process_headset_plugin(struct hda_codec *codec) { unsigned int val; @@ -1178,6 +1237,10 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (!spec) return -ENOMEM; + + INIT_DELAYED_WORK(&spec->change_pinctl_work, change_pinctl_worker); + spec->conexant_codec = codec; + snd_hda_gen_spec_init(&spec->gen); codec->spec = spec; @@ -1187,6 +1250,7 @@ static int cx_probe(struct hda_codec *codec, const struct hda_device_id *id) case 0x14f11f87: spec->is_cx11880_sn6140 = true; snd_hda_jack_detect_enable_callback(codec, 0x19, cx_update_headset_mic_vref); + snd_hda_jack_detect_enable_callback(codec, 0x16, cx_auto_mute_unmute_speaker); break; } -- 2.25.1

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[PATCH] ALSA: au88x0: Fix array bounds warning in EQ drivers
by wangdich9700＠163.com 04 Nov '25

04 Nov '25

From: wangdicheng <wangdich9700(a)163.com> In file included from ../sound/pci/au88x0/au8830.c:15: In function ‘vortex_Eqlzr_SetAllBandsFromActiveCoeffSet’, ../sound/pci/au88x0/au88x0_eq.c:571:9: error: ‘vortex_EqHw_SetRightGainsTarget’ reading 2 bytes from a region of size 0 [-Werror=stringop-overread] vortex_EqHw_SetRightGainsTarget(vortex, &(eq->this130[eq->this10])); Modified the array access in vortex_Eqlzr_SetAllBandsFromActiveCoeffSet() to use pointer arithmetic instead of array indexing. This resolves a compiler warning that incorrectly flagged a buffer overread when accessing the EQ gain array. The this130 array has fixed size 20 and the index is safely within bounds, making the original code correct but confusing to static analysis. Signed-off-by: wangdicheng <wangdicheng(a)kylinos.cn> --- sound/pci/au88x0/au88x0_eq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/au88x0/au88x0_eq.c b/sound/pci/au88x0/au88x0_eq.c index 71c13100d7ef..81a63b5bb31c 100644 --- a/sound/pci/au88x0/au88x0_eq.c +++ b/sound/pci/au88x0/au88x0_eq.c @@ -568,7 +568,7 @@ static int vortex_Eqlzr_SetAllBandsFromActiveCoeffSet(vortex_t * vortex) eqlzr_t *eq = &(vortex->eq); vortex_EqHw_SetLeftGainsTarget(vortex, eq->this130); - vortex_EqHw_SetRightGainsTarget(vortex, &(eq->this130[eq->this10])); + vortex_EqHw_SetRightGainsTarget(vortex, eq->this130 + eq->this10); return 0; } -- 2.25.1

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USB-Audio: Add support for DualSense PS5 controller
by GitHub pull_request - opened 03 Nov '25

03 Nov '25

alsa-project/alsa-ucm-conf pull request #634 was opened from cristicc: The Sony DualSense wireless controller (PS5) has an internal mono speaker and a microphone, in addition to a 3.5mm jack socket for headphone output and headset microphone input. Because this is a UAC1 device, it doesn't advertise any jack detection capability. However, since kernel v6.18-rc1 (see [1]), the controller is able to report HP & MIC insert events via HID, i.e. through a dedicated input device managed by the hid-playstation driver. This is further used by a quirk in the generic USB audio driver (added in v6.17, see [2]), which creates the Headphone Jack & Headset Mic Jack controls and receives hotplug events from the HID driver via input handlers. Moreover, the device provides haptic feedback functionality, which is controlled by a pair of dedicated channels in a quadraphonic audio stream. Ensure proper support for audio switching between headphones/headset and internal speaker/microphone by adding a UCM to split the 4.0 PCM stream into 4 mono channels or a pair of stereo (FL+FR) channels, depending on the active output device/profile. [1] https://lore.kernel.org/all/20250625-dualsense-hid-jack-v2-0-596c0db14128@c… [2] https://lore.kernel.org/all/20250526-dualsense-alsa-jack-v1-0-1a821463b632@… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/634 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/634.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] selftest/alsa: correct grammar in conf_get_bool error string
by Zhang Chujun 03 Nov '25

