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

[GIT PULL] ASoC fixes for v7.0-rc1
by Mark Brown 26 Feb '26

26 Feb '26

The following changes since commit 6de23f81a5e08be8fbf5e8d7e9febc72a5b5f27f: Linux 7.0-rc1 (2026-02-22 13:18:59 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v7.0-rc1 for you to fetch changes up to 32fc4168fa56f6301d858c778a3d712774e9657e: ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table (2026-02-23 14:52:11 +0000) ---------------------------------------------------------------- ASoC: Fixes for v7.0 One quirk and a fix for handling of exotic peripherals on cs42l43. ---------------------------------------------------------------- Azamat Almazbek uulu (1): ASoC: amd: yc: Add ASUS EXPERTBOOK BM1503CDA to quirk table Charles Keepax (1): ASoC: cs42l43: Report insert for exotic peripherals sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/codecs/cs42l43-jack.c | 1 + 2 files changed, 8 insertions(+)

2 1

Re: [PATCH] ALSA: hda/ca0132: Set HP/Speaker auto-detect default from headphone pin verb
by Takashi Iwai 26 Feb '26

26 Feb '26

On Wed, 25 Feb 2026 18:36:45 +0100, Matt DeVillier wrote: > > From 79d54b6b2b2410ac3c169329b93607042b137693 Mon Sep 17 00:00:00 2001 > From: Matt DeVillier <matt.devillier(a)gmail.com> > Date: Wed, 25 Feb 2026 11:25:18 -0600 > Subject: [PATCH] ALSA: hda/ca0132: Set HP/Speaker auto-detect default from > headphone pin verb > > HP/Speaker auto-detect (VNID_HP_ASEL) has been off by default for every > CA0132 device since the driver was added in 2012. vnode_lswitch is > always initialized to 0 in ca0132_init_chip(), and no quirk or other > code path enables it. As a result, headphone jack detection works only > after the user manually turns on "HP/Speaker Auto Detect" in alsamixer, > which is not obvious on laptops with combo jacks (e.g. Google Link, > Alienware). > > Change the default to follow the headphone pin config: if the pin verb > has presence detect enabled (no AC_DEFCFG_MISC_NO_PRESENCE) and the > codec supports it (AC_PINCAP_PRES_DETECT), enable HP_ASEL by default. > This lets firmware (coreboot, UEFI, etc.) express whether the headphone > jack supports insertion detection. Devices with combo jacks can default > to auto-detect; devices with fixed/no jack leave it off. > > Signed-off-by: Matt DeVillier <matt.devillier(a)gmail.com> The code change looks OK, but the patch seems malformed by your mailer, and can't be applied cleanly. Also, the mailing list for the kernel patch submission is linux-sound(a)vger.kernel.org, not alsa-devel ML. Please fix your mail setup and resubmit to the right address. thanks, Takashi > --- > sound/hda/codecs/ca0132.c | 9 +++++++++ > 1 file changed, 9 insertions(+) > > diff --git a/sound/hda/codecs/ca0132.c b/sound/hda/codecs/ca0132.c > index dd054aedd501..ca6cb57347ea 100644 > --- a/sound/hda/codecs/ca0132.c > +++ b/sound/hda/codecs/ca0132.c > @@ -9816,6 +9816,15 @@ static void ca0132_config(struct hda_codec *codec) > spec->dig_in = 0x09; > break; > } > + > + /* Default HP/Speaker auto-detect from headphone pin verb: enable if the > + * pin config indicates presence detect (not AC_DEFCFG_MISC_NO_PRESENCE). > + */ > + if (spec->unsol_tag_hp && > + (snd_hda_query_pin_caps(codec, spec->unsol_tag_hp) & > AC_PINCAP_PRES_DETECT) && > + !(get_defcfg_misc(snd_hda_codec_get_pincfg(codec, spec->unsol_tag_hp)) & > + AC_DEFCFG_MISC_NO_PRESENCE)) > + spec->vnode_lswitch[VNID_HP_ASEL - VNODE_START_NID] = 1; > } > > static int ca0132_prepare_verbs(struct hda_codec *codec) > -- > 2.51.0

1 0

Why does MMAP access ignore start_threshold?
by Mark Hills 25 Feb '26

25 Feb '26

It seems that start_threshold (swparams) is not honoured if MMAP access is used. Whereas I expect that once a total of enough bytes have been committed the device automatically starts the consuming of the buffer, as it does with snd_pcm_writei(). Is this a bug, or intentional design? It does not seem to be the subject of any documentation -- yet for many years I seem to be be working around it. Perhaps I am consistently doing something wrong, here's a test case below. -- Mark #include <alsa/asoundlib.h> #define always(r) _always(r, __LINE__) #define FORMAT SND_PCM_FORMAT_S16 #define ACCESS SND_PCM_ACCESS_MMAP_INTERLEAVED static void _always(int r, unsigned line) { if (r < 0) { fprintf(stderr, "%d: %s\n", line, snd_strerror(r)); abort(); } } static void set_hw(snd_pcm_t *pcm, unsigned rate, unsigned channels, snd_pcm_uframes_t buffer_size, snd_pcm_uframes_t period_size) { int r; snd_pcm_hw_params_t *hw; snd_pcm_hw_params_alloca(&hw); r = snd_pcm_hw_params_any(pcm, hw); always(r); r = snd_pcm_hw_params_set_access(pcm, hw, ACCESS); always(r); r = snd_pcm_hw_params_set_format(pcm, hw, FORMAT); always(r); r = snd_pcm_hw_params_set_rate(pcm, hw, rate, 0); always(r); r = snd_pcm_hw_params_set_channels(pcm, hw, channels); always(r); r = snd_pcm_hw_params_set_buffer_size(pcm, hw, buffer_size); always(r); r = snd_pcm_hw_params_set_period_size(pcm, hw, period_size, 0); always(r); r = snd_pcm_hw_params(pcm, hw); always(r); } static void set_sw(snd_pcm_t *pcm, snd_pcm_uframes_t start_threshold, snd_pcm_uframes_t avail_min) { int r; snd_pcm_sw_params_t *sw; snd_pcm_sw_params_alloca(&sw); r = snd_pcm_sw_params_current(pcm, sw); always(r); r = snd_pcm_sw_params_set_start_threshold(pcm, sw, start_threshold); always(r); r = snd_pcm_sw_params_set_avail_min(pcm, sw, avail_min); always(r); r = snd_pcm_sw_params(pcm, sw); always(r); } snd_pcm_uframes_t do_mmap(snd_pcm_t *pcm) { int r; snd_pcm_uframes_t frames, offset; const snd_pcm_channel_area_t *area; r = snd_pcm_wait(pcm, 1000); always(r); r = snd_pcm_mmap_begin(pcm, &area, &offset, &frames); always(r); r = snd_pcm_mmap_commit(pcm, offset, frames); always(r); return frames; } snd_pcm_uframes_t do_write(snd_pcm_t *pcm, unsigned channels, snd_pcm_uframes_t frames) { int r; void *buf; size_t len = channels * snd_pcm_format_size(FORMAT, frames); buf = alloca(len); bzero(buf, len); r = snd_pcm_writei(pcm, buf, frames); always(r); return frames; } int main(int argc, char *argv[]) { int r; snd_pcm_t *pcm; r = snd_pcm_open(&pcm, argv[1], SND_PCM_STREAM_PLAYBACK, 0); always(r); set_hw(pcm, 48000, 2, 2048, 512); set_sw(pcm, 1024, 512); r = snd_pcm_prepare(pcm); always(r); for (;;) { snd_pcm_uframes_t frames; if (ACCESS == SND_PCM_ACCESS_MMAP_INTERLEAVED) frames = do_mmap(pcm); else frames = do_write(pcm, 2, 512); fprintf(stderr, "written %lu\n", frames); } r = snd_pcm_close(pcm); always(r); return 0; }

1 0

Re: [RFC PATCH] ASoC: Intel: sof_es8336: Add DMI quirk for Huawei BOD-WXX9
by Mark Brown 23 Feb '26

23 Feb '26

On Mon, Feb 23, 2026 at 11:40:45AM +0300, Honey Tydik wrote: > Gentle ping on this RFC. I'm still investigating why GPIO control from > DAPM event doesn't work on Huawei BOD-WXX9, while the same code works > from probe context. Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. If there have been review comments then people may be waiting for those to be addressed. Sending content free pings adds to the mail volume (if they are seen at all) which is often the problem and since they can't be reviewed directly if something has gone wrong you'll have to resend the patches anyway, so sending again is generally a better approach though there are some other maintainers who like them - if in doubt look at how patches for the subsystem are normally handled. Please don't top post, reply in line with needed context. This allows readers to readily follow the flow of conversation and understand what you are talking about and also helps ensure that everything in the discussion is being addressed.

1 0

ucm2: sof-soundwire: Add support for Cirrus Logic CS47L47
by GitHub pull_request - opened 23 Feb '26

23 Feb '26

alsa-project/alsa-ucm-conf pull request #709 was opened from rfvirgil: Add the sof-soundwire and codec files for the CS47L47. This is a SDCA codec with headphone/headset and digital mic. The configuration is based on the CS42L45 files. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/709 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/709.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Audient EVO4
by GitHub pull_request - edited 23 Feb '26

23 Feb '26

alsa-project/alsa-ucm-conf pull request #708 was edited from motolav: This is a 4in/4out device with 1 analog stereo out and 2 analog mono in. The hardware mixed loopback interface takes up the remaining 4 channels. Without this UCM the device is configured as 4.0 surround for in and out. I copied the way Windows configures the device except for the stereo mic mix as I couldn't get it to work somehow(might have been a wiremix issue). The loopback interface requires software to configure what channels are mixed into its recording interface that's only available on Windows and macOS but there is an incomplete script to control the device's hardware registers at github.com/vijay-prema/audient-evo-linux-tools. Closes: #303 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Audient EVO4
by GitHub pull_request - opened 23 Feb '26

23 Feb '26

alsa-project/alsa-ucm-conf pull request #708 was opened from motolav: This is a 4in/4out device with 1 analog stereo out and 2 analog mono in. The hardware mixed loopback interface takes up the remaining 4 channels. Without this UCM the device is configured as 4.0 surround for in and out. I copied the way Windows configures the device except for the stereo mic mix as I couldn't get it to work somehow. The loopback interface requires software to configure what channels are mixed into its recording interface that's only available on Windows and macOS but there is an incomplete script to control the device's hardware registers at github.com/vijay-prema/audient-evo-linux-tools. Closes: #303 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/708.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

fix discards const from pointer target
by GitHub pull_request - opened 20 Feb '26

20 Feb '26

alsa-project/alsa-lib pull request #496 was opened from heitbaum: Since glibc-2.43: For ISO C23, the functions bsearch, memchr, strchr, strpbrk, strrchr, strstr, wcschr, wcspbrk, wcsrchr, wcsstr and wmemchr that return pointers into their input arrays now have definitions as macros that return a pointer to a const-qualified type when the input argument is a pointer to a const-qualified type. https://lists.gnu.org/archive/html/info-gnu/2026-01/msg00005.html fixes: ```c ../../../src/seq/seqmid.c: In function 'snd_seq_parse_address': ../../../src/seq/seqmid.c:449:24: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 449 | if ((p = strpbrk(arg, ":.")) != NULL) { | ^ ../../../src/ucm/main.c: In function 'snd_use_case_get_list': ../../../src/ucm/main.c:2396:22: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 2396 | str1 = strchr(identifier, '/'); | ^ ../../../src/ucm/main.c: In function 'snd_use_case_geti': ../../../src/ucm/main.c:2721:22: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 2721 | str1 = strchr(identifier, '/'); | ^ ../../../src/ucm/main.c: In function 'snd_use_case_set': ../../../src/ucm/main.c:3038:22: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 3038 | str1 = strchr(identifier, '/'); | ^ ../../../src/ucm/ucm_subs.c: In function 'strchr_with_escape': ../../../src/ucm/ucm_subs.c:832:19: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 832 | s = strchr(str, c); | ^ ../../src/conf.c: In function 'snd_config_hook_load_for_all_cards': ../../src/conf.c:4446:51: warning: initialization discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 4446 | char *s = strchr(driver, '.'); | ^~~~~~ ``` Request URL : https://github.com/alsa-project/alsa-lib/pull/496 Patch URL : https://github.com/alsa-project/alsa-lib/pull/496.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

fix discards const from pointer target
by GitHub pull_request - opened 20 Feb '26

20 Feb '26

alsa-project/alsa-utils pull request #322 was opened from heitbaum: Since glibc-2.43: For ISO C23, the functions bsearch, memchr, strchr, strpbrk, strrchr, strstr, wcschr, wcspbrk, wcsrchr, wcsstr and wmemchr that return pointers into their input arrays now have definitions as macros that return a pointer to a const-qualified type when the input argument is a pointer to a const-qualified type. https://lists.gnu.org/archive/html/info-gnu/2026-01/msg00005.html ```c ../../alsactl/init_parse.c: In function 'elemid_get': ../../alsactl/init_parse.c:695:29: warning: initialization discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 695 | char *pos = strchr(attr, ' '); | ^~~~~~ ../../alsactl/init_parse.c: In function 'conf_name_filter': ../../alsactl/init_parse.c:1284:21: warning: initialization discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 1284 | char *ext = strrchr(d->d_name, '.'); | ^~~~~~~ ../../aplay/aplay.c: In function 'create_path': ../../aplay/aplay.c:3226:23: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 3226 | start = strchr(path + 1, '/'); | ^ ../../aplay/aplay.c:3228:23: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 3228 | start = strchr(path, '/'); | ^ ../../topology/pre-process-object.c: In function 'tplg_pp_add_object_tuple_section': ../../topology/pre-process-object.c:562:14: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 562 | type = strchr(token_ref, '.'); | ^ ../../topology/pre-process-dapm.c:175:14: warning: assignment discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers] 175 | args = strchr(string, '.'); | ^ ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/322 Patch URL : https://github.com/alsa-project/alsa-utils/pull/322.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[GIT PULL] ASoC fixes for v7.0-rockchip-i2s-tdm
by Mark Brown 19 Feb '26

19 Feb '26

The following changes since commit f8f774913b4b599169381073f6674e20976e5529: ASoC: SOF: ipc4-control: Set correct error code in refresh_bytes_control (2026-02-09 12:27:28 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v7.0-rockchip-i2s-tdm for you to fetch changes up to 0783052534f547f8f201dd4554b1df9f1f8615b5: ASoC: rockchip: i2s-tdm: Use param rate if not provided by set_sysclk (2026-02-19 11:45:49 +0000) ---------------------------------------------------------------- ASoC: Additional Rockchip fix for v7.0 One more ASoC fix, for a regression with the Rockchip I2S TDM support on a number of platforms. ---------------------------------------------------------------- Alexandre Ferrieux (1): ASoC: codecs: aw88261: Fix erroneous bitmask logic in Awinic init Claudiu Beznea (1): ASoC: renesas: rz-ssi: Fix playback and capture Detlev Casanova (1): ASoC: rockchip: i2s-tdm: Use param rate if not provided by set_sysclk Gustavo Salvini (1): ASoC: amd: yc: Add DMI quirk for ASUS Vivobook Pro 15X M6501RR Jack Yu (1): ASoC: rt721-sdca: Fix issue of fail to detect OMTP jack type Luca Weiss (1): ASoC: qcom: sm8250: Add quinary MI2S support Mark Brown (1): ASoC: Merge up release Niranjan H Y (1): ASoC: tas2783A: add explicit port prepare handling Shengjiu Wang (4): ASoC: dt-bindings: asahi-kasei,ak5558: Reference common DAI properties ASoC: dt-bindings: asahi-kasei,ak4458: set unevaluatedProperties:false ASoC: dt-bindings: asahi-kasei,ak4458: Fix the supply names ASoC: dt-bindings: asahi-kasei,ak5558: Fix the supply names Vijendar Mukunda (2): ASoC: amd: acp: Add ACP7.0 match entries for Realtek parts ASoC: amd: amd_sdw: add machine driver quirk for Lenovo models Ziyi Guo (1): ASoC: fsl_xcvr: Revert fix missing lock in fsl_xcvr_mode_put() .mailmap | 4 +- Documentation/ABI/testing/sysfs-class-tsm | 10 -- Documentation/admin-guide/kernel-parameters.txt | 5 + .../bindings/sound/asahi-kasei,ak4458.yaml | 6 +- .../bindings/sound/asahi-kasei,ak5558.yaml | 9 +- .../devicetree/bindings/sound/ti,tlv320aic3x.yaml | 1 + MAINTAINERS | 53 ++++-- Makefile | 2 +- arch/arm/include/asm/string.h | 5 +- arch/x86/include/asm/kfence.h | 7 +- arch/x86/include/asm/vmware.h | 4 +- arch/x86/kvm/irq.c | 3 +- arch/x86/kvm/svm/avic.c | 4 +- arch/x86/kvm/svm/svm.c | 2 + arch/x86/kvm/vmx/vmx.c | 2 + arch/x86/kvm/x86.c | 30 ++-- arch/x86/kvm/x86.h | 2 + drivers/android/binder.c | 19 ++- drivers/android/binder/rust_binderfs.c | 17 +- drivers/android/binder/thread.rs | 109 +++++++----- drivers/android/binderfs.c | 8 +- drivers/block/loop.c | 45 ++--- drivers/block/rbd.c | 33 ++-- drivers/crypto/ccp/sev-dev-tsm.c | 15 +- drivers/firmware/cirrus/cs_dsp.c | 37 +++++ drivers/firmware/cirrus/cs_dsp.h | 18 ++ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 22 ++- .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 24 ++- drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c | 26 ++- .../firmware/cirrus/test/cs_dsp_test_wmfw_error.c | 24 ++- drivers/firmware/cirrus/test/cs_dsp_tests.c | 1 + drivers/gpio/gpio-loongson-64bit.c | 2 +- drivers/gpio/gpiolib-acpi-core.c | 1 + drivers/gpu/drm/amd/amdgpu/amdgpu_amdkfd_gpuvm.c | 14 +- drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 3 - drivers/gpu/drm/amd/amdgpu/mes_v11_0.c | 2 +- .../gpu/drm/amd/display/dc/dcn30/dcn30_cm_common.c | 37 ++++- .../drm/amd/display/dc/dwb/dcn30/dcn30_cm_common.h | 2 +- .../drm/amd/display/dc/hwss/dcn30/dcn30_hwseq.c | 9 +- .../drm/amd/display/dc/hwss/dcn32/dcn32_hwseq.c | 18 +- .../drm/amd/display/dc/hwss/dcn401/dcn401_hwseq.c | 16 +- drivers/gpu/drm/bridge/imx/imx8mp-hdmi-pai.c | 15 ++ drivers/gpu/drm/gma500/psb_irq.c | 36 ++-- drivers/gpu/drm/mgag200/mgag200_bmc.c | 31 ++-- drivers/gpu/drm/mgag200/mgag200_drv.h | 6 + drivers/gpu/drm/nouveau/include/nvif/client.h | 2 +- drivers/gpu/drm/nouveau/include/nvif/driver.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/device.h | 3 +- drivers/gpu/drm/nouveau/include/nvkm/core/engine.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/object.h | 5 +- drivers/gpu/drm/nouveau/include/nvkm/core/oproxy.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/subdev.h | 4 +- .../drm/nouveau/include/nvkm/core/suspend_state.h | 11 ++ drivers/gpu/drm/nouveau/include/nvkm/subdev/gsp.h | 6 + drivers/gpu/drm/nouveau/nouveau_display.c | 2 - drivers/gpu/drm/nouveau/nouveau_drm.c | 2 +- drivers/gpu/drm/nouveau/nouveau_nvif.c | 10 +- drivers/gpu/drm/nouveau/nvif/client.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/engine.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/ioctl.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/object.c | 20 ++- drivers/gpu/drm/nouveau/nvkm/core/oproxy.c | 2 +- drivers/gpu/drm/nouveau/nvkm/core/subdev.c | 18 +- drivers/gpu/drm/nouveau/nvkm/core/uevent.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/ce/ga100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/ce/priv.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/device/base.c | 22 ++- drivers/gpu/drm/nouveau/nvkm/engine/device/pci.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/device/priv.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/device/user.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/disp/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/disp/chan.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/falcon.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/fifo/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/fifo/uchan.c | 6 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/gf100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv04.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv10.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv20.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv20.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv40.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/mpeg/nv44.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/sec2/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/xtensa.c | 4 +- drivers/gpu/drm/nouveau/nvkm/subdev/acr/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/bar/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/clk/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/devinit/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/subdev/fault/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/fault/user.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gpio/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/gh100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/priv.h | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/fbsr.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/gsp.c | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/rpc.c | 6 + .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r570/fbsr.c | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r570/gsp.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/rm/rm.h | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/tu102.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/i2c/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/instmem/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/pci/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/pmu/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/therm/base.c | 6 +- drivers/gpu/drm/nouveau/nvkm/subdev/timer/base.c | 2 +- drivers/gpu/drm/xe/xe_guc.c | 6 +- drivers/gpu/drm/xe/xe_guc.h | 2 +- drivers/gpu/drm/xe/xe_migrate.c | 2 +- drivers/gpu/drm/xe/xe_pm.c | 13 +- drivers/gpu/drm/xe/xe_query.c | 2 +- drivers/gpu/drm/xe/xe_tlb_inval.c | 2 +- drivers/gpu/drm/xe/xe_tlb_inval_job.c | 2 +- drivers/hwmon/acpi_power_meter.c | 17 +- drivers/hwmon/dell-smm-hwmon.c | 8 + drivers/hwmon/gpio-fan.c | 6 +- drivers/hwmon/occ/common.c | 1 + drivers/i2c/busses/i2c-imx.c | 3 +- drivers/iommu/intel/pasid.h | 2 +- drivers/net/ethernet/adi/adin1110.c | 3 + drivers/net/ethernet/cavium/liquidio/lio_main.c | 39 ++--- drivers/net/ethernet/cavium/liquidio/lio_vf_main.c | 4 +- .../net/ethernet/freescale/dpaa2/dpaa2-switch.c | 10 ++ drivers/net/ethernet/freescale/enetc/enetc.c | 11 +- drivers/net/ethernet/freescale/enetc/enetc4_pf.c | 6 +- drivers/net/ethernet/freescale/enetc/enetc_cbdr.c | 4 - drivers/net/ethernet/freescale/enetc/enetc_hw.h | 17 +- drivers/net/ethernet/google/gve/gve_ethtool.c | 77 ++++++--- drivers/net/ethernet/google/gve/gve_main.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 1 - drivers/net/ethernet/intel/ice/ice_main.c | 26 +-- drivers/net/ethernet/intel/ice/ice_ptp.c | 179 ++++++++++++-------- drivers/net/ethernet/intel/ice/ice_ptp.h | 18 +- drivers/net/ethernet/spacemit/k1_emac.c | 21 ++- drivers/net/ethernet/stmicro/stmmac/stmmac_main.c | 3 +- drivers/net/ethernet/ti/cpsw.c | 41 ++++- drivers/net/ethernet/ti/cpsw_new.c | 34 +++- drivers/net/ethernet/ti/cpsw_priv.h | 1 + drivers/net/macvlan.c | 5 +- drivers/net/phy/sfp.c | 2 + drivers/net/usb/r8152.c | 29 ++-- drivers/net/wireless/intel/iwlwifi/mld/iface.c | 2 - drivers/net/wireless/intel/iwlwifi/mld/mac80211.c | 2 + drivers/net/wireless/intel/iwlwifi/mvm/d3.c | 6 +- drivers/nvme/host/pci.c | 45 +++-- drivers/nvme/target/tcp.c | 17 ++ drivers/pci/ide.c | 10 +- drivers/platform/x86/amd/pmc/pmc-quirks.c | 7 + drivers/platform/x86/classmate-laptop.c | 32 ++++ drivers/platform/x86/hp/hp-bioscfg/bioscfg.c | 5 + drivers/platform/x86/intel/plr_tpmi.c | 2 +- drivers/platform/x86/intel/telemetry/debugfs.c | 4 +- drivers/platform/x86/intel/telemetry/pltdrv.c | 2 +- drivers/platform/x86/intel/vsec.c | 2 + drivers/platform/x86/lg-laptop.c | 11 +- drivers/platform/x86/panasonic-laptop.c | 4 +- drivers/platform/x86/toshiba_haps.c | 2 +- drivers/pmdomain/imx/gpcv2.c | 8 +- drivers/pmdomain/imx/imx8m-blk-ctrl.c | 2 +- drivers/pmdomain/imx/imx8mp-blk-ctrl.c | 30 ++++ drivers/pmdomain/qcom/rpmpd.c | 2 +- drivers/regulator/spacemit-p1.c | 6 +- drivers/soc/qcom/smem.c | 5 +- drivers/spi/spi-tegra114.c | 3 + drivers/spi/spi-tegra20-slink.c | 6 +- drivers/spi/spi-tegra210-quad.c | 56 ++++++- drivers/usb/gadget/function/f_fs.c | 108 ++++++------ drivers/usb/gadget/function/u_fs.h | 2 +- drivers/virt/coco/tsm-core.c | 30 ---- fs/btrfs/raid56.c | 1 + fs/ceph/crypto.c | 9 +- fs/ceph/mds_client.c | 5 +- fs/ceph/mdsmap.c | 26 ++- fs/ceph/mdsmap.h | 1 + fs/ceph/super.h | 16 +- fs/proc/task_mmu.c | 42 +++-- fs/smb/client/cifstransport.c | 4 +- fs/smb/client/smb2file.c | 1 + include/linux/buildid.h | 3 + include/linux/ceph/ceph_fs.h | 6 + include/linux/livepatch.h | 3 + include/linux/pci-ide.h | 4 +- include/linux/rseq_types.h | 6 +- include/linux/skbuff.h | 12 ++ include/linux/tsm.h | 3 - include/trace/events/dma.h | 25 ++- io_uring/fdinfo.c | 13 +- io_uring/zcrx.c | 9 +- kernel/cgroup/dmem.c | 70 +++++++- kernel/dma/contiguous.c | 10 +- kernel/livepatch/core.c | 19 +++ kernel/liveupdate/luo_file.c | 2 - kernel/sched/core.c | 184 ++++++++++++++------- kernel/sched/ext.c | 48 ++++++ kernel/sched/sched.h | 44 ++++- kernel/trace/trace.h | 7 +- kernel/trace/trace_entries.h | 32 ++-- kernel/trace/trace_export.c | 21 ++- lib/buildid.c | 42 +++-- mm/memory-failure.c | 42 +++-- mm/shmem.c | 23 ++- mm/slub.c | 6 +- net/core/filter.c | 8 +- net/core/gro.c | 2 + net/core/link_watch.c | 20 ++- net/core/net-procfs.c | 50 ++++-- net/ethtool/common.c | 3 - net/ethtool/rss.c | 9 +- net/ipv6/ip6_fib.c | 3 +- net/netfilter/nf_tables_api.c | 2 +- net/sched/cls_u32.c | 13 +- net/tipc/crypto.c | 4 +- scripts/livepatch/init.c | 20 +-- scripts/livepatch/klp-build | 4 + scripts/module.lds.S | 9 +- security/lsm.h | 9 - security/lsm_init.c | 7 +- security/min_addr.c | 5 +- sound/drivers/aloop.c | 62 ++++--- sound/hda/codecs/realtek/alc269.c | 1 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 37 +++++ sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 + sound/soc/amd/yc/acp6x-mach.c | 15 +- sound/soc/codecs/aw88261.c | 3 +- sound/soc/codecs/cs42l43-jack.c | 37 ++++- sound/soc/codecs/rt1320-sdw.c | 2 + sound/soc/codecs/rt721-sdca.c | 4 +- sound/soc/codecs/tas2783-sdw.c | 43 +++++ sound/soc/intel/boards/sof_es8336.c | 9 + sound/soc/qcom/sm8250.c | 8 + sound/soc/renesas/rz-ssi.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 10 ++ sound/soc/sof/intel/hda.c | 5 +- sound/usb/mixer_quirks.c | 9 +- tools/objtool/check.c | 4 +- tools/objtool/include/objtool/klp.h | 10 +- tools/objtool/klp-diff.c | 41 ++++- tools/testing/selftests/kvm/Makefile.kvm | 1 + tools/testing/selftests/net/udpgro_fwd.sh | 64 +++++++ virt/kvm/eventfd.c | 44 ++--- 243 files changed, 2272 insertions(+), 1072 deletions(-) create mode 100644 drivers/firmware/cirrus/cs_dsp.h create mode 100644 drivers/gpu/drm/nouveau/include/nvkm/core/suspend_state.h

2 1

control: ucm: add ioctl to retrieve full card components
by GitHub pull_request - edited 19 Feb '26

19 Feb '26

alsa-project/alsa-lib pull request #494 was edited from mstrozek: The fixed-size components field in SNDRV_CTL_IOCTL_CARD_INFO can be too small on systems with many audio devices. The kernel [1] will provide a new ioctl to read the full string while truncating the original in card_info if it grows too big. Make sure alsa-lib can read the full string if the original is truncated. [1] link to v3 Request URL : https://github.com/alsa-project/alsa-lib/pull/494 Patch URL : https://github.com/alsa-project/alsa-lib/pull/494.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v7.0-merge-window
by Mark Brown 19 Feb '26

19 Feb '26

The following changes since commit f8f774913b4b599169381073f6674e20976e5529: ASoC: SOF: ipc4-control: Set correct error code in refresh_bytes_control (2026-02-09 12:27:28 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v7.0-merge-window for you to fetch changes up to 80ca113671a005430207d351cb403c1637106212: ASoC: dt-bindings: asahi-kasei,ak5558: Fix the supply names (2026-02-18 17:12:41 +0000) ---------------------------------------------------------------- ASoC: Fixes for v7.0 merge window A reasonably small set of fixes and quriks that came in during the merge window, there's one more pending that I'll send tomorrow if you didn't send a PR already. ---------------------------------------------------------------- Alexandre Ferrieux (1): ASoC: codecs: aw88261: Fix erroneous bitmask logic in Awinic init Claudiu Beznea (1): ASoC: renesas: rz-ssi: Fix playback and capture Gustavo Salvini (1): ASoC: amd: yc: Add DMI quirk for ASUS Vivobook Pro 15X M6501RR Jack Yu (1): ASoC: rt721-sdca: Fix issue of fail to detect OMTP jack type Luca Weiss (1): ASoC: qcom: sm8250: Add quinary MI2S support Mark Brown (1): ASoC: Merge up release Niranjan H Y (1): ASoC: tas2783A: add explicit port prepare handling Shengjiu Wang (4): ASoC: dt-bindings: asahi-kasei,ak5558: Reference common DAI properties ASoC: dt-bindings: asahi-kasei,ak4458: set unevaluatedProperties:false ASoC: dt-bindings: asahi-kasei,ak4458: Fix the supply names ASoC: dt-bindings: asahi-kasei,ak5558: Fix the supply names Vijendar Mukunda (2): ASoC: amd: acp: Add ACP7.0 match entries for Realtek parts ASoC: amd: amd_sdw: add machine driver quirk for Lenovo models Ziyi Guo (1): ASoC: fsl_xcvr: Revert fix missing lock in fsl_xcvr_mode_put() .mailmap | 4 +- Documentation/ABI/testing/sysfs-class-tsm | 10 -- Documentation/admin-guide/kernel-parameters.txt | 5 + .../bindings/sound/asahi-kasei,ak4458.yaml | 6 +- .../bindings/sound/asahi-kasei,ak5558.yaml | 9 +- .../devicetree/bindings/sound/ti,tlv320aic3x.yaml | 1 + MAINTAINERS | 53 ++++-- Makefile | 2 +- arch/arm/include/asm/string.h | 5 +- arch/x86/include/asm/kfence.h | 7 +- arch/x86/include/asm/vmware.h | 4 +- arch/x86/kvm/irq.c | 3 +- arch/x86/kvm/svm/avic.c | 4 +- arch/x86/kvm/svm/svm.c | 2 + arch/x86/kvm/vmx/vmx.c | 2 + arch/x86/kvm/x86.c | 30 ++-- arch/x86/kvm/x86.h | 2 + drivers/android/binder.c | 19 ++- drivers/android/binder/rust_binderfs.c | 17 +- drivers/android/binder/thread.rs | 109 +++++++----- drivers/android/binderfs.c | 8 +- drivers/block/loop.c | 45 ++--- drivers/block/rbd.c | 33 ++-- drivers/crypto/ccp/sev-dev-tsm.c | 15 +- drivers/firmware/cirrus/cs_dsp.c | 37 +++++ drivers/firmware/cirrus/cs_dsp.h | 18 ++ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 22 ++- .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 24 ++- drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c | 26 ++- .../firmware/cirrus/test/cs_dsp_test_wmfw_error.c | 24 ++- drivers/firmware/cirrus/test/cs_dsp_tests.c | 1 + drivers/gpio/gpio-loongson-64bit.c | 2 +- drivers/gpio/gpiolib-acpi-core.c | 1 + drivers/gpu/drm/amd/amdgpu/amdgpu_amdkfd_gpuvm.c | 14 +- drivers/gpu/drm/amd/amdgpu/amdgpu_drv.c | 3 - drivers/gpu/drm/amd/amdgpu/mes_v11_0.c | 2 +- .../gpu/drm/amd/display/dc/dcn30/dcn30_cm_common.c | 37 ++++- .../drm/amd/display/dc/dwb/dcn30/dcn30_cm_common.h | 2 +- .../drm/amd/display/dc/hwss/dcn30/dcn30_hwseq.c | 9 +- .../drm/amd/display/dc/hwss/dcn32/dcn32_hwseq.c | 18 +- .../drm/amd/display/dc/hwss/dcn401/dcn401_hwseq.c | 16 +- drivers/gpu/drm/bridge/imx/imx8mp-hdmi-pai.c | 15 ++ drivers/gpu/drm/gma500/psb_irq.c | 36 ++-- drivers/gpu/drm/mgag200/mgag200_bmc.c | 31 ++-- drivers/gpu/drm/mgag200/mgag200_drv.h | 6 + drivers/gpu/drm/nouveau/include/nvif/client.h | 2 +- drivers/gpu/drm/nouveau/include/nvif/driver.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/device.h | 3 +- drivers/gpu/drm/nouveau/include/nvkm/core/engine.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/object.h | 5 +- drivers/gpu/drm/nouveau/include/nvkm/core/oproxy.h | 2 +- drivers/gpu/drm/nouveau/include/nvkm/core/subdev.h | 4 +- .../drm/nouveau/include/nvkm/core/suspend_state.h | 11 ++ drivers/gpu/drm/nouveau/include/nvkm/subdev/gsp.h | 6 + drivers/gpu/drm/nouveau/nouveau_display.c | 2 - drivers/gpu/drm/nouveau/nouveau_drm.c | 2 +- drivers/gpu/drm/nouveau/nouveau_nvif.c | 10 +- drivers/gpu/drm/nouveau/nvif/client.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/engine.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/ioctl.c | 4 +- drivers/gpu/drm/nouveau/nvkm/core/object.c | 20 ++- drivers/gpu/drm/nouveau/nvkm/core/oproxy.c | 2 +- drivers/gpu/drm/nouveau/nvkm/core/subdev.c | 18 +- drivers/gpu/drm/nouveau/nvkm/core/uevent.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/ce/ga100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/ce/priv.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/device/base.c | 22 ++- drivers/gpu/drm/nouveau/nvkm/engine/device/pci.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/device/priv.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/device/user.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/disp/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/disp/chan.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/falcon.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/fifo/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/fifo/uchan.c | 6 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/gf100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv04.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv10.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv20.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv20.h | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/gr/nv40.c | 4 +- drivers/gpu/drm/nouveau/nvkm/engine/mpeg/nv44.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/sec2/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/engine/xtensa.c | 4 +- drivers/gpu/drm/nouveau/nvkm/subdev/acr/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/bar/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/clk/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/devinit/base.c | 4 +- drivers/gpu/drm/nouveau/nvkm/subdev/fault/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/fault/user.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gpio/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/gh100.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/priv.h | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/fbsr.c | 2 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/gsp.c | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r535/rpc.c | 6 + .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r570/fbsr.c | 8 +- .../gpu/drm/nouveau/nvkm/subdev/gsp/rm/r570/gsp.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/rm/rm.h | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/gsp/tu102.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/i2c/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/instmem/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/pci/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/pmu/base.c | 2 +- drivers/gpu/drm/nouveau/nvkm/subdev/therm/base.c | 6 +- drivers/gpu/drm/nouveau/nvkm/subdev/timer/base.c | 2 +- drivers/gpu/drm/xe/xe_guc.c | 6 +- drivers/gpu/drm/xe/xe_guc.h | 2 +- drivers/gpu/drm/xe/xe_migrate.c | 2 +- drivers/gpu/drm/xe/xe_pm.c | 13 +- drivers/gpu/drm/xe/xe_query.c | 2 +- drivers/gpu/drm/xe/xe_tlb_inval.c | 2 +- drivers/gpu/drm/xe/xe_tlb_inval_job.c | 2 +- drivers/hwmon/acpi_power_meter.c | 17 +- drivers/hwmon/dell-smm-hwmon.c | 8 + drivers/hwmon/gpio-fan.c | 6 +- drivers/hwmon/occ/common.c | 1 + drivers/i2c/busses/i2c-imx.c | 3 +- drivers/iommu/intel/pasid.h | 2 +- drivers/net/ethernet/adi/adin1110.c | 3 + drivers/net/ethernet/cavium/liquidio/lio_main.c | 39 ++--- drivers/net/ethernet/cavium/liquidio/lio_vf_main.c | 4 +- .../net/ethernet/freescale/dpaa2/dpaa2-switch.c | 10 ++ drivers/net/ethernet/freescale/enetc/enetc.c | 11 +- drivers/net/ethernet/freescale/enetc/enetc4_pf.c | 6 +- drivers/net/ethernet/freescale/enetc/enetc_cbdr.c | 4 - drivers/net/ethernet/freescale/enetc/enetc_hw.h | 17 +- drivers/net/ethernet/google/gve/gve_ethtool.c | 77 ++++++--- drivers/net/ethernet/google/gve/gve_main.c | 4 +- drivers/net/ethernet/intel/i40e/i40e_main.c | 1 - drivers/net/ethernet/intel/ice/ice_main.c | 26 +-- drivers/net/ethernet/intel/ice/ice_ptp.c | 179 ++++++++++++-------- drivers/net/ethernet/intel/ice/ice_ptp.h | 18 +- drivers/net/ethernet/spacemit/k1_emac.c | 21 ++- drivers/net/ethernet/stmicro/stmmac/stmmac_main.c | 3 +- drivers/net/ethernet/ti/cpsw.c | 41 ++++- drivers/net/ethernet/ti/cpsw_new.c | 34 +++- drivers/net/ethernet/ti/cpsw_priv.h | 1 + drivers/net/macvlan.c | 5 +- drivers/net/phy/sfp.c | 2 + drivers/net/usb/r8152.c | 29 ++-- drivers/net/wireless/intel/iwlwifi/mld/iface.c | 2 - drivers/net/wireless/intel/iwlwifi/mld/mac80211.c | 2 + drivers/net/wireless/intel/iwlwifi/mvm/d3.c | 6 +- drivers/nvme/host/pci.c | 45 +++-- drivers/nvme/target/tcp.c | 17 ++ drivers/pci/ide.c | 10 +- drivers/platform/x86/amd/pmc/pmc-quirks.c | 7 + drivers/platform/x86/classmate-laptop.c | 32 ++++ drivers/platform/x86/hp/hp-bioscfg/bioscfg.c | 5 + drivers/platform/x86/intel/plr_tpmi.c | 2 +- drivers/platform/x86/intel/telemetry/debugfs.c | 4 +- drivers/platform/x86/intel/telemetry/pltdrv.c | 2 +- drivers/platform/x86/intel/vsec.c | 2 + drivers/platform/x86/lg-laptop.c | 11 +- drivers/platform/x86/panasonic-laptop.c | 4 +- drivers/platform/x86/toshiba_haps.c | 2 +- drivers/pmdomain/imx/gpcv2.c | 8 +- drivers/pmdomain/imx/imx8m-blk-ctrl.c | 2 +- drivers/pmdomain/imx/imx8mp-blk-ctrl.c | 30 ++++ drivers/pmdomain/qcom/rpmpd.c | 2 +- drivers/regulator/spacemit-p1.c | 6 +- drivers/soc/qcom/smem.c | 5 +- drivers/spi/spi-tegra114.c | 3 + drivers/spi/spi-tegra20-slink.c | 6 +- drivers/spi/spi-tegra210-quad.c | 56 ++++++- drivers/usb/gadget/function/f_fs.c | 108 ++++++------ drivers/usb/gadget/function/u_fs.h | 2 +- drivers/virt/coco/tsm-core.c | 30 ---- fs/btrfs/raid56.c | 1 + fs/ceph/crypto.c | 9 +- fs/ceph/mds_client.c | 5 +- fs/ceph/mdsmap.c | 26 ++- fs/ceph/mdsmap.h | 1 + fs/ceph/super.h | 16 +- fs/proc/task_mmu.c | 42 +++-- fs/smb/client/cifstransport.c | 4 +- fs/smb/client/smb2file.c | 1 + include/linux/buildid.h | 3 + include/linux/ceph/ceph_fs.h | 6 + include/linux/livepatch.h | 3 + include/linux/pci-ide.h | 4 +- include/linux/rseq_types.h | 6 +- include/linux/skbuff.h | 12 ++ include/linux/tsm.h | 3 - include/trace/events/dma.h | 25 ++- io_uring/fdinfo.c | 13 +- io_uring/zcrx.c | 9 +- kernel/cgroup/dmem.c | 70 +++++++- kernel/dma/contiguous.c | 10 +- kernel/livepatch/core.c | 19 +++ kernel/liveupdate/luo_file.c | 2 - kernel/sched/core.c | 184 ++++++++++++++------- kernel/sched/ext.c | 48 ++++++ kernel/sched/sched.h | 44 ++++- kernel/trace/trace.h | 7 +- kernel/trace/trace_entries.h | 32 ++-- kernel/trace/trace_export.c | 21 ++- lib/buildid.c | 42 +++-- mm/memory-failure.c | 42 +++-- mm/shmem.c | 23 ++- mm/slub.c | 6 +- net/core/filter.c | 8 +- net/core/gro.c | 2 + net/core/link_watch.c | 20 ++- net/core/net-procfs.c | 50 ++++-- net/ethtool/common.c | 3 - net/ethtool/rss.c | 9 +- net/ipv6/ip6_fib.c | 3 +- net/netfilter/nf_tables_api.c | 2 +- net/sched/cls_u32.c | 13 +- net/tipc/crypto.c | 4 +- scripts/livepatch/init.c | 20 +-- scripts/livepatch/klp-build | 4 + scripts/module.lds.S | 9 +- security/lsm.h | 9 - security/lsm_init.c | 7 +- security/min_addr.c | 5 +- sound/drivers/aloop.c | 62 ++++--- sound/hda/codecs/realtek/alc269.c | 1 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 37 +++++ sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 + sound/soc/amd/yc/acp6x-mach.c | 15 +- sound/soc/codecs/aw88261.c | 3 +- sound/soc/codecs/cs42l43-jack.c | 37 ++++- sound/soc/codecs/rt1320-sdw.c | 2 + sound/soc/codecs/rt721-sdca.c | 4 +- sound/soc/codecs/tas2783-sdw.c | 43 +++++ sound/soc/intel/boards/sof_es8336.c | 9 + sound/soc/qcom/sm8250.c | 8 + sound/soc/renesas/rz-ssi.c | 2 +- sound/soc/sof/intel/hda.c | 5 +- sound/usb/mixer_quirks.c | 9 +- tools/objtool/check.c | 4 +- tools/objtool/include/objtool/klp.h | 10 +- tools/objtool/klp-diff.c | 41 ++++- tools/testing/selftests/kvm/Makefile.kvm | 1 + tools/testing/selftests/net/udpgro_fwd.sh | 64 +++++++ virt/kvm/eventfd.c | 44 ++--- 242 files changed, 2262 insertions(+), 1072 deletions(-) create mode 100644 drivers/firmware/cirrus/cs_dsp.h create mode 100644 drivers/gpu/drm/nouveau/include/nvkm/core/suspend_state.h

