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[PATCH] ALSA: hda: Add new pci id for AMD GPU display HD audio controller
by Vijendar Mukunda 02 Jun '25

02 Jun '25

Add new pci id for AMD GPU display HD audio controller(device id- 0xab40). Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 512fb22f5e5e..3f8b2bf6eab5 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2715,6 +2715,9 @@ static const struct pci_device_id azx_ids[] = { { PCI_VDEVICE(ATI, 0xab38), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | AZX_DCAPS_PM_RUNTIME }, + { PCI_VDEVICE(ATI, 0xab40), + .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS | + AZX_DCAPS_PM_RUNTIME }, /* GLENFLY */ { PCI_DEVICE(PCI_VENDOR_ID_GLENFLY, PCI_ANY_ID), .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, -- 2.45.2

3 3

snd_pcm_drain stucked.
by GitHub issues - opened 02 Jun '25

02 Jun '25

alsa-project/alsa-lib issue #457 was opened from yangyagami: When snd_pcm_drain is called, if snd_pcm_drop is called in another thread, snd_pcm_drain will get stuck. ```cpp #include <iostream> #include <cmath> #include <thread> #include <alsa/asoundlib.h> #define SAMPLE_RATE 44100 #define FREQUENCY 440.0 // A4 note #define DURATION 5.0 // seconds using namespace std::chrono_literals; int main() { snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int rate = SAMPLE_RATE; int dir; int pcm; pcm = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK, 0); if (pcm < 0) { std::cerr << "Unable to open PCM device: " << snd_strerror(pcm) << std::endl; return 1; } snd_pcm_hw_params_alloca(&params); snd_pcm_hw_params_any(handle, params); snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_rate_near(handle, params, &rate, &dir); snd_pcm_hw_params_set_channels(handle, params, 1); pcm = snd_pcm_hw_params(handle, params); if (pcm < 0) { std::cerr << "Unable to set HW parameters: " << snd_strerror(pcm) << std::endl; return 1; } int samples = static_cast<int>(DURATION * SAMPLE_RATE); int16_t *buffer = new int16_t[samples]; for (int i = 0; i < samples; ++i) { buffer[i] = static_cast<int16_t>(32767 * sin(2 * M_PI * FREQUENCY * i / SAMPLE_RATE)); } pcm = snd_pcm_writei(handle, buffer, samples); if (pcm < 0) { std::cerr << "Playback error: " << snd_strerror(pcm) << std::endl; } std::thread t([handle](){ std::this_thread::sleep_for(500ms); snd_pcm_drop(handle); }); snd_pcm_drain(handle); t.join(); delete[] buffer; snd_pcm_close(handle); return 0; } ``` gdb output ```shell Thread 1 "main" received signal SIGINT, Interrupt. 0x00007ffff71b5e22 in ?? () from /usr/lib/libc.so.6 (gdb) bt #0 0x00007ffff71b5e22 in ?? () from /usr/lib/libc.so.6 #1 0x00007ffff71a9fda in ?? () from /usr/lib/libc.so.6 #2 0x00007ffff71aa64c in ?? () from /usr/lib/libc.so.6 #3 0x00007ffff71acd1e in pthread_cond_wait () from /usr/lib/libc.so.6 #4 0x00007ffff6b9e2b9 in pw_thread_loop_wait () from /usr/lib/libpipewire-0.3.so.0 #5 0x00007ffff7f2ec8f in ?? () from /usr/lib/alsa-lib/libasound_module_pcm_pipewire.so #6 0x00007ffff777c2b8 in ?? () from /usr/lib/libasound.so.2 #7 0x0000555555558d6f in main () at main.cc:116 (gdb) ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/457 Repository URL: https://github.com/alsa-project/alsa-lib

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ucm2: Qualcomm: add ASUS Vivobook S 15 support
by GitHub pull_request - opened 02 Jun '25

