November 2024 - Alsa-devel - mailman.alsa-project.org

`amixer sset Master` doesnt increase and decrease both sides equally
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #279 was opened from lquenti: I look through the issues, man pages and internet and hope I didn't miss anything. Please tell me if its a user error. Whenever I use `amixer sset Master` to increase volume, it will not increase them evenly. Here is a before ![before](https://github.com/user-attachments/assets/4ccfa220-608c-4407-9f6e-fc1d27f1a91b) and after ![after](https://github.com/user-attachments/assets/250ab254-d46a-44e4-a201-58f9224b6293) Interestingly amixer seem to think that it is still evenly balanced, although alsamixer shows that it is unbalanced afterwards. --- Interestingly, when decreasing, the unevenness flips to the other side. [Other people also seem to struggle with it](https://stackoverflow.com/questions/69826139/using-amixer-to-set-audio-… Thank you so much. If it is something somewhat simple to fix, I can also write a fix myself, but for now I am unsure what the problem is. Issue URL : https://github.com/alsa-project/alsa-utils/issues/279 Repository URL: https://github.com/alsa-project/alsa-utils

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Change 26ce:0a08 to list multiple motherboards
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #461 was opened from cybik: The hardware id `26ce:0a08` seems to be re-used by multiple motherboards at ASRock. Identify them within `[brackets]` to point out the specific boards in question. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/461.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ThinkPad P14s Gen 5 AMD micmute LED is not working sometimes.
by GitHub issues - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-utils issue #278 was opened from pma1: The mic and sound output works fine. The mute LED works fine. But when press Fn+F4, the mic-mute led is not working sometimes after bootup. The failrate is like 1/10. Tested with latest alsa-ucm from git, it's reproduced. /sys/class/sound/ctl-led/mic/card1# cat list 7 # amixer -c 1 controls numid=12,iface=CARD,name='Headphone Jack' numid=11,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Phantom Jack' numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=2,iface=MIXER,name='Headphone Playback Switch' numid=1,iface=MIXER,name='Headphone Playback Volume' numid=16,iface=MIXER,name='PCM Playback Volume' numid=17,iface=MIXER,name='Mic ACP LED Capture Switch' numid=8,iface=MIXER,name='Mic Boost Volume' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=5,iface=MIXER,name='Auto-Mute Mode' numid=4,iface=MIXER,name='Speaker Playback Switch' numid=3,iface=MIXER,name='Speaker Playback Volume' numid=15,iface=PCM,name='Capture Channel Map' numid=14,iface=PCM,name='Playback Channel Map' # echo 'Capture Switch' > detach # echo 'Mic ACP LED Capture Switch' > attach /sys/class/sound/ctl-led/mic/card1# cat list 17 Then the micmute LED will work. [alsa-info-amd-micmutled.txt](https://github.com/user-attachments/files/1627… Issue URL : https://github.com/alsa-project/alsa-utils/issues/278 Repository URL: https://github.com/alsa-project/alsa-utils

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USB-Audio: ALC4080: Add support for MSI MPG X870E CARBON WIFI (0db0:0b58)
by GitHub pull_request - opened 11 Nov '24

11 Nov '24

alsa-project/alsa-ucm-conf pull request #460 was opened from xatier: This PR updates the regex for MSI MPG X870E CARBON WIFI motherboard. ``` $ lsusb | ag audio Bus 003 Device 007: ID 0db0:0b58 Micro Star International USB Audio ``` ``` Audio Realtek® ALC4080 Codec ``` https://www.msi.com/Motherboard/MPG-X870E-CARBON-WIFI/Specification Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/460.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 6.1.y] ALSA: hda/realtek: Fix headset mic on TUXEDO Gemini 17 Gen3
by Werner Sembach 09 Nov '24

09 Nov '24

From: Christoffer Sandberg <cs(a)tuxedo.de> Quirk is needed to enable headset microphone on missing pin 0x19. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> Link: https://patch.msgid.link/20241029151653.80726-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a8bc95ffa41a3..da14ab97783f8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10001,6 +10001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x28c1, "Clevo V370VND", ALC2XX_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.43.0

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Odd behavior with Tempotec Serenade PCIe (CM8888)
by GitHub issues - opened 09 Nov '24

09 Nov '24

alsa-project/alsa-lib issue #418 was opened from lss4: This card is powered by CM8888, using `snd-hda-intel` driver, and it does work in Linux, but its behavior is quite odd. 1. Both the RCA connection and 6.35 headphone jack on the card are connected at the moment. Preferably I'd like to be able to switch to use either RCA (to speakers) or headphone, but I can't -- Audio always goes through both outputs, and there's only one analog output from this card. 2. This is the odd behavior I'm talking about -- the card's outputs are muted by default. Playing an audio file will trigger the card to unmute the output after 2-3 seconds followed by an audible click from the relay on the sound card. When the output is active, if no audio is being actively played for at least 10 seconds, the output will be muted again on its own, and this is again followed by an audible click from the relay. Honestly I've never seen an audio card capable of doing something that "intelligent" before. `alsa-info.sh` output attached here. The sound card in question is `card1` (`hwC1D1`). [alsa-info-20241109.txt](https://github.com/user-attachments/files/17687665/… I'm currently using Garuda Linux on this system. Previously I was using Manjaro. Both are Arch-based, and the behaviors are the same. I'm not sure about the support of this chip (CM8888) in general, as the information I could find online about CM8888 on Linux is limited and mostly outdated (several years ago). Issue URL : https://github.com/alsa-project/alsa-lib/issues/418 Repository URL: https://github.com/alsa-project/alsa-lib