2 1

ucm2: tegra: wm8903: fix headphones conflicting device
by GitHub pull_request - opened 19 Feb '26

19 Feb '26

alsa-project/alsa-ucm-conf pull request #707 was opened from clamor-s: Fixes: ea7a065 ("ucm: fix SectionDevice identifiers") Please refer to this pull request for more info https://github.com/alsa-project/alsa-ucm-conf/pull/563 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/707 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/707.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI Z690 has the same problem. Can't get the SPDIF working, front panel doesn't work with microphone.
by GitHub issues - edited 19 Feb '26

19 Feb '26

alsa-project/alsa-ucm-conf issue #706 was edited from Ralfpf: Hope you could help me too, for i saw, that you helped another user with a MSI Z790 and a ALC4080. Her is the thread I read _Originally posted by @perexg in [#664](https://github.com/alsa-project/alsa-ucm-conf/issues/664#issuecomment-3779380548)_ This I posted already on another thread: > I am very new to this subject. I recently switched from Win10 to Linux, so please be patient with my lack of knowledge. I am using a MS690 Unify. Despite the fact, that I can find one in the USB-Audio.conf on line 83 "# 0db0:82c7 MSI MEG Z690I Unify", I still can't get my optical SPDIF working. If i choose digital out, the digital signal is routed to the analog output, which is very ugly to listen to. Also my microphone isn't working. It seems, as if the mic in is configured as sink for monitor out of any other source. sudo dmidecode -t 2 answers dmidecode 3.6 Getting SMBIOS data from sysfs. SMBIOS 3.6.0 present. Handle 0x0002, DMI type 2, 15 bytes Base Board Information Manufacturer: Micro-Star International Co., Ltd. Product Name: MEG Z690 UNIFY (MS-7D28) Version: 1.0 Serial Number: 07D2810_N81E356737 Asset Tag: Default string Features: Board is a hosting board Board is replaceable Location In Chassis: Default string Chassis Handle: 0x0003 Type: Motherboard Contained Object Handles: 0 Hope this is usefull. Please let me know, if you need more information. Of course I have an ALC4080 built in, but it's also recognised as USB-Audio. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/706 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] ASoC: amd: add ACPI mach entries and quirk
by Vijendar Mukunda 19 Feb '26

19 Feb '26

Add ACPI mach entries for ACP7.0 platform for RT1320 + RT722 combination and include machine driver quirk for Lenovo laptops. Vijendar Mukunda (2): ASoC: amd: acp: Add ACP7.0 match entries for Realtek parts ASoC: amd: amd_sdw: add machine driver quirk for Lenovo models sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 ++++++++++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 37 ++++++++++++++++++++++++ 2 files changed, 53 insertions(+) -- 2.45.2

2 3

MSI Z690 has the same problem. Can't get the SPDIF working, front panel doesn't work with micorphone.
by GitHub issues - opened 18 Feb '26

18 Feb '26

alsa-project/alsa-ucm-conf issue #706 was opened from Ralfpf: Hope you could help me too, for i saw, that you helped another user with a MSI Z790 and a ALC4080. Her is the thread I read _Originally posted by @perexg in [#664](https://github.com/alsa-project/alsa-ucm-conf/issues/664#issuecomment-3779380548)_ This I posted already on another thread: > I am very new to this subject. I recently switched from Win10 to Linux, so please be patient with my lack of knowledge. I am using a MS690 Unify. Despite the fact, that I can find one in the USB-Audio.conf on line 83 "# 0db0:82c7 MSI MEG Z690I Unify", I still can't get my optical SPDIF working. If i choose digital out, the digital signal is routed to the analog output, which is very ugly to listen to. Also my microphone isn't working. It seems, as if the mic in is configured as sink for monitor out of any other source. sudo dmidecode -t 2 answers dmidecode 3.6 Getting SMBIOS data from sysfs. SMBIOS 3.6.0 present. Handle 0x0002, DMI type 2, 15 bytes Base Board Information Manufacturer: Micro-Star International Co., Ltd. Product Name: MEG Z690 UNIFY (MS-7D28) Version: 1.0 Serial Number: 07D2810_N81E356737 Asset Tag: Default string Features: Board is a hosting board Board is replaceable Location In Chassis: Default string Chassis Handle: 0x0003 Type: Motherboard Contained Object Handles: 0 Hope this is usefull. Please let me know, if you need more information. Of course I have an ALC4080 built in, but it's also recognised as USB-Audio. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/706 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH v1] ASoC: tas2783A: add explicit port prepare handling
by Mark Brown 17 Feb '26

17 Feb '26

On Sat, 14 Feb 2026 16:17:10 +0530, Niranjan H Y wrote: > TAS2783a required port prepare bits to be set during playback > even when it is using simplified CP_SM. > > Normally, SoundWire core handles prepare sequencing automatically > depending on the type of the device available. For simplified CP_SM > there is no need to set the prepare bits. However, due to a hardware > limitation in TAS2783A, the port must still be explicitly prepared and > de-prepared by the driver to ensure reliable playback. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: tas2783A: add explicit port prepare handling commit: 0d68ad088e334088b031423e59886ae118c11202 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

1 0

Re: [PATCH] ASoC: amd: yc: Add DMI quirk for ASUS Vivobook Pro 15X M6501RR
by Mark Brown 17 Feb '26

17 Feb '26

On Tue, 10 Feb 2026 12:51:56 -0300, Gustavo Salvini wrote: > The ASUS Vivobook Pro 15X (M6501RR) with AMD Ryzen 9 6900HX has an > internal DMIC that is not detected without a DMI quirk entry, as the > BIOS does not set the AcpDmicConnected ACPI _DSD property. > > Adding the DMI entry enables the ACP6x DMIC machine driver to probe > successfully. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: amd: yc: Add DMI quirk for ASUS Vivobook Pro 15X M6501RR commit: ff9cadd1a2c0b2665b7377ac79540d66f212e7e3 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

1 0

ASUS X870-F STRIX - no front panel/headphones audio
by GitHub issues - edited 13 Feb '26

13 Feb '26

alsa-project/alsa-ucm-conf issue #705 was edited from dosiek: Hello, I have an issue where I can't get my headphones/front panel jack to work - it is being detected upon connecting the headphones but no audio is being transmitted. My motherboard is ASUS ROG STRIX X870-F GAMING WIFI, using the ALC4080. `Bus 001 Device 007: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio` I am using Fedora 43 KDE and I have tried a lot of different fixes (such as disabling drivers, modifying the ALC4080 and ALC4080-HiFi files and way more) and nothing has worked out. The motherboard is already in the UCM conf list for the ALC4080 (addressed as X870E-E but it's the same ID), but the front panel audio still does not work. I am at the state where I freshly installed the system, and updated the alsa-ucm-conf using the commands provided (even though the default version already had the ID), and getting this error in alsaucm -i `alsaucm>> listcards ALSA lib parser.c:270:(parse_syntax_field) Incompatible syntax 8 in USB-Audio.conf ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 ALSA lib parser.c:2978:(uc_mgr_scan_master_configs) Unable to open '-hw:0': Invalid argument alsaucm: error failed to get card list: Invalid argument` My alsaucm version is 1.2.14 (upgrading it to 1.2.15 does not fix it), and I am attaching the alsa-info.sh output. [alsa-info.txt](https://github.com/user-attachments/files/25296874/alsa-info… Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/705 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS X870-F STRIX - no front panel/headphones audio
by GitHub issues - opened 13 Feb '26

13 Feb '26

alsa-project/alsa-ucm-conf issue #705 was opened from dosiek: Hello, I have an issue where I can't get my headphones/front panel jack to work - it is being detected upon connecting the headphones but no audio is being transmitted. My motherboard is ASUS ROG STRIX X870-F GAMING WIFI, using the ALC4080. `Bus 001 Device 007: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio` I am using Fedora 43 KDE and I have tried a lot of different fixes (such as disabling drivers, modifying the ALC4080 and ALC4080-HiFi files and way more) and nothing has worked out. The motherboard is already in the UCM conf list for the ALC4080 (addressed as X870E-E but it's the same ID), but the front panel audio still does not work. I am at the state where I freshly installed the system, and updated the alsa-ucm-conf using the commands provided (even though the default version already had the ID), and getting this error in alsaucm -i `alsaucm>> listcards ALSA lib parser.c:270:(parse_syntax_field) Incompatible syntax 8 in USB-Audio.conf ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 ALSA lib parser.c:2978:(uc_mgr_scan_master_configs) Unable to open '-hw:0': Invalid argument alsaucm: error failed to get card list: Invalid argument` My alsaucm version is 1.2.14, and I am attaching the alsa-info.sh output. [alsa-info.txt](https://github.com/user-attachments/files/25296874/alsa-info… Thank you. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/705 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH] ALSA: hda/realtek: Fix speaker output on ASUS ROG Strix G512
by Takashi Iwai 13 Feb '26

13 Feb '26

On Fri, 13 Feb 2026 15:13:11 +0100, Rob Hermann wrote: > > The ASUS ROG Strix G512LW (subsystem ID 0x10431f21) with ALC294 codec > has no speaker output on Linux. The internal speaker amplifier requires > specific codec coefficient writes that differ from the existing > ALC294_FIXUP_ASUS_SPK quirk values. > > The required coefficients were identified by comparing codec register > dumps with and without working speaker output: > - Coefficient index 0x0f needs value 0x7778 (vs 0x7774 in existing quirk) > - Coefficient index 0x40 needs value 0x0800 (vs 0x8800 in existing quirk) > > Without this patch, speakers are completely silent. Headphones and HDMI > audio work correctly. > > This affects at least the following models: > - ASUS ROG Strix G512LW (i7-10750H, RTX 2070) > - ASUS ROG Strix G512LV (reported on Fedora forums) > > Link: https://discussion.fedoraproject.org/t/asus-rog-strix-g512lv-speakers-do-no… > Link: https://forum.endeavouros.com/t/asus-rog-strix-g15-g512lw-no-audio-through-… > Signed-off-by: Rob Hermann <supgeek(a)gmail.com> Could you resubmit to linux-sound(a)vger.kernel.org? That's the right ML for a kernel patch. alsa-devel ML is mainly for user-space stuff. thanks, Takashi

1 0

Qualcomm: qcs615: Remove JackControl from TALOS EVK HiFi config
by GitHub pull_request - opened 12 Feb '26

12 Feb '26

alsa-project/alsa-ucm-conf pull request #704 was opened from QTI-Quill: The EVK board does not support headset or jack detection. Keeping JackControl entries prevents PipeWire (wpctl) from exposing sinks and sources correctly. Remove JackControl from Headphones and Headset devices so PipeWire can enumerate playback and capture nodes normally. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/704 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/704.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Add support for CS42L43B variant
by GitHub pull_request - reopened 11 Feb '26

11 Feb '26

alsa-project/alsa-ucm-conf pull request #665 was reopened from mstrozek: From UCM point of view CS42L43B codec adds 2 new decimators and removes a control to change Decimator 1 mode from base CS42L43. Thus reuse CS42L43 files, and guard the CS42L43B additions behind a condition that the new Decimator controls exist. Link: https://lore.kernel.org/all/20251219100235.1247053-1-mstrozek@opensource.ci… Set as Draft PR until kernel support is merged. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/665 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/665.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt721-sdca: Fix issue of fail to detect OMTP jack type
by jack.yu＠realtek.com 10 Feb '26

10 Feb '26

From: Jack Yu <jack.yu(a)realtek.com> Add related HP-JD settings to fix issue of fail to detect OMTP jack type. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt721-sdca.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c index 8233532a1752..35960c225224 100644 --- a/sound/soc/codecs/rt721-sdca.c +++ b/sound/soc/codecs/rt721-sdca.c @@ -245,12 +245,12 @@ static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721) regmap_write(rt721->mbq_regmap, 0x5b10007, 0x2000); regmap_write(rt721->mbq_regmap, 0x5B10017, 0x1b0f); rt_sdca_index_write(rt721->mbq_regmap, RT721_CBJ_CTRL, - RT721_CBJ_A0_GAT_CTRL1, 0x2a02); + RT721_CBJ_A0_GAT_CTRL1, 0x2205); rt_sdca_index_write(rt721->mbq_regmap, RT721_CAP_PORT_CTRL, RT721_HP_AMP_2CH_CAL4, 0xa105); rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, RT721_UAJ_TOP_TCON14, 0x3b33); - regmap_write(rt721->mbq_regmap, 0x310400, 0x3023); + regmap_write(rt721->mbq_regmap, 0x310400, 0x3043); rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, RT721_UAJ_TOP_TCON14, 0x3f33); rt_sdca_index_write(rt721->mbq_regmap, RT721_VENDOR_ANA_CTL, -- 2.52.0

2 1

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

2 2

[GIT PULL] ASoC updates for v6.20
by Mark Brown 09 Feb '26

09 Feb '26

The following changes since commit 18f7fcd5e69a04df57b563360b88be72471d6b62: Linux 6.19-rc8 (2026-02-01 14:01:13 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.20 for you to fetch changes up to f8f774913b4b599169381073f6674e20976e5529: ASoC: SOF: ipc4-control: Set correct error code in refresh_bytes_control (2026-02-09 12:27:28 +0000) ---------------------------------------------------------------- ASoC: Updates for v7.0 This release is almost all abut driers, there's very little core work here, although some of that driver work is in more generic areas like SDCA and SOF: - Generic SDCA support for reporting jack events. - Continuing platform support, cleanup and feature improements for the AMD, Intel, Qualcomm and SOF code. - Platform description improvements for the Cirrus drivers. - Support for NXP i.MX952, Realtek RT1320 and RT5575, and Sophogo CV1800B. We also pulled in one small SPI API update and some more substantial regmap work (cache description improvements) for use in drivers. ---------------------------------------------------------------- Andy Shevchenko (2): ASoC: Fix acronym for Intel Gemini Lake ASoC: codecs: rt1320-sdw: Refactor to reduce stack frames AngeloGioacchino Del Regno (1): ASoC: dt-bindings: mt8192-afe-pcm: Fix clocks and clock-names Anton D. Stavinskii (7): ASoC: dt-bindings: sophgo,cv1800b: add I2S/TDM controller ASoC: sophgo: add CV1800B I2S/TDM controller driver ASoC: dt-bindings: sophgo,cv1800b: add ADC/DAC codec ASoC: sophgo: add CV1800B internal ADC codec driver ASoC: sophgo: add CV1800B internal DAC codec driver ASoC: sophgo: cv1800b: document DAC overwrite handling ASoC: sophgo: cv1800b: tidy Kconfig spacing Arnd Bergmann (6): ASoC: rt1320: fix 32-bit link failure ASoC: rt1320: fix size_t format string ASoC: rt5575: fix SPI dependency ASoC: sophgo: fix 64-bit division build failure ASoC: fsl_sai: add IMX_SCMI_MISC_DRV dependency ASoC: SOF: Intel: select CONFIG_SND_HDA_EXT_CORE from SND_SOC_SOF_HDA_COMMON Bard Liao (7): ASoC: soc-acpi-intel-arl-match: change rt722 amp endpoint to aggregated ASoC: soc-acpi-intel-ptl-match: use aggregated endpoint in ptl_rt722_l0_rt1320_l23 ASoC: sdw_utils: remove dai registered check ASoC: SOF: Intel: add hda_dsp_stream_pair_get/put helpers ASoC: SOF: Intel: add hda_dma_prepare/cleanup helpers ASoC: SOF: Intel: hda-sdw-bpt: support simultaneous audio and BPT streams ASoC: SOF: Intel: allow module parameter override BT link to 0 Bartosz Golaszewski (1): ASoC: davinci-mcasp: remove unneeded #ifdef Bharadwaj Raju (2): ASoC: dt-bindings: document dvdd-supply property for awinic,aw88261 ASoC: codecs: aw88261: use dvdd-supply regulator Biju Das (5): ASoC: renesas: rz-ssi: Use dev variable in probe() ASoC: renesas: rz-ssi: Remove trailing comma in the terminator entry ASoC: renesas: rz-ssi: Move DMA configuration ASoC: renesas: rz-ssi: Add support for 24 bits sample width ASoC: renesas: rz-ssi: Add support for 32 bits sample width Binbin Zhou (4): ASoC: codecs: es8323: Replace magic numbers with defined macros ASoC: codecs: es8323: remove DAC enablement write from es8323_set_bias_level ASoC: codecs: es8323: Enable proper DAPM widgets for capture ADC ASoC: codecs: es8323: Enable proper DAPM widgets for chip power Boris Faure (1): ASoC: sdca: Fix missing regmap dependencies in Kconfig Brahmajit Das (1): ASoC: Intel: avs: replace strcmp with sysfs_streq Cezary Rojewski (5): ASoC: Intel: catpt: Move IPC error messages one level down ASoC: Intel: catpt: Update CATPT_IPC_ERROR macro ASoC: Intel: catpt: Simplify catpt_stream_read_position() ASoC: Intel: catpt: Specify image names in the device descriptor ASoC: Intel: catpt: Drop superfluous space in PCM code Chancel Liu (2): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX94 ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX952 Charles Keepax (16): ASoC: SDCA: Factor out jack handling into new c file ASoC: SDCA: Add ability to connect SDCA jacks to ASoC jacks ASoC: SDCA: Add ASoC jack hookup in class driver ASoC: SDCA: Add SDCA IRQ enable/disable helpers ASoC: SDCA: Add basic system suspend support ASoC: SDCA: Device boot into the system suspend process ASoC: SDCA: Add lock to serialise the Function initialisation ASoC: SDCA: Tidy up some memory allocations ASoC: SDCA: Handle CONFIG_PM_SLEEP not being set ASoC: SDCA: Add NO_DIRECT_COMPLETE flag to class driver ASoC: SDCA: Remove outdated todo comment ASoC: SDCA: Handle volatile controls correctly ASoC: SDCA: Still process most of the jack detect if control is missing ASoC: SDCA: Rearrange FDL file messages ASoC: SDCA: Add regmap defaults for specification defined values ASoC: SDCA: Limit values user can write to Selected Mode Chen Ni (4): ASoC: codecs: wm8731: Remove unnecessary NULL check before clk_prepare_enable/clk_disable_unprepare ASoC: sun4i-spdif: Add missing check for devm_regmap_init_mmio ASoC: sunxi: sun50i-dmic: Add missing check for devm_regmap_init_mmio ASoC: codecs: max98390: Check return value of devm_gpiod_get_optional() in max98390_i2c_probe() ChiYuan Huang (3): ASoC: codecs: rtq9128: Modify the chip initial setting ASoC: dt-bindings: rtq9128: Add rtq9154 backward compatible ASoC: codecs: rtq9128: Add compatible changes for rtq9154 Chris Morgan (2): ASoC: dt-bindings: aw87390: Add Anbernic RG-DS Amplifier ASoC: codecs: aw87390: Add Anbernic RG-DS amp driver Claudiu Beznea (5): ASoC: codecs: da7213: Move comma operator at the end of the line ASoC: renesas: rz-ssi: Simplify the logic in rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Drop unnecessary if condition ASoC: renesas: rz-ssi: Drop the & operator in front of function name ASoC: renesas: rz-ssi: Drop goto label Cristian Ciocaltea (7): ASoC: nau8821: Fixup nau8821_enable_jack_detect() ASoC: nau8821: Cancel delayed work on component remove ASoC: nau8821: Cancel pending work before suspend ASoC: nau8821: Drop superfluous return statement ASoC: nau8821: Simplify conditional in nau8821_get_osr() ASoC: nau8821: Drop unneeded braces in nau8821_hw_params() ASoC: nau8821: Sort #include directives Cédric Bellegarde (1): ASoC: qcom: q6asm: drop DSP responses for closed data streams Dan Carpenter (1): ASoC: rt1320: Fix retry checking in rt1320_rae_load() David Lin (2): ASoC: Intel: ti-common: support tas2563 amplifier ASoC: Intel: sof_rt5682: add tas2563 speaker amp support Emil Svendsen (1): ASoC: tlv320adcx140: add channel sum control Emil-Juhl (4): ASoC: tlv320adcx140: power on/off the device on demand ASoC: tlv320adcx140: add avdd and iovdd supply ASoC: dt-bindings: clarify areg-supply documentation ASoC: tlv320adcx140: add kcontrol for num biquads Ethan Nelson-Moore (1): ASoC: wcd-mbhc-v2: remove unnecessary module_init/exit functions HariKrishna Sagala (5): ASoC: fsl: fsl_ssi: Replace deprecated strcpy() with strscpy() ASoC: codec: rt274: Use devm_request_threaded_irq to manage IRQ lifetime and fix smatch warning ASoC: mediatek: mt8195: optimize property formatting error handling by using scnprintf() ASoC: codec: rt298: Use devm_request_threaded_irq to manage IRQ lifetime and fix smatch warning ASoC: codec: rt286: Use devm_request_threaded_irq to manage IRQ lifetime and fix smatch warning Harshit Mogalapalli (1): ASoC: cs4271: Fix resource leak in cs4271_soc_resume() Hsieh Hung-En (7): ASoC: es8328: Remove duplicate DAPM routes ASoC: es8328: Propagate error codes from regmap updates ASoC: es8328: Fix DAC deemphasis control handling ASoC: es8328: Propagate errors in set_bias_level() ASoC: es8328: Check errors in set_dai_fmt() ASoC: es8328: Use cached regmap on resume ASoC: es8328: Add error unwind in resume Johan Hovold (2): ASoc: qcom: q6afe: use guards consistently ASoC: codecs: wcd939x-sdw: use devres for regmap allocation Jyri Sarha (4): ASoC: sof: ipc4-topology: Add topology tokens domain_in stack & heap_bytes ASoC: sof: Add domain_id, heap_bytes and stack_bytes to snd_sof_widget ASoC: SOF: ipc4: sof_ipc4_module_init_ext_init structs and macros ASoC: sof ipc4: Add sof_ipc4_widget_setup_msg_payload() and call it Krzysztof Kozlowski (20): ASoC: qcom: q6prm: Fix confusing cleanup.h syntax ASoC: qcom: q6asm: Fix confusing cleanup.h syntax ASoC: qcom: q6apm: Fix confusing cleanup.h syntax ASoC: qcom: q6afe: Fix confusing cleanup.h syntax ASoC: qcom: audioreach: Fix confusing cleanup.h syntax ASoC: qcom: Minor readability improve with new lines ASoC: amd: acp-sdw-legacy: Fix confusing cleanup.h syntax ASoC: amd: acp-sdw-sof: Fix confusing cleanup.h syntax ASoC: SDCA: functions: Fix confusing cleanup.h syntax ASoC: qcom: audioreach: Add support for Speaker Protection module ASoC: qcom: audioreach: Add support for VI Sense module ASoC: qcom: Constify APR callback response data soc: qcom: apr: Use typedef for GPR callback member ASoC: qcom: Constify GPR callback response data ASoC: qcom: Constify APR/GPR result structs ASoC: qcom: topology: Constify pointed topology and vendor structs ASoC: qcom: topology: Constify pointed ar control structs ASoC: qcom: topology: Constify pointed DAPM widget structs ASoC: qcom: topology: Constify pointed snd_soc_tplg_dapm_widget ASoC: qcom: audioreach: Constify function arguments Kuan-Wei Chiu (1): dt-bindings: sound: google,goldfish-audio: Convert to DT schema Kuninori Morimoto (15): ASoC: mediatek: mt8189-nau8825: don't use card->dapm directly ASoC: codecs: tas2780: tidyup format check in tas2780_set_fmt() ASoC: alc5623: tidyup clock inversion in alc5623_set_dai_fmt() ASoC: codecs: es8375: remove unnecessary format check ASoC: codecs: rt1320-sdw: convert to snd_soc_dapm_xxx() ASoC: intel: convert to snd_soc_dapm_xxx() ASoC: tlv320adcx140: don't use snd_soc_component_get_bias_level() ASoC: soc-component: remove snd_soc_component_xxx() wrapper ASoC: soc-component: remove compatibility definition for component ASoC: soc-dapm: remove compatibility definition for dapm ASoC: soc-dapm: remove dev from snd_soc_dapm_context() ASoC: soc-dapm: add snd_soc_dapm_alloc() ASoC: soc-dapm: move struct snd_soc_dapm_context ASoC: soc-dapm: tidyup function naming ASoC: soc-core: add lockdep_assert_held() at snd_soc_unregister_dai() Laurentiu Mihalcea (1): ASoC: dt-bindings: fsl,mqs: make gpr optional for SM-based SoCs Luca Weiss (2): ASoC: codecs: aw88261: Remove AW88261_I2C_NAME macro ASoC: codecs: aw88261: Add devicetree support Mac Chiang (3): ASoC: sdw_utils: Add quirk to ignore RT722 DMIC DAI ASoC: Intel: soc-acpi-intel-ptl-match: drop rt721 related match tables ASoC: SOF: Intel: hda: add SDCA property check Marco Crivellari (1): ASoC: SDCA: Replace use of system_wq with system_dfl_wq Mark Brown (41): ASoC: qcom: Fix confusing cleanup.h Add {24,32}-bit sample width support for RZ/G2L SSI ASoC: amd/sdw: Fix confusing cleanup.h ASoC: Intel: catpt: IPC log improvements and code ASoC: SOF: Support for on-demand DSP boot ASoC: SOF: ipoc4: Support for generic bytes ASoC: codecs: ES8326 : Add Kcontrol ASoC: Intel: add support for TAS2563 amplifier ASoC / soc/qcom: Constify APR/GPR callback response SDCA Jack Fixups ASoC: qcom: Constify few things in audioreach and ASoC: nau8821: Fixes and driver cleanup firmware: cirrus: cs_dsp: Add long-offset WMDR ASoC: rt5575: Add the codec driver for the ALC5575 ASoC: ES8389: Add some members and update ASoC: SOF: ipc4: Send heap/stack bytes via new ASoC: SOF: Use guard()/scoped_guard() for locks when sound: codecs: wm8962: Assorted fixes Add devicetree support for aw88261 amplifier driver & ASoC: Update rtq9128 document and source file ASoC: codecs: aw88261: add dvdd-supply property SDCA System Suspend Support sound: codecs: tlv320adcx140: assorted patches ASoC: codec: es8323: Enable DAPM for ADC and ChipPower ASoC: soc-acpi-intel-arl/ptl-match: use aggregated ASoC: renesas: rz-ssi: Cleanups ASoC: cs35l56: Add support for new Dell laptops ASoC: fsl_audmix: Support the i.MX952 platform ASoC: codec: Remove ak4641/pxa2xx-ac97 and convert to ASoC: capsuling struct snd_soc_dapm_context ASoC: sophgo: add CV1800 I2S controllers support regmap: reg_default_cb for flat cache defaults Anbernic RG-DS AW87391 Speaker Amps ASoC: amd: acp: Add ACP7.0 match entries for Cirrus ASoC: es8328: error handling and resume fixes ASoC: SOF: Intel: reserve link DMA for sdw bpt stream Minor SDCA Fixes ASoC: ti: davinci-mcasp: Add asynchronous mode ASoC: SOF: Support for echoref (virtual DAI) ASoC: cs35l56: More support for new Dell laptops ASoC: rockchip: spdif: Cleanups and port features Nathan Chancellor (1): ASoC: rt1320: Change return type of rt1320_t0_load() to void Niranjan H Y (9): ASoC: tas2783A: sdw_utils: support ch 3 & 4 ASoC: tas2783A: use custom firmware ASoC: tas2783A: update default init writes ASoC: tas2783A: fix error log for calibration data ASoc: tas2783A: fw name based on system details ASoc: tas2783A: acpi match for 4 channel for mtl ASoC: tas2783A: use acpi initialisation table ASoC: tas2783A: read slave properties from acpi table ASoC: tas2783A: fw loading for devices without pci bus Oder Chiou (4): spi: export of_find_spi_controller_by_node() spi: change of_find_spi_controller_by_node() gating to CONFIG_OF ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 Peng Fan (4): ASoC: codec: Remove ak4641 ASoC: codec: Remove pxa2xx-ac97.c ASoC: pxa2xx-ac97: Remove platform_data ASoC: ac97: Convert to GPIO descriptors Peter Ujfalusi (30): ASoC: SOF: ipc4-loader: Remove redundant rpm resume_and_get from load_library ASoC: SOF: control: skip rpm calls in ext_volatile_get if not implemented ASoC: SOF: Add support for on-demand DSP boot ASoC: SOF: sof-client: Add support for on-demand DSP boot ASoC: SOF: Intel: hda-sdw-bpt: Add support for on-demand DSP boot ASoC: SOF: Intel: pci-lnl: Set on_demand_dsp_boot for LNL ASoC: SOF: Intel: pci-ptl: Set on_demand_dsp_boot for PTL and WCL ASoC: SOF: Intel: pci-nvl: Set on_demand_dsp_boot for NVL-S ASoC: SOF: Intel: hda: Only check SSP MCLK mask in case of IPC3 ASoC: SOF: ipc4-control: If there is no data do not send bytes update ASoC: SOF: ipc4-topology: Correct the allocation size for bytes controls ASoC: SOF: ipc4-control: Use the correct size for scontrol->ipc_control_data ASoC: SOF: ipc4-control: Keep the payload size up to date ASoC: SOF: ipc4-topology: Set initial param_id for bytes control type ASoC: SOF: ipc4: Support for sending payload along with LARGE_CONFIG_GET ASoC: SOF: ipc4: Add definition for generic bytes control ASoC: SOF: ipc4-control: Add support for generic bytes control ASoC: SOF: Use guard()/scoped_guard() for mutex locks where it makes sense ASoC: SOF: Intel: Use guard()/scoped_guard() for mutex locks where it makes sense ASoC: SOF: amd: acp-ipc: Use guard() for spinlock_irq() ASoC: SOF: imx: imx-common: Use guard() for spinlock_irqsafe() ASoC: SOF: mediatek: mtk-adsp-common: Use guard() for spinlock_irqsave ASoC: SOF: Intel: Use guard() for spinlocks where it makes sense ASoC: SOF: ipc/ops: Use guard() for spinlocks ASoC: SOF: Intel: hda: Remove MODULE_SOFTDEP for snd-hda-codec-hdmi PCI: Add Intel Nova Lake audio Device ID ASoC: SOF: Intel: add support for Nova Lake NVL ALSA: hda: core: intel-dsp-config: Add support for NVL ALSA: hda: controllers: intel: add support for Nova Lake ASoC: SOF: ipc4-control: Set correct error code in refresh_bytes_control Rafael J. Wysocki (1): ASoC: rockchip: Discard pm_runtime_put() return value Randy Dunlap (2): ASoC: pxa: drop unused Kconfig symbol ASoC: amd: drop unused Kconfig symbols Ranjani Sridharan (10): ASoC: SOF: sof-audio: Add a new op in struct sof_ipc_tplg_ops ASoC: SOF: pcm: Split up widget prepare and setup uapi: sound: sof: tokens: Add missing token for KCPS ASoC: Intel: sof_sdw: Add a DAI link for loopback capture ASoC: SOF: ipc4-topology: Add new tokens for pipeline direction ASoC: SOF: ipc4-topology: Add support for process modules with no input pins ASoC: SOF: sof-audio: Traverse paths with aggregated DAI widgets ASoC: SOF: sof-audio: Add support for loopback capture ASoC: SOF: Intel: hda: Fix NULL pointer dereference ASoC: SOF: Intel: hda: Add a virtual CPU DAI Richard Fitzgerald (20): firmware: cs_dsp: Remove __free() from cs_dsp_debugfs_string_read() firmware: cs_dsp: Don't use __free() in cs_dsp_load() and cs_dsp_load_coeff() ASoC: cs-amp-lib: Replace __free(kfree) with normal kfree() cleanup firmware: cs_dsp: Handle long-offset data blocks firmware: cs_dsp: test_bin: Run test cases with v3 file format firmware: cs_dsp: test_bin: Make patch function a test parameter firmware: cs_dsp: mock_bin: Pass offset32 to cs_dsp_mock_bin_add_raw_block() firmware: cs_dsp: mock_bin: Add function to create long-offset patches firmware: cs_dsp: test: Increase size of XM and YM on Halo Core firmware: cs_dsp: test_bin: Run test cases on long-offset blocks firmware: cs_dsp: test_bin: Add tests for offsets > 0xffff ASoC: cs-amp-lib: Support Dell SSIDExV2 UEFI variable ASoC: cs35l56: Use vendor-specific qualifier in firmware file search ASoC: cs-amp-lib-test: Tests for reading SSIDExV2 ASoC: cs-amp-lib: Add a Kconfig symbol for enabling test hooks ASoC: cs35l56: Add KUnit testing of cs35l56_set_fw_suffix() ASoC: cs35l56-test: Fix missing module namespace import ASoC: cs35l56: Support for reading speaker ID from on-chip GPIOs ASoC: cs35l56-shared: KUnit tests for onchip speaker ID gpios ASoC: cs35l56: KUnit tests for parsing and using onchip GPIOs Rob Herring (Arm) (2): ASoC: dt-bindings: Convert realtek,rt5651 to DT schema ASoC: dt-bindings: Convert ti,tas2552 to DT schema Runrun Liu (1): ASoC: realtek: fix misspelling of "minimum" in comments Sascha Hauer (1): ASoC: dt-bindings: add avdd and iovdd supply Sebastian Andrzej Siewior (1): ASoC: codecs: rt56*: Remove IRQF_ONESHOT Sebastian Krzyszkowiak (2): ASoC: wm8962: Add WM8962_ADC_MONOMIX to "3D Coefficients" mask ASoC: wm8962: Don't report a microphone if it's shorted to ground on plug Sebastian Reichel (5): ASoC: rockchip: spdif: Use device_get_match_data() ASoC: rockchip: spdif: Move DT compatible table ASoC: rockchip: spdif: Fully convert to device managed resources ASoC: rockchip: spdif: Use dev_err_probe ASoC: rockchip: spdif: Convert to FIELD_PREP Sen Wang (4): ASoC: dt-bindings: davinci-mcasp: Add properties for asynchronous mode ASoC: ti: davinci-mcasp: Disambiguate mcasp_is_synchronous function ASoC: ti: davinci-mcasp: Streamline pdir behavior across rx & tx streams ASoC: ti: davinci-mcasp: Add asynchronous mode support Sheetal (4): regmap: Add reg_default_cb callback for flat cache defaults regmap: add KUnit coverage for reg_default_cb callback ASoC: tegra: Add AHUB writeable_reg for RX holes ASoC: tegra: set reg_default_cb callback Shenghao Ding (1): ASoC: tas2781: Put three different calibrated data solution into the same data structure Shengjiu Wang (9): ASoC: dt-bindings: fsl,audmix: Add support for i.MX952 platform ASoC: dt-bindings: fsl,sai: Add AUDMIX mode support on i.MX952 ASoC: fsl_audmix: Add support for i.MX952 platform ASoC: fsl_sai: Add AUDMIX mode support on i.MX952 ASoC: wm8962: add .set_tdm_slot callback function ASoC: dt-bindings: fsl,imx-asrc: Add support for i.MX952 platform ASoC: fsl_asrc_m2m: Add option to start ASRC before DMA device for M2M ASoC: fsl_asrc: Add support for i.MX952 platform ASoC: fsl_asrc_dma: allocate memory from dma device Shuming Fan (4): ASoC: rt1320: support calibration and temperature/r0 loading ASoC: rt1320: support RAE parameters loading ASoC: rt1320: fix the warning the string may be truncated ASoC: rt1320: fix the remainder calculation of r0 value Simon Trimmer (5): ASoC: SDCA: Allow sample width wild cards in set_usage() ASoC: amd: acp: Sort match table into most specific first ASoC: amd: acp: Rename Cirrus Logic component match entries to include link and uid ASoC: amd: acp: Sort Cirrus Logic match entries ASoC: amd: acp: Add ACP7.0 match entries for Cirrus Logic parts Sugar Zhang (5): ASoC: rockchip: spdif: Improve sample rate support ASoC: rockchip: spdif: Swap PCM and DAI component registration order ASoC: rockchip: spdif: Add support for set mclk rate ASoC: rockchip: spdif: Add support for format S32_LE ASoC: rockchip: spdif: Fill IEC958 CS info per params Syed Saba Kareem (1): ASoC: amd: acp: Fix Kconfig dependencies for SND_SOC_ACPI_AMD_SDCA_QUIRKS Thorsten Blum (1): ASoC: codecs: wm0010: Replace cpu_to_be64 + le64_to_cpu with swab64 Uwe Kleine-König (1): ASoC: sdw-mockup: Drop dummy remove function Vijendar Mukunda (2): ASoC: amd: acp: soc-acpi: add is_device_rt712_vb() helper ASoC: amd: maintainer information Zhang Yi (7): ASoC: codecs: ES8326: Add a kcontrol for PGAGAIN ASoC: codecs: ES8326: Add kcontrol for DRE ASoC: dt-bindings: ES8389: Add property about power supply ASoC: codecs: ES8389: Add members related to power supply ASoC: codecs: ES8389: Adjust wakeup configuration ASoC: codecs: ES8389: Add members about the version ASoC: codecs: ES8389: Update clock configuration Ziyi Guo (1): ASoC: fsl: imx-rpmsg: use snd_soc_find_dai_with_mutex() in probe .../devicetree/bindings/goldfish/audio.txt | 17 - .../devicetree/bindings/sound/awinic,aw87390.yaml | 34 +- .../devicetree/bindings/sound/awinic,aw88395.yaml | 13 + .../bindings/sound/davinci-mcasp-audio.yaml | 71 +- .../devicetree/bindings/sound/everest,es8389.yaml | 12 + .../devicetree/bindings/sound/fsl,audmix.yaml | 16 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 1 + .../devicetree/bindings/sound/fsl,mqs.yaml | 12 +- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 22 +- .../devicetree/bindings/sound/fsl,sai.yaml | 16 + .../bindings/sound/google,goldfish-audio.yaml | 38 + .../devicetree/bindings/sound/mt8192-afe-pcm.yaml | 176 ++- .../devicetree/bindings/sound/realtek,rt5575.yaml | 61 + .../devicetree/bindings/sound/realtek,rt5651.yaml | 100 ++ .../devicetree/bindings/sound/richtek,rtq9128.yaml | 12 +- Documentation/devicetree/bindings/sound/rt5651.txt | 63 - .../bindings/sound/sophgo,cv1800b-codecs.yaml | 46 + .../bindings/sound/sophgo,cv1800b-i2s.yaml | 67 ++ .../devicetree/bindings/sound/tas2552.txt | 36 - .../devicetree/bindings/sound/ti,tas2552.yaml | 62 + .../bindings/sound/ti,tlv320adcx140.yaml | 7 +- MAINTAINERS | 9 +- drivers/base/regmap/internal.h | 3 + drivers/base/regmap/regcache-flat.c | 19 + drivers/base/regmap/regcache.c | 3 +- drivers/base/regmap/regmap-kunit.c | 91 ++ drivers/base/regmap/regmap.c | 2 + drivers/firmware/cirrus/cs_dsp.c | 50 +- drivers/firmware/cirrus/test/cs_dsp_mock_bin.c | 38 +- .../firmware/cirrus/test/cs_dsp_mock_mem_maps.c | 8 +- drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c | 16 +- drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 1033 ++++++++++------ .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 14 +- drivers/spi/spi.c | 23 +- include/linux/firmware/cirrus/cs_dsp_test_utils.h | 6 +- include/linux/firmware/cirrus/wmfw.h | 7 +- include/linux/firmware/imx/sm.h | 2 + include/linux/pci_ids.h | 4 +- include/linux/platform_data/davinci_asp.h | 3 +- include/linux/regmap.h | 14 + include/linux/soc/qcom/apr.h | 6 +- include/linux/spi/spi.h | 9 + include/sound/ak4641.h | 23 - include/sound/cs-amp-lib.h | 3 + include/sound/cs35l56.h | 37 + include/sound/sdca_function.h | 4 + include/sound/sdca_interrupts.h | 7 + include/sound/sdca_jack.h | 32 + include/sound/soc-acpi-intel-ssp-common.h | 4 + include/sound/soc-component.h | 29 +- include/sound/soc-dapm.h | 44 +- include/sound/soc.h | 4 +- include/sound/sof.h | 3 + include/sound/sof/ipc4/header.h | 75 ++ include/sound/tas2781.h | 3 +- include/uapi/sound/sof/tokens.h | 6 + sound/arm/Kconfig | 10 - sound/arm/Makefile | 3 - sound/arm/pxa2xx-ac97-lib.c | 50 +- sound/arm/pxa2xx-ac97.c | 286 ----- sound/hda/codecs/side-codecs/tas2781_hda.c | 9 +- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 13 - sound/hda/controllers/intel.c | 3 +- sound/hda/core/intel-dsp-config.c | 8 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 2 - sound/soc/amd/acp/Kconfig | 9 +- sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-sdw-legacy-mach.c | 8 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 8 +- sound/soc/amd/acp/amd-acp70-acpi-match.c | 380 +++++- sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c | 42 + sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h | 25 + sound/soc/codecs/Kconfig | 50 +- sound/soc/codecs/Makefile | 9 +- sound/soc/codecs/ak4641.c | 641 ---------- sound/soc/codecs/alc5623.c | 5 - sound/soc/codecs/aw87390.c | 175 ++- sound/soc/codecs/aw87390.h | 86 ++ sound/soc/codecs/aw88261.c | 16 +- sound/soc/codecs/aw88261.h | 2 - sound/soc/codecs/cs-amp-lib-test.c | 169 +++ sound/soc/codecs/cs-amp-lib.c | 158 ++- sound/soc/codecs/cs35l56-shared-test.c | 680 +++++++++++ sound/soc/codecs/cs35l56-shared.c | 172 +++ sound/soc/codecs/cs35l56-test.c | 639 ++++++++++ sound/soc/codecs/cs35l56.c | 215 +++- sound/soc/codecs/cs35l56.h | 6 + sound/soc/codecs/cs4271.c | 12 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/es8323.c | 294 ++--- sound/soc/codecs/es8323.h | 105 +- sound/soc/codecs/es8326.c | 18 +- sound/soc/codecs/es8326.h | 3 + sound/soc/codecs/es8328.c | 160 ++- sound/soc/codecs/es8375.c | 2 - sound/soc/codecs/es8389.c | 246 ++-- sound/soc/codecs/es8389.h | 8 + sound/soc/codecs/max98390.c | 3 + sound/soc/codecs/nau8821.c | 48 +- sound/soc/codecs/nau8821.h | 1 + sound/soc/codecs/rt1320-sdw.c | 1227 +++++++++++++++++++- sound/soc/codecs/rt1320-sdw.h | 66 ++ sound/soc/codecs/rt274.c | 12 +- sound/soc/codecs/rt286.c | 12 +- sound/soc/codecs/rt298.c | 12 +- sound/soc/codecs/rt5575-spi.c | 118 ++ sound/soc/codecs/rt5575-spi.h | 27 + sound/soc/codecs/rt5575.c | 352 ++++++ sound/soc/codecs/rt5575.h | 58 + sound/soc/codecs/rt5640.c | 4 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5663.c | 4 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rtq9128.c | 114 +- sound/soc/codecs/sdw-mockup.c | 6 - sound/soc/codecs/tas2780.c | 31 +- sound/soc/codecs/tas2781-fmwlib.c | 138 ++- sound/soc/codecs/tas2781-i2c.c | 11 +- sound/soc/codecs/tas2783-sdw.c | 356 +++--- sound/soc/codecs/tas2783.h | 1 + sound/soc/codecs/tlv320adcx140.c | 136 ++- sound/soc/codecs/wcd-mbhc-v2.c | 12 - sound/soc/codecs/wcd939x-sdw.c | 19 +- sound/soc/codecs/wm0010.c | 2 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8962.c | 47 +- sound/soc/fsl/Kconfig | 6 +- sound/soc/fsl/fsl_asrc.c | 38 + sound/soc/fsl/fsl_asrc.h | 18 + sound/soc/fsl/fsl_asrc_common.h | 4 + sound/soc/fsl/fsl_asrc_dma.c | 48 +- sound/soc/fsl/fsl_asrc_m2m.c | 8 +- sound/soc/fsl/fsl_audmix.c | 3 + sound/soc/fsl/fsl_sai.c | 21 + sound/soc/fsl/fsl_sai.h | 4 + sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/intel/avs/board_selection.c | 2 +- sound/soc/intel/avs/core.c | 2 +- sound/soc/intel/avs/path.c | 2 +- sound/soc/intel/boards/Kconfig | 5 + sound/soc/intel/boards/Makefile | 3 + sound/soc/intel/boards/sof_rt5682.c | 6 + sound/soc/intel/boards/sof_sdw.c | 43 +- sound/soc/intel/boards/sof_ti_common.c | 77 ++ sound/soc/intel/boards/sof_ti_common.h | 24 + sound/soc/intel/catpt/core.h | 7 +- sound/soc/intel/catpt/device.c | 6 +- sound/soc/intel/catpt/ipc.c | 12 +- sound/soc/intel/catpt/loader.c | 8 +- sound/soc/intel/catpt/messages.c | 89 +- sound/soc/intel/catpt/pcm.c | 46 +- sound/soc/intel/catpt/sysfs.c | 2 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 23 +- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 14 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 55 +- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 3 + sound/soc/mediatek/mt8189/mt8189-nau8825.c | 6 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 44 +- sound/soc/pxa/Kconfig | 1 - sound/soc/pxa/pxa2xx-ac97.c | 5 +- sound/soc/qcom/qdsp6/audioreach.c | 229 +++- sound/soc/qcom/qdsp6/audioreach.h | 55 +- sound/soc/qcom/qdsp6/q6adm.c | 8 +- sound/soc/qcom/qdsp6/q6afe.c | 26 +- sound/soc/qcom/qdsp6/q6apm.c | 19 +- sound/soc/qcom/qdsp6/q6asm.c | 33 +- sound/soc/qcom/qdsp6/q6core.c | 6 +- sound/soc/qcom/qdsp6/q6prm.c | 22 +- sound/soc/qcom/qdsp6/topology.c | 118 +- sound/soc/renesas/rz-ssi.c | 143 ++- sound/soc/rockchip/Kconfig | 1 + sound/soc/rockchip/rockchip_sai.c | 5 +- sound/soc/rockchip/rockchip_spdif.c | 225 ++-- sound/soc/rockchip/rockchip_spdif.h | 57 +- sound/soc/sdca/Kconfig | 2 + sound/soc/sdca/Makefile | 2 +- sound/soc/sdca/sdca_asoc.c | 91 +- sound/soc/sdca/sdca_class.c | 34 + sound/soc/sdca/sdca_class.h | 2 + sound/soc/sdca/sdca_class_function.c | 141 ++- sound/soc/sdca/sdca_fdl.c | 8 +- sound/soc/sdca/sdca_functions.c | 75 +- sound/soc/sdca/sdca_interrupts.c | 188 +-- sound/soc/sdca/sdca_jack.c | 248 ++++ sound/soc/sdca/sdca_regmap.c | 14 +- sound/soc/sdca/sdca_ump.c | 2 +- sound/soc/sdw_utils/soc_sdw_ti_amp.c | 4 + sound/soc/sdw_utils/soc_sdw_utils.c | 39 +- sound/soc/soc-component.c | 82 -- sound/soc/soc-core.c | 10 + sound/soc/soc-dapm.c | 352 +++--- sound/soc/sof/amd/acp-ipc.c | 4 +- sound/soc/sof/compress.c | 8 + sound/soc/sof/control.c | 13 +- sound/soc/sof/core.c | 1 + sound/soc/sof/debug.c | 7 +- sound/soc/sof/imx/imx-common.c | 8 +- sound/soc/sof/intel/Kconfig | 1 + sound/soc/sof/intel/atom.c | 7 +- sound/soc/sof/intel/bdw.c | 7 +- sound/soc/sof/intel/cnl.c | 11 +- sound/soc/sof/intel/hda-codec.c | 1 - sound/soc/sof/intel/hda-dai-ops.c | 25 +- sound/soc/sof/intel/hda-dai.c | 22 +- sound/soc/sof/intel/hda-ipc.c | 11 +- sound/soc/sof/intel/hda-loader.c | 94 +- sound/soc/sof/intel/hda-mlink.c | 29 +- sound/soc/sof/intel/hda-sdw-bpt.c | 16 +- sound/soc/sof/intel/hda-stream.c | 168 ++- sound/soc/sof/intel/hda.c | 46 +- sound/soc/sof/intel/hda.h | 16 +- sound/soc/sof/intel/mtl.c | 5 +- sound/soc/sof/intel/nvl.c | 24 + sound/soc/sof/intel/pci-apl.c | 2 +- sound/soc/sof/intel/pci-lnl.c | 1 + sound/soc/sof/intel/pci-nvl.c | 32 + sound/soc/sof/intel/pci-ptl.c | 2 + sound/soc/sof/ipc.c | 9 +- sound/soc/sof/ipc3-dtrace.c | 7 +- sound/soc/sof/ipc3-topology.c | 6 +- sound/soc/sof/ipc3.c | 8 +- sound/soc/sof/ipc4-control.c | 198 +++- sound/soc/sof/ipc4-loader.c | 19 +- sound/soc/sof/ipc4-mtrace.c | 24 +- sound/soc/sof/ipc4-pcm.c | 3 +- sound/soc/sof/ipc4-topology.c | 214 +++- sound/soc/sof/ipc4-topology.h | 13 +- sound/soc/sof/ipc4.c | 64 +- sound/soc/sof/mediatek/mtk-adsp-common.c | 4 +- sound/soc/sof/ops.c | 34 +- sound/soc/sof/ops.h | 10 +- sound/soc/sof/pcm.c | 91 +- sound/soc/sof/pm.c | 149 ++- sound/soc/sof/sof-audio.c | 231 ++-- sound/soc/sof/sof-audio.h | 20 + sound/soc/sof/sof-client-ipc-flood-test.c | 7 +- sound/soc/sof/sof-client-ipc-kernel-injector.c | 4 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 14 +- sound/soc/sof/sof-client-probes.c | 26 +- sound/soc/sof/sof-client.c | 46 +- sound/soc/sof/sof-client.h | 3 + sound/soc/sof/sof-priv.h | 3 + sound/soc/sophgo/Kconfig | 47 + sound/soc/sophgo/Makefile | 5 + sound/soc/sophgo/cv1800b-sound-adc.c | 319 +++++ sound/soc/sophgo/cv1800b-sound-dac.c | 208 ++++ sound/soc/sophgo/cv1800b-tdm.c | 716 ++++++++++++ sound/soc/sunxi/sun4i-spdif.c | 4 + sound/soc/sunxi/sun50i-dmic.c | 3 + sound/soc/tegra/tegra186_asrc.c | 1 + sound/soc/tegra/tegra186_dspk.c | 1 + sound/soc/tegra/tegra210_admaif.c | 3 + sound/soc/tegra/tegra210_adx.c | 2 + sound/soc/tegra/tegra210_ahub.c | 60 + sound/soc/tegra/tegra210_ahub.h | 30 + sound/soc/tegra/tegra210_amx.c | 3 + sound/soc/tegra/tegra210_dmic.c | 1 + sound/soc/tegra/tegra210_i2s.c | 2 + sound/soc/tegra/tegra210_mbdrc.c | 1 + sound/soc/tegra/tegra210_mixer.c | 1 + sound/soc/tegra/tegra210_mvc.c | 1 + sound/soc/tegra/tegra210_ope.c | 1 + sound/soc/tegra/tegra210_peq.c | 1 + sound/soc/tegra/tegra210_sfc.c | 1 + sound/soc/ti/davinci-mcasp.c | 517 +++++++-- sound/soc/ti/davinci-mcasp.h | 10 + 272 files changed, 12608 insertions(+), 4067 deletions(-) delete mode 100644 Documentation/devicetree/bindings/goldfish/audio.txt create mode 100644 Documentation/devicetree/bindings/sound/google,goldfish-audio.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5651.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5651.txt create mode 100644 Documentation/devicetree/bindings/sound/sophgo,cv1800b-codecs.yaml create mode 100644 Documentation/devicetree/bindings/sound/sophgo,cv1800b-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 Documentation/devicetree/bindings/sound/ti,tas2552.yaml delete mode 100644 include/sound/ak4641.h create mode 100644 include/sound/sdca_jack.h delete mode 100644 sound/arm/pxa2xx-ac97.c create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h delete mode 100644 sound/soc/codecs/ak4641.c create mode 100644 sound/soc/codecs/cs35l56-shared-test.c create mode 100644 sound/soc/codecs/cs35l56-test.c create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h create mode 100644 sound/soc/intel/boards/sof_ti_common.c create mode 100644 sound/soc/intel/boards/sof_ti_common.h create mode 100644 sound/soc/sdca/sdca_jack.c create mode 100644 sound/soc/sophgo/Kconfig create mode 100644 sound/soc/sophgo/Makefile create mode 100644 sound/soc/sophgo/cv1800b-sound-adc.c create mode 100644 sound/soc/sophgo/cv1800b-sound-dac.c create mode 100644 sound/soc/sophgo/cv1800b-tdm.c