02 Jun '25

alsa-project/alsa-ucm-conf pull request #570 was opened from binarycraft007: S15 supports: - 2 speakers. - 2 dmics - headset with mic. This patch adds support to all these, however only speakers, dmic and headset playback is tested. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/570 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/570.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4080 Mic In not working on ASUS ROG Strix B850-E Gaming WIFI
by GitHub issues - edited 01 Jun '25

01 Jun '25

alsa-project/alsa-ucm-conf issue #569 was edited from kieran4333: [alsa-info.sh](http://alsa-project.org/db/?f=edcbadeae18eb8f200b9625d2b0fb41… `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Analog In - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=11,iface=CARD,name='Clock Source 1 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=22,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=26,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=29,iface=CARD,name='Clock Source 8 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=27,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=12,iface=CARD,name='Line - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=23,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=24,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=28,iface=MIXER,name='PCM Playback Switch',index=2 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=21,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=87,step=0 : values=56,56,56,56,56,56,56,56 | dBminmax-min=-65.25dB,max=0.00dB numid=25,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=59,59 | dBminmax-min=-65.25dB,max=0.00dB numid=13,iface=MIXER,name='Line Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=14,iface=MIXER,name='Line Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=17,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=18,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=35,35 | dBminmax-min=-17.25dB,max=12.00dB numid=9,iface=MIXER,name='Analog In Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Analog In Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=36,36 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR | chmap-fixed=FL,FR,RL,RR | chmap-fixed=FL,FR,FC,LFE,RL,RR | chmap-fixed=FL,FR,FC,LFE,RL,RR,SL,SR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=5,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Capture Channel Map',device=2 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=6,iface=PCM,name='Playback Channel Map',device=2 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=7,iface=PCM,name='Playback Channel Map',device=3 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `alsactl monitor` (Unplugging and plugging the mic) ``` node hw:2, #12 (0,0,0,Line - Input Jack,0) VALUE node hw:2, #12 (0,0,0,Line - Input Jack,0) VALUE ``` `amixer -c events` (Unplugging and plugging mic) ``` event value: numid=12,iface=CARD,name='Line - Input Jack' event value: numid=12,iface=CARD,name='Line - Input Jack' ``` It appears the Mic In is being incorrectly identified as the Line In and I cannot record any sound through this jack. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/569 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ALC4080 Mic In not working on ASUS ROG Strix B580-E Gaming WIFI
by GitHub issues - opened 01 Jun '25

