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[GIT PULL] ASoC fixes for v6.16-rc5
by Mark Brown 17 Jul '25

17 Jul '25

The following changes since commit 3b3312f28ee2d9c386602f8521e419cfc69f4823: ASoC: cs35l56: probe() should fail if the device ID is not recognized (2025-07-03 12:22:27 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.16-rc5 for you to fetch changes up to a609bd74b8680dba62c44f7e6d00d381ddb2d3c0: ASoC: Intel: avs: Fix NULL ptr deref on rmmod (2025-07-08 14:54:40 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.16 Two patches here, one quirk for an AMD system and a fix for an issue on remove of the AVS driver. ---------------------------------------------------------------- Alexandru Andries (1): ASoC: amd: yc: add DMI quirk for ASUS M6501RM Amadeusz Sławiński (1): ASoC: Intel: avs: Fix NULL ptr deref on rmmod sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/intel/avs/pcm.c | 4 +++- 2 files changed, 10 insertions(+), 1 deletion(-)

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[PATCH] ASoC: qcom: x1e80100: Support boards with two speakers
by Krzysztof Kozlowski 17 Jul '25

17 Jul '25

Some Qualcomm X1E laptops have only two speakers. Regardless whether this sound card driver is suitable for them (we could re-use one for some older SoC), we should set reasonable channel map depending on the number of channels, not always 4-speaker setup. This change is necessary for bringing audio support on Lenovo Thinkpad T14s with Qualcomm X1E78100 and only two speakers. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/x1e80100.c | 40 ++++++++++++++++++++++++++++++++++----- 1 file changed, 35 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c index 898b5c26bf1e..8eb57fc12f0d 100644 --- a/sound/soc/qcom/x1e80100.c +++ b/sound/soc/qcom/x1e80100.c @@ -95,23 +95,53 @@ static int x1e80100_snd_hw_params(struct snd_pcm_substream *substream, return qcom_snd_sdw_hw_params(substream, params, &data->sruntime[cpu_dai->id]); } +static int x1e80100_snd_hw_map_channels(unsigned int *ch_map, int num) +{ + switch (num) { + case 1: + ch_map[0] = PCM_CHANNEL_FC; + break; + case 2: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + break; + case 3: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_FR; + ch_map[2] = PCM_CHANNEL_FC; + break; + case 4: + ch_map[0] = PCM_CHANNEL_FL; + ch_map[1] = PCM_CHANNEL_LB; + ch_map[2] = PCM_CHANNEL_FR; + ch_map[3] = PCM_CHANNEL_RB; + break; + default: + return -EINVAL; + } + + return 0; +} + static int x1e80100_snd_prepare(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct x1e80100_snd_data *data = snd_soc_card_get_drvdata(rtd->card); struct sdw_stream_runtime *sruntime = data->sruntime[cpu_dai->id]; - const unsigned int rx_slot[4] = { PCM_CHANNEL_FL, - PCM_CHANNEL_LB, - PCM_CHANNEL_FR, - PCM_CHANNEL_RB }; + unsigned int channels = substream->runtime->channels; + unsigned int rx_slot[4]; int ret; switch (cpu_dai->id) { case WSA_CODEC_DMA_RX_0: case WSA_CODEC_DMA_RX_1: + ret = x1e80100_snd_hw_map_channels(rx_slot, channels); + if (ret) + return ret; + ret = snd_soc_dai_set_channel_map(cpu_dai, 0, NULL, - ARRAY_SIZE(rx_slot), rx_slot); + channels, rx_slot); if (ret) return ret; break; -- 2.43.0

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qcom-lpass: RX HPH Mode is set to a ULP mode which causes distortion
by GitHub issues - opened 17 Jul '25

17 Jul '25

alsa-project/alsa-ucm-conf issue #595 was opened from valpackett: The headphone enable sequence for all Qualcomm devices sets `CLS_H_ULP` mode: https://github.com/alsa-project/alsa-ucm-conf/blob/1b69ade9b6d7ee37a87c08b1… I was trying to figure out why I was getting horrendous distortion on the headphone jack of my Dell Latitude 7455 laptop above a certain relatively-quiet volume level. After lots of fiddling and searching (thanks, [XDA forum audio modders](https://xdaforums.com/t/mod-audio-biquads-utilizing-qualcomms-audi… from several years ago!) I figured it out. Changing that to `CLS_AB_HIFI` has completely fixed the distortion! Should the default be changed to a HIFI mode for everyone, i.e. right in that file? ping: @Srinivas-Kandagatla Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/595 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Alsa speaker-test dose not generate 20kHz Sine Wav
by Harig, George William 16 Jul '25