2 1

[PATCH] ASoC: amd: maintainer information
by Vijendar Mukunda 08 Feb '26

08 Feb '26

Update MAINTAINERS file for AMD ASoC drivers. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- MAINTAINERS | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4a48a11072e4..f27485e54645 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1030,6 +1030,13 @@ L: dmaengine(a)vger.kernel.org S: Supported F: drivers/dma/amd/ae4dma/ +AMD ASoC DRIVERS +M: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> +R: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> +L: linux-sound(a)vger.kernel.org +S: Supported +F: sound/soc/amd/ + AMD AXI W1 DRIVER M: Kris Chaplin <kris.chaplin(a)amd.com> R: Thomas Delev <thomas.delev(a)amd.com> -- 2.45.2

2 1

alsa-ucm-conf upstream update
by GitHub pull_request - opened 06 Feb '26

06 Feb '26

alsa-project/alsa-ucm-conf pull request #703 was opened from jackpot51: Updates to latest version in Ubuntu 26.04, should be compatible with 22.04 and 24.04. It would be good to do audio testing of: - thelio-mega-r2: Upstreamed :heavy_check_mark: - thelio-r3: Not upstreamed :x: - thelio-b5: Not upstreamed :x: - thelio-major-r5: Not upstreamed :x: - thelio-r4: Upstreamed :heavy_check_mark: Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/703 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/703.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: MOTU M6 - fix playback channels
by GitHub pull_request - opened 05 Feb '26

05 Feb '26

alsa-project/alsa-ucm-conf pull request #702 was opened from puleglot: There is only one incorrect instance of playback HWChannels. Fix it. Closes: #700 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/702 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/702.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Motu: M6-HiFi.conf - m6 has only 4 outputs, not 6
by GitHub pull_request - opened 05 Feb '26

05 Feb '26

alsa-project/alsa-ucm-conf pull request #701 was opened from damex: hi, here is the fix for https://github.com/alsa-project/alsa-ucm-conf/issues/700 motu m6 has 6 input and 4 outputs. https://motu.com/products/m-series/m6 'The M6 is a 6-in/4-out USB-C audio interface...' in config it is listed as 6 outputs and that throws ALSA UCM error. adjusting configuration on working system makes error go away. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/701 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/701.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MOTU M6 reports [ALSA UCM error] but seems to be working fine.
by GitHub issues - opened 05 Feb '26

05 Feb '26

alsa-project/alsa-ucm-conf issue #700 was opened from damex: hi, i did fresh fedora 43 install and noticed that my MOTU m6 reports ALSA UCM error on all inputs and outputs but those inputs and outputs work just fine. <img width="659" height="381" alt="Image" src="https://github.com/user-attachments/assets/cdce7782-7581-4c05-b501-2e22d592…" /> <img width="819" height="206" alt="Image" src="https://github.com/user-attachments/assets/2290dd4b-f0af-4f75-a58b-6b5a056e…" /> in some software it is difficult to choose correct input due to presence of this error in device name. i did find https://github.com/alsa-project/alsa-ucm-conf/issues/552 this report to be similar to mine. attaching full alsa info output here (it was requested in that report) [alsa-info.txt](https://github.com/user-attachments/files/25109878/alsa-info… please let me know if i you need any more information from me. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/700 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

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

1 1

[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

4 5

[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(-)

3 2

[PATCH] ASoC: amd: acp: soc-acpi: add is_device_rt712_vb() helper
by Vijendar Mukunda 13 Jan '26

13 Jan '26

Add a filter to skip the RT172 VB configuration if a SmartMic Function is not found in the SDCA descriptors. If the ACPI information is incorrect this can only be quirked further with DMI information. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/Kconfig | 6 +++ sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 50 ++++++++++++++++++++ sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c | 42 ++++++++++++++++ sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h | 14 ++++++ 5 files changed, 114 insertions(+) create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c create mode 100644 sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index c2a60bc80ee6..ed2fa055f7f6 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -15,8 +15,14 @@ config SND_SOC_AMD_ACP_COMMON config SND_SOC_ACPI_AMD_MATCH tristate + select SND_SOC_ACPI_AMD_SDCA_QUIRKS select SND_SOC_ACPI if ACPI +config SND_SOC_ACPI_AMD_SDCA_QUIRKS + tristate + depends on ACPI + depends on SND_SOC_SDCA + if SND_SOC_AMD_ACP_COMMON config SND_SOC_AMD_ACP_PDM diff --git a/sound/soc/amd/acp/Makefile b/sound/soc/amd/acp/Makefile index 08220b9a3802..81d23aded348 100644 --- a/sound/soc/amd/acp/Makefile +++ b/sound/soc/amd/acp/Makefile @@ -27,6 +27,7 @@ snd-soc-acpi-amd-match-y := amd-acp63-acpi-match.o amd-acp70-acpi-match.o snd-acp-sdw-mach-y := acp-sdw-mach-common.o snd-acp-sdw-sof-mach-y += acp-sdw-sof-mach.o snd-acp-sdw-legacy-mach-y += acp-sdw-legacy-mach.o +snd-soc-acpi-amd-sdca-quirks-y += soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_SOC_AMD_ACP_PCM) += snd-acp-pcm.o obj-$(CONFIG_SND_SOC_AMD_ACP_I2S) += snd-acp-i2s.o @@ -40,6 +41,7 @@ obj-$(CONFIG_SND_AMD_ASOC_REMBRANDT) += snd-acp-rembrandt.o obj-$(CONFIG_SND_AMD_ASOC_ACP63) += snd-acp63.o obj-$(CONFIG_SND_AMD_ASOC_ACP70) += snd-acp70.o +obj-$(CONFIG_SND_SOC_ACPI_AMD_SDCA_QUIRKS) += snd-soc-acpi-amd-sdca-quirks.o obj-$(CONFIG_SND_AMD_SOUNDWIRE_ACPI) += snd-amd-sdw-acpi.o obj-$(CONFIG_SND_SOC_AMD_MACH_COMMON) += snd-acp-mach.o obj-$(CONFIG_SND_SOC_AMD_LEGACY_MACH) += snd-acp-legacy-mach.o diff --git a/sound/soc/amd/acp/amd-acp70-acpi-match.c b/sound/soc/amd/acp/amd-acp70-acpi-match.c index 871b4f054a84..fa39f18578ca 100644 --- a/sound/soc/amd/acp/amd-acp70-acpi-match.c +++ b/sound/soc/amd/acp/amd-acp70-acpi-match.c @@ -7,6 +7,7 @@ */ #include <sound/soc-acpi.h> +#include "soc-acpi-amd-sdca-quirks.h" #include "../mach-config.h" static const struct snd_soc_acpi_endpoint single_endpoint = { @@ -44,6 +45,39 @@ static const struct snd_soc_acpi_endpoint spk_3_endpoint = { .group_id = 1 }; +static const struct snd_soc_acpi_endpoint jack_amp_g1_dmic_endpoints[] = { + /* Jack Endpoint */ + { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, + /* Amp Endpoint, work as spk_l_endpoint */ + { + .num = 1, + .aggregated = 1, + .group_position = 0, + .group_id = 1, + }, + /* DMIC Endpoint */ + { + .num = 2, + .aggregated = 0, + .group_position = 0, + .group_id = 0, + }, +}; + +static const struct snd_soc_acpi_adr_device rt712_vb_1_group1_adr[] = { + { + .adr = 0x000130025D071201ull, + .num_endpoints = ARRAY_SIZE(jack_amp_g1_dmic_endpoints), + .endpoints = jack_amp_g1_dmic_endpoints, + .name_prefix = "rt712" + } +}; + static const struct snd_soc_acpi_adr_device rt711_rt1316_group_adr[] = { { .adr = 0x000030025D071101ull, @@ -254,6 +288,15 @@ static const struct snd_soc_acpi_link_adr acp70_cs35l56x4_l1[] = { {} }; +static const struct snd_soc_acpi_link_adr acp70_alc712_vb_l1[] = { + { + .mask = BIT(1), + .num_adr = ARRAY_SIZE(rt712_vb_1_group1_adr), + .adr_d = rt712_vb_1_group1_adr, + }, + {} +}; + static const struct snd_soc_acpi_link_adr acp70_rt722_only[] = { { .mask = BIT(0), @@ -308,6 +351,12 @@ struct snd_soc_acpi_mach snd_soc_acpi_amd_acp70_sdw_machines[] = { .links = acp70_cs35l56x4_l1, .drv_name = "amd_sdw", }, + { + .link_mask = BIT(1), + .links = acp70_alc712_vb_l1, + .machine_check = snd_soc_acpi_amd_sdca_is_device_rt712_vb, + .drv_name = "amd_sdw", + }, {}, }; EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sdw_machines); @@ -327,3 +376,4 @@ EXPORT_SYMBOL(snd_soc_acpi_amd_acp70_sof_sdw_machines); MODULE_DESCRIPTION("AMD ACP7.0 & ACP7.1 tables and support for ACPI enumeration"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Vijendar.Mukunda(a)amd.com"); +MODULE_IMPORT_NS("SND_SOC_ACPI_AMD_SDCA_QUIRKS"); diff --git a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c new file mode 100644 index 000000000000..63bf9e3c0ae1 --- /dev/null +++ b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.c @@ -0,0 +1,42 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-acpi-amd-sdca-quirks.c - tables and support for SDCA quirks + * + * Copyright(c) 2025 Advanced Micro Devices, Inc. All rights reserved. + * + */ + +#include <linux/soundwire/sdw_amd.h> +#include <sound/sdca.h> +#include <sound/soc-acpi.h> +#include "soc-acpi-amd-sdca-quirks.h" + +/* + * Pretend machine quirk. The argument type is not the traditional + * 'struct snd_soc_acpi_mach' pointer but instead the sdw_amd_ctx + * which contains the peripheral information required for the + * SoundWire/SDCA filter on the SMART_MIC setup and interface + * revision. When the return value is false, the entry in the + * 'snd_soc_acpi_mach' table needs to be skipped. + */ +bool snd_soc_acpi_amd_sdca_is_device_rt712_vb(void *arg) +{ + struct sdw_amd_ctx *ctx = arg; + int i; + + if (!ctx) + return false; + + for (i = 0; i < ctx->peripherals->num_peripherals; i++) { + if (sdca_device_quirk_match(ctx->peripherals->array[i], + SDCA_QUIRKS_RT712_VB)) + return true; + } + + return false; +} +EXPORT_SYMBOL_NS(snd_soc_acpi_amd_sdca_is_device_rt712_vb, "SND_SOC_ACPI_AMD_SDCA_QUIRKS"); + +MODULE_DESCRIPTION("ASoC ACPI AMD SDCA quirks"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS("SND_SOC_SDCA"); diff --git a/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h new file mode 100644 index 000000000000..7e345a236da1 --- /dev/null +++ b/sound/soc/amd/acp/soc-acpi-amd-sdca-quirks.h @@ -0,0 +1,14 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * soc-acpi-amd-sdca-quirks.h - tables and support for SDCA quirks + * + * Copyright(c) 2025 Advanced Micro Devices, Inc. All rights reserved. + * + */ + +#ifndef _SND_SOC_ACPI_AMD_SDCA_QUIRKS +#define _SND_SOC_ACPI_AMD_SDCA_QUIRKS + +bool snd_soc_acpi_amd_sdca_is_device_rt712_vb(void *arg); + +#endif -- 2.45.2

5 5

UCM HiFi profile fails on hardware with Master volume instead of Headphone volume controls (ASUS ZenBook UX425EA)
by GitHub issues - opened 13 Jan '26

13 Jan '26

alsa-project/alsa-ucm-conf issue #687 was opened from akincano: ### Hardware Information - Laptop: ASUS ZenBook UX425EA_UX425EA - Audio Controller: Intel Tiger Lake-LP Smart Sound Technology - Audio Controller (PCI ID: 0x8086:0xa0c8) - Codec: Realtek ALC294 (Vendor ID: 0x10ec0294, Subsystem ID: 0x10431cee) - Kernel: 6.18.5.arch1-1 - Distribution: Arch Linux ### Software Versions ```bash alsa-ucm-conf 1.2.15.2-1 sof-firmware 2025.12-1 linux 6.18.5.arch1-1 ``` ### Problem Description The SOF HiFi profile fails to create speaker/headphone audio devices on this hardware. WirePlumber reports: ```bash spa.alsa: No sink and source at HiFi: Headphones ``` The HiFi profile shows as available: no and only HDMI outputs are exposed, with no analog audio devices. ### Root Cause **This is a UCM configuration issue**, not a kernel driver issue. The problem is that /usr/share/alsa/ucm2/HDA/HiFi-analog.conf checks for ALSA controls that don't exist on this hardware. **ALSA controls exposed by this hardware (identical for both SOF and HDA):** ```bash numid=3,iface=CARD,name='Speaker Phantom Jack' numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' [HDMI/IEC958 controls omitted for brevity] ``` **Controls the UCM config expects but don't exist:** - Headphone Playback Volume ❌ (only Master Playback Volume exists) - Headphone Playback Switch ❌ (only Master Playback Switch exists) - Headphone Jack ❌ (only Speaker Phantom Jack exists) **UCM config logic that fails:** In /usr/share/alsa/ucm2/HDA/HiFi-analog.conf: ```bash If.hpvol { Condition { Type ControlExists Control "name='Headphone Playback Volume'" } False.Define.hpvol "" } If.hp { Condition { Type String Empty "${var:hpvol}" } False.SectionDevice."Headphones" { # ... creates headphone device } } ``` Because Headphone Playback Volume doesn't exist, hpvol is set to empty string, and the If.hp condition prevents the headphone device from being created. Similarly, the jack detection logic checks for: ```bash Define.hpjack "Headphone Jack" If.hpjack { Condition { Type ControlExists Control "iface=CARD,name='Headphone Mic Jack'" } # ... falls back to checking 'Front Headphone Jack' } ``` But never checks for Speaker Phantom Jack, which is what this hardware exposes. ### Working Configuration (Legacy HDA Driver) With the legacy snd_hda_intel driver (forced via blacklisting SOF), PulseAudio's built-in profile logic correctly handles this hardware: **Working profile:** output:analog-stereo+input:analog-stereo **Exposed ports:** - analog-output-speaker: Speakers (uses Master Playback Volume/Switch) - analog-output-headphones: Headphones (uses Master Playback Volume/Switch) - Automatic jack detection via Speaker Phantom Jack The legacy HDA driver works because PulseAudio has fallback logic that handles hardware with only Master volume controls, but SOF's UCM-based configuration doesn't have this fallback. ### Proposed Solution The UCM configuration needs to be updated to handle hardware that only exposes Master Playback Volume/Switch instead of dedicated Headphone controls. **Option 1: Add fallback logic to HiFi-analog.conf** Check for Master Playback Volume when Headphone Playback Volume doesn't exist: ```bash If.hpvol { Condition { Type ControlExists Control "name='Headphone Playback Volume'" } False.If.master { Condition { Type ControlExists Control "name='Master Playback Volume'" } True.Define.hpvol "Master" } } ``` Similarly for jack detection, check Speaker Phantom Jack: ```bash If.hpjack { Condition { Type ControlExists Control "iface=CARD,name='Headphone Mic Jack'" } True.Define.hpjack "Headphone Mic Jack" False.If.front { Condition { Type ControlExists Control "iface=CARD,name='Front Headphone Jack'" } True.Define.hpjack "Front Headphone Jack" False.If.speaker { Condition { Type ControlExists Control "iface=CARD,name='Speaker Phantom Jack'" } True.Define.hpjack "Speaker Phantom Jack" } } } ``` **Option 2: Create a specific profile for this hardware** Create a UCM profile specifically for Tiger Lake systems with Realtek ALC294 that only expose Master controls. **Option 3: Update SOF topology** If this is a SOF topology issue, update the topology to expose proper Headphone controls instead of just Master. ### Additional Debug Information **Pin 0x21 configuration (headphone jack):** ```bash Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Pincap 0x0001001c: OUT HP EAPD Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 ``` The pin is capable of jack detection (EAPD Detect in capabilities) but is misconfigured as "N/A" with "NO_PRESENCE". However, the jack detection still works via the Speaker Phantom Jack ALSA control. **Full codec information:** Available upon request **Full WirePlumber logs:** Available upon request ### Impact This affects all ASUS ZenBook UX425EA users (and potentially other laptops with similar hardware configurations) running SOF. Users must either: 1. Use pro-audio profile (no automatic headphone/speaker switching) 2. Blacklist SOF and force legacy HDA driver (not a long-term solution) ### Related Issues This was initially reported in #686 which was closed as a kernel issue, but further investigation revealed it's actually a UCM configuration problem. The kernel driver is exposing the ALSA controls correctly; the UCM config just doesn't know how to handle this hardware configuration. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/687 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALSA 1.2.8 release
by Jaroslav Kysela 13 Jan '26

13 Jan '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.

2 12

snd_seq_drain_output change breaks existing code.
by GitHub issues - edited 13 Jan '26

13 Jan '26

alsa-project/alsa-lib issue #493 was edited from Hoosier-Beagler: The little gem shown below has broken a lot of code. Previously, no positive results were returned and software relied on -EAGAIN to detect remaining data. That result is no longer returned if some data was processed. This has broken the old and stable version of my application written in C(++) which is no longer being maintained. I have ported my app to python but this change has also broken public python wrappers for the alsa library. In [alsa-python / pyalsa / alsaseq.c](https://github.com/alsa-project/alsa-python/blob/master/pyalsa/a… below line 2800 and also [python-alsa-midi / alsa_midi / client.py](https://github.com/Jajcus/python-alsa-midi/blob/main/alsa_midi/c… at line 626 and possibly the sub-classes as well (lines 1679, 1871). None of these wrappers expected a valid return value and do not pass along said value returned from the c library call. Consequently, there is no longer an indication that the output has not been fully drained. ``` --- alsa-lib-1.2.14/src/seq/seq.c 2025-04-14 11:42:04.000000000 -0500 +++ alsa-lib-1.2.15.2/src/seq/seq.c 2026-01-08 08:27:57.000000000 -0600 @@ -4445,21 +4445,22 @@ int snd_seq_drain_output(snd_seq_t *seq) { ssize_t result, processed = 0; assert(seq); while (seq->obufused > 0) { result = seq->ops->write(seq, seq->obuf, seq->obufused); if (result < 0) { - if (result == -EAGAIN && processed) + if (result == -EAGAIN && processed > 0) return seq->obufused; return result; } if ((size_t)result < seq->obufused) memmove(seq->obuf, seq->obuf + result, seq->obufused - result); seq->obufused -= result; + processed += result; } return 0; } /** * \brief extract the first event in output buffer * \param seq sequencer handle ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/493 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_seq_drain_output change breaks existing code.
by GitHub issues - opened 13 Jan '26

13 Jan '26

alsa-project/alsa-lib issue #493 was opened from Hoosier-Beagler: The little gem shown below has broken a lot of code. Previously, no positive results were returned and software relied on -EAGAIN to detect remaining data. That result is no longer returned if some data was processed. This has broken the old and stable version of my application written in C(++) which is no longer being maintained. I have ported my app to python but this change has also broken public python wrappers for the alsa library. In [alsa-python / pyalsa / alsaseq.c](https://github.com/alsa-project/alsa-python/blob/master/pyalsa/a… below line 2800 and also [python-alsa-midi / alsa_midi / client.py](https://github.com/Jajcus/python-alsa-midi/blob/main/alsa_midi/c… at line 626 and possibly the sub-classes as well (lines 1679, 1871). None of these wrappers expected a valid return value and do not pass along said value returned from the c library call. ``` --- alsa-lib-1.2.14/src/seq/seq.c 2025-04-14 11:42:04.000000000 -0500 +++ alsa-lib-1.2.15.2/src/seq/seq.c 2026-01-08 08:27:57.000000000 -0600 @@ -4445,21 +4445,22 @@ int snd_seq_drain_output(snd_seq_t *seq) { ssize_t result, processed = 0; assert(seq); while (seq->obufused > 0) { result = seq->ops->write(seq, seq->obuf, seq->obufused); if (result < 0) { - if (result == -EAGAIN && processed) + if (result == -EAGAIN && processed > 0) return seq->obufused; return result; } if ((size_t)result < seq->obufused) memmove(seq->obuf, seq->obuf + result, seq->obufused - result); seq->obufused -= result; + processed += result; } return 0; } /** * \brief extract the first event in output buffer * \param seq sequencer handle ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/493 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

SOF HiFi profile fails to detect headphone jack on ASUS ZenBook UX425EA (ALC294)
by GitHub issues - opened 12 Jan '26

12 Jan '26

alsa-project/alsa-ucm-conf issue #686 was opened from akincano: Component: Sound(ALSA) → PCI Description: Hardware: Laptop: ASUS ZenBook UX425EA_UX425EA Audio Controller: Intel Tiger Lake-LP Smart Sound Technology Audio Controller (PCI ID: 0x8086:0xa0c8) Codec: Realtek ALC294 (Vendor ID: 0x10ec0294, Subsystem ID: 0x10431cee) Kernel: 6.18.4-arch1-1 Distribution: Arch Linux ALSA components: HDA:80862812,80860101,00100000 HDA:10ec0294,10431cee,00100004 cfg-dmics:2 Problem: When using the SOF (Sound Open Firmware) driver, the HiFi profile fails to create audio sinks/sources for the headphone jack. WirePlumber logs show: spa.alsa: No sink and source at HiFi: Headphones The HiFi profile shows as available: no and the active profile defaults to off. Attempting to switch to HiFi results in only HDMI outputs being available, with no analog speaker or headphone outputs. The headphone jack (pin 0x21) is misconfigured: Node 0x21 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0001001c: OUT HP EAPD Detect EAPD 0x2: EAPD Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x02 0x03* The key issues are: Pin Default 0x411111f0 - configured as "N/A" instead of headphone jack Misc = NO_PRESENCE - no jack detection enabled Pin-ctls: 0x00 - pin not configured for output Unsolicited: tag=00, enabled=0 - jack detection events disabled This causes PipeWire/WirePlumber to only create HDMI outputs in the HiFi profile, with no analog audio devices. Expected Behavior: The HiFi profile should detect and expose: Internal speakers Headphone jack with jack detection Automatic switching between speakers and headphones when plugging/unplugging Digital microphone Current Workaround: Blacklisting SOF and forcing legacy HDA driver works perfectly: ```bash # /etc/modprobe.d/alsa-base.conf blacklist snd_sof_pci_intel_tgl blacklist snd_sof_intel_hda_common blacklist snd_sof options snd-intel-dspcfg dsp_driver=1 ``` With legacy HDA driver, the same pin 0x21 is properly configured: ```bash Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Control: name="Headphone Playback Switch", index=0, device=0 ``` The `output:analog-stereo` profile works perfectly with automatic jack detection and switching. **Reproduction:** 1. Boot ASUS ZenBook UX425EA with default SOF driver (no blacklisting) 2. Check `pactl list cards` - shows: - HiFi profile: `available: no` - Active profile: `off` - Only HDMI ports listed under HiFi profile, no analog outputs 3. Check `journalctl --user -u wireplumber` - see error: `spa.alsa: No sink and source at HiFi: Headphones` 4. Attempting to use HiFi or even pro-audio profile results in no working analog audio **SOF Firmware/Topology Information:** ```bash sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware file: intel/sof/sof-tgl.ri sof-audio-pci-intel-tgl 0000:00:1f.3: Topology file: intel/sof-tplg/sof-hda-generic-2ch.tplg sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:1 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:1 sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 ``` **Card Configuration (SOF):** ``` Name: alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_generic api.alsa.card.name = "sof-hda-dsp" alsa.long_card_name = "ASUSTeKCOMPUTERINC.-ZenBookUX425EA_UX425EA-1.0-UX425EA" alsa.driver_name = "snd_soc_skl_hda_dsp" alsa.mixer_name = "Realtek ALC294" Profiles: off: Off (sinks: 0, sources: 0, priority: 0, available: yes) HiFi: Play HiFi quality Music (sinks: 3, sources: 1, priority: 9900, available: no) pro-audio: Pro Audio (sinks: 5, sources: 3, priority: 1, available: yes) Active Profile: off HiFi profile ports (no analog outputs): [Out] HDMI3: HDMI / DisplayPort 3 Output [Out] HDMI2: HDMI / DisplayPort 2 Output [Out] HDMI1: HDMI / DisplayPort 1 Output [In] Mic1: Digital Microphone Request: The ASUS ZenBook UX425EA needs either: A UCM configuration that properly maps pin 0x21 as a headphone jack with presence detection, OR A kernel quirk to set the correct pin configuration for this laptop model, OR An updated SOF topology file that includes the analog outputs The hardware clearly supports headphone jack detection (the legacy HDA driver proves this), but SOF is not utilizing it correctly. Impact: This affects all ASUS ZenBook UX425EA users running modern kernels with SOF as the default driver. Users either have no audio or must manually blacklist SOF to use the legacy HDA driver. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/686 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Laptop built-in speakers not detected after c8b5541
by GitHub issues - opened 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf issue #685 was opened from Xeltalliv: Updating from v1.2.15.1 to v1.2.15.2 broke sound on ASUS Zenbook 14 UM425UA laptop. It has no external headphone jacks. I narrowed it down the cause to the commit https://github.com/alsa-project/alsa-ucm-conf/commit/c8b55418ff36036fc96073… With v1.2.15.1 `wpctl status` shows: ``` Audio ├─ Devices: │ 48. Renoir/Cezanne HDMI/DP Audio Controller [alsa] │ 49. Ryzen HD Audio Controller [alsa] │ ├─ Sinks: │ * 56. Ryzen HD Audio Controller Headphones [vol: 0.40] │ ├─ Sources: │ * 57. Ryzen HD Audio Controller Stereo Microphone [vol: 1.00] │ 58. Ryzen HD Audio Controller Digital Microphone [vol: 1.00] │ ├─ Filters: │ └─ Streams: ``` With v1.2.15.2 `wpctl status` shows: ``` Audio ├─ Devices: │ 48. Renoir/Cezanne HDMI/DP Audio Controller [alsa] │ 49. Ryzen HD Audio Controller [alsa] │ ├─ Sinks: │ * 34. Dummy Output [vol: 1.00] │ ├─ Sources: │ * 56. Ryzen HD Audio Controller Digital Microphone [vol: 0.97] │ ├─ Filters: │ └─ Streams: ``` Not sure if any of the following helps: In both cases `dmesg | grep -iE 'snd|alsa|pipewire'` shows: ``` [ 8.235764] snd_rn_pci_acp3x 0000:04:00.5: enabling device (0000 -> 0002) [ 8.383393] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client [ 8.404286] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 8.434270] snd_hda_codec_alc269 hdaudioC1D0: autoconfig for ALC294: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 8.434280] snd_hda_codec_alc269 hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.434284] snd_hda_codec_alc269 hdaudioC1D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.434288] snd_hda_codec_alc269 hdaudioC1D0: mono: mono_out=0x0 [ 8.434290] snd_hda_codec_alc269 hdaudioC1D0: inputs: ``` `aplay -l` shows: ``` **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Generic [HD-Audio Generic], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic_1 [HD-Audio Generic], device 0: ALC294 Analog [ALC294 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `amixer -c 1 scontrols` shows: ``` Simple mixer control 'Master',0 Simple mixer control 'Mic ACP LED',0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/685 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for UR24C firmware version channel differences, fix UR22C
by GitHub pull_request - edited 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf pull request #684 was edited from jwlademann: Steinberg UR24C has a similar change to the number of capture channels for version 3 firmware that UR22C has. During implementation I found an issue with the condition used for UR22C which I fixed also. See: https://github.com/alsa-project/alsa-ucm-conf/pull/554 and https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for UR24C firmware version channel differences, fix UR22C
by GitHub pull_request - edited 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf pull request #684 was edited from jwlademann: Steinberg UR24C has a similar change to the number of capture channels for version 3 firmware that UR22C. During implementation I found an issue with the condition used for UR22C which I fixed also. See: https://github.com/alsa-project/alsa-ucm-conf/pull/554 and https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for UR24C firmware version channel differences, fix UR22C
by GitHub pull_request - opened 11 Jan '26

11 Jan '26

alsa-project/alsa-ucm-conf pull request #684 was opened from jwlademann: Steinberg UR24C has a similar change to the number of capture channels fot version 3 firmware. During implementation I found an issue with the condition used for UR22C which I fixed also. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/684.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASUS ROG Maximus XIII Hero (Tiger Lake, ALC4082) - Microphone not working - Missing USB ID 0b05:19ac
by GitHub issues - opened 09 Jan '26