01 Jun '25

alsa-project/alsa-ucm-conf issue #569 was opened from kieran4333: [alsa-info.sh](http://alsa-project.org/db/?f=edcbadeae18eb8f200b9625d2b0fb41… `lsusb | grep -i audio` ``` Bus 001 Device 002: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio ``` `amixer -c 2 contents` ``` numid=8,iface=CARD,name='Analog In - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=11,iface=CARD,name='Clock Source 1 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=15,iface=CARD,name='Clock Source 3 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=19,iface=CARD,name='Clock Source 4 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=22,iface=CARD,name='Clock Source 5 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=26,iface=CARD,name='Clock Source 6 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=29,iface=CARD,name='Clock Source 8 Validity' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=27,iface=CARD,name='Headphone - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=12,iface=CARD,name='Line - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=16,iface=CARD,name='Mic - Input Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=23,iface=CARD,name='Speaker - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=20,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=24,iface=MIXER,name='PCM Playback Switch',index=1 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=28,iface=MIXER,name='PCM Playback Switch',index=2 ; type=BOOLEAN,access=rw------,values=1 : values=off numid=21,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=8,min=0,max=87,step=0 : values=56,56,56,56,56,56,56,56 | dBminmax-min=-65.25dB,max=0.00dB numid=25,iface=MIXER,name='PCM Playback Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=59,59 | dBminmax-min=-65.25dB,max=0.00dB numid=13,iface=MIXER,name='Line Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=14,iface=MIXER,name='Line Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=17,iface=MIXER,name='Mic Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=18,iface=MIXER,name='Mic Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=35,35 | dBminmax-min=-17.25dB,max=12.00dB numid=9,iface=MIXER,name='Analog In Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off numid=10,iface=MIXER,name='Analog In Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=39,step=0 : values=36,36 | dBminmax-min=-17.25dB,max=12.00dB numid=1,iface=PCM,name='Capture Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=4,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR | chmap-fixed=FL,FR,RL,RR | chmap-fixed=FL,FR,FC,LFE,RL,RR | chmap-fixed=FL,FR,FC,LFE,RL,RR,SL,SR numid=2,iface=PCM,name='Capture Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=5,iface=PCM,name='Playback Channel Map',device=1 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=3,iface=PCM,name='Capture Channel Map',device=2 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=6,iface=PCM,name='Playback Channel Map',device=2 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR numid=7,iface=PCM,name='Playback Channel Map',device=3 ; type=INTEGER,access=r--v-R--,values=2,min=0,max=36,step=0 : values=0,0 | container | chmap-fixed=FL,FR ``` `alsactl monitor` (Unplugging and plugging the mic) ``` node hw:2, #12 (0,0,0,Line - Input Jack,0) VALUE node hw:2, #12 (0,0,0,Line - Input Jack,0) VALUE ``` `amixer -c events` (Unplugging and plugging mic) ``` event value: numid=12,iface=CARD,name='Line - Input Jack' event value: numid=12,iface=CARD,name='Line - Input Jack' ``` It appears the Mic In is being incorrectly identified as the Line In and I cannot record any sound through this jack. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/569 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Arturia AudioFuse support
by GitHub issues - opened 01 Jun '25

01 Jun '25

alsa-project/alsa-ucm-conf issue #568 was opened from saaj: I own [Arturia AudioFuse][1]. I am on Linux Mint and I got faux "Analogue Surround 7.1 Output" probably since Linux Mint 20. The version before that was property split input multiple devices in the system sound settings. Not a big deal, but occasionally I want just plug it in to listen to music on a good DAC and headphone (without setting up Jack and ingestigating XRUNs). I didn't find a simple workaround and just used Jack (in my instrument/recording setup). I recently updated to Linux Mint 22.1 (based on Ubuntu 24.04) and wanted to explore if there's a solution, since LM's audio is now based in PipeWire. Out of the box I got the same "Analogue Surround 7.1 Output". After a lot of searching, I found that some Arturia products are already in my `/usr/share/alsa/ucm2/USB-Audio/Arturia` (alsa-ucm-conf 1.2.10-1ubuntu5.4). Then I just applied https://github.com/alsa-project/alsa-ucm-conf/commit/b68aa52, `cp Minifuse-12.conf Audiofuse.conf`, `cp Minifuse-12-HiFi.conf Audiofuse-HiFi.conf`, added an include in `USB-Audio.conf` and `s/minifuse12/audiofuse` in the files. `systemctl --user restart pipewire pipewire-pulse wireplumber` and viola -- I've got the split back! :-) I'm only interested in "Main Output L/R AudioFuse", which is working. The rest may not, and I still plan to use Jack for recording. But I'm happy to validate things on the interface. Attaching my patch over Minifuse-12 for completeness. [audiofuse.patch.gz](https://github.com/user-attachments/files/20541119/audi… [1]: https://www.arturia.com/products/audio/audiofuse/overview Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/568 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

aseqsend silently fails (ENODEV) with hardware port
by GitHub issues - opened 01 Jun '25