03 Nov '25

The phrase "an bool" is grammatically incorrect; it should be "a bool". Signed-off-by: Zhang Chujun <zhangchujun(a)cmss.chinamobile.com> diff --git a/tools/testing/selftests/alsa/conf.c b/tools/testing/selftests/alsa/conf.c index 5b7c83fe87b3..317212078e36 100644 --- a/tools/testing/selftests/alsa/conf.c +++ b/tools/testing/selftests/alsa/conf.c @@ -448,7 +448,7 @@ int conf_get_bool(snd_config_t *root, const char *key1, const char *key2, int de ksft_exit_fail_msg("key '%s'.'%s' search error: %s\n", key1, key2, snd_strerror(ret)); ret = snd_config_get_bool(cfg); if (ret < 0) - ksft_exit_fail_msg("key '%s'.'%s' is not an bool\n", key1, key2); + ksft_exit_fail_msg("key '%s'.'%s' is not a bool\n", key1, key2); return !!ret; } -- 2.50.1.windows.1

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bug in snd_seq_query_subscribe_get_time_real
by Erwin P 02 Nov '25

02 Nov '25

Hello, I think there is a bug in snd_seq_query_subscribe_get_time_real(). This function returns (info->flags & SNDRV_SEQ_PORT_SUBS_TIMESTAMP), but it should be (info->flags & SNDRV_SEQ_PORT_SUBS_TIME_REAL), as it is in snd_seq_port_subscribe_get_time_real(). Regards, Erwin.

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Reg: Internal Microphone Broken on ASUS X509UA (ALC3251, SSID 1043:1961) Starting AFTER Kernel 6.8.0-31
by jo 02 Nov '25

02 Nov '25

*System/Hardware Details:* Parameter Value *Laptop Model* *ASUS VivoBook X509UA* *CPU Generation* Intel Core i3-7020U (Kaby Lake / Sunrise Point-LP) *Audio Codec* Realtek ALC3251 (Variant of ALC256) *Subsystem ID (SSID)* *1043:1961* (ASUSTeK Computer Inc.) *PCI Device ID* Intel Corporation Sunrise Point-LP HD Audio [8086:9d71] *Working Kernel Version* *Linux Kernel 6.8.0-31* and previous versions. *Broken Kernel Version* *Starts AFTER 6.8.0-31* (e.g., 6.8.0-32, 6.8.0-35, etc.). Description of the Issue The internal microphone on the ASUS X509UA is *no longer functional* on Linux kernels newer than version 6.8.0-31. The input device is detected by ALSA/PulseAudio, but the capture level remains at zero (no sound is recorded). The issue is a *clear regression*, as the microphone worked correctly on kernel 6.8.0-31 and all previous versions (e.g., in Ubuntu 22.04 LTS). It also works correctly under Chrome OS Flex, which uses a custom-patched kernel. Diagnostic Steps Performed All standard user-level and common developer workarounds have failed on the affected kernel versions: 1. *ALSA Mixer Levels:* Confirmed that the "Capture," "Mic," and "Internal Mic Boost" levels are correctly set (not muted—"OO" instead of "MM") and boosted (to 80-100%). 2. *hdajackretask (Pin Remapping):* All available pins, including 0x19, 0x1b, and "Not Connected" pins (0x12, 0x13, 0x18, 0x1a), were tested with the *"Internal mic"* function. None of these overrides corrected the issue. 3. *hda-verb Commands:* Standard GPIO-based *quirks* commonly used to fix Realtek mic issues (e.g., hda-verb 0x01/0x02/0x04) were executed and failed to activate the microphone. 4. *ALSA Model Quirk Options:* The following common and targeted model= options were added to /etc/modprobe.d/test-audio.conf (one at a time, followed by sudo alsa force-reload) and did not resolve the issue: - options snd-hda-intel model=alc256-asus-mic1 - options snd-hda-intel model=laptop-amic - options snd-hda-intel model=dell-headset-multi (This caused the microphone to disappear entirely.) 5. *Old Configuration Files:* Reverting to or applying the exact configuration files (hda.conf) that *worked* on the pre-6.8 kernels failed to activate the microphone on the post-6.8.0-31 kernels, suggesting a deeper structural change in the driver code. Request The failure of all standard workarounds indicates that the regression is due to a *missing or incorrect pin-mapping configuration* for the *1043:1961 SSID* within the snd-hda-intel driver, likely introduced in a patch after *6.8.0-31*. A patch is needed to re-introduce the correct microphone *quirk* for the *ASUS X509UA (1043:1961)* to restore functionality on the latest Linux kernel series.