09 Jan '26

alsa-project/alsa-ucm-conf issue #683 was opened from FallonBlue: ## Hardware - **Motherboard**: ASUS ROG Maximus XIII Hero (Z490/Z590 chipset, Tiger Lake) - **Audio Codec**: Realtek ALC4082 (connected via USB) - **USB Device ID**: `0b05:19ac` - **OS**: Bazzite (Fedora-based), kernel 6.17.7 - **Audio Stack**: PipeWire + WirePlumber ## Problem The microphone input on the rear panel does not work. The device is detected as "Generic USB Audio" but ALSA UCM does not recognize it, so proper audio routing is not configured. Speaker/headphone output works fine. ## Root Cause The USB device ID `0b05:19ac` is missing from the UCM configuration files: - `/usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf` - `/usr/share/alsa/ucm2/USB-Audio/Realtek/ALC4080-HiFi.conf` ## Requested Fix Please add `0b05:19ac` (ASUS ROG Maximus XIII Hero) to the ALC4080 UCM configuration, similar to how issue #165 was resolved for the Z690 Hero (`0b05:1a27`). The device appears in lsusb as: ``` Bus 001 Device 002: ID 0b05:19ac ASUSTek Computer, Inc. USB Audio ``` ## Additional Info - Microphone works correctly in Windows - Workaround: Using external USB microphone - Related boards already in config: `0b05:1996` (Maximus XIII), `0b05:1a27` (Z690 Hero) Thank you! Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/683 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [BUG] Samsung Galaxy Book 4 Ultra - MAX98390 speakers not supported on MTL platform
by Péter Ujfalusi 08 Jan '26

08 Jan '26

Hi José, The machine likely similar to Book 4 Pro, it is an HDA+companion amp setup: https://github.com/thesofproject/linux/issues/5002 The bounty for Book 4 Pro resulted (as far as I know) working audio, which can be replicated: https://github.com/stephanlensky/galaxy-book4-linux-sound You might need to add the quirk for the Ultra and by luck it might work as well. SOF is not needed, used in these machines as they are pure HDA devices. -- Péter On 08/01/2026 00:26, José Augusto de Almeida Neto wrote: > Hi ALSA developers, > > I'm reporting a hardware support issue with the Samsung Galaxy Book 4 Ultra > (NP960XGL-XG1BR) running on Intel Meteor Lake (MTL) platform. > > SUMMARY: > The laptop has MAX98390 smart speaker amplifiers that are not supported in > the current Linux kernel. Headphones work (ALC298 HDA codec) but speakers > are completely non-functional. > > HARDWARE: > - System: Samsung Galaxy Book 4 Ultra (NP960XGL-XG1BR) > - CPU: Intel Meteor Lake-P > - Audio Controller: Intel MTL HD Audio (8086:7728) > - HDA Codec: Realtek ALC298 (working for headphones) > - Speaker Amps: 4x Maxim MAX98390 on I2C bus 2 (addresses 0x38, 0x39, > 0x3C, 0x3D) > - Kernel: 6.17.9 (also tested on 6.8.0 mainline) > > ROOT CAUSE: > 1. No machine driver match in soc-acpi-intel-mtl-match.c for Samsung + > MAX98390 > 2. No SOF topology file for MTL + HDA + MAX98390 combination > 3. NHLT only shows SSP2 (Bluetooth), no SSP configured for speakers > 4. SOF falls back to generic skl_hda_dsp_generic (2-channel HDA only) > > VERIFICATION: > - MAX98390 chips respond on I2C (revision 0x42 confirmed on all 4 chips) > - ACPI device MAX98390:00 present (status 15 - enabled) > - ALC298 speaker DAC receives audio but no output from physical speakers > - No I2S/TDM connection to MAX98390 amplifiers > > I have collected detailed technical information including: > - DSDT and NHLT ACPI tables > - Full HDA codec dump > - I2C bus scans and MAX98390 register reads > - dmesg logs showing current SOF behavior > > Full detailed report: BUG_REPORT.md attached > > QUESTION: > Is there ongoing work to support MAX98390 on Meteor Lake platforms? > I'm willing to test patches and provide additional debugging information. > > Thank you, > Jose >

1 0

Re: [BUG REPORT] arm64: dts: qcom: x1e80100: Audio routing issues on ThinkPad T14s Gen 6
by Johan Hovold 07 Jan '26

07 Jan '26

On Wed, Jan 07, 2026 at 03:09:10PM +0100, Simone Flavio Paris wrote: > I am testing the latest experimental kernels on the Lenovo ThinkPad > T14s Gen 6 (Snapdragon X Elite). > > I have encountered a specific issue with the audio subsystem: while > the DSP and ALSA stack appear fully functional (firmware loads, > streams are processed without errors), there is no analog output on > speakers or headphones. > > I have performed extensive diagnostics and manual mixer path > configurations to rule out userspace/UCM issues. Below is a detailed > technical report of the current status, routing attempts, and logs. > > I hope this information helps in refining the machine driver or > topology profiles for this specific device. I am available to test > patches if needed. > > [Technical Report] > > Target Hardware: Lenovo ThinkPad T14s Gen 6 (21N1) - Snapdragon X > Elite (X1E80100) > OS: Ubuntu 25.10 (Debian-based) > Kernel: 6.17.0-8-generic #8-Ubuntu SMP PREEMPT_DYNAMIC aarch64 You need to report this to your distro as this is not a mainline issue. IIUC, Ubuntu disables audio by default until you pass a kernel parameter due to the lack of speaker protection. Johan

1 0

Alsaloop fix short options
by GitHub pull_request - opened 07 Jan '26

07 Jan '26

alsa-project/alsa-utils pull request #317 was opened from jacmet: Various minor fixes to the alsaloop command line parsing Request URL : https://github.com/alsa-project/alsa-utils/pull/317 Patch URL : https://github.com/alsa-project/alsa-utils/pull/317.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v11 0/4] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 07 Jan '26

07 Jan '26

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v11: - Patch 1/4: - minor fixes - Patch 2/4: - change of_find_spi_controller_by_node() gating to CONFIG_OF - Patch 3/4: - nothing - Patch 4/4: - achieve reverse-christmas-tree notation - remove formal version check - revise check whether the firmware boots from SPI or not - minor fixes Changes in v10: - Patch 1/3: - export of_find_spi_controller_by_node() - Patch 2/3: - remove realtek,rt5575-use-spi - add spi-parent for firmware-loading - Patch 3/3: - use of_find_spi_controller_by_node() to get the SPI controller and add the spi device for firmware-loading - Link to v10: https://lore.kernel.org/all/20251216071853.3929135-1-oder_chiou@realtek.com/ Changes in v9: - Patch 1/2: - modify the comment - change the compatible name to "realtek,rt5575-use-spi" - Patch 2/2: - remove the standalone rt5575_spi_driver module and integrate its functionality into the I2C driver - move the SPI firmware-loading function to rt5575-spi.c - use the match data to distinguish between w/wo flash - minor fixes - Link to v9: https://lore.kernel.org/all/20251211110130.2925541-1-oder_chiou@realtek.com/ Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2: - add compatible enum "realtek,rt5575-with-spi" - Link to v8: https://lore.kernel.org/all/20251201105926.1714341-1-oder_chiou@realtek.com/ Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2: - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2: - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (4): spi: export of_find_spi_controller_by_node() spi: change of_find_spi_controller_by_node() gating to CONFIG_OF ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 .../bindings/sound/realtek,rt5575.yaml | 61 +++ drivers/spi/spi.c | 23 +- include/linux/spi/spi.h | 9 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 118 ++++++ sound/soc/codecs/rt5575-spi.h | 27 ++ sound/soc/codecs/rt5575.c | 352 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 +++ 9 files changed, 651 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

3 4

ucm2: da7213: Add ADC switch in HeadphoneMic2 sequences
by GitHub pull_request - opened 07 Jan '26

07 Jan '26

alsa-project/alsa-ucm-conf pull request #682 was opened from QTI-Quill: Enable and disable the ADC switch in HeadphoneMic2EnableSeq.conf and HeadphoneMic2DisableSeq.conf to ensure headset microphone audio works properly on Talos EVK with DA7213 codec. Without this change, the headset mic path remains muted and capture does not function. Fixes: 5ccdd19 ("Qualcomm: qcs615: Add TALOS EVK HiFi config") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/682 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/682.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v7 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 06 Jan '26

06 Jan '26

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 86 ++++++++ sound/soc/codecs/rt5575-spi.h | 16 ++ sound/soc/codecs/rt5575.c | 370 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++++ 6 files changed, 540 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index a0dfef57200c..a3ea5febd1e0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -211,6 +211,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1767,6 +1768,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate "Realtek ALC5575 Codec - I2C" + depends on I2C + +config SND_SOC_RT5575_SPI + tristate "Realtek ALC5575 Codec - SPI" + depends on SPI_MASTER + select SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 39138d96a720..82f660cbe8ec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -252,6 +252,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-spi-y := rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -684,6 +686,8 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o +obj-$(CONFIG_SND_SOC_RT5575_SPI) += snd-soc-rt5575-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..9dbc8170fb76 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,86 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_CMD_BURST_WRITE 5 +#define RT5575_SPI_BUF_LEN 240 + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +bool rt5575_spi_ready; +static struct spi_device *rt5575_spi; + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @addr: Start address. + * @txbuf: Data buffer for writing. + * @len: Data length. + * + */ +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + + memcpy(&buf.data, &txbuf[offset], end); + + spi_write(rt5575_spi, &buf, sizeof(buf)); + + offset += RT5575_SPI_BUF_LEN; + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5575_spi_burst_write); + +static int rt5575_spi_probe(struct spi_device *spi) +{ + rt5575_spi = spi; + + rt5575_spi_ready = true; + + return 0; +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct spi_driver rt5575_spi_driver = { + .driver = { + .name = "rt5575", + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_spi_probe, +}; +module_spi_driver(rt5575_spi_driver); + +MODULE_DESCRIPTION("ALC5575 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..cafe49a7ecc2 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,16 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +extern bool rt5575_spi_ready; + +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..58508d643273 --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,370 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +#include "rt5575-spi.h" +#endif + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +static void rt5575_fw_load_by_spi(const struct firmware *fw, void *context) +{ + struct rt5575_priv *rt5575 = context; + struct i2c_client *i2c = rt5575->i2c; + const struct firmware *firmware; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw2.bin", + "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin" + }; + static const u32 fw_addr[] = { 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + int i, ret; + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + rt5575_spi_burst_write(0x5f400000, fw->data, fw->size); + release_firmware(fw); + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], &i2c->dev); + if (!ret) { + rt5575_spi_burst_write(fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } else { + dev_err(&i2c->dev, "Request firmware failure: %d\n", ret); + return; + } + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + + if (ret) + dev_err(&i2c->dev, "Run firmware failure: %d\n", ret); +} +#endif + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker CH-01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker CH-23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + + rt5575->component = component; + + dev_info(component->dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, "realtek/rt5575/rt5575_fw1.bin", + component->dev, GFP_KERNEL, rt5575, rt5575_fw_load_by_spi); +#endif + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val; + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + if (!rt5575_spi_ready) + return -EPROBE_DEFER; +#endif + + rt5575 = devm_kzalloc(&i2c->dev, sizeof(struct rt5575_priv), + GFP_KERNEL); + if (rt5575 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(&i2c->dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_ID_1, &val); + if (!val) { + dev_err(&i2c->dev, "This is not formal version\n"); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..11149612274a --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,54 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.51.2

4 5

[PATCH v5 1/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 06 Jan '26

06 Jan '26

This codec driver is for the ALC5575 that has a built-in audio DSP. The firmware can be loaded by the SPI from the SOC or the external flash. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 110 +++++++++ sound/soc/codecs/rt5575-spi.h | 37 +++ sound/soc/codecs/rt5575.c | 421 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 +++++ 6 files changed, 639 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 18131144ebbd..4c8f61a610a6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -210,6 +210,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1739,6 +1740,14 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate + depends on I2C + +config SND_SOC_RT5575_SPI + tristate + depends on SPI_MASTER + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bd95a7c911d5..2317b830b79a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -251,6 +251,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-spi-y := rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -682,6 +684,8 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o +obj-$(CONFIG_SND_SOC_RT5575_SPI) += snd-soc-rt5575-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..54c23c2d9c34 --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,110 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright 2022 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/uaccess.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5575-spi.h" + +static struct spi_device *rt5575_spi; + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length. + * + * + * Returns true for success. + */ +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5575_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int end, offset = 0; + + write_buf = kmalloc(RT5575_SPI_BUF_LEN + 6, GFP_KERNEL); + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0x000000ff) >> 0; + write_buf[2] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[3] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[4] = ((addr + offset) & 0xff000000) >> 24; + + memcpy(&write_buf[5], &txbuf[offset], end); + + if (end % 8) + end = (end / 8 + 1) * 8; + + spi_write(rt5575_spi, write_buf, end + 6); + + offset += RT5575_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5575_spi_burst_write); + +static int rt5575_spi_probe(struct spi_device *spi) +{ + rt5575_spi = spi; + + return 0; +} + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct spi_driver rt5575_spi_driver = { + .driver = { + .name = "rt5575", + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_spi_probe, +}; +module_spi_driver(rt5575_spi_driver); + +MODULE_DESCRIPTION("ALC5575 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..59ac348b101f --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,37 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright 2022 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +#define RT5575_SPI_BUF_LEN 240 + +/* SPI Command */ +enum { + RT5575_SPI_CMD_16_READ = 0, + RT5575_SPI_CMD_16_WRITE, + RT5575_SPI_CMD_32_READ, + RT5575_SPI_CMD_32_WRITE, + RT5575_SPI_CMD_BURST_READ, + RT5575_SPI_CMD_BURST_WRITE, +}; + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); +#else +static inline int rt5575_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + return -EINVAL; +} +#endif + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..0444a3a0a890 --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,421 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright 2022 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/acpi.h> +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/firmware.h> +#include <linux/property.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#include "rt5575-spi.h" + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +static int rt5575_spi_fw_run_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + +static int rt5575_spi_fw_run_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + const struct firmware *firmware; + int i, ret; + + if (!ucontrol->value.bytes.data[0]) + return 0; + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + ret = request_firmware(&firmware, "realtek/rt5575/0x70000000.dat", + component->dev); + if (ret == 0) { + rt5575_spi_burst_write(0x5f400000, firmware->data, + firmware->size); + release_firmware(firmware); + } + + ret = request_firmware(&firmware, "realtek/rt5575/0x70200000.dat", + component->dev); + if (ret == 0) { + rt5575_spi_burst_write(0x5f600000, firmware->data, + firmware->size); + release_firmware(firmware); + } + + ret = request_firmware(&firmware, "realtek/rt5575/0x703fe000.dat", + component->dev); + if (ret == 0) { + rt5575_spi_burst_write(0x5f7fe000, firmware->data, + firmware->size); + release_firmware(firmware); + } + + ret = request_firmware(&firmware, "realtek/rt5575/0x703ff000.dat", + component->dev); + if (ret == 0) { + rt5575_spi_burst_write(0x5f7ff000, firmware->data, + firmware->size); + release_firmware(firmware); + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + ret = 1; + for (i = 0; i < 100 && ret; i++) { + regmap_read(rt5575->regmap, RT5575_SW_INT, &ret); + msleep(100); + } + + if (ret) { + dev_err(component->dev, "Firmware failure\n"); + return -EINVAL; + } + + return 0; +} +#endif + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker01 Playback Volume", RT5575_SPK01_VOL, 17, 1, + 167, 0, ob_tlv), + SOC_DOUBLE("Speaker23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker23 Playback Volume", RT5575_SPK23_VOL, 17, 1, + 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, + 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, + 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, + 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, + RT5575_MIXR_VOL, 1, 127, 0, ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, + 0, ob_tlv), +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) + SND_SOC_BYTES_EXT("SPI FW run", 1, rt5575_spi_fw_run_get, + rt5575_spi_fw_run_put), +#endif +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_getuuid(struct rt5575_priv *rt5575) +{ + int efuse_uuid_low, efuse_uuid_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &efuse_uuid_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &efuse_uuid_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)efuse_uuid_high << 32) | (long long)efuse_uuid_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + + rt5575->component = component; + + dev_info(component->dev, "UUID: %llx\n", rt5575_getuuid(rt5575)); + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); + + return 0; +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val; + + rt5575 = devm_kzalloc(&i2c->dev, sizeof(struct rt5575_priv), + GFP_KERNEL); + if (rt5575 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(&i2c->dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_ID_1, &val); + if (!val) { + dev_err(&i2c->dev, "This is not formal version\n"); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, &rt5575_soc_component_dev, + rt5575_dai, ARRAY_SIZE(rt5575_dai)); +} + +static const struct of_device_id rt5575_of_match[] = { + {.compatible = "realtek,rt5575"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5575_of_match), + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..3203b1653573 --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,58 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright 2022 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.34.1

4 4

USB-Audio: Add support for the SSL 2 MK II and SSL2+ MK II
by GitHub pull_request - edited 06 Jan '26

06 Jan '26

alsa-project/alsa-ucm-conf pull request #681 was edited from MTRNord: This adds support for https://solidstatelogic.com/products/ssl-2-mkii and the https://solidstatelogic.com/products/ssl-2-plus-mkii and solves #555 The MK II shows up the same way as the SSL2, but as a different USB Device. I only own the SSL2 MK2, but not the SSL2+ MK2. So I can only verify it for the SSL2 MK2. However, I am fairly certain that the + version likely is also behaving identically. There are no visible spec changes between the v1 and MK2 of these devices, from all I was able to tell based on the internet. So, likely only the USB ID changed (at least for the things alsa-ucm is concerened about) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/681 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/681.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for the SSL 2 MK II
by GitHub pull_request - opened 06 Jan '26

06 Jan '26

alsa-project/alsa-ucm-conf pull request #681 was opened from MTRNord: This adds support for https://solidstatelogic.com/products/ssl-2-mkii and the https://solidstatelogic.com/products/ssl-2-plus-mkii and solves #555 The MK II shows up the same way as the SSL2, but as a different USB Device. I only own the SSL2 MK2, but not the SSL2+ MK2. So I can only verify it for the SSL2 MK2. However, I am fairly certain that the + version likely is also behaving identically. There are no visible spec changes between the v1 and MK2 of these devices, from all I was able to tell based on the internet. So, likely only the USB ID changed (at least for the things alsa-ucm is concerened about) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/681 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/681.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Replace common/direct.conf included file with DirectUseCase macro (directm.conf)
by GitHub pull_request - opened 06 Jan '26

06 Jan '26

alsa-project/alsa-ucm-conf pull request #680 was opened from perexg: There is a requirement to have more direct profiles with different channel counts (e.g. for Scarlett 18i20). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/680 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/680.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[bug] Commit 36a111a is interfearing with and crashing steam games with native Dualsense support
by GitHub issues - opened 06 Jan '26

06 Jan '26

alsa-project/alsa-ucm-conf issue #677 was opened from tamodolo: [The commit](https://github.com/alsa-project/alsa-ucm-conf/commit/36a111ad007f2f… is interfearing with games that have native support for the Sony's Dualsense controller. On Arch after this update games cannot send haptics properly to the controller over USB and crash after some time or at launch. In my case, Final Fantasy XIV Online became unplayable after this update because it is crashing at launch. When it works haptics don't. The problems does not happen when the profile is deleted from the system. Solution found by [this reddit user](https://www.reddit.com/r/ffxiv/comments/1q480jc/comment/nxqtn2v/) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/677 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Geometry Dash: A Fast-Paced Rhythm Platformer Full of Challenge
by wallaceholcomb610＠gmail.com 06 Jan '26

06 Jan '26

Geometry Dash is a popular rhythm-based platform game developed by RobTop Games. Known for its intense difficulty, energetic music, and colorful geometric visuals, Geometry Dash challenges players to test their reflexes, timing, and patience. Since its release, the game has attracted millions of players and built a strong global community. https://geometry-games.io In Geometry Dash, players control a small geometric character that automatically moves forward through obstacle-filled levels. The main action is tapping or clicking to jump, fly, or switch gravity, depending on the game mode. Although the controls are very simple, the gameplay is extremely challenging. One small mistake can cause the player to crash and restart the level from the beginning. What makes Geometry Dash unique is its strong connection between music and gameplay. Each level is synchronized with an electronic soundtrack, and obstacles often match the rhythm and beat of the music. As players become more experienced, they start to rely on the music to time their jumps and movements. This creates an immersive experience where playing feels like following a fast musical pattern. The game includes various modes such as cube, ship, ball, UFO, wave, robot, and spider. Each mode changes how the character moves, forcing players to adapt quickly. Speed changes, gravity portals, and visual effects add more complexity and keep the gameplay exciting and unpredictable. Geometry Dash also features a powerful level editor that allows players to create and share their own custom levels. This has led to millions of user-made levels, ranging from simple designs to extremely difficult challenges. The creative community is one of the reasons the game remains popular after many years. Although Geometry Dash can be frustrating, it is also very rewarding. Completing a difficult level after many failed attempts gives players a strong sense of achievement. The game teaches persistence, focus, and patience, making every success feel well-earned. In conclusion, Geometry Dash is more than just a game about jumping over spikes. It is a rhythm-driven challenge that combines music, precision, and creativity into a thrilling experience. With its simple controls, deep gameplay, and passionate community, Geometry Dash continues to be one of the most memorable and influential platform games ever created.

1 0

Dinosaur Game: A Simple Offline Game with Endless Fun
by wallaceholcomb610＠gmail.com 06 Jan '26

06 Jan '26

Dinosaur Game is a simple yet iconic browser game that appears in Google Chrome when there is no internet connection. At first glance, it looks very basic, but behind its minimalist design lies an addictive and surprisingly challenging experience that has entertained millions of players worldwide. https://dinosaur-game.io The game starts when a small pixelated dinosaur appears on the screen along with a message saying there is no internet connection. By pressing the spacebar or tapping the screen, the dinosaur begins to run automatically across a desert landscape. The goal is simple: avoid obstacles for as long as possible and achieve the highest score you can. Gameplay in Dinosaur Game is extremely easy to understand. The dinosaur runs forward nonstop, and players must jump over cacti and later dodge flying pterodactyls. As the game continues, the speed gradually increases, making obstacles appear faster and closer together. This rising difficulty keeps the game exciting and forces players to stay focused at all times. One of the most interesting aspects of Dinosaur Game is its visual style. The game uses black-and-white pixel graphics that are clean and simple. There are no unnecessary details, which helps players clearly see obstacles and react quickly. Occasionally, the background switches between day and night, adding a small but enjoyable visual change. Unlike many modern games, Dinosaur Game has no background music. This silence allows players to concentrate fully on the timing of their jumps. The lack of sound also makes the game suitable for playing anywhere, such as at school, work, or in public places. Dinosaur Game is popular because of its accessibility. It requires no download, no account, and no internet connection. Anyone using Google Chrome can play it instantly. This makes it enjoyable for players of all ages, including people who do not usually play video games. Although Dinosaur Game is mainly designed for fun, it also helps improve reaction time and concentration. Players must learn to anticipate obstacles and respond quickly as the speed increases. Because each round restarts instantly after failure, the game encourages repeated attempts and self-improvement. In conclusion, Dinosaur Game proves that a great game does not need complex graphics or advanced mechanics. With simple controls, clean visuals, and increasing difficulty, it turns a moment of frustration—losing internet—into a fun and memorable experience. The running dinosaur has become a symbol of offline entertainment and remains one of the most beloved browser games of all time.

1 0

Discover the Fun: How to Play the Connections Game
by marginal.mammal.xmif＠protectsmail.net 06 Jan '26

06 Jan '26

In the ever-evolving world of online gaming, some games stand out due to their unique blend of challenge and fun. One such game is the Connections Game. It’s accessible, intriguing, and offers players a fantastic way to flex their cognitive muscles while enjoying their time. In this article, we will delve into how to play the Connections Game, some effective tips to enhance your experience, and why it’s worth a try. https://connectionsgamefree.com Gameplay The Connections Game is all about linking words in groups based on their connections. At first glance, it may seem simple, but as you dive deeper, you’ll find layers of complexity that make it both fun and engaging. Here’s a breakdown of how to play: 1. Objective: The main aim of the game is to find and connect a series of related words that belong to a specific category. Think of it as a mixed bag of words with hidden relations that you need to uncover. 2. Setup: Upon starting the game, you’ll be presented with a grid of words. These words can belong to various themes or categories—anything from animals to famous landmarks, and everything in between. 3. Finding Connections: Your job is to scan through the grid and identify groups of words that are connected. You’ll click or tap on the words to select them, and once you have selected four related words, the game will confirm that they are indeed a correct cluster. 4. Winning the Game: The more words you manage to connect, the higher your score. You can either play against the clock or take your time, depending on your preference. As you progress, the game may introduce more complex word sets or require more skilled connections. 5. Variety: Another exciting aspect of the Connections Game is that each round can be different. With numerous sets of words to choose from, every game provides a fresh experience, keeping players engaged and challenged. Tips for Playing the Connections Game Now that you have a grasp of the gameplay, let’s explore some tips to improve your skills and enhance your experience: 1. Start With Familiar Words: When you first look at the grid, focus on words that you recognize. Connecting familiar terms can give you the confidence to explore more unfamiliar words and connections later. 2. Think Categorically: As you scan through the words, try to think in categories. Is one of the words related to animals? Perhaps another word hints at food? By grouping ideas in your mind first, you can more easily spot connections. 3. Practice Makes Perfect: Like any game, the more you play the Connections Game, the better you will become. Familiarity with various types of word relationships will allow you to identify connections quicker. 4. Play With Friends: If you’re looking to add a social aspect to your gaming experience, invite friends to play alongside you. Friendly competition can heighten the enjoyment, and you can discuss strategies on how to find word connections. 5. Stay Focused: Sometimes, it’s easy to get distracted during gameplay. Make sure to focus your attention on the task at hand to enhance your performance. A clear mind can spot connections that may previously have escaped your notice. 6. Review Past Games: After finishing a game, take a look at the words you missed. Reviewing these words can help you understand the connections you previously overlooked and can boost your learning for future games. Conclusion The Connections Game offers a delightful blend of mental challenge and entertainment. It encourages players to think critically and in connections, providing an enriching way to strengthen cognitive skills. Whether you enjoy playing alone or wish to share the experience with friends, this game has something for everyone. The enjoyment lies in both the challenge and the satisfaction that comes with successfully identifying the connections among seemingly random words. So why not take a break from your usual routine and give the Connections Game a try? You might just find yourself hooked on the challenge and fun it presents. Start connecting words today! Visit Connections Game and experience the thrill firsthand. With each round, you’ll find yourself becoming more adept at spotting those crucial connections. Happy gaming!

1 0

SSL2.conf Defines Incorrect Number of Capture Channels
by GitHub issues - opened 06 Jan '26

06 Jan '26

alsa-project/alsa-ucm-conf issue #676 was opened from TheBakedPotato: The file at `ucm2/USB-Audio/SolidStateLabs/SSL2.conf` has the line ``` Define.DirectCaptureChannels 4 ``` However, I believe that should actually be ``` Define.DirectCaptureChannels 2 ``` since the SSL2 (**not** the SSL2+) only has 2 physical inputs. As well, now, with that change, a third audio "profile" appears ago for the SSL2 output device of `Direct SSL 2`. <img width="416" height="78" alt="Image" src="https://github.com/user-attachments/assets/14998079-d81d-444f-80b1-b198ad04…" /> The `[ALSA UCM error]` still appears, but now the correct "profile" is available. As well, now my music player, Amberol, is able to persist the volume change between songs. Before if I changed the volume and proceed to the next song the audio in the Amberol application would get set to max. The output in `journalctl` is still present ``` Jan 05 18:11:39 endeavour-zone wireplumber[1533]: spa.alsa: Error in ALSA UCM profile for _ucm0001.hw:S2,0 (HiFi: Mic2: source): CaptureChannels=4 > avail 2 Jan 05 18:11:39 endeavour-zone wireplumber[1533]: spa.alsa: Errors in ALSA UCM profile for card SSL 2 Jan 05 18:11:40 endeavour-zone wireplumber[1533]: spa.bluez5: BlueZ system service is not available ``` if the HiFi profile is selected. I'm assuming because the file `SSL2-HiFi.conf` needs to be updated for the correct number of hardware capture channels. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/676 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Fix scarlett 18i20 channel detection
by GitHub pull_request - opened 05 Jan '26

05 Jan '26

alsa-project/alsa-ucm-conf pull request #675 was opened from perexg: It's a follow-up for #660 . Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/675 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/675.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH] ASoC: amd: acp: Add support for Huawei Matebook D14 (NBM-WXX9)
by Mark Brown 05 Jan '26

05 Jan '26

On Sun, Dec 28, 2025 at 04:53:18PM -0800, Andres Ballesteros Gallegos wrote: > This patch adds the DMI quirk entries for the Huawei Matebook D14 > (2020 AMD Renoir model), identified as NBM-WXX9. > diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c > index 8e45f2a..1234567 100644 --- a/sound/soc/amd/acp-config.c +++ > b/sound/soc/amd/acp-config.c @@ -148,6 +148,17 @@ static const struct > config_entry config_table[] = { {} }, }, > > { > .flags = FLAG_AMD_LEGACY, This has been badly mangled by your mail client, it looks like at a minimum all tabs have been converted into spaces. If it's hard to sort your mail client it might be easier to use b4's web submission: https://b4.docs.kernel.org/en/latest/ > > Signed-off-by: ANDRÉS BALLESTEROS GALLEGOS ANDRES.BALLESTEROSGALLEGOS(a)GMAIL.COM This should come at the end of the changelog before the actual patch.

1 0

Bluetooth: Suddenly corrupted SCO packet
by GitHub issues - opened 03 Jan '26

03 Jan '26

alsa-project/alsa-ucm-conf issue #674 was opened from wwidesoft: Bluetooth headset stopped working after upgrading package alsa-ucm-conf to version 1.2.15.1-1 (Arch Linux kernel 6.18.2-zen2-1-zen 64-bit) downgrading to version 1.2.14-2 solved the issue **dmesg** kernel: Bluetooth: hci0: corrupted SCO packet kernel: Bluetooth: hci0: SCO packet for unknown connection handle 257 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/674 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Generic HDA UCM (HDA/HiFi-analog.conf + HiFi-mic.conf) creates phantom devices on jackless Framework Laptop 16 (ALC285, f111:000d) causing wrong PipeWire defaults
by GitHub issues - opened 03 Jan '26

03 Jan '26

alsa-project/alsa-ucm-conf issue #673 was opened from dniku: **DISCLOSURE**: the text below was written with heavy assistance from a language model. --- ### Summary On Framework Laptop 16 (AMD Ryzen AI 300 series) with `alsa.mixer_name = "Realtek ALC285"`, `alsa.components = "HDA:10ec0285,f111000d,00100002"`, the generic HDA UCM profile creates a `Headphones` playback device even though the codec exposes no headphone jack/controls. WirePlumber then selects a phantom `Headphones` sink by default (higher UCM priority than `Speaker`), requiring a manual switch to `Speaker`. Separately: UCM also exposes an HDA “Stereo Microphone” capture path (`Mic2`) which records silence; the real microphone is provided by the ACP/DMIC path. This is reproducible after wiping WirePlumber state and restarting PipeWire/WirePlumber. ### Hardware / software - Machine: Framework Laptop 16 (AMD Ryzen AI 300 series) - Codec: Realtek ALC285, subsystem id f111:000d - Distro: NixOS 25.11 - PipeWire: 1.4.9 - WirePlumber: 0.5.12 - alsa-ucm-conf: 1.2.14 ### Steps to reproduce 1. Ensure you are using the stock alsa-ucm-conf UCM2 files (no overrides). 2. Clear WP state and restart user audio services: ```sh wpctl clear-default rm -rf ~/.local/state/wireplumber systemctl --user restart wireplumber pipewire pipewire-pulse ``` 3. Observe the default sink: ```sh wpctl status -n wpctl inspect @DEFAULT_AUDIO_SINK@ ``` ### Observed behavior (before fix) Default sink becomes the phantom Headphones profile: ```sh $ wpctl status -n # ... ├─ Sinks: │ * 59. alsa_output.pci-0000_c2_00.6.HiFi__Headphones__sink [vol: 0.40] ``` ```sh $ wpctl inspect @DEFAULT_AUDIO_SINK@ # ... device.profile.name = "HiFi: Headphones: sink" * node.name = "alsa_output.pci-0000_c2_00.6.HiFi__Headphones__sink" ``` But ALSA exposes no headphone controls on that codec/card: ```sh $ amixer -c 2 scontrols Simple mixer control 'Master',0 Simple mixer control 'Speaker',0 Simple mixer control 'Bass Speaker',0 Simple mixer control 'PCM',0 $ amixer -c 2 controls numid=7,iface=CARD,name='Speaker Front Phantom Jack' numid=8,iface=CARD,name='Speaker Surround Phantom Jack' numid=6,iface=MIXER,name='Master Playback Switch' numid=5,iface=MIXER,name='Master Playback Volume' numid=13,iface=MIXER,name='PCM Playback Volume' numid=4,iface=MIXER,name='Bass Speaker Playback Switch' numid=3,iface=MIXER,name='Bass Speaker Playback Volume' numid=2,iface=MIXER,name='Speaker Playback Switch' numid=1,iface=MIXER,name='Speaker Playback Volume' numid=10,iface=PCM,name='Capture Channel Map' numid=9,iface=PCM,name='Playback Channel Map' numid=11,iface=PCM,name='Capture Channel Map',device=2 numid=12,iface=PCM,name='Capture Channel Map',index=1,device=2 ``` The codec dump also contains no “Headphone/Headset” pins: ```sh $ grep -nEi 'Headphone|Headset|Front Headphone' /proc/asound/card2/codec\#0 # (no output) ``` ### Root cause hypothesis [`ucm2/HDA/HiFi-analog.conf`](https://github.com/alsa-project/alsa-ucm-conf/blob/5d175e1f7e3df15973945a9cc15e82fa7ca0d7b6/ucm2/HDA/HiFi-analog.conf) defines `SectionDevice."Headphones"` unconditionally and assigns it `PlaybackPriority 200`, while `Speaker` is `PlaybackPriority 100`. On jackless hardware with no headphone controls/jack, this results in a “Headphones” UCM device that WirePlumber prefers. Similar problem exists in [`ucm2/HDA/HiFi-mic.conf`](https://github.com/alsa-project/alsa-ucm-conf/blob/5d175e1f7e3df15973945a9cc15e82fa7ca0d7b6/ucm2/HDA/HiFi-mic.conf) ### Proposed fix Guard creation of `SectionDevice."Headphones"` so it is only instantiated when relevant headphone controls/jack controls exist (e.g. `Headphone Playback Volume`, `Headphone Playback Switch`, or `iface=CARD,name='Headphone Jack'`). Similarly guard the HDA analog capture device on this platform when it is not functional, leaving the working ACP/DMIC capture path as the only mic. --- (end of section written with language model assistance) I have proposed a patch for Framework 16 in https://github.com/NixOS/nixos-hardware/pull/1718. Please see the diagnostics shared in that PR. I would appreciate feedback on the specific workaround that is implemented there from someone familiar with `alsa-ucm-conf`. For the output of `nix shell nixpkgs#alsa-utils -c alsa-info --no-upload --stdout` before/after applying the patch, as well as the diff between the two, please see [this gist](https://gist.github.com/dniku/d9b5095aef0e926399ebd772d0b40bd0). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/673 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v11 3/4] ASoC: dt-bindings: realtek,rt5575: add support for ALC5575
by Oder Chiou 02 Jan '26

02 Jan '26

Audio codec with I2S, I2C and SPI. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org> --- .../bindings/sound/realtek,rt5575.yaml | 61 +++++++++++++++++++ 1 file changed, 61 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml new file mode 100644 index 000000000000..981ebc39b195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5575.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5575 audio CODEC + +maintainers: + - Oder Chiou <oder_chiou(a)realtek.com> + +description: + The device supports both I2C and SPI. I2C is mandatory, while SPI is + optional depending on the hardware configuration. SPI is used for + firmware loading if present. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5575 + + reg: + maxItems: 1 + + spi-parent: + description: + Optional phandle reference to the SPI controller used for firmware + loading. The argument specifies the chip select. + $ref: /schemas/types.yaml#/definitions/phandle-array + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + # I2C-only node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + }; + }; + + # I2C + optional SPI node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + spi-parent = <&spi0 0>; /* chip-select 0 */ + }; + }; -- 2.52.0

2 3

Remove outdated DisableSequence elements
by GitHub pull_request - opened 02 Jan '26

02 Jan '26

alsa-project/alsa-ucm-conf pull request #671 was opened from mstrozek: cs42l43's 'Headset Microphone' and 'Microphone' devices were both used to control mic mute LED in the past, but this was changed [1] to only allow the built-in microphones to control the LED. Thus clean up the DisableSequences to remove the switching off that is no longer needed. Also, cs42l45 had those Sequences copied over from cs42l43, so remove them here as well. [1]: https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/common/ctl/l… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/671 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/671.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Qualcomm: disable broken companders and low power modes (fix #595)
by GitHub pull_request - opened 02 Jan '26

02 Jan '26

alsa-project/alsa-ucm-conf pull request #670 was opened from valpackett: It's 2026. Let's finally land the fixes for #595 maybe? :) // Not (yet?) tested on wcd934x, wcd937x and wcd939x.. but let's see if we can get any testers with different devices to show up.. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/670 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/670.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] MOTU Midi Express XT
by holger＠dehnhardt.org 02 Jan '26

02 Jan '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

4 14

alsa-lib 1.2.15.1 | only stereo fallback mode for Realtek ALC1220
by GitHub issues - opened 01 Jan '26

01 Jan '26

alsa-project/alsa-lib issue #492 was opened from Eiseiche: It's similar to https://github.com/alsa-project/alsa-lib/issues/488. After updating from _alsa-lib 1.2.14-2_ to _alsa-lib 1.2.15.1-1_ , I only got the stereo fallback mode. [1.2.14-2 _alsa-info.txt](https://github.com/user-attachments/files/24400680/1.2.14-2… [1.2.15.1-1_alsa-info.txt](https://github.com/user-attachments/files/2440068… Issue URL : https://github.com/alsa-project/alsa-lib/issues/492 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: Qualcomm: add Asus vivobook 16 support
by GitHub pull_request - edited 29 Dec '25

29 Dec '25

alsa-project/alsa-ucm-conf pull request #668 was edited from kellermanrivero: Add support for Asus Vivobook 16 (F1607QA) with Snapdragon X1 SoC. This device topology is nearly identical to Lenovo Thinkpad T14s or Asus Vivobook S15. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/668 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/668.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add support for the Behringer ULM200D wireless microphones.
by GitHub pull_request - opened 29 Dec '25

29 Dec '25

alsa-project/alsa-ucm-conf pull request #669 was opened from kevincox: This is a USB receiver for two wireless microphones. Each microphone appears on a different channel. By default these would be detected as left and right. Just a disclaimer, I don't know what I am doing here. I used the UMC202HD as a reference and changed the values where it seemed reasonable. This device also has no output so I deleted everything that seemed output related. Seems to work. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/669 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/669.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] Applied "ASoC: tlv320aic32x4: Model CODEC_CLKIN in CCF" to the asoc tree
by Mark Brown 29 Dec '25

29 Dec '25

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

8 9

Asus vivobook 16 x126100
by GitHub pull_request - opened 29 Dec '25

29 Dec '25

alsa-project/alsa-ucm-conf pull request #668 was opened from kellermanrivero: Add support for Asus Vivobook 16 (F1607QA) with Snapdragon X1 SoC. This device topology is nearly identical to Lenovo Thinkpad T14s or Asus Vivobook S15. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/668 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/668.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Experience the Thrills of FNAF: A Guide to Playing
by reluctant.lobster.knax＠protectsmail.net 27 Dec '25

27 Dec '25

The world of horror games has spawned countless titles, but few have left a mark as significant as Fnaf Short for Five Nights at Freddy’s, this game invites players into a chilling atmosphere filled with suspense, strategy, and the occasional scare. If you’re looking to experience this unique blend of gameplay and story, you're in the right place! Let’s dive into how to play FNAF and get the most out of your experience. Gameplay: Surviving Five Nights https://fnafgamefree.com At its core, FNAF is a survival horror game where you take on the role of a nighttime security guard at Freddy Fazbear’s Pizza, a seemingly innocent family restaurant. As you settle in for your shift, the animatronic characters take on a life of their own. Your goal is to survive five nights without being caught by these eerie machines. The gameplay revolves around monitoring security cameras to track the movements of the animatronics. You'll be seated in a dimly lit office where you can control the cameras and manage limited resources like power. The key to winning is not only keeping an eye on the animatronics but also making strategic decisions about when to close doors and conserve power. If your power runs out, you’ll be left vulnerable, making it critical to maintain your resources wisely. As the nights progress, things get more intense. The animatronics become quicker and more unpredictable, and you’ll need to master the art of resource management and timing to survive until 6 AM. The tension builds day by day, and just when you think you’ve got a handle on things, the game introduces new surprises. Tips for a Successful Experience 1. Monitor Camera Feeds Wisely: Spend some time familiarizing yourself with the layout of the restaurant and the movements of the animatronics. Some will stay in specific locations longer, while others will be more unpredictable. 2. Use Your Resources: Conserve your power! Constantly shutting doors or using cameras drains your battery faster. It’s a balancing act of vigilance and resource management. 3. Listen Carefully: Audio cues are vital to your survival. Sounds can indicate the whereabouts of various animatronics, helping you decide the best course of action. 4. Practice Patience: Each night can feel like a marathon. Take your time to assess the situation; sometimes, waiting and watching is the best strategy. 5. Learn from Mistakes: Every time you get caught, use it as a learning opportunity. Each failure provides insights into the timing and strategies that will help you succeed in future attempts. Conclusion: Embrace the Fear Playing FNAF is not just about surviving; it’s an experience rich with tension and excitement that keeps players coming back for more. Each night presents new challenges, and the thrill of anticipation makes it a captivating choice for horror enthusiasts and gamers alike. Whether you’re a seasoned gamer or a newcomer to the horror genre, FNAF offers a unique experience that blends strategy with fear. So dim the lights, put on your headphones, and prepare to immerse yourself in the eerie world of animatronics and suspense. You can discover all aspects of the game and even play it for free at Fnaf. Enjoy the thrill, and remember to keep an eye on those cameras!