01 Jun '25

alsa-project/alsa-utils issue #300 was opened from michaelforney: When I use `aseqsend` to send a sysex message to a device, it doesn't seem to work, though there is no error message. Running with strace, it seems it is failing with ENODEV. ```sh $ strace aseqsend -p 32:0 'F0 43 7D 10 41 30 01 00 00 01 F7' ... open("/dev/snd/seq", O_WRONLY|O_LARGEFILE|O_CLOEXEC) = 3 fcntl(3, F_SETFD, FD_CLOEXEC) = 0 ioctl(3, SNDRV_SEQ_IOCTL_PVERSION, 0x7fffef1ae678) = 0 ioctl(3, SNDRV_SEQ_IOCTL_USER_PVERSION, 0x7fffef1ae67c) = 0 mmap(NULL, 20480, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x7f6ae25f7000 ioctl(3, SNDRV_SEQ_IOCTL_CLIENT_ID, 0x7fffef1ae67c) = 0 ioctl(3, SNDRV_SEQ_IOCTL_RUNNING_MODE, 0x7fffef1ae680) = 0 ioctl(3, SNDRV_SEQ_IOCTL_GET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_GET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_SET_CLIENT_INFO, 0x7fffef1ae940) = 0 ioctl(3, SNDRV_SEQ_IOCTL_CREATE_PORT, 0x7fffef1ae960) = 0 write(3, "\202\4\0\375\0\0\0\0\0\0\0\0\0\0 \0\v\0\0\0\320\212WN\361U\0\0\360C}\20"..., 39) = -1 ENODEV (No such device) nanosleep({tv_sec=0, tv_nsec=1000000}, 0x7fffef1aea20) = 0 close(3) = 0 munmap(0x7f6ae25f7000, 20480) = 0 exit_group(0) = ? +++ exited with 0 +++ $ ``` `amidi -S ...` works as expected. If I modify `aseqsend` to first subscribe to the port, it seems to work: ```diff diff --git a/seq/aseqsend/aseqsend.c b/seq/aseqsend/aseqsend.c index 92354eb..ad0a636 100644 --- a/seq/aseqsend/aseqsend.c +++ b/seq/aseqsend/aseqsend.c @@ -364,6 +364,7 @@ int main(int argc, char *argv[]) char do_port_list = 0; char verbose = 0; int k; + int err; while ((c = getopt_long(argc, argv, "hi:Vvlp:s:u:", long_options, NULL)) != -1) { switch (c) { @@ -439,6 +440,8 @@ int main(int argc, char *argv[]) error("Unable to parse port name!"); exit(EXIT_FAILURE); } + err = snd_seq_connect_to(seq, 0, addr.client, addr.port); + check_snd("connect to port", err); sent_data_c = 0; //counter of actually sent bytes ``` From what I've read, I don't think it should be necessary to subscribe to a port to send it messages. However, I did find one sentence in the alsa-lib docs that seem to indicate that it is needed for hardware ports: > There is another subscription type for opposite direction: Suppose a MIDI sequencer program which sends events to a MIDI output device. In ALSA system, MIDI device is not opened until the associated MIDI port is accessed. Thus, in order to activate MIDI device, we have to subscribe to MIDI port for write. After this connection is established, events will be properly sent to MIDI output device. This seems to be a pretty basic use of `aseqsend`, so I am a bit surprised that it wasn't working. Is the subscription the right way to fix the problem, or am I doing something else wrong? Issue URL : https://github.com/alsa-project/alsa-utils/issues/300 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v1] ALSA: hda/tas2781: Add compatible for hardware id TIAS2781 and TXNW2781
by Baojun Xu 31 May '25