16 Jul '25

Hello Alsa-Dev, I hope that I'm writing to the proper place to report a problem, please advise if not... I'm not able to produce 20kHz sine wav with speaker-test 1.2.13 Running on Tinycore 16.0 . Below is what I submitted to Tinycore Team, and they suggested that I contact you with my findings. The hardware I'm using is Winsystems E3825 dual-core, 1.33GHz SATA 2.0 channel mSATA socket (shared with miniPCIe) 2 GB of Soldered Down DDR3 Link to product: https://winsystems.com/product/ppm-c407/ Tinycore Team; My problem today is that speaker-test does not produce a 20kHz sine wave on Tinycore 16.0 to the line out. speaker-test does produce something just not a 20kHz sine wave. Code: speaker-test -t sine -f 20000 -X -l 0 However, using the exact same hardware running Ubuntu 20.24 or 22.04 the following code works and a 20kHz sine wave is produced. The difference being that the alsa versions are different between the 3 Operating Systems. Tinycore 16.0 Alsa version is 1.2.13 - failure Ubuntu 20.04 Alsa version is 1.2.2 - success Ubuntu 22.04 Alsa version is 1.2.6 - success Through my testing, starting at 8000 Hz, and incrementing by 1000 Hz, speaker-test does produce a sine wave, until I went over 15000 Hz then no more sine wave. :) :) Code: speaker-test -t sine -f 8000 -X -l 0 speaker-test -t sine -f 15000 -X -l 0 :( Code: speaker-test -t sine -f 16000 -X -l 0 The work around for the time being is to configure speaker-test to play a 20kHz wav file, this dose work and produces a beautiful sine wav :) Code: speaker-test -t wav -f /opt/20kHz.wav -l 0 Now for the people who will say why on earth do you want to produce a sound that no one can hear, and my reply is "Exactly". I send the 20kHz sine wav across the audio line as a way to supervise that the audio line is fully functioning. The second the 20kHz tone stops we are notified and send someone out to fix the problem, whether it be hardware or someone accidentality cuts the wire, we want to know. Basically the 20 kHz sine wave does not interferer with paging in the system. My first thoughts on why the 20 kHz sine is not produced, is because of missing driver. But I made sure that every sound driver that was loaded on Ubuntu was loaded on Tinycore. Code: sudo modprobe snd_seq_midi snd_seq_midi_event snd_intel_dspcfg I also loaded AlsaPlugins.tcz just incase, but that did not make a difference. Code: tceload -i alsa-plugins.tcz The last thing I tried after was to set the format sample -F parameter to 96000 Code: speaker-test -t sine -f 20000 -X -l 0 -F 96000 At first when I did this I thought I was so smart. A sine wave appeared, however I did not have a speaker hooked up, but when the speaker got plugged in I heard a high pitched noise. So the parameter affected the output but not the desired output that I need. Another thought that pass through my mind is could this be a configuration setting during the build of alsa that need to be turned on. After all 15000 Hz does work. This is slightly related to a previous post linked below. https://forum.tinycorelinux.net/index.php/topic,27673.0.html Thank you Any help is greatly appreciated gharig