1 0

Experience the Thrills of FNAF: A Guide to Playing
by reluctant.lobster.knax＠protectsmail.net 27 Dec '25

27 Dec '25

The world of horror games has spawned countless titles, but few have left a mark as significant as Fnaf Short for Five Nights at Freddy’s, this game invites players into a chilling atmosphere filled with suspense, strategy, and the occasional scare. If you’re looking to experience this unique blend of gameplay and story, you're in the right place! Let’s dive into how to play FNAF and get the most out of your experience. Gameplay: Surviving Five Nights https://fnafgamefree.com At its core, FNAF is a survival horror game where you take on the role of a nighttime security guard at Freddy Fazbear’s Pizza, a seemingly innocent family restaurant. As you settle in for your shift, the animatronic characters take on a life of their own. Your goal is to survive five nights without being caught by these eerie machines. The gameplay revolves around monitoring security cameras to track the movements of the animatronics. You'll be seated in a dimly lit office where you can control the cameras and manage limited resources like power. The key to winning is not only keeping an eye on the animatronics but also making strategic decisions about when to close doors and conserve power. If your power runs out, you’ll be left vulnerable, making it critical to maintain your resources wisely. As the nights progress, things get more intense. The animatronics become quicker and more unpredictable, and you’ll need to master the art of resource management and timing to survive until 6 AM. The tension builds day by day, and just when you think you’ve got a handle on things, the game introduces new surprises. Tips for a Successful Experience 1. Monitor Camera Feeds Wisely: Spend some time familiarizing yourself with the layout of the restaurant and the movements of the animatronics. Some will stay in specific locations longer, while others will be more unpredictable. 2. Use Your Resources: Conserve your power! Constantly shutting doors or using cameras drains your battery faster. It’s a balancing act of vigilance and resource management. 3. Listen Carefully: Audio cues are vital to your survival. Sounds can indicate the whereabouts of various animatronics, helping you decide the best course of action. 4. Practice Patience: Each night can feel like a marathon. Take your time to assess the situation; sometimes, waiting and watching is the best strategy. 5. Learn from Mistakes: Every time you get caught, use it as a learning opportunity. Each failure provides insights into the timing and strategies that will help you succeed in future attempts. Conclusion: Embrace the Fear Playing FNAF is not just about surviving; it’s an experience rich with tension and excitement that keeps players coming back for more. Each night presents new challenges, and the thrill of anticipation makes it a captivating choice for horror enthusiasts and gamers alike. Whether you’re a seasoned gamer or a newcomer to the horror genre, FNAF offers a unique experience that blends strategy with fear. So dim the lights, put on your headphones, and prepare to immerse yourself in the eerie world of animatronics and suspense. You can discover all aspects of the game and even play it for free at Fnaf. Enjoy the thrill, and remember to keep an eye on those cameras!

1 0

[PATCH v11 4/4] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 26 Dec '25

26 Dec '25

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 118 ++++++++++++ sound/soc/codecs/rt5575-spi.h | 27 +++ sound/soc/codecs/rt5575.c | 352 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 ++++++ 6 files changed, 568 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 93a266a12a3b..90f09d4bdf8e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -212,6 +212,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1783,6 +1784,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate "Realtek ALC5575 Codec - I2C" + depends on I2C + +config SND_SOC_RT5575_SPI + tristate "Realtek ALC5575 Codec - SPI" + depends on SPI_MASTER && I2C + depends on SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d687d4f74363..a6406bc907a9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -253,6 +253,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-$(CONFIG_SND_SOC_RT5575_SPI) += rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -686,6 +688,7 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..9a349965435b --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,118 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_CMD_BURST_WRITE 5 +#define RT5575_SPI_BUF_LEN 240 + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +struct spi_device *rt5575_spi_get_device(struct device *dev) +{ + struct spi_device *spi; + struct spi_controller *ctlr; + struct device_node *spi_np; + u32 cs; + + spi_np = of_parse_phandle(dev->of_node, "spi-parent", 0); + if (!spi_np) { + dev_err(dev, "Failed to get spi-parent phandle\n"); + return NULL; + } + + if (of_property_read_u32_index(dev->of_node, "spi-parent", 1, &cs)) + cs = 0; + + ctlr = of_find_spi_controller_by_node(spi_np); + of_node_put(spi_np); + if (!ctlr) { + dev_err(dev, "Failed to get spi_controller\n"); + return NULL; + } + + if (cs >= ctlr->num_chipselect) { + dev_err(dev, "Chip select has wrong number %d\n", cs); + spi_controller_put(ctlr); + return NULL; + } + + spi = spi_new_device(ctlr, &(struct spi_board_info){ + .modalias = "rt5575", + .chip_select = cs, + .max_speed_hz = 10000000, + }); + + spi_controller_put(ctlr); + return spi; +} + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @spi: SPI device. + * @addr: Start address. + * @txbuf: Data buffer for writing. + * @len: Data length. + * + */ +static void rt5575_spi_burst_write(struct spi_device *spi, u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE, + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + memcpy(&buf.data, &txbuf[offset], end); + spi_write(spi, &buf, sizeof(buf)); + + offset += RT5575_SPI_BUF_LEN; + } +} + +int rt5575_spi_fw_load(struct spi_device *spi) +{ + struct device *dev = &spi->dev; + const struct firmware *firmware; + int i, ret; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw1.bin", + "realtek/rt5575/rt5575_fw2.bin", + "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin", + }; + static const u32 fw_addr[] = { 0x5f400000, 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], dev); + if (ret) { + dev_err(dev, "Request firmware failure: %d\n", ret); + return ret; + } + + rt5575_spi_burst_write(spi, fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } + + return 0; +} diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..084638820b6f --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,27 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +struct spi_device *rt5575_spi_get_device(struct device *dev); +int rt5575_spi_fw_load(struct spi_device *spi); +#else +static inline struct spi_device *rt5575_spi_get_device(struct device *dev) +{ + return NULL; +} + +static inline int rt5575_spi_fw_load(struct spi_device *spi) +{ + return -EINVAL; +} +#endif + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..c5525ad195ee --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,352 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#include "rt5575-spi.h" + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_BOOT: + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker CH-01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker CH-23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; + + rt5575->component = component; + + dev_info(dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + return regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + return regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt5575_fw_load_by_spi(struct rt5575_priv *rt5575) +{ + struct i2c_client *i2c = rt5575->i2c; + struct spi_device *spi; + struct device *dev = &i2c->dev; + int ret; + + spi = rt5575_spi_get_device(dev); + if (!spi) { + dev_err(dev, "Failed to get spi_device\n"); + return -ENODEV; + } + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + ret = rt5575_spi_fw_load(spi); + if (ret) { + dev_err(dev, "Load firmware failure: %d\n", ret); + return -ENODEV; + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + if (ret) { + dev_err(dev, "Run firmware failure: %d\n", ret); + return -ENODEV; + } + + return 0; +} + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val, boot; + struct device *dev = &i2c->dev; + + rt5575 = devm_kzalloc(dev, sizeof(struct rt5575_priv), GFP_KERNEL); + if (!rt5575) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(dev, "Failed to allocate DSP register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_BOOT, &boot); + if ((boot & RT5575_BOOT_MASK) == RT5575_BOOT_SPI) { + if (!IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI)) { + dev_err(dev, "Please enable CONFIG_SND_SOC_RT5575_SPI\n"); + return -ENODEV; + } + + if (rt5575_fw_load_by_spi(rt5575)) + return -ENODEV; + } + + return devm_snd_soc_register_component(dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = rt5575_of_match, + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..752a3c8f5aa9 --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,58 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_BOOT 0x8004 +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +#define RT5575_BOOT_MASK 0x3 +#define RT5575_BOOT_SPI 0x0 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.52.0

1 0

[PATCH v11 0/4] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 26 Dec '25

26 Dec '25

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v11: - Patch 1/4: - minor fixes - Patch 2/4: - change of_find_spi_controller_by_node() gating to CONFIG_OF - Patch 3/4: - nothing - Patch 4/4: - achieve reverse-christmas-tree notation - remove formal version check - revise check whether the firmware boots from SPI or not - minor fixes Changes in v10: - Patch 1/3: - export of_find_spi_controller_by_node() - Patch 2/3: - remove realtek,rt5575-use-spi - add spi-parent for firmware-loading - Patch 3/3: - use of_find_spi_controller_by_node() to get the SPI controller and add the spi device for firmware-loading - Link to v10: https://lore.kernel.org/all/20251216071853.3929135-1-oder_chiou@realtek.com/ Changes in v9: - Patch 1/2: - modify the comment - change the compatible name to "realtek,rt5575-use-spi" - Patch 2/2: - remove the standalone rt5575_spi_driver module and integrate its functionality into the I2C driver - move the SPI firmware-loading function to rt5575-spi.c - use the match data to distinguish between w/wo flash - minor fixes - Link to v9: https://lore.kernel.org/all/20251211110130.2925541-1-oder_chiou@realtek.com/ Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2: - add compatible enum "realtek,rt5575-with-spi" - Link to v8: https://lore.kernel.org/all/20251201105926.1714341-1-oder_chiou@realtek.com/ Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2: - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2: - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (4): spi: export of_find_spi_controller_by_node() spi: change of_find_spi_controller_by_node() gating to CONFIG_OF ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 .../bindings/sound/realtek,rt5575.yaml | 61 +++ drivers/spi/spi.c | 23 +- include/linux/spi/spi.h | 9 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 118 ++++++ sound/soc/codecs/rt5575-spi.h | 27 ++ sound/soc/codecs/rt5575.c | 352 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 +++ 9 files changed, 651 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

1 0

[PATCH 4/4] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 26 Dec '25

26 Dec '25

The ALC5575 integrates an audio DSP that typically loads its firmware from an external flash via its own SPI host interface. In certain hardware configurations, the firmware can alternatively be loaded through the SPI client interface. The driver provides basic mute and volume control functions. When the SPI client interface is enabled, firmware loading is handled by the SPI driver. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 118 ++++++++++++ sound/soc/codecs/rt5575-spi.h | 27 +++ sound/soc/codecs/rt5575.c | 352 ++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 ++++++ 6 files changed, 568 insertions(+) create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 93a266a12a3b..90f09d4bdf8e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -212,6 +212,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1305 imply SND_SOC_RT1308 imply SND_SOC_RT5514 + imply SND_SOC_RT5575 imply SND_SOC_RT5616 imply SND_SOC_RT5631 imply SND_SOC_RT5640 @@ -1783,6 +1784,15 @@ config SND_SOC_RT5514_SPI_BUILTIN bool # force RT5514_SPI to be built-in to avoid link errors default SND_SOC_RT5514=y && SND_SOC_RT5514_SPI=m +config SND_SOC_RT5575 + tristate "Realtek ALC5575 Codec - I2C" + depends on I2C + +config SND_SOC_RT5575_SPI + tristate "Realtek ALC5575 Codec - SPI" + depends on SPI_MASTER && I2C + depends on SND_SOC_RT5575 + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d687d4f74363..a6406bc907a9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -253,6 +253,8 @@ snd-soc-rt286-y := rt286.o snd-soc-rt298-y := rt298.o snd-soc-rt5514-y := rt5514.o snd-soc-rt5514-spi-y := rt5514-spi.o +snd-soc-rt5575-y := rt5575.o +snd-soc-rt5575-$(CONFIG_SND_SOC_RT5575_SPI) += rt5575-spi.o snd-soc-rt5616-y := rt5616.o snd-soc-rt5631-y := rt5631.o snd-soc-rt5640-y := rt5640.o @@ -686,6 +688,7 @@ obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5514_SPI_BUILTIN) += snd-soc-rt5514-spi.o +obj-$(CONFIG_SND_SOC_RT5575) += snd-soc-rt5575.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5575-spi.c b/sound/soc/codecs/rt5575-spi.c new file mode 100644 index 000000000000..9a349965435b --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.c @@ -0,0 +1,118 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575-spi.c -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/firmware.h> +#include <linux/of.h> +#include <linux/spi/spi.h> + +#include "rt5575-spi.h" + +#define RT5575_SPI_CMD_BURST_WRITE 5 +#define RT5575_SPI_BUF_LEN 240 + +struct rt5575_spi_burst_write { + u8 cmd; + u32 addr; + u8 data[RT5575_SPI_BUF_LEN]; + u8 dummy; +} __packed; + +struct spi_device *rt5575_spi_get_device(struct device *dev) +{ + struct spi_device *spi; + struct spi_controller *ctlr; + struct device_node *spi_np; + u32 cs; + + spi_np = of_parse_phandle(dev->of_node, "spi-parent", 0); + if (!spi_np) { + dev_err(dev, "Failed to get spi-parent phandle\n"); + return NULL; + } + + if (of_property_read_u32_index(dev->of_node, "spi-parent", 1, &cs)) + cs = 0; + + ctlr = of_find_spi_controller_by_node(spi_np); + of_node_put(spi_np); + if (!ctlr) { + dev_err(dev, "Failed to get spi_controller\n"); + return NULL; + } + + if (cs >= ctlr->num_chipselect) { + dev_err(dev, "Chip select has wrong number %d\n", cs); + spi_controller_put(ctlr); + return NULL; + } + + spi = spi_new_device(ctlr, &(struct spi_board_info){ + .modalias = "rt5575", + .chip_select = cs, + .max_speed_hz = 10000000, + }); + + spi_controller_put(ctlr); + return spi; +} + +/** + * rt5575_spi_burst_write - Write data to SPI by rt5575 address. + * @spi: SPI device. + * @addr: Start address. + * @txbuf: Data buffer for writing. + * @len: Data length. + * + */ +static void rt5575_spi_burst_write(struct spi_device *spi, u32 addr, const u8 *txbuf, size_t len) +{ + struct rt5575_spi_burst_write buf = { + .cmd = RT5575_SPI_CMD_BURST_WRITE, + }; + unsigned int end, offset = 0; + + while (offset < len) { + if (offset + RT5575_SPI_BUF_LEN <= len) + end = RT5575_SPI_BUF_LEN; + else + end = len % RT5575_SPI_BUF_LEN; + + buf.addr = cpu_to_le32(addr + offset); + memcpy(&buf.data, &txbuf[offset], end); + spi_write(spi, &buf, sizeof(buf)); + + offset += RT5575_SPI_BUF_LEN; + } +} + +int rt5575_spi_fw_load(struct spi_device *spi) +{ + struct device *dev = &spi->dev; + const struct firmware *firmware; + int i, ret; + static const char * const fw_path[] = { + "realtek/rt5575/rt5575_fw1.bin", + "realtek/rt5575/rt5575_fw2.bin", + "realtek/rt5575/rt5575_fw3.bin", + "realtek/rt5575/rt5575_fw4.bin", + }; + static const u32 fw_addr[] = { 0x5f400000, 0x5f600000, 0x5f7fe000, 0x5f7ff000 }; + + for (i = 0; i < ARRAY_SIZE(fw_addr); i++) { + ret = request_firmware(&firmware, fw_path[i], dev); + if (ret) { + dev_err(dev, "Request firmware failure: %d\n", ret); + return ret; + } + + rt5575_spi_burst_write(spi, fw_addr[i], firmware->data, firmware->size); + release_firmware(firmware); + } + + return 0; +} diff --git a/sound/soc/codecs/rt5575-spi.h b/sound/soc/codecs/rt5575-spi.h new file mode 100644 index 000000000000..084638820b6f --- /dev/null +++ b/sound/soc/codecs/rt5575-spi.h @@ -0,0 +1,27 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575-spi.h -- ALC5575 SPI driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_SPI_H__ +#define __RT5575_SPI_H__ + +#if IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI) +struct spi_device *rt5575_spi_get_device(struct device *dev); +int rt5575_spi_fw_load(struct spi_device *spi); +#else +static inline struct spi_device *rt5575_spi_get_device(struct device *dev) +{ + return NULL; +} + +static inline int rt5575_spi_fw_load(struct spi_device *spi) +{ + return -EINVAL; +} +#endif + +#endif /* __RT5575_SPI_H__ */ diff --git a/sound/soc/codecs/rt5575.c b/sound/soc/codecs/rt5575.c new file mode 100644 index 000000000000..c5525ad195ee --- /dev/null +++ b/sound/soc/codecs/rt5575.c @@ -0,0 +1,352 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * rt5575.c -- ALC5575 ALSA SoC audio component driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#include <linux/i2c.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "rt5575.h" +#include "rt5575-spi.h" + +static bool rt5575_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5575_BOOT: + case RT5575_ID: + case RT5575_ID_1: + case RT5575_MIXL_VOL: + case RT5575_MIXR_VOL: + case RT5575_PROMPT_VOL: + case RT5575_SPK01_VOL: + case RT5575_SPK23_VOL: + case RT5575_MIC1_VOL: + case RT5575_MIC2_VOL: + case RT5575_WNC_CTRL: + case RT5575_MODE_CTRL: + case RT5575_I2S_RATE_CTRL: + case RT5575_SLEEP_CTRL: + case RT5575_ALG_BYPASS_CTRL: + case RT5575_PINMUX_CTRL_2: + case RT5575_GPIO_CTRL_1: + case RT5575_DSP_BUS_CTRL: + case RT5575_SW_INT: + case RT5575_DSP_BOOT_ERR: + case RT5575_DSP_READY: + case RT5575_DSP_CMD_ADDR: + case RT5575_EFUSE_DATA_2: + case RT5575_EFUSE_DATA_3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(ob_tlv, -9525, 75, 0); + +static const struct snd_kcontrol_new rt5575_snd_controls[] = { + SOC_DOUBLE("Speaker CH-01 Playback Switch", RT5575_SPK01_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-01 Playback Volume", RT5575_SPK01_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Speaker CH-23 Playback Switch", RT5575_SPK23_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Speaker CH-23 Playback Volume", RT5575_SPK23_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic1 Capture Switch", RT5575_MIC1_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic1 Capture Volume", RT5575_MIC1_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE("Mic2 Capture Switch", RT5575_MIC2_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Mic2 Capture Volume", RT5575_MIC2_VOL, 17, 1, 167, 0, ob_tlv), + SOC_DOUBLE_R("Mix Playback Switch", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 31, 1, 1), + SOC_DOUBLE_R_TLV("Mix Playback Volume", RT5575_MIXL_VOL, RT5575_MIXR_VOL, 1, 127, 0, + ob_tlv), + SOC_DOUBLE("Prompt Playback Switch", RT5575_PROMPT_VOL, 31, 15, 1, 1), + SOC_DOUBLE_TLV("Prompt Playback Volume", RT5575_PROMPT_VOL, 17, 1, 167, 0, ob_tlv), +}; + +static const struct snd_soc_dapm_widget rt5575_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3RX", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3TX", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF4RX", "AIF4 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF4TX", "AIF4 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_INPUT("INPUT"), + SND_SOC_DAPM_OUTPUT("OUTPUT"), +}; + +static const struct snd_soc_dapm_route rt5575_dapm_routes[] = { + { "AIF1TX", NULL, "INPUT" }, + { "AIF2TX", NULL, "INPUT" }, + { "AIF3TX", NULL, "INPUT" }, + { "AIF4TX", NULL, "INPUT" }, + { "OUTPUT", NULL, "AIF1RX" }, + { "OUTPUT", NULL, "AIF2RX" }, + { "OUTPUT", NULL, "AIF3RX" }, + { "OUTPUT", NULL, "AIF4RX" }, +}; + +static long long rt5575_get_priv_id(struct rt5575_priv *rt5575) +{ + int priv_id_low, priv_id_high; + + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0xa0000000); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_2, &priv_id_low); + regmap_read(rt5575->regmap, RT5575_EFUSE_DATA_3, &priv_id_high); + regmap_write(rt5575->regmap, RT5575_EFUSE_PID, 0); + + return ((long long)priv_id_high << 32) | (long long)priv_id_low; +} + +static int rt5575_probe(struct snd_soc_component *component) +{ + struct rt5575_priv *rt5575 = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; + + rt5575->component = component; + + dev_info(dev, "Private ID: %llx\n", rt5575_get_priv_id(rt5575)); + + return 0; +} + +#define RT5575_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5575_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver rt5575_dai[] = { + { + .name = "rt5575-aif1", + .id = RT5575_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif2", + .id = RT5575_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif3", + .id = RT5575_AIF3, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, + { + .name = "rt5575-aif4", + .id = RT5575_AIF4, + .playback = { + .stream_name = "AIF4 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + .capture = { + .stream_name = "AIF4 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = RT5575_STEREO_RATES, + .formats = RT5575_FORMATS, + }, + }, +}; + +static const struct snd_soc_component_driver rt5575_soc_component_dev = { + .probe = rt5575_probe, + .controls = rt5575_snd_controls, + .num_controls = ARRAY_SIZE(rt5575_snd_controls), + .dapm_widgets = rt5575_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5575_dapm_widgets), + .dapm_routes = rt5575_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5575_dapm_routes), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static const struct regmap_config rt5575_dsp_regmap = { + .name = "dsp", + .reg_bits = 32, + .val_bits = 32, + .reg_stride = 2, +}; + +static int rt5575_i2c_read(void *context, unsigned int reg, unsigned int *val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + return regmap_read(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); +} + +static int rt5575_i2c_write(void *context, unsigned int reg, unsigned int val) +{ + struct i2c_client *client = context; + struct rt5575_priv *rt5575 = i2c_get_clientdata(client); + + return regmap_write(rt5575->dsp_regmap, reg | RT5575_DSP_MAPPING, val); +} + +static const struct regmap_config rt5575_regmap = { + .reg_bits = 16, + .val_bits = 32, + .reg_stride = 4, + .max_register = 0xfffc, + .readable_reg = rt5575_readable_register, + .reg_read = rt5575_i2c_read, + .reg_write = rt5575_i2c_write, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt5575_fw_load_by_spi(struct rt5575_priv *rt5575) +{ + struct i2c_client *i2c = rt5575->i2c; + struct spi_device *spi; + struct device *dev = &i2c->dev; + int ret; + + spi = rt5575_spi_get_device(dev); + if (!spi) { + dev_err(dev, "Failed to get spi_device\n"); + return -ENODEV; + } + + regmap_write(rt5575->dsp_regmap, 0xfafafafa, 0x00000004); + regmap_write(rt5575->dsp_regmap, 0x18008064, 0x00000000); + regmap_write(rt5575->dsp_regmap, 0x18008068, 0x0002ffff); + + ret = rt5575_spi_fw_load(spi); + if (ret) { + dev_err(dev, "Load firmware failure: %d\n", ret); + return -ENODEV; + } + + regmap_write(rt5575->dsp_regmap, 0x18000000, 0x00000000); + regmap_update_bits(rt5575->regmap, RT5575_SW_INT, 1, 1); + + regmap_read_poll_timeout(rt5575->regmap, RT5575_SW_INT, ret, !ret, 100000, 10000000); + if (ret) { + dev_err(dev, "Run firmware failure: %d\n", ret); + return -ENODEV; + } + + return 0; +} + +static int rt5575_i2c_probe(struct i2c_client *i2c) +{ + struct rt5575_priv *rt5575; + int ret, val, boot; + struct device *dev = &i2c->dev; + + rt5575 = devm_kzalloc(dev, sizeof(struct rt5575_priv), GFP_KERNEL); + if (!rt5575) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5575); + + rt5575->i2c = i2c; + + rt5575->dsp_regmap = devm_regmap_init_i2c(i2c, &rt5575_dsp_regmap); + if (IS_ERR(rt5575->dsp_regmap)) { + ret = PTR_ERR(rt5575->dsp_regmap); + dev_err(dev, "Failed to allocate DSP register map: %d\n", ret); + return ret; + } + + rt5575->regmap = devm_regmap_init(dev, NULL, i2c, &rt5575_regmap); + if (IS_ERR(rt5575->regmap)) { + ret = PTR_ERR(rt5575->regmap); + dev_err(dev, "Failed to allocate register map: %d\n", ret); + return ret; + } + + regmap_read(rt5575->regmap, RT5575_ID, &val); + if (val != RT5575_DEVICE_ID) { + dev_err(dev, "Device with ID register %08x is not rt5575\n", val); + return -ENODEV; + } + + regmap_read(rt5575->regmap, RT5575_BOOT, &boot); + if ((boot & RT5575_BOOT_MASK) == RT5575_BOOT_SPI) { + if (!IS_ENABLED(CONFIG_SND_SOC_RT5575_SPI)) { + dev_err(dev, "Please enable CONFIG_SND_SOC_RT5575_SPI\n"); + return -ENODEV; + } + + if (rt5575_fw_load_by_spi(rt5575)) + return -ENODEV; + } + + return devm_snd_soc_register_component(dev, &rt5575_soc_component_dev, rt5575_dai, + ARRAY_SIZE(rt5575_dai)); +} + +static const struct i2c_device_id rt5575_i2c_id[] = { + { "rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5575_i2c_id); + +static const struct of_device_id rt5575_of_match[] = { + { .compatible = "realtek,rt5575" }, + { } +}; +MODULE_DEVICE_TABLE(of, rt5575_of_match); + +static struct i2c_driver rt5575_i2c_driver = { + .driver = { + .name = "rt5575", + .owner = THIS_MODULE, + .of_match_table = rt5575_of_match, + }, + .probe = rt5575_i2c_probe, + .id_table = rt5575_i2c_id, +}; +module_i2c_driver(rt5575_i2c_driver); + +MODULE_DESCRIPTION("ASoC ALC5575 driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt5575.h b/sound/soc/codecs/rt5575.h new file mode 100644 index 000000000000..752a3c8f5aa9 --- /dev/null +++ b/sound/soc/codecs/rt5575.h @@ -0,0 +1,58 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt5575.h -- ALC5575 ALSA SoC audio driver + * + * Copyright(c) 2025 Realtek Semiconductor Corp. + * + */ + +#ifndef __RT5575_H__ +#define __RT5575_H__ + +#define RT5575_DEVICE_ID 0x10ec5575 +#define RT5575_DSP_MAPPING 0x18000000 + +#define RT5575_BOOT 0x8004 +#define RT5575_ID 0x8008 +#define RT5575_ID_1 0x800c +#define RT5575_MIXL_VOL 0x8a14 +#define RT5575_MIXR_VOL 0x8a18 +#define RT5575_PROMPT_VOL 0x8a84 +#define RT5575_SPK01_VOL 0x8a88 +#define RT5575_SPK23_VOL 0x8a8c +#define RT5575_MIC1_VOL 0x8a98 +#define RT5575_MIC2_VOL 0x8a9c +#define RT5575_WNC_CTRL 0x80ec +#define RT5575_MODE_CTRL 0x80f0 +#define RT5575_I2S_RATE_CTRL 0x80f4 +#define RT5575_SLEEP_CTRL 0x80f8 +#define RT5575_ALG_BYPASS_CTRL 0x80fc +#define RT5575_PINMUX_CTRL_2 0x81a4 +#define RT5575_GPIO_CTRL_1 0x8208 +#define RT5575_DSP_BUS_CTRL 0x880c +#define RT5575_SW_INT 0x0018 +#define RT5575_DSP_BOOT_ERR 0x8e14 +#define RT5575_DSP_READY 0x8e24 +#define RT5575_DSP_CMD_ADDR 0x8e28 +#define RT5575_EFUSE_DATA_2 0xc638 +#define RT5575_EFUSE_DATA_3 0xc63c +#define RT5575_EFUSE_PID 0xc660 + +#define RT5575_BOOT_MASK 0x3 +#define RT5575_BOOT_SPI 0x0 + +enum { + RT5575_AIF1, + RT5575_AIF2, + RT5575_AIF3, + RT5575_AIF4, + RT5575_AIFS, +}; + +struct rt5575_priv { + struct i2c_client *i2c; + struct snd_soc_component *component; + struct regmap *dsp_regmap, *regmap; +}; + +#endif /* __RT5575_H__ */ -- 2.52.0

1 0

[PATCH 3/4] ASoC: dt-bindings: realtek,rt5575: add support for ALC5575
by Oder Chiou 26 Dec '25

26 Dec '25

Audio codec with I2S, I2C and SPI. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reviewed-by: Rob Herring (Arm) <robh(a)kernel.org> --- .../bindings/sound/realtek,rt5575.yaml | 61 +++++++++++++++++++ 1 file changed, 61 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml new file mode 100644 index 000000000000..981ebc39b195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5575.yaml @@ -0,0 +1,61 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5575.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: ALC5575 audio CODEC + +maintainers: + - Oder Chiou <oder_chiou(a)realtek.com> + +description: + The device supports both I2C and SPI. I2C is mandatory, while SPI is + optional depending on the hardware configuration. SPI is used for + firmware loading if present. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5575 + + reg: + maxItems: 1 + + spi-parent: + description: + Optional phandle reference to the SPI controller used for firmware + loading. The argument specifies the chip select. + $ref: /schemas/types.yaml#/definitions/phandle-array + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + # I2C-only node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + }; + }; + + # I2C + optional SPI node + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@57 { + compatible = "realtek,rt5575"; + reg = <0x57>; + spi-parent = <&spi0 0>; /* chip-select 0 */ + }; + }; -- 2.52.0

1 0

[PATCH 0/4] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 26 Dec '25

26 Dec '25

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v11: - Patch 1/4: - minor fixes - Patch 2/4: - change of_find_spi_controller_by_node() gating to CONFIG_OF - Patch 3/4: - nothing - Patch 4/4: - achieve reverse-christmas-tree notation - remove formal version check - revise check whether the firmware boots from SPI or not - minor fixes Changes in v10: - Patch 1/3: - export of_find_spi_controller_by_node() - Patch 2/3: - remove realtek,rt5575-use-spi - add spi-parent for firmware-loading - Patch 3/3: - use of_find_spi_controller_by_node() to get the SPI controller and add the spi device for firmware-loading - Link to v10: https://lore.kernel.org/all/20251216071853.3929135-1-oder_chiou@realtek.com/ Changes in v9: - Patch 1/2: - modify the comment - change the compatible name to "realtek,rt5575-use-spi" - Patch 2/2: - remove the standalone rt5575_spi_driver module and integrate its functionality into the I2C driver - move the SPI firmware-loading function to rt5575-spi.c - use the match data to distinguish between w/wo flash - minor fixes - Link to v9: https://lore.kernel.org/all/20251211110130.2925541-1-oder_chiou@realtek.com/ Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2: - add compatible enum "realtek,rt5575-with-spi" - Link to v8: https://lore.kernel.org/all/20251201105926.1714341-1-oder_chiou@realtek.com/ Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2: - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2: - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (4): spi: export of_find_spi_controller_by_node() spi: change of_find_spi_controller_by_node() gating to CONFIG_OF ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 .../bindings/sound/realtek,rt5575.yaml | 61 +++ drivers/spi/spi.c | 23 +- include/linux/spi/spi.h | 9 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 118 ++++++ sound/soc/codecs/rt5575-spi.h | 27 ++ sound/soc/codecs/rt5575.c | 352 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 58 +++ 9 files changed, 651 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

1 0

envy24control: update README
by GitHub pull_request - opened 26 Dec '25

26 Dec '25

alsa-project/alsa-tools pull request #44 was opened from qaqland: A minor change to prevent downstream from mistakenly removing this GTK4-based package when dropping GTK2. Request URL : https://github.com/alsa-project/alsa-tools/pull/44 Patch URL : https://github.com/alsa-project/alsa-tools/pull/44.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

ucm2: Qualcomm: Add Lenovo Yoga Air 14s support
by GitHub pull_request - opened 25 Dec '25

25 Dec '25

alsa-project/alsa-ucm-conf pull request #667 was opened from shawnguo2: Lenovo Yoga Air 14s laptop is basically a Slim 7 for China, and they share the same audio configuration. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/667 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/667.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add Focusrite Vocaster interfaces
by GitHub pull_request - opened 24 Dec '25

24 Dec '25

alsa-project/alsa-ucm-conf pull request #666 was opened from geoffreybennett: Add split configs for the Focusrite Vocaster One and Two interfaces. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/666 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/666.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Enable Mute LED for Lenovo platform
by Kailang 23 Dec '25

23 Dec '25

Hi Takashi, Attach patch was for enable mute led functions of the Lenovo platform. Many Thanks. BR, Kailang

2 1

A Fun Guide to Mastering the Addictive World of Slope
by homely.scallop.ripl＠protectsmail.net 23 Dec '25

23 Dec '25

Looking for a simple yet maddeningly addictive game to kill some time? Then look no further than Slope. This fast-paced, minimalist game is easy to pick up, but incredibly difficult to master. It's a perfect blend of reflexes, anticipation, and a touch of luck, making it an instant classic for anyone looking for a quick and challenging gaming fix. Let's dive into what makes Slope so compelling and how you can improve your gameplay. What is Slope All About? Slope is, at its core, an endless runner. You control a ball rolling down a randomly generated, 3D slope. The goal is simple: survive as long as possible. The longer you last, the faster the ball travels and the more treacherous the obstacles become. There are no levels to complete, no boss fights to conquer. It’s just you, the ball, the slope, and the relentless pursuit of a higher score. Gameplay: Rolling with the Punches (and Cubes) The controls are incredibly straightforward. You use the left and right arrow keys (or the A and D keys) to steer the ball. That’s it! However, simplicity doesn't equate to ease. The slope is riddled with red blocks, gaps, and unpredictable twists and turns. Contact with a red block ends your run instantly. Successfully navigating the course requires quick reflexes, precise steering, and a good understanding of the game's physics. The real challenge lies in anticipating what's coming next. The landscape scrolls forward, giving you a brief glimpse of the obstacles ahead. This allows you to plan your movements and steer accordingly. However, as the speed increases, the time you have to react shrinks dramatically. What initially seems manageable quickly becomes a blur of obstacles demanding lightning-fast decisions. Tips and Tricks to Enhance Your Slope Game https://slopegame-online.com While Slope may seem purely based on luck, there are definitely strategies you can employ to improve your performance and climb the leaderboard. Here are a few tips to help you get started: Practice Makes Perfect: This is the most obvious, but also the most important. The more you play, the better you'll become at anticipating the obstacles and reacting quickly. Pay attention to patterns in the level generation; you might start to recognize recurring layouts and learn how to navigate them effectively. Small Adjustments are Key: Avoid making large, sweeping movements. Instead, use small, incremental adjustments to stay on course. Overcorrecting is a common mistake that often leads to collisions with red blocks. A gentle touch is far more effective than frantic steering. Look Ahead, Not Down: Focus your attention on the upcoming section of the slope, not directly at the ball. This will give you more time to react to upcoming obstacles and plan your movements. It's like driving; you wouldn't stare directly at your hood, would you? Embrace the Curves: Don't be afraid to use the curves in the slope to your advantage. Sometimes, the best way to avoid an obstacle is to ride the outer edge of a curve, using it to steer you around a danger zone. Learn to Predict: Although the levels are randomly generated, the game does have some underlying logic to how obstacles are placed. Playing enough Slope will help you pick up on these patterns, letting you predict, to some extent, what’s coming next. Master the Art of the “Brake” (Sort Of): While there's no dedicated brake button, you can effectively slow down by gently tapping the opposite direction of your current trajectory. This will subtly reduce your speed, giving you a fraction of a second more to react. It's a risky maneuver, but it can be useful in tight situations. Take Breaks: Slope can be incredibly addictive and frustrating. If you find yourself getting increasingly stressed or frustrated, take a break. Stepping away for a few minutes and coming back with a fresh perspective can significantly improve your performance. After all, nobody wants to be raging over a rolling ball! Speaking of which, you can always check out Slope again at Slope. Final Thoughts: Keep on Rolling! Slope is a wonderfully simple, yet incredibly challenging game that offers endless replayability. Its addictive nature stems from its straightforward gameplay loop: play, crash, analyze, improve, repeat. The thrill of surpassing your previous high score, even by a tiny margin, is immensely satisfying. So, go ahead, give it a try. Just be warned: you might find yourself spending hours trying to conquer that treacherous slope! Happy rolling!