31 May '25

TIAS2781 is unofficial hardware id in acpi for tas2781 in HDA, has been used for several projects. TXNW is the official hardware id for TI, will be used in new projects, including device on SPI bus. This patch will support both TIAS2781 and TXNW2781 in acpi with tas2781 under HDA. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/patch_realtek.c | 21 +++++++++++++++++---- sound/pci/hda/tas2781_hda_i2c.c | 10 +++++++--- 2 files changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd0d7ba73..ff32ac25d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7323,7 +7323,7 @@ static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fix alc_fixup_inv_dmic(cdc, fix, action); } -static void tas2781_fixup_i2c(struct hda_codec *cdc, +static void tas2781_fixup_tias_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "i2c", "TIAS2781", "-%s:00", 1); @@ -7334,6 +7334,12 @@ static void tas2781_fixup_spi(struct hda_codec *cdc, const struct hda_fixup *fix comp_generic_fixup(cdc, action, "spi", "TXNW2781", "-%s:00-tas2781-hda.%d", 2); } +static void tas2781_fixup_txnw_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "TXNW2781", "-%s:00-tas2781-hda.%d", 1); +} + static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -8001,6 +8007,7 @@ enum { ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, ALC287_FIXUP_TAS2781_I2C, ALC245_FIXUP_TAS2781_SPI_2, + ALC287_FIXUP_TXNW2781_I2C, ALC287_FIXUP_YOGA7_14ARB7_I2C, ALC245_FIXUP_HP_MUTE_LED_COEFBIT, ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT, @@ -10251,7 +10258,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC287_FIXUP_TAS2781_I2C] = { .type = HDA_FIXUP_FUNC, - .v.func = tas2781_fixup_i2c, + .v.func = tas2781_fixup_tias_i2c, .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, @@ -10261,6 +10268,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, + [ALC287_FIXUP_TXNW2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_txnw_i2c, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, + }, [ALC287_FIXUP_YOGA7_14ARB7_I2C] = { .type = HDA_FIXUP_FUNC, .v.func = yoga7_14arb7_fixup_i2c, @@ -11318,8 +11331,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x390d, "Lenovo Yoga Pro 7 14ASP10", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC), - SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x391f, "Yoga S990-16 pro Quad YC Quad", ALC287_FIXUP_TXNW2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3920, "Yoga S990-16 pro Quad VECO Quad", ALC287_FIXUP_TXNW2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index d91eed9f7..48c7af191 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -583,16 +583,19 @@ static int tas2781_hda_i2c_probe(struct i2c_client *clt) if (!tas_hda->priv) return -ENOMEM; + hda_priv->save_calibration = tas2781_save_calibration; + tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; if (strstr(dev_name(&clt->dev), "TIAS2781")) { device_name = "TIAS2781"; - hda_priv->save_calibration = tas2781_save_calibration; - tas_hda->priv->global_addr = TAS2781_GLOBAL_ADDR; + } else if (strstr(dev_name(&clt->dev), "TXNW2781:00-tas2781-hda.0")) { + device_name = "TXNW2781"; } else if (strstr(dev_name(&clt->dev), "INT8866")) { device_name = "INT8866"; hda_priv->save_calibration = tas2563_save_calibration; tas_hda->priv->global_addr = TAS2563_GLOBAL_ADDR; - } else + } else { return -ENODEV; + } tas_hda->priv->irq = clt->irq; ret = tas2781_read_acpi(tas_hda->priv, device_name); @@ -723,6 +726,7 @@ static const struct i2c_device_id tas2781_hda_i2c_id[] = { static const struct acpi_device_id tas2781_acpi_hda_match[] = { {"TIAS2781", 0 }, + {"TXNW2781", 0 }, {"INT8866", 0 }, {} }; -- 2.43.0

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[PATCH AUTOSEL 5.10 08/10] ASoC: tegra210_ahub: Add check to of_device_get_match_data()
by Sasha Levin 30 May '25