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[PATCH REQUEST] ASUS ROG Zephyrus G14 GA403W - CS35L56 volume controls not exposed correctly
by John DelTorchio 02 Nov '25

02 Nov '25

After extensive debugging, the issue has been isolated to incorrect ALSA mixer control mapping. Technical Details Hardware Configuration: - Laptop Model: ASUS ROG Zephyrus G14 (GA403W) - Audio Codec: Realtek ALC285 (Subsystem ID: 0x10431024) - Amplifiers: Dual Cirrus Logic CS35L56 smart amplifiers - ALSA Card: Card 2 (HD-Audio Generic) - Driver: snd_hda_intel with cs35l56-hda module Amplifier Status: The CS35L56 amplifiers are being detected and initialized correctly: cs35l56-hda i2c-CSC3556:00-cs35l56-hda.0: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.16 cs35l56-hda i2c-CSC3556:00-cs35l56-hda.1: DSP1: Firmware: 1a00d6 vendor: 0x2 v3.11.16 Firmware files loaded: - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1.wmfw - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1-amp1.bin - cirrus/cs35l56-b0-dsp1-misc-10431024-spkid1-amp2.bin The Actual Problem Current (broken) behavior: - The volume slider controls Master Playback Volume (numid=24) on the ALC285 codec - This control only affects the analog signal level going into the CS35L56 amplifiers - Lowering this control degrades signal quality (adds muffling/distortion) without actually reducing speaker output volume - The CS35L56 amplifiers remain at a fixed gain, so perceived loudness doesn't change significantly Correct behavior should be: - The volume slider should control AMP1 Speaker Playback Volume (numid=6) and AMP2 Speaker Playback Volume (numid=12) - The codec's Master and Speaker controls should remain at 100% for optimal signal-to-noise ratio Verified working configuration: bash # Codec output (should be fixed at these values) amixer -c2 sset 'Master' 100 % amixer -c2 sset 'Speaker' 100 % # Amplifier controls (should be user-adjustable via volume slider) amixer -c2 sset 'AMP1 Speaker' 85 % amixer -c2 sset 'AMP2 Speaker' 85 % `` ` Manual testing confirms: - ✅ ` AMP1 Speaker ` and ` AMP2 Speaker ` controls properly adjust speaker loudness - ✅ Volume range 0 -100% works without distortion when codec is at 100 % - ❌ PipeWire/desktop environment uses the wrong mixer controls ### PipeWire Configuration Current sink properties show it's using ACP ( ALSA Card Profiles ) : ` ` ` api.alsa.use-acp = "true" api.alsa.path = "front:2" ` ` ` The automatically selected mixer path is choosing ` Master ` as the primary volume control instead of the CS35L56 amp controls. ### Required Fix This needs one of the following solutions: 1 . **ALSA UCM ( Use Case Manager ) profile** for device ID ` 10431024 ` : - Lock codec controls at 100 % - Expose AMP1/AMP2 Speaker controls as primary volume 2 . **Custom ALSA mixer path** at ` /usr/share/alsa/mixer/paths/ ` : - Define AMP1/AMP2 Speaker as volume elements - Set Master/Speaker to ` volume = ignore ` or ` volume-limit = 0.0 ` 3 . **Kernel quirk** in ` sound/pci/hda/patch_realtek.c ` : - Add model-specific configuration for GA403W - Similar to how other ASUS models with CS35L41/CS35L56 are handled 4 . **PipeWire/WirePlumber profile update**: - Device-specific configuration to use correct mixer elements ### Relevant Files **ALSA Controls Output:** ` `` numid = 6 ,iface = MIXER,name = 'AMP1 Speaker Playback Volume' : values = 400 | dBscale-min = -100.00dB,step = 0 .25dB,mute = 0 | Range: 0 -448 numid = 12 ,iface = MIXER,name = 'AMP2 Speaker Playback Volume' : values = 400 | dBscale-min = -100.00dB,step = 0 .25dB,mute = 0 | Range: 0 -448 numid = 24 ,iface = MIXER,name = 'Master Playback Volume' : values = 39 | dBscale-min = -65.25dB,step = 0 .75dB,mute = 0 | Range: 0 -87 Codec Information: - See full codec dump at: https://paste.centos.org/view/6c3a3640 - Full diagnostics at: https://paste.centos.org/view/c0d20bf3 Temporary Workaround Users can manually set correct levels on boot with this autostart script: ~/.config/autostart/audio-fix.desktop: desktop [Desktop Entry] Type=Application Name=Audio Fix Exec=/usr/bin/bash -c "sleep 5 && amixer -c2 sset 'Master' 100%% && amixer -c2 sset 'Speaker' 100%% && amixer -c2 sset 'AMP1 Speaker' 85%% && amixer -c2 sset 'AMP2 Speaker' 85%%" Hidden=false NoDisplay=false However, the volume slider will still not function correctly. Sent with [Proton Mail](https://proton.me/mail/home) secure email.

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