1 0

[GIT PULL] ASoC fixes for v6.19-rc1
by Mark Brown 21 Dec '25

21 Dec '25

The following changes since commit 8f0b4cce4481fb22653697cced8d0d04027cb1e8: Linux 6.19-rc1 (2025-12-14 16:05:07 +1200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.19-rc1 for you to fetch changes up to 56d953a8d0da5e53c2594edde23465ec49385b1c: ASoC: Intel: mtl-match: Add 6-amp matches for CS35L56 (2025-12-19 13:25:16 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.19 We've been quite busy with fixes since the merge window, though not in any particularly exciting ways - the standout thing is the fix for _SX controls which were broken by a change to how we do clamping, otherwise it's all fairly run of the mill fixes and quirks. ---------------------------------------------------------------- Alexander Stein (4): ASoC: fsl_easrc: fix duplicate debugfs directory error ASoC: fsl_asrc_dma: fix duplicate debugfs directory error ASoC: fsl_sai: Add missing registers to cache default ASoC: fsl_xcvr: provide regmap names Andrew Elantsev (1): ASoC: amd: yc: Add quirk for Honor MagicBook X16 2025 Bard Liao (2): ASoC: sdw_utils: subtract the endpoint that is not present ASoC: SOF: ipc4-topology: set playback channel mask Chancel Liu (2): ASoC: fsl_sai: Constrain sample rates from audio PLLs only in master mode ASoC: fsl-asoc-card: Use of_property_present() for non-boolean properties Mark Brown (4): ASoC: Intel: common / SOF: Use function topologies for ASoC: SOF: ipc4-topology: fixes for 'exotic' format ASoC: fsl_easrc: Fix duplicate debufs entries ASoC: Intel: mtl-match: Add 6-amp matches for CS35L56 Peter Ujfalusi (6): ASoC: SOF: ipc4-topology: Prefer 32-bit DMIC blobs for 8-bit formats as well ASoC: SOF: ipc4-topology: Convert FLOAT to S32 during blob selection ASoC: SOF: topology: Add context when sink or source widget is missing ASoC: SOF: Intel: pci-mtl: Change the topology path to intel/sof-ipc4-tplg ASoC: soc-acpi / SOF: Add best_effort flag to get_function_tplg_files op ASoC: Intel: soc-acpi-intel-nvl-match: Drop rt722 l3 from the match table Robert Oscilowski (1): ASoC: qcom: sdm845: set quaternary MI2S codec DAI to I2S format Shengjiu Wang (1): ASoC: ak4458: remove the reset operation in probe and remove Shuming Fan (1): ASoC: rt1320: update VC blind write settings Stefan Binding (3): ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L56 with feedback ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L63 with feedback ASoC: ops: fix snd_soc_get_volsw for sx controls sheetal (1): ASoC: tegra: Fix uninitialized flat cache warning in tegra210_ahub include/sound/soc-acpi.h | 5 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/ak4458.c | 4 - sound/soc/codecs/rt1320-sdw.c | 16 ++-- sound/soc/fsl/fsl-asoc-card.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 3 + sound/soc/fsl/fsl_easrc.c | 3 + sound/soc/fsl/fsl_sai.c | 13 ++- sound/soc/fsl/fsl_xcvr.c | 3 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 104 +++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-nvl-match.c | 49 ---------- sound/soc/intel/common/sof-function-topology-lib.c | 5 +- sound/soc/intel/common/sof-function-topology-lib.h | 2 +- sound/soc/qcom/sdm845.c | 2 + sound/soc/sdw_utils/soc_sdw_utils.c | 8 +- sound/soc/soc-ops.c | 32 ++++--- sound/soc/sof/intel/pci-mtl.c | 6 +- sound/soc/sof/ipc4-topology.c | 49 ++++++---- sound/soc/sof/topology.c | 26 +++++- sound/soc/tegra/tegra210_ahub.c | 6 +- 20 files changed, 238 insertions(+), 113 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.19-rc1
by Mark Brown 20 Dec '25

20 Dec '25

The following changes since commit 8f0b4cce4481fb22653697cced8d0d04027cb1e8: Linux 6.19-rc1 (2025-12-14 16:05:07 +1200) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.19-rc1 for you to fetch changes up to 56d953a8d0da5e53c2594edde23465ec49385b1c: ASoC: Intel: mtl-match: Add 6-amp matches for CS35L56 (2025-12-19 13:25:16 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.19 We've been quite busy with fixes since the merge window, though not in any particularly exciting ways - the standout thing is the fix for _SX controls which were broken by a change to how we do clamping, otherwise it's all fairly run of the mill fixes and quirks. ---------------------------------------------------------------- Alexander Stein (4): ASoC: fsl_easrc: fix duplicate debugfs directory error ASoC: fsl_asrc_dma: fix duplicate debugfs directory error ASoC: fsl_sai: Add missing registers to cache default ASoC: fsl_xcvr: provide regmap names Andrew Elantsev (1): ASoC: amd: yc: Add quirk for Honor MagicBook X16 2025 Bard Liao (2): ASoC: sdw_utils: subtract the endpoint that is not present ASoC: SOF: ipc4-topology: set playback channel mask Chancel Liu (2): ASoC: fsl_sai: Constrain sample rates from audio PLLs only in master mode ASoC: fsl-asoc-card: Use of_property_present() for non-boolean properties Mark Brown (4): ASoC: Intel: common / SOF: Use function topologies for ASoC: SOF: ipc4-topology: fixes for 'exotic' format ASoC: fsl_easrc: Fix duplicate debufs entries ASoC: Intel: mtl-match: Add 6-amp matches for CS35L56 Peter Ujfalusi (6): ASoC: SOF: ipc4-topology: Prefer 32-bit DMIC blobs for 8-bit formats as well ASoC: SOF: ipc4-topology: Convert FLOAT to S32 during blob selection ASoC: SOF: topology: Add context when sink or source widget is missing ASoC: SOF: Intel: pci-mtl: Change the topology path to intel/sof-ipc4-tplg ASoC: soc-acpi / SOF: Add best_effort flag to get_function_tplg_files op ASoC: Intel: soc-acpi-intel-nvl-match: Drop rt722 l3 from the match table Robert Oscilowski (1): ASoC: qcom: sdm845: set quaternary MI2S codec DAI to I2S format Shengjiu Wang (1): ASoC: ak4458: remove the reset operation in probe and remove Shuming Fan (1): ASoC: rt1320: update VC blind write settings Stefan Binding (3): ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L56 with feedback ASoC: Intel: soc-acpi-intel-mtl-match: Add 6 amp CS35L63 with feedback ASoC: ops: fix snd_soc_get_volsw for sx controls sheetal (1): ASoC: tegra: Fix uninitialized flat cache warning in tegra210_ahub include/sound/soc-acpi.h | 5 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/ak4458.c | 4 - sound/soc/codecs/rt1320-sdw.c | 16 ++-- sound/soc/fsl/fsl-asoc-card.c | 8 +- sound/soc/fsl/fsl_asrc_dma.c | 3 + sound/soc/fsl/fsl_easrc.c | 3 + sound/soc/fsl/fsl_sai.c | 13 ++- sound/soc/fsl/fsl_xcvr.c | 3 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 104 +++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-nvl-match.c | 49 ---------- sound/soc/intel/common/sof-function-topology-lib.c | 5 +- sound/soc/intel/common/sof-function-topology-lib.h | 2 +- sound/soc/qcom/sdm845.c | 2 + sound/soc/sdw_utils/soc_sdw_utils.c | 8 +- sound/soc/soc-ops.c | 32 ++++--- sound/soc/sof/intel/pci-mtl.c | 6 +- sound/soc/sof/ipc4-topology.c | 49 ++++++---- sound/soc/sof/topology.c | 26 +++++- sound/soc/tegra/tegra210_ahub.c | 6 +- 20 files changed, 238 insertions(+), 113 deletions(-)

1 0

[PATCH v10 0/3] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 19 Dec '25

19 Dec '25

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v10: - Patch 1/3: - export of_find_spi_controller_by_node() - Patch 2/3: - remove realtek,rt5575-use-spi - add spi-parent for firmware-loading - Patch 3/3: - use of_find_spi_controller_by_node() to get the SPI controller and add the spi device for firmware-loading Changes in v9: - Patch 1/2: - modify the comment - change the compatible name to "realtek,rt5575-use-spi" - Patch 2/2: - remove the standalone rt5575_spi_driver module and integrate its functionality into the I2C driver - move the SPI firmware-loading function to rt5575-spi.c - use the match data to distinguish between w/wo flash - minor fixes - Link to v9: https://lore.kernel.org/all/20251211110130.2925541-1-oder_chiou@realtek.com/ Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2: - add compatible enum "realtek,rt5575-with-spi" - Link to v8: https://lore.kernel.org/all/20251201105926.1714341-1-oder_chiou@realtek.com/ Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2: - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2: - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (3): spi: export of_find_spi_controller_by_node() ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 .../bindings/sound/realtek,rt5575.yaml | 61 +++ drivers/spi/spi.c | 3 +- include/linux/spi/spi.h | 11 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 119 ++++++ sound/soc/codecs/rt5575-spi.h | 26 ++ sound/soc/codecs/rt5575.c | 359 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++ 9 files changed, 645 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

5 9

ucm2 USB-Audio: Antelope Audio Zen Go: add UCM2 configuration files
by GitHub pull_request - edited 19 Dec '25

19 Dec '25

alsa-project/alsa-ucm-conf pull request #659 was edited from ryodeushii: Add support for Antelope Audio Zen Go SC [alsa-info.txt](https://github.com/user-attachments/files/24074814/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/659 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/659.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof: hdmi: Add 5.1 and 7.1 variants of HDMI devices when using IPC4
by GitHub pull_request - reopened 19 Dec '25

19 Dec '25

alsa-project/alsa-ucm-conf pull request #633 was reopened from ujfalusi: Systems using IPC4 can support up to 8 channels of audio (and passthrough) via HDMI. In UCM the default PlaybackChannels is set to 2, which prevents users from selecting multichannel configurations. When probing the card, Pipewire will drop configurations that are not supported either by the PCM device or based on ELD information. This means that if the equipment supports only stereo then the 5.1 and 7.1 variants should not be visible, if the equipment is 5.1 capable, then only the 7.1 variant is removed. The kernel will refine the PCM parameters based on the ELD information as wall when https://lore.kernel.org/linux-sound/20251029073600.13624-1-peter.ujfalusi@l… is applied. @perexg, @wtay, @ford-prefect, is this something which can help PW and user space to handle the HDMI a bit better with SOF? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/633.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Add support for CS42L43B variant
by GitHub pull_request - opened 19 Dec '25

19 Dec '25

alsa-project/alsa-ucm-conf pull request #665 was opened from mstrozek: From UCM point of view CS42L43B codec adds 2 new decimators and removes a control to change Decimator 1 mode from base CS42L43. Thus reuse CS42L43 files, and guard the CS42L43B additions behind a condition that the new Decimator controls exist. Link: https://lore.kernel.org/all/20251219100235.1247053-1-mstrozek@opensource.ci… Set as Draft PR until kernel support is merged. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/665 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/665.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI Z790I EDGE WiFI not detecting front panel audio
by GitHub issues - opened 19 Dec '25

19 Dec '25

alsa-project/alsa-ucm-conf issue #664 was opened from abhisas: This Motherboard is in already in the UCM conf list for the ALC4080, but the front panel audio still does not work, and I am not sure why. I am on Fedora version 43. Any help would be appreciated. Below is output from lsusb ` us 001 Device 007: ID 0db0:62a4 Micro Star International USB Audio ` Alsa info.sh output: http://alsa-project.org/db/?f=01026cd96771a91f619f9c36abb51b6bfe9321f6 Output from plugging in front panel jack (alsactl monitor) ```node hw:3, #14 (0,0,0,Mic - Input Jack,0) VALUE node hw:3, #25 (0,0,0,Headphone - Output Jack,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #19 (2,0,0,PCM Playback Volume,0) VALUE node hw:3, #14 (0,0,0,Mic - Input Jack,0) VALUE ``` alsaucm -c hw:2 dump text" ``` ALSA lib parser.c:305:(error_node) [error.ucm] UCM is not supported for this HDA model (HDA NVidia at 0x82080000 irq 17) ALSA lib main.c:1804:(snd_use_case_mgr_open) [error.ucm] failed to import hw:2 use case configuration -6 alsaucm: error failed to open sound card hw:2: No such device or address ``` aplay -l: ``` **** List of PLAYBACK Hardware Devices **** card 0: Stream [JBL Quantum Stream], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [Odyssey G50A] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/664 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/8] ASoC: tas2783A: sdw_utils: support ch 3 & 4
by Niranjan H Y 18 Dec '25

18 Dec '25

Currently the machine driver for tas2783A can only support 2 channels. This patch adds support for 2 channel playback with 4 device setup. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- v2: - no change --- sound/soc/sdw_utils/soc_sdw_ti_amp.c | 4 ++++ sound/soc/sdw_utils/soc_sdw_utils.c | 22 ++++++++++++++++++---- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_ti_amp.c b/sound/soc/sdw_utils/soc_sdw_ti_amp.c index cbd60faec..488ef2ef4 100644 --- a/sound/soc/sdw_utils/soc_sdw_ti_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_ti_amp.c @@ -58,6 +58,10 @@ int asoc_sdw_ti_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, strscpy(speaker, "Left Spk", sizeof(speaker)); } else if (!strncmp(prefix, "tas2783-2", strlen("tas2783-2"))) { strscpy(speaker, "Right Spk", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-3", strlen("tas2783-3"))) { + strscpy(speaker, "Left Spk2", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-4", strlen("tas2783-4"))) { + strscpy(speaker, "Right Spk2", sizeof(speaker)); } else { ret = -EINVAL; dev_err(card->dev, "unhandled prefix %s", prefix); diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index f169d9589..1d1ba7a25 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -40,11 +40,25 @@ static const struct snd_soc_dapm_widget lr_spk_widgets[] = { SND_SOC_DAPM_SPK("Right Spk", NULL), }; +static const struct snd_soc_dapm_widget lr_4spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk2", NULL), + SND_SOC_DAPM_SPK("Right Spk2", NULL), +}; + static const struct snd_kcontrol_new lr_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new lr_4spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + SOC_DAPM_PIN_SWITCH("Left Spk2"), + SOC_DAPM_PIN_SWITCH("Right Spk2"), +}; + static const struct snd_soc_dapm_widget rt700_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), @@ -69,10 +83,10 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = asoc_sdw_ti_amp_init, .rtd_init = asoc_sdw_ti_spk_rtd_init, - .controls = lr_spk_controls, - .num_controls = ARRAY_SIZE(lr_spk_controls), - .widgets = lr_spk_widgets, - .num_widgets = ARRAY_SIZE(lr_spk_widgets), + .controls = lr_4spk_controls, + .num_controls = ARRAY_SIZE(lr_4spk_controls), + .widgets = lr_4spk_widgets, + .num_widgets = ARRAY_SIZE(lr_4spk_widgets), }, }, .dai_num = 1, -- 2.43.0

3 9

Dell XPS 14 9440 speaker don't work after update to v1.2.15
by GitHub issues - opened 17 Dec '25

17 Dec '25

alsa-project/alsa-ucm-conf issue #663 was opened from taaem: Hello, I have a Dell XPS 14 (9440) and after the last update (on openSUSE Tumbleweed) of `alsa-ucm-conf` the speakers disappeared in the system and pipewire only reports a `dummy-output`. For reference, the downstream bug, with all the logs is https://bugzilla.suse.com/show_bug.cgi?id=1255123, if needed I can provide additional logs or info! Thanks Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/663 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ALSA: usb-mixer: us16x08: validate meter packet indices
by Shipei Qu 17 Dec '25

17 Dec '25

Hi, resending with a properly formatted diff (the previous email had a malformed patch header). The change itself is the same: while fuzzing a USB gadget that emulates a Tascam US-16x08 we found that get_meter_levels_from_urb() in mixer_us16x08.c uses a channel index taken directly from the 64-byte meter packet to index meter_level[], comp_level[] and master_level[] without any bounds checking. A malformed packet can therefore cause out-of-bounds writes in the snd_us16x08_meter_store. A malicious USB audio device (or USB gadget implementation) that pretends to be a US-16x08-compatible interface can trigger this by sending crafted meter packets. We have a small USB gadget-based PoC for this behaviour and can share it if that would be helpful. This driver is used by common distributions (e.g. Ubuntu) when a US-16x08 or compatible USB audio device is present. The same pattern is present in current mainline kernels. This issue was first reported via security(a)kernel.org. The kernel security team explained that, in the upstream threat model, USB endpoints are expected to be trusted (i.e. only trusted devices should be bound to drivers), so they consider this a normal bug rather than a security vulnerability, and asked us to send a fix to the development lists. The patch below adds simple range checks before updating these arrays. Reported-by: DARKNAVY (@DarkNavyOrg) <vr(a)darknavy.com> Signed-off-by: Shipei Qu <qu(a)darknavy.com> --- sound/usb/mixer_us16x08.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 1c5712c31..f9df40730 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -655,17 +655,25 @@ static void get_meter_levels_from_urb(int s, u8 *meter_urb) { int val = MUC2(meter_urb, s) + (MUC3(meter_urb, s) << 8); + int ch = MUB2(meter_urb, s) - 1; + + if (ch < 0) + return; if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) { - if (MUC0(meter_urb, s) == 0x72) - store->meter_level[MUB2(meter_urb, s) - 1] = val; - if (MUC0(meter_urb, s) == 0xb2) - store->comp_level[MUB2(meter_urb, s) - 1] = val; + if (ch < SND_US16X08_MAX_CHANNELS) { + if (MUC0(meter_urb, s) == 0x72) + store->meter_level[ch] = val; + if (MUC0(meter_urb, s) == 0xb2) + store->comp_level[ch] = val; + } } if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && - MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) - store->master_level[MUB2(meter_urb, s) - 1] = val; + MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) { + if (ch < ARRAY_SIZE(store->master_level)) + store->master_level[ch] = val; + } } /* Function to retrieve current meter values from the device. -- 2.45.1

2 2

[PATCH v1] ASoC: amd: acp: Audio is not resuming after s0ix
by Raghavendra Prasad Mallela 17 Dec '25

17 Dec '25

From: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> Audio fails to resume after system exits suspend mode due to accessing incorrect ring buffer address during resume. This patch resolves issue by selecting correct address based on the ACP version. Signed-off-by: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> Signed-off-by: Raghavendra Prasad Mallela <raghavendraprasad.mallela(a)amd.com> --- sound/soc/amd/acp/acp-legacy-common.c | 32 +++++++++++++++++++++------ 1 file changed, 25 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 3078f459e005..da80c761d657 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -208,7 +208,7 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, struct acp_resource *rsrc = chip->rsrc; struct acp_stream *stream = substream->runtime->private_data; u32 reg_dma_size, reg_fifo_size, reg_fifo_addr; - u32 phy_addr, acp_fifo_addr, ext_int_ctrl; + u32 phy_addr = 0, acp_fifo_addr, ext_int_ctrl; unsigned int dir = substream->stream; switch (dai->driver->id) { @@ -219,7 +219,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, SP_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_TX_FIFOADDR(chip); reg_fifo_size = ACP_I2S_TX_FIFOSIZE(chip); - phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_SP_TX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_I2S_TX_RINGBUFADDR(chip)); } else { reg_dma_size = ACP_I2S_RX_DMA_SIZE(chip); @@ -227,7 +230,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, SP_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_RX_FIFOADDR(chip); reg_fifo_size = ACP_I2S_RX_FIFOSIZE(chip); - phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_SP_RX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_I2S_RX_RINGBUFADDR(chip)); } break; @@ -238,7 +244,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, BT_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_TX_FIFOADDR(chip); reg_fifo_size = ACP_BT_TX_FIFOSIZE(chip); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_BT_TX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_BT_TX_RINGBUFADDR(chip)); } else { reg_dma_size = ACP_BT_RX_DMA_SIZE(chip); @@ -246,7 +255,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, BT_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_RX_FIFOADDR(chip); reg_fifo_size = ACP_BT_RX_FIFOSIZE(chip); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_BT_RX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_BT_RX_RINGBUFADDR(chip)); } break; @@ -257,7 +269,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, HS_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_HS_TX_FIFOADDR; reg_fifo_size = ACP_HS_TX_FIFOSIZE; - phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_HS_TX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_HS_TX_RINGBUFADDR); } else { reg_dma_size = ACP_HS_RX_DMA_SIZE; @@ -265,7 +280,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, HS_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_HS_RX_FIFOADDR; reg_fifo_size = ACP_HS_RX_FIFOSIZE; - phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_HS_RX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_HS_RX_RINGBUFADDR); } break; -- 2.17.1

5 4

[PATCH 1/3] ALSA: usb-audio: Update for native DSD support quirks
by Jussi Laako 17 Dec '25

17 Dec '25

Maintenance patch for native DSD support. Add set of missing device and vendor quirks; TEAC, Esoteric, Luxman and Musical Fidelity. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 61bd61ffb1b2..94a8fdc9c6d3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2230,6 +2230,12 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { DEVICE_FLG(0x0644, 0x806b, /* TEAC UD-701 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), + DEVICE_FLG(0x0644, 0x807d, /* TEAC UD-507 */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), + DEVICE_FLG(0x0644, 0x806c, /* Esoteric XD */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2388,6 +2394,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x30be, 0x0101, /* Schiit Hel */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x3255, 0x0000, /* Luxman D-10X */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x339b, 0x3a07, /* Synaptics HONOR USB-C HEADSET */ QUIRK_FLAG_MIXER_PLAYBACK_MIN_MUTE), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ @@ -2431,6 +2439,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2622, /* IAG Limited devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x2772, /* Musical Fidelity devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x278b, /* Rotel? */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x292b, /* Gustard/Ess based devices */ -- 2.43.0

3 4

Behringer UMC404HD Line A/B volume too low when used in "Pro Audio" mode
by GitHub issues - edited 16 Dec '25

16 Dec '25

alsa-project/alsa-ucm-conf issue #662 was edited from mrbumpy409: The Behringer UMC404HD shows five volume faders in alsamixer: * **Line A** and **UMC404HD 192k Output Front**: These faders are linked (changing one changes the other) and control the level of **L & R MAIN OUT** and **PLAYBACK OUTPUT A 1/L & 2/R** on the back of the unit. * **Line B** and **UMC404HD 192k Output Back**: These faders are linked and control the level of **PLAYBACK OUTPUT B 3/L & 4/R**. * **UMC404HD 192k Output 1**: The primary volume control for the unit. How these sliders are used by Pipewire depends on the mode selected for the Behringer: * **Default**: Two output devices are presented to the user: - **UMC404HD 192k Line A**: Controls the "Line A" volume in alsamixer. - **UMC404HD 192k Line B**: Controls the "Line B" volume in alsamixer. - The "UMC404HD 192k Output 1" slider in alsamixer appears to be unused. * **Direct UMC404HD 192k**: One output device is presented to the user: - **Direct UMC404HD 192k**: Doesn't seem to be manipulating any of the hardware output levels in alsamixer. - The "Line A" and "Line B" channels are not available via Pipewire in this mode. * **Pro Audio**: One output device is presented to the user: - **UMC404HD 192k Pro**: Controls the "UMC404HD 192k Output 1" output level in alsamixer. - The "Line A" and "Line B" channels are not available via Pipewire in this mode. I would imagine most people are using a Behringer UMC404HD for recording purposes, and therefore would likely be using it in "Pro Audio" mode (or at least they should be if they care about latency). Since a recent update (Fedora 43 KDE), the levels of the "Line A" and "Line B" channels (and their linked pairs) are being set by default to **41**, which results in audio that is far too quiet when using the line outputs (headphone output is unaffected). Furthermore, these low levels throw off the balance between the computer's output and the UMC404HD's live input monitoring. "Line A" and "Line B" channels should both be at **100** by default to match the Windows driver (and previous ALSA version?) and provide correct balance with the live monitoring feature. Currently, I have to run alsamixer on every boot to adjust "Line A" and "Line B" to **100**. Am I right to report this to ALSA, or is this something Pipewire should be handling? Pipewire doesn't appear to touch the "Line A" or "Line B" levels at all when in "Pro Audio" mode, so I'd assume this is ALSA's doing, but I don't really know. ### My System * **OS:** Fedora Linux 43 Plasma Desktop Edition (Plasma Desktop 6.5.4, KDE Frameworks 6.21.0, Qt 6.10.1) * **Linux Kernel:** 6.17.11-300.fc43.x86_64 (64-bit) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/662 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Behringer UMC404HD Line A/B volume too low when used in "Pro Audio" mode
by GitHub issues - opened 16 Dec '25

16 Dec '25

alsa-project/alsa-ucm-conf issue #662 was opened from mrbumpy409: The Behringer UMC404HD shows five volume faders in alsamixer: * **Line A** and **UMC404HD 192k Output Front**: These faders are linked (changing one changes the other) and control the level of **L & R MAIN OUT** and **PLAYBACK OUTPUT A 1/L & 2/R** on the back of the unit. * **Line B** and **UMC404HD 192k Output Back**: These faders are linked and control the level of **PLAYBACK OUTPUT B 3/L & 4/R**. * **UMC404HD 192k Output 1**: The primary volume control for the unit. How these sliders are used by Pipewire depends on the mode selected for the Behringer: * **Default**: Two output devices are presented to the user: - **UMC404HD 192k Line A**: Controls the "Line A" volume in alsamixer. - **UMC404HD 192k Line B**: Controls the "Line B" volume in alsamixer. - The "UMC404HD 192k Output 1" slider in alsamixer appears to be unused. * **Direct UMC404HD 192k**: One output device is presented to the user: - **Direct UMC404HD 192k**: Doesn't seem to be manipulating any of the hardware output levels in alsamixer. - The "Line A" and "Line B" channels are not available via Pipewire in this mode. * **Pro Audio**: One output device is presented to the user: - **UMC404HD 192k Pro**: Controls the "UMC404HD 192k Output 1" output level in alsamixer. - The "Line A" and "Line B" channels are not available via Pipewire in this mode. I would imagine most people are using a Behringer UMC404HD for recording purposes, and therefore would likely be using it in "Pro Audio" mode (or at least they should be if they care about latency). Since a recent update (Fedora 43 KDE), the levels of the "Line A" and "Line B" channels (and their linked pairs) are being set by default to **41**, which results in audio that is far too quiet when using the line outputs (headphone output is unaffected). Furthermore, these low levels throw off the balance between the computer's output and the UMC404HD's live input monitoring. "Line A" and "Line B" channels should both be at **100** by default to match the Windows driver (and previous ALSA version?) and provide correct balance with the live monitoring feature. Currently, I have to run alsamixer on every boot to adjust "Line A" and "Line B" to **100**. Am I right to report this to ALSA, or is this something Pipewire should be handling? Pipewire doesn't appear to touch the "Line A" or "Line B" levels at all when in "Pro Audio" mode, so I'd assume this is ALSA's doing, but I don't really know. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/662 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib 1.2.15: only fallback stereo available (bisected)
by GitHub issues - opened 16 Dec '25

16 Dec '25

alsa-project/alsa-lib issue #491 was opened from Just4GH: With alsa-lib 1.2.15 only stereo is available. ``` git bisect bad Bisecting: 0 revisions left to test after this (roughly 0 steps) [ eda76146c5653ff1d5bc4b4c53f7a2d5ccc17da2 ] conf: fix load_for_all_cards() - do not merge the card specific contents ``` Reverting eda76146c5653ff1d5bc4b4c53f7a2d5ccc17da2 allows the usual profiles to load and 5.1 output. ``` $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC1150 Analog [ALC1150 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC1150 Digital [ALC1150 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/491 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

aplay: add missing break before the default case
by GitHub pull_request - opened 16 Dec '25

16 Dec '25

alsa-project/alsa-utils pull request #315 was opened from bardliao: Add the break before the default case back. Otherwise, all cases will fall into the default/error case. Fixes: e78583ab7cde ("aplay: reorganize format handling in begin_wave()") Request URL : https://github.com/alsa-project/alsa-utils/pull/315 Patch URL : https://github.com/alsa-project/alsa-utils/pull/315.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

mixer: fix cache coherency issue in multi-application scenario
by GitHub pull_request - edited 16 Dec '25

16 Dec '25

alsa-project/alsa-lib pull request #490 was edited from DevreeseJorik: Remove conditional hardware writes in set_volume_ops and set_switch_ops to ensure hardware state is always updated. This fixes an issue where setting controls to the same value repeatedly does nothing, even if the hardware state has changed due to actions from other applications. --- Given this code has been in alsa-lib for 20 years, I assume there is a good reason for it, or perhaps a workaround exists that I could not find. The issue we faced is that after an initial write, if the same value is written again, but in the meantime a different application altered the hardware state, the new change will not occur, yet no error code is provided. Also, reading the current state returns the current application's state, rather than the actual hardware state. Is alsa-lib not designed with multi-application uses in mind? Request URL : https://github.com/alsa-project/alsa-lib/pull/490 Patch URL : https://github.com/alsa-project/alsa-lib/pull/490.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

mixer: fix cache coherency issue in multi-application scenario
by GitHub pull_request - opened 16 Dec '25

16 Dec '25

alsa-project/alsa-lib pull request #490 was opened from DevreeseJorik: Remove conditional hardware writes in set_volume_ops and set_switch_ops to ensure hardware state is always updated. This fixes an issue where setting controls to the same value repeatedly does nothing, even if the hardware state has changed due to actions from other applications. Request URL : https://github.com/alsa-project/alsa-lib/pull/490 Patch URL : https://github.com/alsa-project/alsa-lib/pull/490.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

HP OMEN Transcend 14 (board 8C58, Meteor Lake) – SoundWire speakers not configured (missing UCM profile)
by GitHub issues - opened 15 Dec '25

15 Dec '25

alsa-project/alsa-ucm-conf issue #661 was opened from Farnaz03: ### **Hardware** * Model: HP OMEN Transcend Gaming Laptop 14-fb0xxx * Board ID: **8C58** * CPU: Intel Meteor Lake-P * Audio: Intel SOF + SoundWire * Codec(s): rt711-sdca (headset), rt1316 (speaker amp) * OS: Ubuntu 24.04 LTS ### **Kernels tested** * 6.8.0-88-generic * 6.8.0-90-generic * 6.14.0-1008-oem * 6.14.0-1014-oem --- ### **Description** Internal speakers produce no sound on Linux. SOF firmware and SoundWire devices are detected successfully, and a Speaker PCM exists, but no UCM profile configures or enables the speaker path. Audio reportedly worked earlier (May–July 2025) and stopped after an update around August 2025. --- ### **DMI / board identification** ``` sys_vendor : HP product_name : OMEN Transcend Gaming Laptop 14-fb0xxx board_name : 8C58 board_vendor : HP ``` --- ### **SOF topology in use** ``` intel/sof-ace-tplg/sof-mtl-rt711-l0-rt1316-l3-2ch.tplg ``` --- ### **aplay -l** ``` card 1: sofsoundwire [sof-soundwire], device 0: Jack Out card 1: sofsoundwire [sof-soundwire], device 2: Speaker card 1: sofsoundwire [sof-soundwire], device 31: Deepbuffer Jack Out ... ``` --- ### **ALSA direct tests (PipeWire bypassed)** ``` aplay -D hw:1,2 /usr/share/sounds/alsa/Front_Center.wav aplay -D hw:1,0 /usr/share/sounds/alsa/Front_Center.wav aplay -D hw:1,31 /usr/share/sounds/alsa/Front_Center.wav ``` All fail with: ``` aplay: set_params: Channels count non available ``` --- ### **UCM status** ``` /usr/share/alsa/ucm2/sof-soundwire/ -> empty (no profiles) ``` --- ### **Expected result** Internal speakers work via ALSA / PipeWire. ### **Actual result** Speaker PCM exists but is unusable due to missing UCM configuration. ### **Request** Add a UCM profile for HP OMEN Transcend 14 (board 8C58) under `sof-soundwire`, enabling the speaker path for the existing SOF topology. ### Optional notes / suggestions This platform uses SOF SoundWire with rt1316 speaker amp and rt711 headset codec, and the loaded topology already supports a 2-channel speaker path: sof-mtl-rt711-l0-rt1316-l3-2ch.tplg It may be possible to adapt or clone an existing HP SoundWire UCM profile for Meteor Lake platforms using rt1316 (if available), matching DMI board ID `HP / 8C58`. I am happy to test patches or provide additional logs if needed. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/661 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add conditional channel count on Scarlett 18i20 version
by GitHub pull_request - opened 15 Dec '25

15 Dec '25

alsa-project/alsa-ucm-conf pull request #660 was opened from dathegreat: Resolves #559 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/660 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/660.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

snd_lib_log_set_handler
by GitHub issues - opened 14 Dec '25

14 Dec '25

alsa-project/alsa-utils issue #314 was opened from sidrat: Function snd_lib_log_filter is still showing up after it was removed from alsa-lib, can some one make a patch to replace snd_lib_log_filter with snd_lib_error_set_handler, thank you alsactl/alsactl.c alsactl.c: In function ‘main’: alsactl.c:465:9: error: implicit declaration of function ‘snd_lib_log_set_handler’; did you mean ‘snd_lib_error_set_handler’? [-Wimplicit-function-declaration] 465 | snd_lib_log_set_handler(log_handler); | ^~~~~~~~~~~~~~~~~~~~~~~ | snd_lib_error_set_handler utils.c: In function ‘log_handler’: utils.c:202:14: error: implicit declaration of function ‘snd_lib_log_filter’ [-Wimplicit-function-declaration] 202 | if (!snd_lib_log_filter(prio, interface, NULL)) | ^~~~~~~~~~~~~~~~~~ make[2]: *** [Makefile:638: alsactl-alsactl.o] Error 1 make[2]: *** Waiting for unfinished jobs.... utils.c:205:17: error: implicit declaration of function ‘snd_lib_log_priority’ [-Wimplicit-function-declaration] 205 | text1 = snd_lib_log_priority(prio); | ^~~~~~~~~~~~~~~~~~~~ utils.c:205:15: error: assignment to ‘const char *’ from ‘int’ makes pointer from integer without a cast [-Wint-conversion] 205 | text1 = snd_lib_log_priority(prio); | ^ utils.c:206:17: error: implicit declaration of function ‘snd_lib_log_interface’ [-Wimplicit-function-declaration] 206 | text2 = snd_lib_log_interface(interface); | ^~~~~~~~~~~~~~~~~~~~~ utils.c:206:15: error: assignment to ‘const char *’ from ‘int’ makes pointer from integer without a cast [-Wint-conversion] 206 | text2 = snd_lib_log_interface(interface); Issue URL : https://github.com/alsa-project/alsa-utils/issues/314 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

amixer decrease volume fails when volume is 100 and increment is 1 or 2
by GitHub issues - opened 14 Dec '25

14 Dec '25

alsa-project/alsa-utils issue #313 was opened from vExcess: If the volume is at 100%, running either of the following fails to decrease the volume ``` amixer -D pulse sset Master 2%- amixer -D pulse sset Master 1%- ``` Running `amixer -D pulse sset Master 2%-` outputs ``` Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 64225 [98%] [on] Front Right: Playback 64225 [98%] [on] ``` It looks like it has decreased the volume by 2%, but it actually hasn't. When I run the command repeatedly it stays at 98% and doesn't go lower. But even though amixer says the volume is at 98%, Linux Mint still reports it as 100%. I'm using alsa-utils 1.2.9-1ubuntu5 ``` OS: Linux Mint 22.2 x86_64 Host: 83DD IdeaPad Slim 5 16AHP9 Kernel: 6.11.0-29-generic ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/313 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

tascam us 322 / 366
by GitHub issues - opened 13 Dec '25

13 Dec '25

alsa-project/alsa-lib issue #489 was opened from riramau: Is it possible to add support for tascam 322 / 366 ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/489 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-mixer: us16x08: validate meter packet indices
by Vulnerability Report 13 Dec '25

13 Dec '25

Hi, while fuzzing a USB gadget that emulates a Tascam US-16x08 we found that get_meter_levels_from_urb() in mixer_us16x08.c uses a channel index taken directly from the 64-byte meter packet to index meter_level[], comp_level[] and master_level[] without any bounds checking. A malformed packet can therefore cause out-of-bounds writes in the snd_us16x08_meter_store. A malicious USB audio device (or USB gadget implementation) that pretends to be a US-16x08-compatible interface can trigger this by sending crafted meter packets. We have a small USB gadget-based PoC for this behaviour and can share it if that would be helpful. This driver is used by common distributions (e.g. Ubuntu) when a US-16x08 or compatible USB audio device is present. The same pattern is present in current mainline kernels. This issue was first reported via security(a)kernel.org. The kernel security team explained that, in the upstream threat model, USB endpoints are expected to be trusted (i.e. only trusted devices should be bound to drivers), so they consider this a normal bug rather than a security vulnerability, and asked us to send a fix to the development lists. The patch below adds simple range checks before updating these arrays. Suggested patch: ---  sound/usb/mixer_us16x08.c | 15 +++++++++++++--  1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index XXXXXXXX..YYYYYYYY 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -647,15 +647,26 @@ static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store)  /* retrieve the meter level values from URB message */  static void get_meter_levels_from_urb(int s,   struct snd_us16x08_meter_store *store,   u8 *meter_urb)  {   int val = MUC2(meter_urb, s) + (MUC3(meter_urb, s) << 8); + int ch = MUB2(meter_urb, s) - 1; + + if (ch < 0) + return;     if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && - MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) { - if (MUC0(meter_urb, s) == 0x72) - store->meter_level[MUB2(meter_urb, s) - 1] = val; - if (MUC0(meter_urb, s) == 0xb2) - store->comp_level[MUB2(meter_urb, s) - 1] = val; - } + MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) { + if (ch < SND_US16X08_MAX_CHANNELS) { + if (MUC0(meter_urb, s) == 0x72) + store->meter_level[ch] = val; + if (MUC0(meter_urb, s) == 0xb2) + store->comp_level[ch] = val; + } + }   if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 && - MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) - store->master_level[MUB2(meter_urb, s) - 1] = val; + MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62) { + if (ch < ARRAY_SIZE(store->master_level)) + store->master_level[ch] = val; + }  } Reported-by: DARKNAVY (@DarkNavyOrg) <vr(a)darknavy.com> Signed-off-by: DARKNAVY (@DarkNavyOrg) <vr(a)darknavy.com>

2 1

[BUG] Asus Expertbook B3402FV (1043:1e03) ALC256 - Headphone Low Vol/Noise & Internal Mic Broken
by shahab ghodsi 12 Dec '25

12 Dec '25

Dear ALSA and Intel HD Audio Developers, I am reporting an audio issue affecting the headphone output and internal microphone on my new Asus Expertbook B3402FV running Fedora 43 (Kernel 6.17.9). *Hardware Summary:* - Laptop Model: Asus Expertbook B3402FV - Subsystem ID: 1043:1e03 - Controller: Intel Raptor Lake-P/U/H cAVS [8086:51ca] - Codec: Realtek ALC256 - Kernel: 6.17.9-300.fc43.x86_64 *Problem Details:* 1. Headphone Output (3.5mm Jack): The output is extremely low volume and heavily distorted with noise, even at 100%. Only the loudest parts of the audio track are audible. This strongly suggests the signal is bypassing the headphone amplifier. 2. Hands-free (3.5mm jack) microphone: WORKS correctly. 3. Internal Microphone (Built-in): The internal microphone does NOT work at all. 3. Internal Speakers: Work correctly. *Troubleshooting Attempted (All failed to resolve the issue):* - Forcing legacy driver: `options snd-intel-dspcfg dsp_driver=1` - Trying known ALC256 model quirks: `model=alc256-asus-mic`, `model=dell-headset-multi`, `model=alc256-asus-headset force_model=1` - Manual pin retasking via `hdajackretask`. - Manual gain adjustments using `hda-verb`. The issue appears to be a missing or incorrect initialization quirk for the specific Subsystem ID 1043:1e03 that is required to correctly map the DAC to the headphone amplifier and enable the internal mic array. The full diagnostic report is available here: http://alsa-project.org/db/?f=3847f92b33024055a4ce7d79016aff1715f3a4f9 Thank you for your time and assistance in looking into this specific hardware configuration. Best regards, Shahab Ghodsi

1 0

[PATCH v9 0/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 12 Dec '25

12 Dec '25

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v9: - Patch 1/2: - modify the comment - change the compatible name to "realtek,rt5575-use-spi" - Patch 2/2: - remove the standalone rt5575_spi_driver module and integrate its functionality into the I2C driver - move the SPI firmware-loading function to rt5575-spi.c - use the match data to distinguish between w/wo flash - minor fixes Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2: - add compatible enum "realtek,rt5575-with-spi" - Link to v8: https://lore.kernel.org/all/20251201105926.1714341-1-oder_chiou@realtek.com/ Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2: - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2: - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (2): ASoC: dt-bindings: realtek,rt5575: add support for ALC5575 ASoC: rt5575: Add the codec driver for the ALC5575 .../bindings/sound/realtek,rt5575.yaml | 44 +++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/rt5575-spi.c | 102 +++++ sound/soc/codecs/rt5575-spi.h | 26 ++ sound/soc/codecs/rt5575.c | 363 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 59 +++ 7 files changed, 607 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

2 5

alsa-info --upload should say HTTPS, not HTTP
by GitHub issues - opened 12 Dec '25

12 Dec '25

alsa-project/alsa-utils issue #312 was opened from jidanni: ``` # alsa-info --upload Your ALSA information is located at http://alsa-project.org/db/?f=a87cd8ba94ebd2f08fde83d50e41782402f117f6 Please inform the person helping you. ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/312 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH REQUEST] Add support for Pioneer DJM-900nexus (missing quirk, NXS2 already supported)
by immoralimmortality 11 Dec '25

11 Dec '25

Hi ALSA maintainers! I'm requesting support for the Pioneer DJM-900nexus USB audio interface (VID:PID 08e4:0158). BACKGROUND: The DJM-900nexus is the predecessor to the DJM-900NXS2, which is already supported in the kernel (quirks-table.h, mixer_quirks.c). CURRENT BEHAVIOR: - Device enumerates successfully (lsusb shows 08e4:0158) - snd-usb-audio loads but probe fails with "cannot find UAC_HEADER" - Kernel logs: "snd-usb-audio 3-4:1.0: probe with driver snd-usb-audio failed with error -22" - Falls back to MIDI-only: "Quirk or no altset; falling back to MIDI 1.0" - No ALSA PCM devices created despite isochronous endpoints being present DEVICE DETAILS: - Works correctly on Windows with Pioneer's driver - USB descriptor shows vendor-specific audio streaming interface (Interface 0, Class 0xFF) - Alternate Setting 1 has isochronous endpoints (EP5 OUT 0x05, EP6 IN 0x86) for audio - MIDI interface (Interface 2) works correctly on Linux TESTING PERFORMED: - Attempted adding to new_id: echo "08e4 0158" > /sys/bus/usb/drivers/snd-usb-audio/new_id Result: Driver attempted probe but still failed with "cannot find UAC_HEADER" - Multiple replug/reload cycles - no change REQUESTED FIX: Add quirk entry for 08e4:0158 (DJM-900nexus) following the pattern of the existing DJM-900NXS2 (2b73:000a) support. Based on the descriptor layout, this device should need similar vendor-specific handling. The fix should involve adding USB_DEVICE_VENDOR_SPEC(0x08e4, 0x0158) entry in sound/usb/quirks-table.h with appropriate audio format definitions for: - Interface 0, Alternate Setting 1 - Playback: EP 0x05 (OUT) - Capture: EP 0x86 (IN) - Format: S24_3LE, 48kHz stereo (2 channels) - Also include MIDI on Interface 2 I've attached a proposed patch based on the DJM-900NXS2 quirk pattern. SUPPORTING DATA ATTACHED: - djm-900nexus-quirk.patch - Proposed kernel patch - djm-dmesg.log - Kernel probe failure logs - djm-asound-cards.txt - Current ALSA state - djm-arecord.txt - arecord -l output ENVIRONMENT: - Kernel: 6.17.10-300.fc43.x86_64 (Fedora 43) - Device firmware: bcdDevice=10.05 Since the newer DJM-900NXS2 already has full support, I believe adding support for the original nexus should be straightforward. The devices are very similar in functionality. I'm happy to test any patches or provide additional information/traces as needed. Thank you for considering this request! Kind regards, f

1 0

[PATCH v1] ALSA: hda/tas2781: Add new quirk for HP new project
by Baojun Xu 11 Dec '25

11 Dec '25

Add new vendor_id and subsystem_id in quirk for HP new project (NexusX). Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 9510aa2f1bda..171a71457ec3 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6770,6 +6770,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8e60, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e61, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e62, "HP Trekker ", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8e8a, "HP NexusX", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8e9d, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8e9e, "HP 17 Turbine OmniBook X UMA", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8eb6, "HP Abe A6U", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_GPIO), -- 2.25.1

1 0

[GIT PULL] ASoC fixes for v6.19-merge-window
by Mark Brown 11 Dec '25

11 Dec '25

The following changes since commit c5fae31f60a91dbe884ef2789fb3440bb4cddf05: ASoC: fsl_micfil: Set default quality and channel (2025-11-29 00:59:00 +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-merge-window for you to fetch changes up to f34836a8ddf9216ff919927cddb705022bf30aab: ASoC: amd: acp: update tdm channels for specific DAI (2025-12-10 09:17:53 +0900) ---------------------------------------------------------------- ASoC: Fixes for v6.19 A small pile of fixes that came in during the merge window, it's all fairly standard device specific stuff. ---------------------------------------------------------------- Eric Biggers (1): ASoC: cros_ec_codec: Remove unnecessary selection of CRYPTO Eric Naim (1): ASoC: cs35l41: Always return 0 when a subsystem ID is found Haotian Zhang (1): ASoC: bcm: bcm63xx-pcm-whistler: Check return value of of_dma_configure() Hemalatha Pinnamreddy (2): ASoC: amd: acp: Audio is not resuming after s0ix ASoC: amd: acp: update tdm channels for specific DAI Johan Hovold (4): ASoC: codecs: wcd937x: fix OF node leaks on probe failure ASoC: codecs: wcd938x: fix OF node leaks on probe failure ASoC: codecs: wcd939x: fix OF node leaks on probe failure ASoc: qcom: q6afe: fix bad guard conversion Krzysztof Kozlowski (2): ASoC: codecs: nau8325: Silence uninitialized variables warnings ASoC: rockchip: Fix Wvoid-pointer-to-enum-cast warning (again) Mark Brown (2): ASoC: ak4458 & ak5558: disable regulator if error ASoC: codecs: wcd93xx: fix OF node leaks on probe Richard Fitzgerald (2): ASoC: cs-amp-lib: Revert use of __free(kfree) back to normal C cleanup ASoC: cs35l56: Fix incorrect select SND_SOC_CS35L56_CAL_SYSFS_COMMON Shengjiu Wang (3): ASoC: dt-bindings: cirrus,cs42xx8: Reference common DAI properties ASoC: ak4458: Disable regulator when error happens ASoC: ak5558: Disable regulator when error happens .../devicetree/bindings/sound/cirrus,cs42xx8.yaml | 5 +++- sound/soc/amd/acp/acp-i2s.c | 2 ++ sound/soc/amd/acp/acp-legacy-common.c | 30 +++++++++++++++++----- sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 ++- sound/soc/codecs/Kconfig | 3 +-- sound/soc/codecs/ak4458.c | 10 +++++++- sound/soc/codecs/ak5558.c | 10 +++++++- sound/soc/codecs/cs-amp-lib.c | 29 ++++++++++++--------- sound/soc/codecs/cs35l41.c | 7 ++--- sound/soc/codecs/nau8325.c | 4 +-- sound/soc/codecs/wcd937x.c | 4 +-- sound/soc/codecs/wcd938x.c | 3 +-- sound/soc/codecs/wcd939x.c | 3 +-- sound/soc/qcom/qdsp6/q6afe.c | 4 +-- sound/soc/rockchip/rockchip_pdm.c | 2 +- 15 files changed, 82 insertions(+), 38 deletions(-)