30 May '25

From: Yuanjun Gong <ruc_gongyuanjun(a)163.com> [ Upstream commit 04cb269c204398763a620d426cbee43064854000 ] In tegra_ahub_probe(), check the result of function of_device_get_match_data(), return an error code in case it fails. Signed-off-by: Yuanjun Gong <ruc_gongyuanjun(a)163.com> Link: https://patch.msgid.link/20250513123744.3041724-1-ruc_gongyuanjun@163.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- **YES** This commit should be backported to stable kernel trees. **Extensive Analysis:** **1. Nature of the Bug Fix:** This commit fixes a critical null pointer dereference vulnerability in the tegra210_ahub driver. The code calls `of_device_get_match_data(&pdev->dev)` but never validates the return value before using `ahub->soc_data`. Looking at the kernel tree, we can see that immediately after the assignment, the code accesses `ahub->soc_data->regmap_config` and later `ahub->soc_data->cmpnt_drv`, `ahub->soc_data->dai_drv`, and `ahub->soc_data->num_dais` without any null checking. **2. Severity and Impact:** - **High severity**: A null pointer dereference in the kernel probe function can cause kernel panic - **User impact**: Systems using Tegra210 audio hardware could crash during device initialization - **Subsystem criticality**: Audio subsystem failure affects multimedia functionality **3. Code Analysis:** The fix is minimal and surgical: ```c ahub->soc_data = of_device_get_match_data(&pdev->dev); +if (!ahub->soc_data) + return -ENODEV; ``` This follows established kernel patterns and returns an appropriate error code (-ENODEV) when device match data cannot be found. **4. Consistency with Similar Fixes:** Examining the kernel tree reveals that other Tegra drivers have already been fixed with identical patterns: - `tegra30_ahub.c` already has this null check (returns -EINVAL) - `tegra30_i2s.c` already has this null check (returns -ENODEV) This indicates that tegra210_ahub was an oversight that should be corrected for consistency and safety. **5. Historical Reference Analysis:** While the provided historical commits all have "Backport Status: NO", they were primarily code cleanup and API migration commits (switching to `of_device_get_match_data`), not security/stability fixes. This commit is fundamentally different - it's a critical bug fix that prevents kernel crashes. **6. Stable Tree Criteria Compliance:** - ✅ **Important bugfix**: Prevents kernel panic - ✅ **Minimal risk**: Two-line addition with well-established pattern - ✅ **Small and contained**: No architectural changes - ✅ **Clear purpose**: Prevents null pointer dereference - ✅ **Subsystem confined**: Only affects tegra210_ahub driver **7. Risk Assessment:** - **Regression risk**: Extremely low - only adds a safety check - **Side effects**: None - failure case now returns clean error instead of crashing - **Testing impact**: Improves robustness without changing successful code paths **Conclusion:** This is a textbook example of a commit that should be backported to stable trees. It fixes a critical null pointer dereference that can cause kernel panics, uses minimal code changes following established patterns, and has virtually no risk of introducing regressions. The fix brings tegra210_ahub in line with other Tegra drivers that already have this protection. sound/soc/tegra/tegra210_ahub.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index 1b2f7cb8c6adc..686c8ff46ec8a 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -607,6 +607,8 @@ static int tegra_ahub_probe(struct platform_device *pdev) return -ENOMEM; ahub->soc_data = of_device_get_match_data(&pdev->dev); + if (!ahub->soc_data) + return -ENODEV; platform_set_drvdata(pdev, ahub); -- 2.39.5

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[PATCH AUTOSEL 5.10 04/10] ASoC: tas2770: Power cycle amp on ISENSE/VSENSE change
by Sasha Levin 30 May '25