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Missing analog-stereo profile for Dell Latitude 3420 (ALC3204, SOF, Fedora 42)
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #594 was opened from KrzysztofWisiecki: **Summary:** On my Dell Latitude 3420 (Intel Tiger Lake, Realtek ALC3204, sof-hda-dsp), running Fedora 42 with PipeWire and WirePlumber (no PulseAudio), I am unable to select a standard "analog-stereo" profile for the internal sound card. Only "pro-audio" and "off" profiles are available. This makes it impossible to use speakers and the internal microphone together in a typical desktop setup. **System details:** Laptop: Dell Latitude 3420 Codec: Realtek ALC3204 Sound driver: sof-hda-dsp (SOF, not snd_hda_intel) OS: Fedora 42, fully updated PipeWire: Compiled with libpipewire 1.4.6 Linked with libpipewire 1.4.6 WirePlumber: Compiled with libwireplumber 0.5.8 Linked with libwireplumber 0.5.8 Kernel: 6.14.0-63.fc42.x86_64 **What is the issue?** No "analog-stereo" (or other typical analog) profiles are available. Only these profiles are listed: off pro-audio Profile: off: Off (sinks: 0, sources: 0, priority: 0, available: yes) pro-audio: Pro Audio (sinks: 5, sources: 3, priority: 1, available: yes) Active Profile: pro-audio ALSA devices are present: cat /proc/asound/cards 0 [sofhdadsp ]: sof-hda-dsp - sof-hda-dsp DellInc.-Latitude3420--03RT7P aplay -l card 0: sofhdadsp [sof-hda-dsp], device 0: HDA Analog (*) [] ... card 0: sofhdadsp [sof-hda-dsp], device 6: DMIC (*) [] ... The analog inputs/outputs exist at the ALSA layer. There is no file /usr/share/alsa-card-profile/mixer/profile-sets/sof-hda-dsp.conf by default. Attempting to copy or create a minimal sof-hda-dsp.conf with analog-stereo mapping does not enable the analog profile – only "pro-audio" appears. What have I tried? Created a custom /usr/share/alsa-card-profile/mixer/profile-sets/sof-hda-dsp.conf (based on default.conf/analog-only.conf/minimal analog-stereo mapping). Set api.acp.auto-profile and api.acp.auto-port to true for this device in /etc/wireplumber/wireplumber.conf.d/99-auto-profile.conf. Restarted PipeWire and WirePlumber. Reinstalled firmware (alsa-sof-firmware), alsa-card-profiles, alsa-ucm-conf. Checked for conflicting blacklist or modprobe options. Verified that everything works as expected with snd_hda_intel legacy driver (dsp_driver=1), but I want to use SOF Tried updating to the latest Fedora 42 and checked for updates in alsa-card-profiles/ucm-conf. Expected: "analog-stereo" (or similar) profile should be available for the internal sound card (sof-hda-dsp) so that both speakers and microphones can be used in a standard desktop configuration. Automatic profile switching should work (i.e., plugging in headphones/switching between speakers and headphones, using internal mic). Actual: Only "pro-audio" and "off" profiles are available. Manual switching is possible in "pro-audio", but lacks the convenience and user experience of standard profiles. No "analog-stereo", "analog-duplex" or similar options are listed. Request / What I'd like to see: Please add or enable standard analog profiles ("analog-stereo", "analog-duplex", etc.) for Dell Latitude 3420 with Realtek ALC3204 and sof-hda-dsp, if possible. Any guidance on manually enabling such profiles in the meantime would be appreciated. Let me know if you need full logs, system info, or additional tests! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/594 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ASoC: rt5660: Fix the dmic data source from GPIO2
by Oder Chiou 14 Jul '25

14 Jul '25

The patch fixes an issue with the dmic data pin connected to GPIO2. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5660.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 82b92e83be4c..44c3a3b92f98 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1315,14 +1315,17 @@ static int rt5660_i2c_probe(struct i2c_client *i2c) regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, RT5660_GP1_PIN_MASK, RT5660_GP1_PIN_DMIC1_SCL); - if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) + if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_GPIO2) { regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, RT5660_SEL_DMIC_DATA_MASK, RT5660_SEL_DMIC_DATA_GPIO2); - else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) + regmap_update_bits(rt5660->regmap, RT5660_GPIO_CTRL1, + RT5660_GP2_PIN_MASK, RT5660_GP2_PIN_DMIC1_SDA); + } else if (rt5660->pdata.dmic1_data_pin == RT5660_DMIC1_DATA_IN1P) { regmap_update_bits(rt5660->regmap, RT5660_DMIC_CTRL1, RT5660_SEL_DMIC_DATA_MASK, RT5660_SEL_DMIC_DATA_IN1P); + } } return devm_snd_soc_register_component(&i2c->dev, -- 2.34.1

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I'm using Ubuntu 24.04.2 LTS with Wayland on an iMac 27" (2017) equipped with the Cirrus Logic CS8409 audio codec. After compiling the driver manually (https://github.com/egorenar/snd-hda-codec-cs8409), ALSA detects the audio device correctly and it works properly with PipeWire and PulseAudio. However, it breaks with each kernel update, requiring recompilation. **Request**: Please consider adding native support for CS8409 in `alsa-ucm-conf`. Modern systems like Ubuntu 24.04.2 handle the device well, and i
by GitHub issues - edited 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #592 was edited from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/592 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Feature request: Add native support for CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #593 was opened from Marek-mahav: I've tested the CS8409 driver on Ubuntu 24.04.2 with Wayland… [celý text tu] Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/593 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Feature request: Add native support for Cirrus Logic CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #592 was opened from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/592 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Feature request: Add native support for Cirrus Logic CS8409 codec
by GitHub issues - opened 14 Jul '25

14 Jul '25

alsa-project/alsa-ucm-conf issue #591 was opened from Marek-mahav: -w Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/591 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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