2 1

USB-Audio: Antelope Audio Zen Go: add UCM2 configuration files
by GitHub pull_request - opened 10 Dec '25

10 Dec '25

alsa-project/alsa-ucm-conf pull request #659 was opened from ryodeushii: Add support for Antelope Audio Zen Go SC [alsa-info.txt](https://github.com/user-attachments/files/24074814/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/659 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/659.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v8 0/2] ASoC: rt5575: Add the codec driver for the ALC5575
by Oder Chiou 10 Dec '25

10 Dec '25

Hi all, This patch series adds support for the Realtek ALC5575 audio codec. Changes in v8: - Patch 1/2: - remove the variable rt5575_spi_ready - use the multiple compatible names to distinguish between w/wo flash - Patch 2/2 - add compatible enum "realtek,rt5575-with-spi" Changes in v7: - Patch 1/2: - add a caption for the tristates - remove the redundant enum of the SPI command - add the error log in the request firmware failure - change the function name rt5575_spi_fw_loaded to rt5575_fw_load_by_spi - minor fixes - Patch 2/2 - modify commit message - Link to v7: https://lore.kernel.org/all/20251121084112.743518-1-oder_chiou@realtek.com/ Changes in v6: - Patch 1/2: - modify commit message - add select SND_SOC_RT5575 to config SND_SOC_RT5575_SPI in the Kconfig - revise the boiler plate in the head of the file - sort the include files - use a structure to transfer the spi data - use the poll() related function instead the for-loop - revise the UUID to the private ID - minor fixes - Patch 2/2 - modify description - Link to v6: https://lore.kernel.org/all/20251031073245.3629060-1-oder_chiou@realtek.com/ Changes in v2 to v5: - Patch 1/2: - move the firmware to the subdirectory - remove the empty functions - remove the cache_type in the regmap_config - add the error log in the run firmware failure - Patch 2/2: - nothing - Link to v5: https://lore.kernel.org/all/20251015103404.3075684-1-oder_chiou@realtek.com/ Oder Chiou (2): ASoC: rt5575: Add the codec driver for the ALC5575 ASoC: dt-bindings: realtek,rt5575: add bindings for ALC5575 .../bindings/sound/realtek,rt5575.yaml | 44 ++ sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5575-spi.c | 84 ++++ sound/soc/codecs/rt5575-spi.h | 16 + sound/soc/codecs/rt5575.c | 375 ++++++++++++++++++ sound/soc/codecs/rt5575.h | 54 +++ 7 files changed, 587 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5575.yaml create mode 100644 sound/soc/codecs/rt5575-spi.c create mode 100644 sound/soc/codecs/rt5575-spi.h create mode 100644 sound/soc/codecs/rt5575.c create mode 100644 sound/soc/codecs/rt5575.h -- 2.52.0

3 18

[PATCH] ASoC: soc-compress: Fix race condition in soc_compr_open_fe
by 이승빈 10 Dec '25

10 Dec '25

Hello Maintainers, Takashi and Jaroslav, I am writing to respectfully follow up on a patch I submitted to the alsa-devel mailing list and Patchwork I would like to bring up an important issue regarding the interaction between soc_compr_trigger_fe and snd_soc_dpcm_runtime_update. If the soc_compr_trigger_fe function is called during the execution of snd_soc_dpcm_runtime_update, it may lead to unintended modification of the FE's DPCM runtime update variable. As a result, the BE list associated with the FE could become detached, preventing proper closure of the BE. Ex)When repeatedly switching between deep playback and offload playback during MP3 playback To prevent this race condition, access to this critical section should be protected using snd_soc_dpcm_mutex_lock. Ensuring mutual exclusion during these operations will maintain the integrity of the BE list and avoid potential resource leaks. I understand that you have a large backlog of patches to review. If you have had a chance to look at it and found any issues, or if there's anything I need to fix or clarify, please let me know. I would be happy to prepare a v2 version if necessary. Thank you very much for your time and attention to this matter. Best regards, Change-Id: Ibfd1712e6baf41a30afa357a91ada83d205f968b Signed-off-by: Seungbin Lee <seungbin.lee(a)samsung.com> --- sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index fc8db54352541..1a2a3fe6a2340 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -286,6 +286,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) snd_soc_card_mutex_lock(fe->card); + snd_soc_dpcm_mutex_lock(fe); + ret = snd_soc_dai_compr_trigger(cpu_dai, cstream, cmd); if (ret < 0) goto out; @@ -315,6 +317,8 @@ static int soc_compr_trigger_fe(struct snd_compr_stream *cstream, int cmd) out: fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; + snd_soc_dpcm_mutex_unlock(fe); + snd_soc_card_mutex_unlock(fe->card); return ret; } -- 2.34.1

1 0

[PATCH v1] ASoC: amd: acp: update tdm channels for specific DAI
by Raghavendra Prasad Mallela 10 Dec '25

10 Dec '25

From: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> TDM channel updates were applied to all DAIs, causing configurations to overwrite for unrelated streams. The logic is modified to update channels only for targeted DAI. This prevents corruption of other DAI settings and resolves audio issues observed during system suspend and resume cycles. Fixes: 12229b7e50cf ("ASoC: amd: acp: Add TDM support for acp i2s stream") Signed-off-by: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> Signed-off-by: Raghavendra Prasad Mallela <raghavendraprasad.mallela(a)amd.com> --- sound/soc/amd/acp/acp-i2s.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 4ba0a66981ea..283a674c7e2c 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -157,6 +157,8 @@ static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mas spin_lock_irq(&chip->acp_lock); list_for_each_entry(stream, &chip->stream_list, list) { + if (dai->id != stream->dai_id) + continue; switch (chip->acp_rev) { case ACP_RN_PCI_ID: case ACP_RMB_PCI_ID: -- 2.17.1

2 1

alsa-lib-1.2.15: no sound
by GitHub issues - opened 09 Dec '25

09 Dec '25

alsa-project/alsa-lib issue #488 was opened from andreygrozin: After upgrading alsa-lib to 1.2.15 I get grozin@localhost /home/Music $ qmmp <song>.mp3 ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.15/work/alsa-lib-1.2.15/src/pcm/pcm.c:2722:(snd_pcm_open_noupdate) [error.pcm] Unknown PCM cards.pcm.default OutputALSA::initialize: error opening PCM device default OutputWriter::initialize: unable to initialize output and no sound. Similarly, grozin@localhost /home/Music $ mpv <song>.mp3 [ffmpeg/demuxer] mp3: Estimating duration from bitrate, this may be inaccurate ● Audio --aid=1 (mp3 2ch 44100 Hz 192 kbps) File tags: Genre: Other ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.15/work/alsa-lib-1.2.15/src/pcm/pcm.c:2722:(snd_pcm_open_noupdate) [error.pcm] Unknown PCM cards.pcm.default [ao/alsa] Playback open error: No such file or directory [ao] Failed to initialize audio driver 'alsa' Could not open/initialize audio device -> no sound. Audio: no audio Exiting... (Errors when loading file) After downgrading alsa-lib to 1.2.14 everything works fine again. This is on Gentoo Linux, see the Gentoo bug https://bugs.gentoo.org/967264 Issue URL : https://github.com/alsa-project/alsa-lib/issues/488 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm: clean up data types
by GitHub pull_request - opened 09 Dec '25

09 Dec '25

alsa-project/alsa-lib pull request #487 was opened from floppym: int should be used for file descriptors. sysconf returns long. waitpid returns pid_t. Request URL : https://github.com/alsa-project/alsa-lib/pull/487 Patch URL : https://github.com/alsa-project/alsa-lib/pull/487.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sof-soundwire: third fix for multi-codec
by GitHub pull_request - opened 09 Dec '25

09 Dec '25

alsa-project/alsa-ucm-conf pull request #658 was opened from mstrozek: Ensure MultiCodec-based actions are only done when MultiCodec1 is not empty, otherwise Speaker/Mic/HeadsetCodec1 vars can be cleared without reason and prevent other devices from being recognized. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/658 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/658.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Enable Mute LED for HP platform
by Kailang 09 Dec '25

09 Dec '25

Hi Takashi, Attach patch was enable mute led functions. Many Thanks. Kailang

3 2

ucm2: codecs: rt722: add condition to SetLED for mic
by GitHub pull_request - opened 09 Dec '25

09 Dec '25

alsa-project/alsa-ucm-conf pull request #657 was opened from shumingfan: Some SKUs don't have the internal mic, so the patch adds the condition to check whether the SKU has the internal mic or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/657 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/657.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

BUG: snd_usb_audio: iFi Zen DAC 3 hangs on query, defaults to S16_LE (Fedora kernel 6.17.9-300.fc43.x86_64)
by Dustin Fenster 09 Dec '25

09 Dec '25

1 0

Build Failure Due to Undefined Reference to close_range with alsa-lib 1.2.15
by GitHub issues - opened 08 Dec '25

08 Dec '25

alsa-project/alsa-lib issue #485 was opened from darkshram: ### Description I encountered a build failure while compiling **alsa-lib 1.2.15** due to an undefined reference to `close_range`. My environment uses **glibc 2.30**, which should support this function. ### Error Message ``` ucm/.libs/libucm.a(ucm_exec.o): In function `uc_mgr_exec': /builddir/build/BUILD/alsa-lib-1.2.15/src/ucm/ucm_exec.c:263: undefined reference to `close_range' collect2: error: ld returned 1 exit status ``` ### Solution To resolve the issue, I applied the following patch, which effectively reverses [the change that introduced](https://github.com/alsa-project/alsa-lib/pull/459) `close_range`: ``` diff -Naur alsa-lib-1.2.15.orig/src/ucm/ucm_exec.c alsa-lib-1.2.15/src/ucm/ucm_exec.c --- alsa-lib-1.2.15.orig/src/ucm/ucm_exec.c 2025-12-07 13:41:56.000000000 -0600 +++ alsa-lib-1.2.15/src/ucm/ucm_exec.c 2025-12-08 15:03:30.632324127 -0600 @@ -259,12 +259,8 @@ close(f); -#if defined(_GNU_SOURCE) - close_range(3, maxfd, 0); -#else for (f = 3; f < maxfd; f++) close(f); -#endif /* install default handlers for the forked process */ signal(SIGINT, SIG_DFL); ``` This workaround restored compatibility and allowed the build to complete successfully. ### Additional Information Please consider this issue for further review. If close_range is required, clearer compatibility notes might be beneficial. Thank you! Issue URL : https://github.com/alsa-project/alsa-lib/issues/485 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: USB-Audio: Add Rodecaster Pro II
by GitHub pull_request - opened 08 Dec '25

08 Dec '25

alsa-project/alsa-ucm-conf pull request #656 was opened from parzival-space: This PR adds support for the RODECaster Pro II Split PCM configuration. The Rodecaster reports as two audio devices, device 0 is always the mic/chat track and device 1 can be configure by the user to either be a multitrack device or a simple stereo input/output device. Using the USB PID its possible to detect the current channel configuration (similar to the GoXLR). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/656 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/656.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: uapi: Fix typo in asound.h comment
by Andres J Rosa 08 Dec '25

08 Dec '25

Fix 'level-shit' to 'level-shift' in struct snd_cea_861_aud_if comment. Signed-off-by: Andres J Rosa <andyrosa(a)gmail.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 5a049eeae..d3ce75ba9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -60,7 +60,7 @@ struct snd_cea_861_aud_if { unsigned char db2_sf_ss; /* sample frequency and size */ unsigned char db3; /* not used, all zeros */ unsigned char db4_ca; /* channel allocation code */ - unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */ + unsigned char db5_dminh_lsv; /* downmix inhibit & level-shift values */ }; /**************************************************************************** -- 2.52.0.windows.1

2 1

ucm2: sof-soundwire: Simplify cs42l45 configs
by GitHub pull_request - opened 08 Dec '25

08 Dec '25

alsa-project/alsa-ucm-conf pull request #655 was opened from mstrozek: Continued from https://github.com/alsa-project/alsa-ucm-conf/pull/644 Simplify cs42l45 configs following machine driver changes [1] which bring normal switches instead of mute switches. [1] https://lore.kernel.org/all/20251127163426.2500633-2-ckeepax@opensource.cir… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/655 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/655.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsactl: fix build when in subdirectory
by GitHub pull_request - opened 08 Dec '25

08 Dec '25

alsa-project/alsa-utils pull request #311 was opened from heitbaum: create the conf directory as part of the edit recipe fixes: https://github.com/alsa-project/alsa-utils/commit/f047a8a66917960ef5a9ff0e4… Request URL : https://github.com/alsa-project/alsa-utils/pull/311 Patch URL : https://github.com/alsa-project/alsa-utils/pull/311.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[for alsa-lib 1.2.15] USB-Audio: add support for conf.d configurations
by GitHub pull_request - edited 07 Dec '25

07 Dec '25

alsa-project/alsa-ucm-conf pull request #654 was edited from perexg: It may be useful to add new or override specific hardware configurations until they are merged to the main USB-Audio.conf file. BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/609 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/654 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/654.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[for alsa-lib 1.2.15] USB-Audio: Steinberg UR22C - fix capture channels for older firmware
by GitHub pull_request - edited 07 Dec '25

07 Dec '25

alsa-project/alsa-ucm-conf pull request #554 was edited from perexg: It seems that firmware version 3.20 added 4 capture channels while previous firmware has only 2 capture channels. Fixes: https://github.com/alsa-project/alsa-ucm-conf/issues/547 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/554.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[for alsa-lib-1.2.15]: USB-Audio: GoXLR: enable detection of beta firmware (25 channels)
by GitHub pull_request - edited 07 Dec '25

07 Dec '25

alsa-project/alsa-ucm-conf pull request #524 was edited from perexg: 'Syntax 8' is required (alsa-lib 1.2.14+) Fixes: 17f9b4f ("USB-Audio: GoXLR - fix the channel detection for mini, cleanups") Link: https://github.com/alsa-project/alsa-ucm-conf/issues/444 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/524.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 05 Dec '25

05 Dec '25

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

4 8

FreeCine APK TV
by FatmanVincent＠hotmail.com 05 Dec '25

05 Dec '25

FreeCine APK tornou se uma das aplicações de streaming click : https://freecine.com.br/ mais procuradas graças ao seu conjunto de funcionalidades que oferecem aos utilizadores uma forma simples e rápida de aceder aos seus filmes e séries favoritos. O grande destaque da aplicação está na forma como combina praticidade qualidade de imagem e uma navegação fluida criando uma experiência de entretenimento completa para qualquer dispositivo Android

1 0

DRAFT [for alsa-lib 1.2.15] USB-Audio: add support for conf.d configurations
by GitHub pull_request - opened 04 Dec '25

04 Dec '25

alsa-project/alsa-ucm-conf pull request #654 was opened from perexg: It may be useful to add new or override specific hardware configurations until they are merged to the main USB-Audio.conf file. BugLink: https://github.com/alsa-project/alsa-ucm-conf/issues/609 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/654 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/654.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Update cs42l45 configurations
by GitHub pull_request - edited 04 Dec '25

04 Dec '25

alsa-project/alsa-ucm-conf pull request #644 was edited from mstrozek: Update JackControls to follow machine driver changes [1] [1] https://github.com/thesofproject/linux/pull/5590 Keeping this PR as a draft until the SOF PR is finalized and merged Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 04 Dec '25

04 Dec '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

3 2

[PATCH] ASoC: SOF: Intel: Retry codec probing if it fails
by Kai Vehmanen 04 Dec '25

04 Dec '25

From: Hui Wang <hui.wang(a)canonical.com> On the latest Lenovo Thinkstation laptops, we often experience the speaker failure after rebooting, check the dmesg, we could see: sof-audio-pci-intel-tgl 0000:00:1f.3: codec #0 probe error, ret: -5 The analogue codec on the machine is ALC287, then we designed a testcase to reboot and check the codec probing result repeatedly, we found the analogue codec probing always failed at least once within several minutes to several hours (roughly 1 reboot per min). This issue happens on all laptops of this Thinkstation model, but with legacy HDA driver, we couldn't reproduce this issue on those laptops. And so far, this issue is not reproduced on machines which don't belong to this model. We tried to make the hda_dsp_ctrl_init_chip() same as hda_intel_init_chip() which is the controller init routine in the legacy HDA driver, but it didn't help. We found when issue happens, the resp is -1, and if we let driver re-run send_cmd() and get_response(), it will get the correct response 10ec0287, then driver continues the rest work, finally boot to the desktop and all audio function work well. Here adding codec probing retries to 3 times, it could fix the issue on this Thinkstation model, and it doesn't bring impact to other machines. Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/intel/hda-codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) Notes: - patch for-5.16 on top of for-5.16 8a724d5f60 diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index 6744318de612..13cd96e6724a 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -22,6 +22,7 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 +#define CODEC_PROBE_RETRIES 3 /* load the legacy HDA codec driver */ static int request_codec_module(struct hda_codec *codec) @@ -121,12 +122,15 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; u32 resp = -1; - int ret; + int ret, retry = 0; + + do { + mutex_lock(&hbus->core.cmd_mutex); + snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); + snd_hdac_bus_get_response(&hbus->core, address, &resp); + mutex_unlock(&hbus->core.cmd_mutex); + } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); - mutex_lock(&hbus->core.cmd_mutex); - snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); - snd_hdac_bus_get_response(&hbus->core, address, &resp); - mutex_unlock(&hbus->core.cmd_mutex); if (resp == -1) return -EIO; dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", base-commit: 8a724d5f60904084e09de2685ff9e77bc8a40815 -- 2.34.1

3 2

Asus ROG Crosshair X870E hero (ACL4082) - No audio from rear microphone
by GitHub issues - opened 03 Dec '25

03 Dec '25

alsa-project/alsa-ucm-conf issue #652 was opened from fixed77: The Rear Microphone is detected as Line In when I plug in my mic. There is no sound from the microphone. I don't use the Front Audio connection as my case doesn't have I/O for it, I just used the 2 audio ports in the rear. Motherboard: Asus ROG Crosshair X870E hero Audio Device: ALC4082 Distro: Ubuntu 24.04 Window Manager: Gnome-Shell Packages: ``` alsa-base/noble,noble,now 1.0.25+dfsg-0ubuntu7 alsa-topology-conf/noble,noble,now 1.2.5.1-2 alsa-ucm-conf/noble-updates,noble-updates,now 1.2.10-1ubuntu5.7 alsa-utils/noble,now 1.2.9-1ubuntu5 gstreamer1.0-alsa/noble-updates,noble-security,now 1.24.2-1ubuntu0.3 gstreamer1.0-pipewire/noble-updates,now 1.0.5-1ubuntu3.1 gstreamer1.0-pulseaudio/noble-updates,noble-security,now 1.24.2-1ubuntu1.2 libpipewire-0.3-0t64/noble-updates,now 1.0.5-1ubuntu3.1 libpipewire-0.3-common/noble-updates,noble-updates,now 1.0.5-1ubuntu3.1 libpipewire-0.3-modules/noble-updates,now 1.0.5-1ubuntu3.1 pipewire-alsa/noble-updates,now 1.0.5-1ubuntu3.1 pipewire-audio/noble-updates,noble-updates,now 1.0.5-1ubuntu3.1 pipewire-bin/noble-updates,now 1.0.5-1ubuntu3.1 pipewire-pulse/noble-updates,now 1.0.5-1ubuntu3.1 pipewire/noble-updates,now 1.0.5-1ubuntu3.1 pulseaudio-utils/noble-updates,now 1:16.1+dfsg1-2ubuntu10.1 ``` `lsusb | grep -i audio` ``` Bus 001 Device 009: ID 0b05:1b7c ASUSTek Computer, Inc. USB Audio ``` `alsactl monitor` (unplugging and plugging 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 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 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 7 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=30,iface=CARD,name='IEC958 In - Output Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on 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=on 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=255,step=0 : values=215,215 | dBminmax-min=-127.50dB,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=39,39 | dBminmax-min=-17.25dB,max=12.00dB numid=9,iface=MIXER,name='Analog In Capture Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=10,iface=MIXER,name='Analog In 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=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=3,4 | 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=3,4 | 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 ``` `arecord -l` ``` **** List of CAPTURE Hardware Devices **** card 2: Audio [USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 1: USB Audio [USB Audio #1] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: Audio [USB Audio], device 2: USB Audio [USB Audio #2] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `alsaucm -c hw:2 dump text` ``` Verb.HiFi { Comment "HiFi 2.0 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 2 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.SPDIF { Comment "S/PDIF Output" Values { CaptureCTL "_ucm0001.hw:Audio" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,2 PlaybackPCM "_ucm0001.hw:Audio,3" PlaybackPriority 100 TQ HiFi } } Device.Line1 { Comment "Line Input" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Line,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 100 JackControl "Line - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } Device.Mic1 { Comment Microphone Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,2" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 5+1" { Comment "HiFi 5.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 6 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.SPDIF { Comment "S/PDIF Output" Values { CaptureCTL "_ucm0001.hw:Audio" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,2 PlaybackPCM "_ucm0001.hw:Audio,3" PlaybackPriority 100 TQ HiFi } } Device.Line1 { Comment "Line Input" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Line,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 100 JackControl "Line - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } Device.Mic1 { Comment Microphone Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,2" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } Verb."HiFi 7+1" { Comment "HiFi 7.1 channels" Device.Speaker { Comment Speakers Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Speaker - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackChannels 8 PlaybackMixerElem PCM,0 PlaybackPCM "_ucm0001.hw:Audio" PlaybackPriority 200 TQ HiFi } } Device.Headphones { Comment "Front Headphones" Values { CaptureCTL "_ucm0001.hw:Audio" JackControl "Headphone - Output Jack" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,1 PlaybackPCM "_ucm0001.hw:Audio,1" PlaybackPriority 300 TQ HiFi } } Device.SPDIF { Comment "S/PDIF Output" Values { CaptureCTL "_ucm0001.hw:Audio" PlaybackCTL "_ucm0001.hw:Audio" PlaybackMixerElem PCM,2 PlaybackPCM "_ucm0001.hw:Audio,3" PlaybackPriority 100 TQ HiFi } } Device.Line1 { Comment "Line Input" Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Line,0 CapturePCM "_ucm0001.hw:Audio,1" CapturePriority 100 JackControl "Line - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } Device.Mic1 { Comment Microphone Values { CaptureCTL "_ucm0001.hw:Audio" CaptureMixerElem Mic,0 CapturePCM "_ucm0001.hw:Audio,2" CapturePriority 300 JackControl "Mic - Input Jack" PlaybackCTL "_ucm0001.hw:Audio" TQ HiFi } } } ``` [alsa-info.txt](https://github.com/user-attachments/files/23915492/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/652 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: amd: acp: Audio is not resuming after s0ix
by Raghavendra Prasad Mallela 03 Dec '25

03 Dec '25

From: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> Audio fails to resume after system exits suspend mode due to accessing incorrect ring buffer address during resume. This patch resolves issue by selecting correct address based on the ACP version. Fixes: f6f7d25b11033 ("ASoC: amd: acp: Add pte configuration for ACP7.0 platform") Signed-off-by: Hemalatha Pinnamreddy <hemalatha.pinnamreddy2(a)amd.com> Signed-off-by: Raghavendra Prasad Mallela <raghavendraprasad.mallela(a)amd.com> --- Changes since v1: - Include Fixes commit - drop unnecessary variable initialization sound/soc/amd/acp/acp-legacy-common.c | 30 +++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-common.c b/sound/soc/amd/acp/acp-legacy-common.c index 3078f459e005..4e477c48d4bd 100644 --- a/sound/soc/amd/acp/acp-legacy-common.c +++ b/sound/soc/amd/acp/acp-legacy-common.c @@ -219,7 +219,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, SP_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_TX_FIFOADDR(chip); reg_fifo_size = ACP_I2S_TX_FIFOSIZE(chip); - phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_SP_TX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_I2S_TX_RINGBUFADDR(chip)); } else { reg_dma_size = ACP_I2S_RX_DMA_SIZE(chip); @@ -227,7 +230,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, SP_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_I2S_RX_FIFOADDR(chip); reg_fifo_size = ACP_I2S_RX_FIFOSIZE(chip); - phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_SP_RX_MEM_WINDOW_START; + else + phy_addr = I2S_SP_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_I2S_RX_RINGBUFADDR(chip)); } break; @@ -238,7 +244,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, BT_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_TX_FIFOADDR(chip); reg_fifo_size = ACP_BT_TX_FIFOSIZE(chip); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_BT_TX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_BT_TX_RINGBUFADDR(chip)); } else { reg_dma_size = ACP_BT_RX_DMA_SIZE(chip); @@ -246,7 +255,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, BT_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_BT_RX_FIFOADDR(chip); reg_fifo_size = ACP_BT_RX_FIFOSIZE(chip); - phy_addr = I2S_BT_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_BT_RX_MEM_WINDOW_START; + else + phy_addr = I2S_BT_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_BT_RX_RINGBUFADDR(chip)); } break; @@ -257,7 +269,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, HS_PB_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_HS_TX_FIFOADDR; reg_fifo_size = ACP_HS_TX_FIFOSIZE; - phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_HS_TX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_TX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_HS_TX_RINGBUFADDR); } else { reg_dma_size = ACP_HS_RX_DMA_SIZE; @@ -265,7 +280,10 @@ static int set_acp_i2s_dma_fifo(struct snd_pcm_substream *substream, HS_CAPT_FIFO_ADDR_OFFSET; reg_fifo_addr = ACP_HS_RX_FIFOADDR; reg_fifo_size = ACP_HS_RX_FIFOSIZE; - phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; + if (chip->acp_rev >= ACP70_PCI_ID) + phy_addr = ACP7x_I2S_HS_RX_MEM_WINDOW_START; + else + phy_addr = I2S_HS_RX_MEM_WINDOW_START + stream->reg_offset; writel(phy_addr, chip->base + ACP_HS_RX_RINGBUFADDR); } break; -- 2.17.1

1 0

[PATCH] ALSA: uapi: Fix typo in [asound.h](http://_vscodecontentref_/0) comment
by Andres Rosa 03 Dec '25

03 Dec '25

Fix 'level-shit' to 'level-shift' in struct snd_cea_861_aud_if comment. Signed-off-by: Andy Rosa <andyrosa(a)gmail.com> --- include/uapi/sound/asound.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 5a049eeae..d3ce75ba9 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -60,7 +60,7 @@ struct snd_cea_861_aud_if { unsigned char db2_sf_ss; /* sample frequency and size */ unsigned char db3; /* not used, all zeros */ unsigned char db4_ca; /* channel allocation code */ - unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */ + unsigned char db5_dminh_lsv; /* downmix inhibit & level-shift values */ }; /**************************************************************************** -- 2.52.0.windows.1

1 0

Add Audient iD14 MK2 support
by GitHub pull_request - opened 03 Dec '25

03 Dec '25

alsa-project/alsa-ucm-conf pull request #651 was opened from VolanDeVovan: Add UCM2 configuration for the Audient iD14 MK2 USB audio interface. Device: Audient iD14 MK2 USB ID: 2708:0008 This is a minimal configuration exposing only the main analog I/O: - Microphone/line inputs (2 mono channels) - Headphone and monitor outputs Full 12-channel capture (ADAT, S/PDIF, loopback) is not exposed. Playback devices: - Headphones (priority 200) - Line1/Monitor 1-2 (priority 100) Capture devices: - Mic1/Input 1 (priority 200) - Mic2/Input 2 (priority 100) Tested on NixOS with PipeWire. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/651 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/651.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC updates for v6.19
by Mark Brown 02 Dec '25