30 May '25

From: Hector Martin <marcan(a)marcan.st> [ Upstream commit f529c91be8a34ac12e7599bf87c65b6f4a2c9f5c ] The ISENSE/VSENSE blocks are only powered up when the amplifier transitions from shutdown to active. This means that if those controls are flipped on while the amplifier is already playing back audio, they will have no effect. Fix this by forcing a power cycle around transitions in those controls. Reviewed-by: Neal Gompa <neal(a)gompa.dev> Signed-off-by: Hector Martin <marcan(a)marcan.st> Signed-off-by: James Calligeros <jcalligeros99(a)gmail.com> Link: https://patch.msgid.link/20250406-apple-codec-changes-v5-1-50a00ec850a3@gma… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- **YES** This commit should be backported to stable kernel trees based on my analysis of both the commit message and code changes. Here's my extensive explanation: ## Bug Fix Analysis This commit addresses a specific functional bug in the tas2770 audio codec driver where ISENSE/VSENSE controls don't work properly when changed during active playback. The commit message clearly states: "if those controls are flipped on while the amplifier is already playing back audio, they will have no effect." ## Code Changes Analysis The fix is contained and minimal, involving three key changes to `sound/soc/codecs/tas2770.c`: 1. **Addition of `sense_event` function (lines +21 to +43)**: This function implements a power cycling mechanism that forces the amplifier through a shutdown state when ISENSE/VSENSE controls are changed. This ensures the changes take effect regardless of playback state. 2. **Modified DAPM widget definitions (lines +45 to +48)**: The ISENSE and VSENSE switches are changed from simple `SND_SOC_DAPM_SWITCH` to `SND_SOC_DAPM_SWITCH_E` with event handling, connecting them to the new `sense_event` function. 3. **Event triggers**: The widgets respond to `SND_SOC_DAPM_PRE_REG` and `SND_SOC_DAPM_POST_REG` events to perform the power cycling around register changes. ## Why This Should Be Backported 1. **User-Affecting Bug**: This fixes a real functional issue where audio controls don't work as expected during playback, which directly impacts user experience. 2. **Small and Contained**: The fix is confined to a single driver file (`tas2770.c`) and doesn't affect other subsystems. The changes are surgical and targeted. 3. **Low Risk**: The fix follows established ASoC patterns using standard DAPM event handling. Similar power cycling approaches are used throughout the ASoC subsystem. 4. **No New Features**: This purely fixes existing functionality rather than adding new features. 5. **Comparison with Similar Commits**: Looking at the provided examples, this commit is very similar to "Similar Commit #2" and "Similar Commit #5" which were both marked as backportable (YES). Like commit #2, it fixes incorrect hardware behavior with a small register/control change. Like commit #5, it addresses power state management issues in audio hardware. 6. **Hardware-Specific Fix**: The commit addresses a hardware limitation specific to the tas2770 chip where ISENSE/VSENSE blocks are only powered up during shutdown-to-active transitions. This is documented in the commit message and is a legitimate hardware workaround. The fix ensures that software speaker protection functionality works correctly by guaranteeing that IVSENSE controls are functional, which is critical for protecting audio hardware from damage. sound/soc/codecs/tas2770.c | 30 ++++++++++++++++++++++++++++-- 1 file changed, 28 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 1928c1616a52d..629cc24d51c3d 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -158,11 +158,37 @@ static const struct snd_kcontrol_new isense_switch = static const struct snd_kcontrol_new vsense_switch = SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); +static int sense_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct tas2770_priv *tas2770 = snd_soc_component_get_drvdata(component); + + /* + * Powering up ISENSE/VSENSE requires a trip through the shutdown state. + * Do that here to ensure that our changes are applied properly, otherwise + * we might end up with non-functional IVSENSE if playback started earlier, + * which would break software speaker protection. + */ + switch (event) { + case SND_SOC_DAPM_PRE_REG: + return snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + case SND_SOC_DAPM_POST_REG: + return tas2770_update_pwr_ctrl(tas2770); + default: + return 0; + } +} + static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2770_asi1_mux), - SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, &isense_switch), - SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, &vsense_switch), + SND_SOC_DAPM_SWITCH_E("ISENSE", TAS2770_PWR_CTRL, 3, 1, &isense_switch, + sense_event, SND_SOC_DAPM_PRE_REG | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_SWITCH_E("VSENSE", TAS2770_PWR_CTRL, 2, 1, &vsense_switch, + sense_event, SND_SOC_DAPM_PRE_REG | SND_SOC_DAPM_POST_REG), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_OUTPUT("OUT"), -- 2.39.5

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