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[Question] About Roland Rubix series, alsa-ucm-conf and snd_usb_audio
by GitHub issues - opened 08 Nov '24

08 Nov '24

alsa-project/alsa-ucm-conf issue #459 was opened from sim590: I want to understand what `alsa_ucm-conf` is about exactly regarding the support for specific devices. I'm thinking to buy a Roland Rubix series audio interface (either 22 or 24) and I'm wondering if both my XLR inputs are gonna appear in alsa/pulseaudio/pipewire as separate input lines. Does that require a potential change in `snd_usb_audio` kernel module or only a configuration file on this repository? I've looked at the configuration files on this repository and I have found a Roland directory, but nothing about the Rubix series explicitly. Presently, I have a Focusrite Scarlett 2i2 *fake clone* (unfortunately) which I'm gonna return shortly because it's counterfeit. As of right now, I don't see the two separate outputs, but only the following input: ![image](https://github.com/user-attachments/assets/7bf58e54-67d1-4e20-989c-5e1ec0828850) which takes audio from my first XLR input port. I guess that's the class compliant interface behaviour. Correct me if I'm wrong, I'm trying to learn about all of this. So, I just wanna be sure that if I go and buy my Roland Rubix, that the device is actually gonna show up the two input lines, and if not that I will know what I will have to do to make it so. Is there gonna be a configuration file required in `alsa-ucm-conf`? Any potentiel change in `snd_usb_audio` kernel module? I'm a bit confused about the details regarding the support of the audio interface devices. Thanks. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/459 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ASoC Kernel 6.1: Query regarding dynamic dai link/dapm registration during bootup
by Shalini Manjunatha (Consultant) (QUIC) 08 Nov '24

08 Nov '24

Hi All, On kernel6.1 based Qualcomm target for automotive platform based on Android U, we have a feature early services to launch certain early apps at early init of android bootup cycle. Hence in order to launch early chime sound of audio at early init of android boot up cycle, we need to load all required audio kernel modules at early init stage in order to get sound card registered. We have certain KPI goals for every early apps to get completed within lesser duration so that we can achieve final KPI goal of android boot complete as per the target specification. In this journey, while optimizing the sound card registration through machine driver probe to achieve lower KPI for audio early chime sound, we find that for early chime launch we required only selected pcm front end dai links only. Hence if we keep FE dai link structure in our machine driver with only required items for early chime, overall sound card registrations gets completed within 200msec from the time it starts, which is a very good KPI number to achieve. So we were trying out code change to split overall FE dai link structure into two, one for early chime and another for all other required dai link's, then some how we need to register early chime related dai link during early init so that sound card registration process gets completed quickly, later stage we want to dynamically load all other FE dai links. Could you please help us to suggest on how can we achieve this?. Is there any possible way out to avoid static FE dai link registration for all required ones from machine driver during machine driver probe, instead register only limited list of dai links. Later can we dynamically register the dai links in order to get PCM devices entries created based on use case. Currently we have this API devm_snd_soc_register_card() being called with snd_soc_card parameter which has all possible FE dai links populated at once. Queries: 1) can we register sound card twice during bootup, once with limited dai links and next with all required ones? 2) can we register dai links/dapm dynamically based on use case requriement? 3) If it is possible, could you share any reference from any other vendor who is able to achieve the same?. Regards, Shalini M

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[PATCH RESEND v4 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 08 Nov '24

08 Nov '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: dt-bindings: irondevice,sma1307: Add initial DT ASoC: sma1307: Add driver for Iron Device SMA1307 .../bindings/sound/irondevice,sma1307.yaml | 53 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2052 +++++++++++++++++ sound/soc/codecs/sma1307.h | 444 ++++ 5 files changed, 2561 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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[GIT PULL] ASoC fixes for v6.12-rc6
by Mark Brown 08 Nov '24

08 Nov '24

The following changes since commit 2db63e92186d7201ee1cb2f5af11757c5e5a1020: wcd937x codec fixes (2024-10-29 19:18:48 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc6 for you to fetch changes up to 48b86532c10128cf50c854a90c2d5b1410f4012d: ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits (2024-11-07 15:35:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.12 A moderately large pile of small changes here, split fairly evenly between fixes and ID additions/quirks and all of it driver specific. ---------------------------------------------------------------- Amelie Delaunay (1): ASoC: stm32: spdifrx: fix dma channel release in stm32_spdifrx_remove Jyri Sarha (1): ASoC: SOF: sof-client-probes-ipc4: Set param_size extension bits Kuninori Morimoto (1): MAINTAINERS: Generic Sound Card section Luo Yifan (2): ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate() ASoC: stm: Prevent potential division by zero in stm32_sai_get_clk_div() Markus Petri (1): ASoC: amd: yc: Support dmic on another model of Lenovo Thinkpad E14 Gen 6 Mingcong Bai (1): ASoC: amd: yc: fix internal mic on Xiaomi Book Pro 14 2022 Shenghao Ding (1): ASoC: tas2781: Add new driver version for tas2563 & tas2781 qfn chip Venkata Prasad Potturu (1): ASoC: SOF: amd: Fix for incorrect DMA ch status register offset MAINTAINERS | 9 +++++++++ sound/soc/amd/yc/acp6x-mach.c | 14 ++++++++++++++ sound/soc/codecs/tas2781-fmwlib.c | 1 + sound/soc/sof/amd/acp.c | 10 +++++++++- sound/soc/sof/sof-client-probes-ipc4.c | 1 + sound/soc/stm/stm32_sai_sub.c | 6 +++--- sound/soc/stm/stm32_spdifrx.c | 2 +- 7 files changed, 38 insertions(+), 5 deletions(-)

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