02 Dec '25

The following changes since commit ac3fd01e4c1efce8f2c054cdeb2ddd2fc0fb150d: Linux 6.18-rc7 (2025-11-23 14:53:16 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.19 for you to fetch changes up to c5fae31f60a91dbe884ef2789fb3440bb4cddf05: ASoC: fsl_micfil: Set default quality and channel (2025-11-29 00:59:00 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.19 This is a very large set of updates, as well as some more extensive cleanup work from Morimto-san we've also added a generic SCDA class driver for SoundWire devices enabling us to support many chips with no custom code. There's also a batch of new drivers added for both SoCs and CODECs. - Added a SoundWire SCDA generic class driver, pulling in a little regmap work to support it. - A *lot* of cleaup and API improvement work from Morimoto-san. - Lots of work on the existing Cirrus, Intel, Maxim and Qualcomm drivers. - Support for Allwinner A523, Mediatek MT8189, Qualcomm QCM2290, QRB2210 and SM6115, SpacemiT K1, and TI TAS2568, TAS5802, TAS5806, TAS5815, TAS5828 and TAS5830. This also pulls in some gpiolib changes supporting shared GPIOs in the core there so we can convert some of the ASoC drivers open coding handling of that to the core functionality. ---------------------------------------------------------------- Alex Tran (1): ASoC: codecs: rt5670: use SOC_VALUE_ENUM_SINGLE_DECL for DAC2 L/R MX-1B Alexander Stein (1): ASoC: fsl_xcvr: use dev_err_probe() replacing dev_err() + return Alexey Klimov (2): ASoC: dt-bindings: qcom,sm8250: add QRB2210 soundcard ASoC: qcom: sm8250: add qrb2210-sndcard compatible string Balamurugan C (1): ASoC: Intel: sof_rt5682: Add quirk override support Baojun Xu (6): ASoC: tas2781: Add TAS5802, TAS5815, and TAS5828 ASoC: dt-bindings: ti,tas2781: Add TAS5802, TAS5815, and TAS5828 ASoC: tas2781: Add tas5822 support ASoC: dt-bindings: ti,tas2781: Add TAS5822 support ASoC: dt-bindings: ti,tas2781: Add TAS2568/2574/5806M/5806MD/5830 support ASoC: tas2781: Add tas2568/2574/5806m/5806md/5830 support Bard Liao (22): ASoC: SOF: Don't print the monolithic topology name if function topology may be used ASoC: soc-acpi: make some variables of acpi adr and link adr non-const ASoC: soc_sdw_utils: add name_prefix to asoc_sdw_codec_info struct ASoC: Intel: export sof_sdw_get_tplg_files ASoC: soc_sdw_utils: export asoc_sdw_get_dai_type ASoC: SOF: add platform name into sof_intel_dsp_desc ASoC: SOF: don't check the existence of dummy topology ASoC: SOF: Intel: use sof_sdw as default SDW machine driver ASoC: sof-function-topology-lib: escalate the log when missing function topoplogy ASoC: SOF: Don't print the monolithic topology name if function topology may be used ASoC: soc-acpi: make some variables of acpi adr and link adr non-const ASoC: soc_sdw_utils: add name_prefix to asoc_sdw_codec_info struct ASoC: Intel: export sof_sdw_get_tplg_files ASoC: soc_sdw_utils: export asoc_sdw_get_dai_type ASoC: SOF: add platform name into sof_intel_dsp_desc ASoC: SOF: don't check the existence of dummy topology ASoC: SOF: Intel: use sof_sdw as default SDW machine driver ASoC: sof-function-topology-lib: escalate the log when missing function topoplogy ASoC: soc_sdw_utils: add cs35l57 support ASoC: SOF: Intel: select SND_SOC_SDW_UTILS in SND_SOC_SOF_HDA_GENERIC ASoC: SOF: Intel: remove hyphen from AMP<index> name_prexix ASoC: Intel: sof_sdw: create BT dai link if bt_link_mask is set Bartosz Golaszewski (8): string: provide strends() gpiolib: define GPIOD_FLAG_SHARED gpiolib: implement low-level, shared GPIO support gpio: shared-proxy: implement the shared GPIO proxy driver gpiolib: support shared GPIOs in core subsystem code gpio: provide gpiod_is_shared() ASoC: wsa881x: drop GPIOD_FLAGS_BIT_NONEXCLUSIVE flag from GPIO lookup ASoC: wsa883x: drop GPIOD_FLAGS_BIT_NONEXCLUSIVE flag from GPIO lookup Cezary Rojewski (8): ASoC: Intel: avs: Allow the topology to carry NHLT data ASoC: Intel: avs: Honor NHLT override when setting up a path ASoC: Intel: catpt: Fix offset checks ASoC: Intel: catpt: Switch to resource_xxx() API ASoC: Intel: catpt: Fix error path in hw_params() ASoC: Intel: catpt: Fix probing order of driver components ASoC: Intel: catpt: Do not ignore errors on runtime resume ASoC: Intel: catpt: Do not block the system from suspending Chancel Liu (2): ASoC: fsl_micfil: Add default quality for different platforms ASoC: fsl_micfil: Set channel range control Charles Keepax (38): ASoC: SDCA: Rename SoundWire struct device variables regmap: sdw-mbq: Don't assume the regmap device is the SoundWire slave ASoC: SDCA: Add manual PM runtime gets to IRQ handlers ASoC: SDCA: Pass SoundWire slave to HID ASoC: SDCA: Pass device register map from IRQ alloc to handlers ASoC: SDCA: Update externally_requested flag to cover all requests ASoC: SDCA: Factor out a helper to find SDCA IRQ data ASoC: SDCA: Rely less on the ASoC component in IRQ handling ASoC: SDCA: Force some SDCA Controls to be volatile ASoC: SDCA: Parse XU Entity properties ASoC: SDCA: Parse Function Reset max delay ASoC: SDCA: Add UMP buffer helper functions ASoC: SDCA: Add completion for FDL start and stop ASoC: SDCA: Add UMP timeout handling for FDL ASoC: SDCA: Add early IRQ handling ASoC: SDCA: Add HID button IRQ ASoC: SDCA: Add companion amp Function ASoC: SDCA: Add stubs for FDL helper functions ASoC: SDCA: Remove duplicated module macros ASoC: SDCA: Fix missing dash in HIDE DisCo property ASoC: SDCA: Add missing forward declaration in header ASoC: SDCA: Correct FDL locking in sdca_fdl_process() ASoC: SDCA: Add comment for function reset polling ASoC: SDCA: Move most of the messages from info to debug ASoC: SDCA: Use helper macros for control identification ASoC: SDCA: Factor out helper to process Control defaults ASoC: SDCA: Populate regmap cache for readable Controls ASoC: SDCA: Add helper to write initialization writes ASoC: SDCA: Add basic SDCA class driver ASoC: SDCA: Add basic SDCA function driver ASoC: SDCA: Align mute controls to ALSA expectations ASoC: SDCA: Add terminal type into input/output widget name ASoC: sdw_utils: Add codec_conf for every DAI ASoC: sdw_utils: Move codec_name to dai info ASoC: intel: sof_sdw: Add ability to have auxiliary devices ASoC: sdw_utils: Add cs42l45 support functions ASoC: intel: sof_sdw: Add codec_info for cs42l45 ASoC: SDCA: Fixup some more Kconfig issues Chen-Yu Tsai (4): ASoC: SOF: Fix function topology name check in profile info output ASoC: dt-bindings: allwinner,sun4i-a10-i2s: Add compatible for A523 ASoC: dt-bindings: allwinner,sun4i-a10-spdif: Add compatible for A523 ASoC: sun4i-spdif: Support SPDIF output on A523 family Christophe JAILLET (2): ASoC: renesas: fsi: Constify struct fsi_stream_handler ASoC: fsl_spdif: Constify some structures Cyril Chao (10): ASoC: mediatek: mt8189: add common header ASoC: mediatek: mt8189: support audio clock control ASoC: mediatek: mt8189: support ADDA in platform driver ASoC: mediatek: mt8189: support I2S in platform driver ASoC: mediatek: mt8189: support TDM in platform driver ASoC: mediatek: mt8189: support PCM in platform driver ASoC: dt-bindings: mediatek,mt8189-afe-pcm: add audio afe document ASoC: mediatek: mt8189: add platform driver ASoC: dt-bindings: mediatek,mt8189-nau8825: add mt8189-nau8825 document ASoC: mediatek: mt8189: add machine driver with nau8825 Dan Carpenter (6): ASoC: codecs: va-macro: Clean up on error path in probe() ASoC: codecs: simplify aw87390_init() argument a bit ASoC: codecs: aw88261: pass pointer directly instead of passing the address ASoC: SDCA: Fix NULL vs IS_ERR() bug in sdca_dev_register_functions() ASoC: mediatek: mt8189: remove unnecessary NULL check ASoC: nau8325: Delete a stray tab Dario Binacchi (1): ASoC: mxs-saif: support usage with simple-audio-card Francesco Lavra (1): ASoC: tegra: remove Kconfig dependency on TEGRA20_APB_DMA Frank Li (1): ASoC: dt-bindings: consolidate simple audio codec to trivial-codec.yaml Goko Mell (1): ASoC: spacemit: fix incorrect error check for sspa clock HariKrishna Sagala (5): ASoC: codec: wm8400: replace printk() calls with dev_*() device aware logging ASoC: soc-core: check ops & auto_selectable_formats in snd_soc_dai_get_fmt() to prevent dereference error ASoC: tas2781: Replace deprecated strcpy() with strscpy() ASoC: SOF: sof-client-probes: Replace snprintf() with scnprintf() ASoC: Intel: avs: Replace snprintf() with scnprintf() Herve Codina (3): ASoC: cs4271: Disable regulators in component_probe() error path ASoC: dt-bindings: cirrus,cs4271: Document mclk clock ASoC: cs4271: Add support for the external mclk Jaroslav Kysela (2): ASoC: nau8325: use simple i2c probe function ASoC: nau8325: add missing build config Johan Hovold (5): ASoC: stm32: sai: fix device leak on probe ASoC: stm32: sai: fix clk prepare imbalance on probe failure ASoC: stm32: sai: fix OF node leak on probe ASoC: stm32: sai: clean up probe error path ASoC: codecs: wcd939x: fix regmap leak on probe failure Jonathan Marek (8): ASoC: codecs: lpass-wsa-macro: remove unnecessary bounds check ASoC: codecs: lpass-wsa-macro: remove useless gain read/write sequence ASoC: codecs: lpass-wsa-macro: remove mix path event ASoC: codecs: lpass-wsa-macro: fix path clock dependencies ASoC: codecs: lpass-wsa-macro: add volume controls for mix path ASoC: codecs: lpass-wsa-macro: remove main path event ASoC: codecs: lpass-wsa-macro: remove unused WSA_MACRO_RX_MIX enum ASoC: codecs: lpass-rx-macro: fix mute_stream affecting all paths Kai Vehmanen (1): ASoC: Intel: boards: fix HDMI playback lookup when HDMI-In capture used Konrad Dybcio (1): ASoC: codecs: va-macro: Rework version checking Krzysztof Kozlowski (2): ASoC: codecs: pm4125: Fix potential conflict when probing two devices ASoC: codecs: pm4125: Remove irq_chip on component unbind Kuninori Morimoto (431): ASoC: greybus: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: atmel: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: 88pm860x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: ab8500: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: adav80x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: ak4458: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: ak4619: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: ak4641: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: arizona: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw87390: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw88081: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw88166: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw88261: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw88395: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: aw88399: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: bd28623: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cros_ec: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs35l36: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs35l45: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs4234: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs4270: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs4271: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs42l42: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs42l43: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs42l51: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs42l84: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs43130: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs47l15: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs48l32: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: cs530x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da7210: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da7213: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da7218: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da7219: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da732x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: da9055: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: es8328: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: fs210x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: idt821034: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: lpass-rx-macro: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: lpass-tx-macro: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: lpass-va-macro: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: lpass-wsa-macro: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: madera: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max9759: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max9768: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max98088: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max98090: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max98095: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max98390: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max9867: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: max98925: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: msm8916-wcd-digital: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: mt6358: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: mt6359: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: mt6660: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: nau8810: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: nau8822: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: ntp8835: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: pcm1681: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: pcm512x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: pcm6240: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: peb2466: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: pm4125: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt1011: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt1015: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt1318: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt5631: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt5659: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt5665: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt5670: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt711-sdca: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: rt9123: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: sgtl5000: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: sma1303: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: sma1307: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: sta32x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: sta350: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas2562: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas2781: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas5086: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas571x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas5720: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tas5805m: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tfa989x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tlv320adc3xxx: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tlv320adcx140: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tlv320aic23: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tlv320dac33: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tscs42xx: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: tscs454: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: twl4030: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: twl6040: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: uda1334: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wcd9335: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wcd934x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wcd937x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wcd938x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wcd939x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm2000: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm5102: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm5110: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8350: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8400: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8580: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8731: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8753: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8903: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8904: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8955: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8958: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8960: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8962: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8983: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8985: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8990: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8991: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8994: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm8996: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm9081: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm_adsp: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wm_hubs: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wsa881x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wsa883x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: codecs: wsa884x: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: fsl: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: intel: atom: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: intel: catpt: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: common: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: mt8183: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: mt8186: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: mt8188: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: mt8192: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: mediatek: mt8195: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: rockchip: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: tegra: tegra186: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: tegra: tegra210: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: uniphier: use snd_kcontrol_chip() instead of snd_soc_kcontrol_component() ASoC: soc.h: remove snd_soc_kcontrol_component() ASoC: arizona-haptics: convert to snd_soc_dapm_xxx() ASoC: arizona-micsupp: convert to snd_soc_dapm_xxx() ASoC: audio_codec: convert to snd_soc_dapm_xxx() ASoC: audio_helper: convert to snd_soc_dapm_xxx() ASoC: audio_topology: convert to snd_soc_dapm_xxx() ASoC: soc.h: convert to snd_soc_dapm_xxx() ASoC: asoc.h: convert to snd_soc_dapm_xxx() ASoC: amd: acp-mach-common: convert to snd_soc_dapm_xxx() ASoC: amd: acp3x-es83xx: convert to snd_soc_dapm_xxx() ASoC: amd: acp5x-mach: convert to snd_soc_dapm_xxx() ASoC: atmel: sam9g20_wm8731: convert to snd_soc_dapm_xxx() ASoC: atmel: tse850-pcm5142: convert to snd_soc_dapm_xxx() ASoC: codecs: ab8500: convert to snd_soc_dapm_xxx() ASoC: codecs: 88pm860x: convert to snd_soc_dapm_xxx() ASoC: codecs: ad1836: convert to snd_soc_dapm_xxx() ASoC: codecs: ad193x: convert to snd_soc_dapm_xxx() ASoC: codecs: adau1761: convert to snd_soc_dapm_xxx() ASoC: codecs: adau1781: convert to snd_soc_dapm_xxx() ASoC: codecs: adau17x1: convert to snd_soc_dapm_xxx() ASoC: codecs: adau1977: convert to snd_soc_dapm_xxx() ASoC: codecs: adau7118: convert to snd_soc_dapm_xxx() ASoC: codecs: adav80x: convert to snd_soc_dapm_xxx() ASoC: codecs: ak4641: convert to snd_soc_dapm_xxx() ASoC: codecs: alc5623: convert to snd_soc_dapm_xxx() ASoC: codecs: arizona-jack: convert to snd_soc_dapm_xxx() ASoC: codecs: audio-iio-aux: convert to snd_soc_dapm_xxx() ASoC: codecs: aw88261: convert to snd_soc_dapm_xxx() ASoC: codecs: aw88395: convert to snd_soc_dapm_xxx() ASoC: codecs: cpcap: convert to snd_soc_dapm_xxx() ASoC: codecs: cs35l33: convert to snd_soc_dapm_xxx() ASoC: codecs: cs35l41: convert to snd_soc_dapm_xxx() ASoC: codecs: cs35l45: convert to snd_soc_dapm_xxx() ASoC: codecs: cs35l56: convert to snd_soc_dapm_xxx() ASoC: codecs: cs4234: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l43-jack: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l43: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l51: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l52: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l56: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42l73: convert to snd_soc_dapm_xxx() ASoC: codecs: cs42xx8: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l15: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l24: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l35: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l85: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l90: convert to snd_soc_dapm_xxx() ASoC: codecs: cs47l92: convert to snd_soc_dapm_xxx() ASoC: codecs: cs48l32: convert to snd_soc_dapm_xxx() ASoC: codecs: cs530x: convert to snd_soc_dapm_xxx() ASoC: codecs: cs53l30: convert to snd_soc_dapm_xxx() ASoC: codecs: cx20442: convert to snd_soc_dapm_xxx() ASoC: codecs: cs2072x: convert to snd_soc_dapm_xxx() ASoC: codecs: da7213: convert to snd_soc_dapm_xxx() ASoC: codecs: da7218: convert to snd_soc_dapm_xxx() ASoC: codecs: da7219-aad: convert to snd_soc_dapm_xxx() ASoC: codecs: da7219: convert to snd_soc_dapm_xxx() ASoC: codecs: da732x: convert to snd_soc_dapm_xxx() ASoC: codecs: da9055: convert to snd_soc_dapm_xxx() ASoC: codecs: es7134: convert to snd_soc_dapm_xxx() ASoC: codecs: es8311: convert to snd_soc_dapm_xxx() ASoC: codecs: es8316: convert to snd_soc_dapm_xxx() ASoC: codecs: es8326: convert to snd_soc_dapm_xxx() ASoC: codecs: es8328: convert to snd_soc_dapm_xxx() ASoC: codecs: es8389: convert to snd_soc_dapm_xxx() ASoC: codecs: hda: convert to snd_soc_dapm_xxx() ASoC: codecs: hdac_hda: convert to snd_soc_dapm_xxx() ASoC: codecs: hdac_hdmi: convert to snd_soc_dapm_xxx() ASoC: codecs: hdmi-codec: convert to snd_soc_dapm_xxx() ASoC: codecs: jz4740: convert to snd_soc_dapm_xxx() ASoC: codecs: jz4760: convert to snd_soc_dapm_xxx() ASoC: codecs: jz4770: convert to snd_soc_dapm_xxx() ASoC: codecs: lm49453: convert to snd_soc_dapm_xxx() ASoC: codecs: lpass-rx-macro: convert to snd_soc_dapm_xxx() ASoC: codecs: lpass-tx-macro: convert to snd_soc_dapm_xxx() ASoC: codecs: lpass-va-macro: convert to snd_soc_dapm_xxx() ASoC: codecs: lpass-wsa-macro: convert to snd_soc_dapm_xxx() ASoC: codecs: madera: convert to snd_soc_dapm_xxx() ASoC: codecs: max98088: convert to snd_soc_dapm_xxx() ASoC: codecs: max98090: convert to snd_soc_dapm_xxx() ASoC: codecs: max98095: convert to snd_soc_dapm_xxx() ASoC: codecs: max98373: convert to snd_soc_dapm_xxx() ASoC: codecs: max98390: convert to snd_soc_dapm_xxx() ASoC: codecs: max98396: convert to snd_soc_dapm_xxx() ASoC: codecs: max9850: convert to snd_soc_dapm_xxx() ASoC: codecs: max9867: convert to snd_soc_dapm_xxx() ASoC: codecs: ml26124: convert to snd_soc_dapm_xxx() ASoC: codecs: mt6357: convert to snd_soc_dapm_xxx() ASoC: codecs: mt6358: convert to snd_soc_dapm_xxx() ASoC: codecs: mt6359: convert to snd_soc_dapm_xxx() ASoC: codecs: nau8810: convert to snd_soc_dapm_xxx() ASoC: codecs: nau8821: convert to snd_soc_dapm_xxx() ASoC: codecs: nau8822: convert to snd_soc_dapm_xxx() ASoC: codecs: nau8824: convert to snd_soc_dapm_xxx() ASoC: codecs: nau8825: convert to snd_soc_dapm_xxx() ASoC: codecs: pcm186x: convert to snd_soc_dapm_xxx() ASoC: codecs: pcm512x: convert to snd_soc_dapm_xxx() ASoC: codecs: rk3308: convert to snd_soc_dapm_xxx() ASoC: codecs: rt1011: convert to snd_soc_dapm_xxx() ASoC: codecs: rt1015: convert to snd_soc_dapm_xxx() ASoC: codecs: rt274: convert to snd_soc_dapm_xxx() ASoC: codecs: rt286: convert to snd_soc_dapm_xxx() ASoC: codecs: rt298: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5514: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5516: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5631: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5640: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5645: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5651: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5659: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5660: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5663: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5665: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5668: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5670: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5677: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5682: convert to snd_soc_dapm_xxx() ASoC: codecs: rt5682s: convert to snd_soc_dapm_xxx() ASoC: codecs: rt700: convert to snd_soc_dapm_xxx() ASoC: codecs: rt711-sdca: convert to snd_soc_dapm_xxx() ASoC: codecs: rt711: convert to snd_soc_dapm_xxx() ASoC: codecs: rt712-sdca-dmic: convert to snd_soc_dapm_xxx() ASoC: codecs: rt712-sdca: convert to snd_soc_dapm_xxx() ASoC: codecs: rt715-sdca: convert to snd_soc_dapm_xxx() ASoC: codecs: rt715: convert to snd_soc_dapm_xxx() ASoC: codecs: rt721-sdca: convert to snd_soc_dapm_xxx() ASoC: codecs: simple-mux: convert to snd_soc_dapm_xxx() ASoC: codecs: sma1303: convert to snd_soc_dapm_xxx() ASoC: codecs: sma1307: convert to snd_soc_dapm_xxx() ASoC: codecs: ssm2518: convert to snd_soc_dapm_xxx() ASoC: codecs: ssm2602: convert to snd_soc_dapm_xxx() ASoC: codecs: ssm4567: convert to snd_soc_dapm_xxx() ASoC: codecs: sta32x: convert to snd_soc_dapm_xxx() ASoC: codecs: sta350: convert to snd_soc_dapm_xxx() ASoC: codecs: sta529: convert to snd_soc_dapm_xxx() ASoC: codecs: tas571x: convert to snd_soc_dapm_xxx() ASoC: codecs: tas6424: convert to snd_soc_dapm_xxx() ASoC: codecs: tlv320adc3xxx: convert to snd_soc_dapm_xxx() ASoC: codecs: tlv320aic31xx: convert to snd_soc_dapm_xxx() ASoC: codecs: tlv320aic32x4: convert to snd_soc_dapm_xxx() ASoC: codecs: tlv320aic3x: convert to snd_soc_dapm_xxx() ASoC: codecs: tlv320dac33: convert to snd_soc_dapm_xxx() ASoC: codecs: twl4030: convert to snd_soc_dapm_xxx() ASoC: codecs: twl6040: convert to snd_soc_dapm_xxx() ASoC: codecs: uda1380: convert to snd_soc_dapm_xxx() ASoC: codecs: wcd9335: convert to snd_soc_dapm_xxx() ASoC: codecs: wcd934x: convert to snd_soc_dapm_xxx() ASoC: codecs: wcd937x: convert to snd_soc_dapm_xxx() ASoC: codecs: wm0010: convert to snd_soc_dapm_xxx() ASoC: codecs: wm5100: convert to snd_soc_dapm_xxx() ASoC: codecs: wm5102: convert to snd_soc_dapm_xxx() ASoC: codecs: wm5110: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8350: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8400: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8510: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8523: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8580: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8711: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8728: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8731: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8737: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8750: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8753: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8770: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8776: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8804: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8900: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8903: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8904: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8940: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8955: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8960: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8961: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8962: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8971: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8974: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8978: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8983: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8985: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8988: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8990: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8991: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8993: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8994: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8995: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8996: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8997: convert to snd_soc_dapm_xxx() ASoC: codecs: wm8998: convert to snd_soc_dapm_xxx() ASoC: codecs: wm9081: convert to snd_soc_dapm_xxx() ASoC: codecs: wm9090: convert to snd_soc_dapm_xxx() ASoC: codecs: wm9712: convert to snd_soc_dapm_xxx() ASoC: codecs: wm9713: convert to snd_soc_dapm_xxx() ASoC: codecs: wm_adsp: convert to snd_soc_dapm_xxx() ASoC: codecs: wm_hubs: convert to snd_soc_dapm_xxx() ASoC: fsl: fsl-asoc-card: convert to snd_soc_dapm_xxx() ASoC: fsl: imx-rpmsg: convert to snd_soc_dapm_xxx() ASoC: generic: audio-graph-card: convert to snd_soc_dapm_xxx() ASoC: intel: atom: sst-atom-controls: convert to snd_soc_dapm_xxx() ASoC: intel: avs: da7219: convert to snd_soc_dapm_xxx() ASoC: intel: avs: es8336: convert to snd_soc_dapm_xxx() ASoC: intel: avs: nau8825: convert to snd_soc_dapm_xxx() ASoC: intel: avs: rt274: convert to snd_soc_dapm_xxx() ASoC: intel: avs: rt5514: convert to snd_soc_dapm_xxx() ASoC: intel: avs: rt5640: convert to snd_soc_dapm_xxx() ASoC: intel: avs: control: convert to snd_soc_dapm_xxx() ASoC: intel: avs: pcm: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bdw-rt5677: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bytcht_cx2072x: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bytcht_es8316: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bytcr_rt5640: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bytcr_rt5651: convert to snd_soc_dapm_xxx() ASoC: intel: boards: bytcr_wm5102: convert to snd_soc_dapm_xxx() ASoC: intel: boards: cht_bsw_max98090_ti: convert to snd_soc_dapm_xxx() ASoC: intel: boards: cht_bsw_rt5645: convert to snd_soc_dapm_xxx() ASoC: intel: boards: cht_bsw_rt5672: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_board_helpers: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_cirrus_common: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_da7219: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_es8336: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_maxim_common: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_nau8825: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_nuvoton_common: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_pcm512x: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_realtek_common: convert to snd_soc_dapm_xxx() ASoC: intel: boards: sof_rt5682: convert to snd_soc_dapm_xxx() ASoC: mediatek: mtk-afe-platform-driver: convert to snd_soc_dapm_xxx() ASoC: mediatek: mtk-dsp-sof-common: convert to snd_soc_dapm_xxx() ASoC: mediatek: mt8186-mt6366-common: convert to snd_soc_dapm_xxx() ASoC: mediatek: mt8186-mt6366: convert to snd_soc_dapm_xxx() ASoC: mediatek: mt8188-mt6359: convert to snd_soc_dapm_xxx() ASoC: mediatek: mt8195-mt6359: convert to snd_soc_dapm_xxx() ASoC: mediatek: mt8365-afe-pcm: convert to snd_soc_dapm_xxx() ASoC: meson: aiu-acodec-ctrl: convert to snd_soc_dapm_xxx() ASoC: meson: aiu-codec-ctrl: convert to snd_soc_dapm_xxx() ASoC: meson: axg-spdifout: convert to snd_soc_dapm_xxx() ASoC: meson: axg-tdm-interface: convert to snd_soc_dapm_xxx() ASoC: meson: g12a-toacodec: convert to snd_soc_dapm_xxx() ASoC: meson: g12a-tohdmitx: convert to snd_soc_dapm_xxx() ASoC: meson: t9015: convert to snd_soc_dapm_xxx() ASoC: pxa: spitz: convert to snd_soc_dapm_xxx() ASoC: qcom: q6routing: convert to snd_soc_dapm_xxx() ASoC: qcom: q6usb: convert to snd_soc_dapm_xxx() ASoC: qcom: topology: convert to snd_soc_dapm_xxx() ASoC: qcom: sc7180: convert to snd_soc_dapm_xxx() ASoC: rockchip: rk3288_hdmi_analog: convert to snd_soc_dapm_xxx() ASoC: rockchip: rockchip_max98090: convert to snd_soc_dapm_xxx() ASoC: samsung: aries_wm8994: convert to snd_soc_dapm_xxx() ASoC: samsung: bells: convert to snd_soc_dapm_xxx() ASoC: samsung: littlemill: convert to snd_soc_dapm_xxx() ASoC: samsung: lowland: convert to snd_soc_dapm_xxx() ASoC: samsung: midas_wm1811: convert to snd_soc_dapm_xxx() ASoC: samsung: smdk_wm8994: convert to snd_soc_dapm_xxx() ASoC: samsung: speyside: convert to snd_soc_dapm_xxx() ASoC: samsung: tm2_wm5110: convert to snd_soc_dapm_xxx() ASoC: samsung: tobermory: convert to snd_soc_dapm_xxx() ASoC: sdca: sdca_asoc: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_bridge_cs35l56: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_cs42l42: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_cs42l43: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_cs_amp: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_dmic: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_maxim: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt5682: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt700: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt711: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt_amp: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt_mf_sdca: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_rt_sdca_jack_common: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_ti_amp: convert to snd_soc_dapm_xxx() ASoC: sdw_utils: soc_sdw_utils: convert to snd_soc_dapm_xxx() ASoC: soc-core: convert to snd_soc_dapm_xxx() ASoC: soc-dapm: convert to snd_soc_dapm_xxx() ASoC: soc-jack: convert to snd_soc_dapm_xxx() ASoC: soc-pcm: convert to snd_soc_dapm_xxx() ASoC: soc-topology: convert to snd_soc_dapm_xxx() ASoC: sof-client-probes: convert to snd_soc_dapm_xxx() ASoC: sunxi: sun4i-codec: convert to snd_soc_dapm_xxx() ASoC: sunxi: sun50i-codec-analog: convert to snd_soc_dapm_xxx() ASoC: sunxi: sun8i-codec-analog: convert to snd_soc_dapm_xxx() ASoC: sunxi: sun8i-codec: convert to snd_soc_dapm_xxx() ASoC: tegra: tegra210_ahub: convert to snd_soc_dapm_xxx() ASoC: tegra: tegra_asoc_machine: convert to snd_soc_dapm_xxx() ASoC: tegra: tegra_wm8903: convert to snd_soc_dapm_xxx() ASoC: ti: ams-delta: convert to snd_soc_dapm_xxx() ASoC: ti: davinci-evm: convert to snd_soc_dapm_xxx() ASoC: ti: j721e-evm: convert to snd_soc_dapm_xxx() ASoC: ti: n810: convert to snd_soc_dapm_xxx() ASoC: ti: omap-abe-twl6040: convert to snd_soc_dapm_xxx() ASoC: ti: omap-twl4030: convert to snd_soc_dapm_xxx() ASoC: ti: omap3pandora: convert to snd_soc_dapm_xxx() ASoC: ti: rx51: convert to snd_soc_dapm_xxx() ASoC: ux500: mop500_ab8500: convert to snd_soc_dapm_xxx() ASoC: codecs: arizona: convert to snd_soc_dapm_xxx() Laurentiu Mihalcea (1): ASoC: SOF: imx9: use SCMI API for LM management Lukas Bulwahn (1): MAINTAINERS: refer to trivial-codec.yaml in relevant sections Mac Chiang (4): ASoC: Intel: soc-acpi-intel-ptl-match: Add support for rt722_l0_rt1320_l23 ASoC: Intel: sof_sdw: Add quirk to exclude RT722 speaker ASoC: Intel: soc-acpi-ptl-match: add cs42l43_agg_l3_cs35l56_2 support ASoC: Intel: sof_sdw: add codec speaker support for the SKU Maciej Strozek (3): ASoC: SDCA: Add SDCA FDL data parsing ASoC: SDCA: Add FDL library for XU entities ASoC: SDCA: Add FDL-specific IRQ processing Marco Crivellari (1): ASoC: replace use of system_wq with system_dfl_wq Mark Brown (36): ASoC: use sof_sdw as default Intel SOF SDW machine Add tegra264 audio device tree support ASoC: use sof_sdw as default Intel SOF SDW machine Add Audio Support for Kaanapali MTP Boards ASoC: amd: ps: Propagate the PCI subsystem Vendor and ASoC: spacemit: add i2s support to K1 SoC ASoC: use snd_kcontrol_chip() instead of ASoC: Merge up fixes ALSA: cs35l56: Add support for factory calibration Sndcard compatible for qrb2210/qcm2290 allwinner: a523: Enable I2S and SPDIF TX ASoC: spacemit: fix build warning and error Add support for Cirrus Logic CS530x DAC and CODEC Add SDCA UMP/FDL support ASoC: tas2783: Fix build for SoundWire API update ASoC: qcom: sdw: fix memory leak Add support for an external Master Clock in the ASoC: qcom: q6dsp: fixes and updates ASoC: Merge up fixes ASoC: Intel: machine driver updates for 6.19 ASoC: cs35l56: Support for restoring calibration on ASoC: codecs: lpass-macro: complete sm6115 support ASoC: codecs: pm4125: Two minor fixes for ASoC: Intel: avs: Allow for NHLT configuration ASoC: convert to snd_soc_dapm_xxx() ASoC: mediatek: Add support for MT8189 SoC gpio: improve support for shared GPIOs firmware: cs_dsp: Cleanup debugfs for wmfw and bin Add SDCA class driver ASoC: cs35l56: Set access permissions on volatile nau8325 build fixes ASoC: stm32: sai: fix device and OF node leaks on Add support for cs42l45 into the Intel machine driver ASoC: Intel: catpt: Round of fixes and PM changes firmware: cs_dsp: Remove need for clients to supply ASoC: fsl_micfil: Set default quality and channel Markus Elfring (1): ASoC: rockchip: i2s-tdm: Omit a variable reassignment in rockchip_i2s_tdm_probe() Peter Ujfalusi (3): ASoC: SOF: pcm: Set the PCM device name for HDMI ASoC: soc-pcm: Preserve hw parameters from components in dpcm_runtime_setup_fe ASOC: Intel: sof_sdw: add quirk for Avell B.ON (OEM rebranded NUC15) Pierre-Louis Bossart (1): ASoC: SDCA: add function devices Prasad Kumpatla (3): ASoC: dt-bindings: qcom,sm8250: Add kaanapali sound card ASoC: qcom: sc8280xp: Add support for Kaanapali ASoC: dt-bindings: qcom: Add Kaanapali LPASS macro codecs Randy Dunlap (1): firmware: cs_dsp: fix kernel-doc warnings in a header file Ranganath V N (1): ASoC: dt-bindings: ti,pcm1862: convert to dtschema Richard Fitzgerald (29): ASoC: cs35l56: Read silicon ID during initialization and save it ASoC: cs-amp-lib: Add helpers for factory calibration ASoC: cs35l56: Add common code for factory calibration ASoC: cs35l56: Create debugfs files for factory calibration ALSA: hda/cs35l56: Create debugfs files for factory calibration ASoC: cs-amp-lib-test: Add cases for factory calibration helpers ASoC: cs-amp-lib: Return attributes from cs_amp_get_efi_variable() ASoC: cs-amp-lib: Add function to write calibration to UEFI ASoC: cs35l56: Add calibration command to store into UEFI ALSA: hda/cs35l56: Set cal_index to the amp index ASoC: cs-amp-lib-test: Add test cases for cs_amp_set_efi_calibration_data() ASoC: cs35l56: Alter error codes for calibration routine ASoC: cs35l56: Add control to read CAL_SET_STATUS ASoC: cs35l56: Allow restoring factory calibration through ALSA control firmware: cs_dsp: Factor out common debugfs string read firmware: cs_dsp: Append \n to debugfs string during read ASoC: soc.h: Add SOC_ENUM_EXT_ACC() to allow setting access flags ASoC: soc.h: Add SND_SOC_BYTES_E_ACC() to allow setting access flags ASoC: cs35l56: Use SOC_ENUM_EXT_ACC() for CAL_SET_STATUS control ASoC: cs35l56: Use SND_SOC_BYTES_E_ACC() for CAL_DATA_RB control firmware: cs_dsp: Store control length as 32-bit ASoC: cs-amp-lib: Remove redundant calls to kunit_deactivate_static_stub() firmware: cs_dsp: Remove redundant download buffer allocator firmware: cs_dsp: Take pwr_lock around reading controls debugfs firmware: cs_dsp: Use kvzalloc() to allocate control caches ASoC: cs-amp-lib: Use __free(kfree) instead of manual freeing firmware: cs_dsp: Don't require client to provide a struct cs_dsp_client_ops firmware: cs_dsp: Add test cases for client_ops == NULL ASoC: cs35l56: Log a message if firmware is missing Sharique Mohammad (6): ASoC: max98090/91: added DAPM widget for digital output for max98091 ASoC: max98090/91: fixing a space ASoC: max98090/91: adding DAPM routing for digital output for max98091 ASoC: max98090/91: fixing the stream index ASoC: max98090/91: adding two virtual Mux widgets for digital mics ASoC: max98090/91: adding the two virtual Mux widgets in the routes Shengjiu Wang (1): ASoC: fsl_aud2htx: add IEC958_SUBFRAME_LE format in supported list Shuming Fan (3): ASoC: SOF: Intel: add hyphen between name and index to amp name_prefix ASoC: sdw_utils: add name_prefix for rt1321 part id ASoC: SDCA: support Q7.8 volume format Simon Trimmer (5): ASoC: amd: ps: Propagate the PCI subsystem Vendor and Device IDs ASoC: amd: amd_sdw: Propagate the PCI subsystem Vendor and Device IDs ASoC: cs530x: Add CODEC and DAC support ASoC: cs530x: Check the DEVID matches the devtype ASoC: cs530x: Rename i2c related structures Srinivas Kandagatla (32): ASoC: qcom: sdw: fix memory leak for sdw_stream_runtime ASoC: qcom: sdw: remove redundant code ASoC: qcom: sdm845: make use of common helpers ASoC: qcom: sc7280: make use of common helpers ASoC: codecs: va-macro: fix revision checking ASoC: qcom: q6apm-dai: set flags to reflect correct operation of appl_ptr ASoC: qcom: q6adm: the the copp device only during last instance ASoC: qcom: qdsp6: q6asm-dai: set 10 ms period and buffer alignment. ASoC: qcom: q6asm-dai: perform correct state check before closing ASoC: qcom: q6asm: handle the responses after closing ASoC: qcom: q6asm-dai: schedule all available frames to avoid dsp under-runs ASoC: qcom: q6asm: add q6asm_get_hw_pointer ASoC: qcom: q6asm-dai: use q6asm_get_hw_pointer ASoC: qcom: q6asm: set runtime correctly for each stream ASoc: qcom: audioreach: remove unused variables ASoc: qcom: audioreach: Use automatic cleanup of kfree() ASoc: qcom: q6adm: Use automatic cleanup of kfree() ASoc: qcom: q6afe: Use automatic cleanup of kfree() ASoc: qcom: q6apm: Use automatic cleanup of kfree() ASoc: qcom: q6prm: Use automatic cleanup of kfree() ASoc: qcom: q6asm: Use automatic cleanup of kfree() ASoC: qcom: q6afe: Use guard() for spin locks ASoC: qcom: q6apm-dai: Use guard() for spin locks ASoC: qcom: q6asm-dai: Use guard() for spin locks ASoC: qcom: q6asm: Use guard() for spin locks ASoC: codecs: pm4125: remove duplicate code ASoC: codecs: lpass-tx-macro: fix SM6115 support ASoC: dt-bindings: qcom,lpass-rx-macro: Add sm6115 LPASS RX ASoC: dt-bindings: qcom,lpass-va-macro: re-arrange clock-names ASoC: dt-bindings: qcom,lpass-va-macro: Add sm6115 LPASS VA ASoC: codecs: lpass-va-macro: add SM6115 compatible ASoC: codecs: lpass-rx-macro: add SM6115 compatible Tang Bin (2): ASoc: tas2783A: Remove unneeded variable assignment ASoC: codecs: Fix the error of excessive semicolons Thomas Weißschuh (1): ASoC: stm32: dfsdm: don't use %pK through printk Thorsten Blum (1): ASoC: Intel: atom: Replace strcpy() with strscpy() Troy Mitchell (5): ASoC: dt-bindings: Add bindings for SpacemiT K1 ASoC: spacemit: add i2s support for K1 SoC ASoC: spacemit: add failure check for spacemit_i2s_init_dai() ASoC: spacemit: use `depends on` instead of `select` ASoC: spacemit: add failure check for spacemit_i2s_init_dai() Vitaly Rodionov (9): ASoC: cs530x: Update the copyright headers ASoC: cs530x: Sort #include directives and tydy up whitespaces ASoC: cs530x: Remove unused struct members and constants ASoC: cs530x: Correct constant naming ASoC: dt-bindings: sound: cirrus: cs530x: Add cs530x ASoC: cs530x: Rename bitfield to reflect common use for ADC and DAC ASoC: cs530x: Correct MCLK reference frequency values ASoC: dt-bindings: sound: cirrus: cs530x: Add SPI bus support ASoC: cs530x: Add SPI bus support for cs530x parts Weidong Wang (1): ASoC: codecs: Modify awinic amplifier dsp read and write functions Wolfram Sang (1): ASoC: dt-bindings: don't check node names jempty.liang (1): ASoC: soc-core: Pre-check zero CPU/codec DAIs, handle early rtd->dais alloc failure sheetal (1): dt-bindings: sound: Update ADMAIF bindings for tegra264 .../devicetree/bindings/sound/adi,max98363.yaml | 60 - .../devicetree/bindings/sound/adi,ssm2602.txt | 19 - .../devicetree/bindings/sound/adi,ssm3515.yaml | 49 - .../bindings/sound/allwinner,sun4i-a10-i2s.yaml | 4 +- .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 44 +- .../devicetree/bindings/sound/cirrus,cs4271.yaml | 10 + .../devicetree/bindings/sound/cirrus,cs530x.yaml | 8 + Documentation/devicetree/bindings/sound/cs4265.txt | 29 - Documentation/devicetree/bindings/sound/cs4341.txt | 22 - Documentation/devicetree/bindings/sound/cs4349.txt | 19 - Documentation/devicetree/bindings/sound/da9055.txt | 22 - .../bindings/sound/mediatek,mt8189-afe-pcm.yaml | 178 + .../bindings/sound/mediatek,mt8189-nau8825.yaml | 101 + .../devicetree/bindings/sound/nuvoton,nau8540.yaml | 40 - .../devicetree/bindings/sound/nuvoton,nau8810.yaml | 45 - .../bindings/sound/nvidia,tegra210-admaif.yaml | 106 +- .../devicetree/bindings/sound/nxp,tfa9879.yaml | 44 - .../devicetree/bindings/sound/nxp,uda1342.yaml | 42 - .../devicetree/bindings/sound/pcm1789.txt | 22 - .../devicetree/bindings/sound/pcm179x.txt | 27 - .../devicetree/bindings/sound/pcm186x.txt | 42 - .../devicetree/bindings/sound/pcm5102a.txt | 13 - .../bindings/sound/qcom,lpass-rx-macro.yaml | 19 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 1 + .../bindings/sound/qcom,lpass-va-macro.yaml | 50 +- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 1 + .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 + .../devicetree/bindings/sound/qcom,wcd934x.yaml | 2 +- .../devicetree/bindings/sound/spacemit,k1-i2s.yaml | 87 + .../devicetree/bindings/sound/ti,pcm1862.yaml | 76 + .../devicetree/bindings/sound/ti,tas2781.yaml | 58 +- .../devicetree/bindings/sound/trivial-codec.yaml | 79 + .../devicetree/bindings/sound/wlf,wm8510.yaml | 41 - .../devicetree/bindings/sound/wlf,wm8523.yaml | 40 - .../devicetree/bindings/sound/wlf,wm8580.yaml | 42 - .../devicetree/bindings/sound/wlf,wm8711.yaml | 40 - .../devicetree/bindings/sound/wlf,wm8728.yaml | 40 - .../devicetree/bindings/sound/wlf,wm8737.yaml | 40 - .../devicetree/bindings/sound/wlf,wm8750.yaml | 42 - .../devicetree/bindings/sound/wlf,wm8753.yaml | 62 - .../devicetree/bindings/sound/wlf,wm8776.yaml | 41 - .../devicetree/bindings/sound/wlf,wm8961.yaml | 43 - .../devicetree/bindings/sound/wlf,wm8974.yaml | 41 - Documentation/devicetree/bindings/sound/wm8770.txt | 16 - MAINTAINERS | 8 +- drivers/base/regmap/regmap-sdw-mbq.c | 23 +- drivers/firmware/cirrus/cs_dsp.c | 175 +- .../firmware/cirrus/test/cs_dsp_test_callbacks.c | 1 + drivers/gpio/Kconfig | 17 + drivers/gpio/Makefile | 2 + drivers/gpio/gpio-shared-proxy.c | 333 + drivers/gpio/gpiolib-shared.c | 558 + drivers/gpio/gpiolib-shared.h | 71 + drivers/gpio/gpiolib.c | 70 +- drivers/gpio/gpiolib.h | 2 + drivers/input/misc/arizona-haptics.c | 14 +- drivers/regulator/arizona-micsupp.c | 8 +- drivers/staging/greybus/audio_codec.c | 16 +- drivers/staging/greybus/audio_helper.c | 9 +- drivers/staging/greybus/audio_topology.c | 24 +- include/linux/firmware/cirrus/cs_dsp.h | 6 +- include/linux/firmware/cirrus/cs_dsp_test_utils.h | 18 +- include/linux/gpio/consumer.h | 9 + include/linux/regmap.h | 21 +- include/linux/soundwire/sdw_registers.h | 2 + include/linux/string.h | 18 + include/sound/cs-amp-lib.h | 25 +- include/sound/cs35l56.h | 50 + include/sound/sdca.h | 19 + include/sound/sdca_fdl.h | 105 + include/sound/sdca_function.h | 126 +- include/sound/sdca_hid.h | 21 +- include/sound/sdca_interrupts.h | 19 +- include/sound/sdca_regmap.h | 2 + include/sound/sdca_ump.h | 50 + include/sound/soc-acpi.h | 8 +- include/sound/soc.h | 39 +- include/sound/soc_sdw_utils.h | 19 +- include/sound/tas2781.h | 6 + include/trace/events/asoc.h | 4 +- include/uapi/sound/intel/avs/tokens.h | 6 + lib/tests/string_kunit.c | 13 + sound/hda/codecs/side-codecs/Kconfig | 15 + sound/hda/codecs/side-codecs/cs35l56_hda.c | 116 +- sound/hda/codecs/side-codecs/cs35l56_hda.h | 6 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/acp/acp-mach-common.c | 40 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 23 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 12 +- sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 10 +- sound/soc/amd/ps/acp63.h | 2 + sound/soc/amd/ps/pci-ps.c | 9 + sound/soc/amd/vangogh/acp5x-mach.c | 3 +- sound/soc/atmel/atmel-pdmic.c | 4 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 32 +- sound/soc/codecs/88pm860x-codec.c | 11 +- sound/soc/codecs/Kconfig | 49 + sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/ab8500-codec.c | 18 +- sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad193x.c | 4 +- sound/soc/codecs/adau1761.c | 9 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau17x1.c | 14 +- sound/soc/codecs/adau1977.c | 5 +- sound/soc/codecs/adau7118.c | 7 +- sound/soc/codecs/adav80x.c | 10 +- sound/soc/codecs/ak4458.c | 4 +- sound/soc/codecs/ak4619.c | 4 +- sound/soc/codecs/ak4641.c | 7 +- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/arizona-jack.c | 8 +- sound/soc/codecs/arizona.c | 22 +- sound/soc/codecs/audio-iio-aux.c | 2 +- sound/soc/codecs/aw87390.c | 14 +- sound/soc/codecs/aw88081.c | 24 +- sound/soc/codecs/aw88166.c | 159 +- sound/soc/codecs/aw88166.h | 5 - sound/soc/codecs/aw88261.c | 34 +- sound/soc/codecs/aw88395/aw88395.c | 30 +- sound/soc/codecs/aw88395/aw88395_device.c | 39 +- sound/soc/codecs/aw88395/aw88395_device.h | 8 +- sound/soc/codecs/aw88399.c | 253 +- sound/soc/codecs/aw88399.h | 5 - sound/soc/codecs/bd28623.c | 6 +- sound/soc/codecs/cpcap.c | 21 +- sound/soc/codecs/cros_ec_codec.c | 10 +- sound/soc/codecs/cs-amp-lib-test.c | 1522 ++- sound/soc/codecs/cs-amp-lib.c | 372 +- sound/soc/codecs/cs35l33.c | 4 +- sound/soc/codecs/cs35l36.c | 6 +- sound/soc/codecs/cs35l41.c | 2 +- sound/soc/codecs/cs35l45.c | 12 +- sound/soc/codecs/cs35l56-shared.c | 439 +- sound/soc/codecs/cs35l56.c | 252 +- sound/soc/codecs/cs35l56.h | 6 + sound/soc/codecs/cs4234.c | 7 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 36 +- sound/soc/codecs/cs42l42.c | 2 +- sound/soc/codecs/cs42l43-jack.c | 14 +- sound/soc/codecs/cs42l43.c | 38 +- sound/soc/codecs/cs42l51.c | 6 +- sound/soc/codecs/cs42l52.c | 5 +- sound/soc/codecs/cs42l56.c | 5 +- sound/soc/codecs/cs42l73.c | 3 +- sound/soc/codecs/cs42l84.c | 4 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/cs47l15.c | 11 +- sound/soc/codecs/cs47l24.c | 4 +- sound/soc/codecs/cs47l35.c | 5 +- sound/soc/codecs/cs47l85.c | 5 +- sound/soc/codecs/cs47l90.c | 5 +- sound/soc/codecs/cs47l92.c | 11 +- sound/soc/codecs/cs48l32.c | 40 +- sound/soc/codecs/cs530x-i2c.c | 24 +- sound/soc/codecs/cs530x-spi.c | 92 + sound/soc/codecs/cs530x.c | 526 +- sound/soc/codecs/cs530x.h | 90 +- sound/soc/codecs/cs53l30.c | 9 +- sound/soc/codecs/cx20442.c | 5 +- sound/soc/codecs/cx2072x.c | 6 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 21 +- sound/soc/codecs/da7218.c | 21 +- sound/soc/codecs/da7219-aad.c | 10 +- sound/soc/codecs/da7219.c | 29 +- sound/soc/codecs/da732x.c | 7 +- sound/soc/codecs/da9055.c | 6 +- sound/soc/codecs/es7134.c | 2 +- sound/soc/codecs/es8311.c | 3 +- sound/soc/codecs/es8316.c | 4 +- sound/soc/codecs/es8326.c | 16 +- sound/soc/codecs/es8328.c | 8 +- sound/soc/codecs/es8389.c | 4 +- sound/soc/codecs/fs210x.c | 2 +- sound/soc/codecs/hda.c | 2 +- sound/soc/codecs/hdac_hda.c | 4 +- sound/soc/codecs/hdac_hdmi.c | 35 +- sound/soc/codecs/hdmi-codec.c | 2 +- sound/soc/codecs/idt821034.c | 8 +- sound/soc/codecs/jz4740.c | 3 +- sound/soc/codecs/jz4760.c | 7 +- sound/soc/codecs/jz4770.c | 8 +- sound/soc/codecs/lm49453.c | 3 +- sound/soc/codecs/lpass-rx-macro.c | 116 +- sound/soc/codecs/lpass-tx-macro.c | 19 +- sound/soc/codecs/lpass-va-macro.c | 104 +- sound/soc/codecs/lpass-wsa-macro.c | 169 +- sound/soc/codecs/madera.c | 48 +- sound/soc/codecs/max9759.c | 8 +- sound/soc/codecs/max9768.c | 4 +- sound/soc/codecs/max98088.c | 18 +- sound/soc/codecs/max98090.c | 54 +- sound/soc/codecs/max98095.c | 27 +- sound/soc/codecs/max98373.c | 3 +- sound/soc/codecs/max98390.c | 22 +- sound/soc/codecs/max98396.c | 11 +- sound/soc/codecs/max9850.c | 3 +- sound/soc/codecs/max9867.c | 15 +- sound/soc/codecs/max98925.c | 2 +- sound/soc/codecs/ml26124.c | 3 +- sound/soc/codecs/msm8916-wcd-digital.c | 6 +- sound/soc/codecs/mt6357.c | 2 +- sound/soc/codecs/mt6358.c | 25 +- sound/soc/codecs/mt6359.c | 18 +- sound/soc/codecs/mt6660.c | 3 +- sound/soc/codecs/nau8325.c | 5 +- sound/soc/codecs/nau8810.c | 7 +- sound/soc/codecs/nau8821.c | 28 +- sound/soc/codecs/nau8822.c | 20 +- sound/soc/codecs/nau8824.c | 10 +- sound/soc/codecs/nau8825.c | 6 +- sound/soc/codecs/ntp8835.c | 4 +- sound/soc/codecs/pcm1681.c | 4 +- sound/soc/codecs/pcm186x.c | 6 +- sound/soc/codecs/pcm512x.c | 25 +- sound/soc/codecs/pcm6240.c | 9 +- sound/soc/codecs/peb2466.c | 6 +- sound/soc/codecs/pm4125-sdw.c | 64 +- sound/soc/codecs/pm4125.c | 104 +- sound/soc/codecs/pm4125.h | 18 +- sound/soc/codecs/rk3308_codec.c | 3 +- sound/soc/codecs/rt1011.c | 36 +- sound/soc/codecs/rt1015.c | 17 +- sound/soc/codecs/rt1318.c | 4 +- sound/soc/codecs/rt274.c | 5 +- sound/soc/codecs/rt286.c | 8 +- sound/soc/codecs/rt298.c | 9 +- sound/soc/codecs/rt5514.c | 8 +- sound/soc/codecs/rt5616.c | 5 +- sound/soc/codecs/rt5631.c | 10 +- sound/soc/codecs/rt5640.c | 19 +- sound/soc/codecs/rt5645.c | 15 +- sound/soc/codecs/rt5651.c | 13 +- sound/soc/codecs/rt5659.c | 13 +- sound/soc/codecs/rt5660.c | 5 +- sound/soc/codecs/rt5663.c | 10 +- sound/soc/codecs/rt5665.c | 6 +- sound/soc/codecs/rt5668.c | 3 +- sound/soc/codecs/rt5670.c | 43 +- sound/soc/codecs/rt5677.c | 14 +- sound/soc/codecs/rt5682.c | 10 +- sound/soc/codecs/rt5682s.c | 5 +- sound/soc/codecs/rt700.c | 28 +- sound/soc/codecs/rt711-sdca.c | 13 +- sound/soc/codecs/rt711.c | 28 +- sound/soc/codecs/rt712-sdca-dmic.c | 9 +- sound/soc/codecs/rt712-sdca.c | 20 +- sound/soc/codecs/rt715-sdca.c | 9 +- sound/soc/codecs/rt715.c | 35 +- sound/soc/codecs/rt721-sdca.c | 9 +- sound/soc/codecs/rt722-sdca-sdw.c | 4 +- sound/soc/codecs/rt9123.c | 6 +- sound/soc/codecs/sgtl5000.c | 8 +- sound/soc/codecs/simple-mux.c | 7 +- sound/soc/codecs/sma1303.c | 31 +- sound/soc/codecs/sma1307.c | 95 +- sound/soc/codecs/ssm2518.c | 3 +- sound/soc/codecs/ssm2602.c | 4 +- sound/soc/codecs/ssm4567.c | 3 +- sound/soc/codecs/sta32x.c | 10 +- sound/soc/codecs/sta350.c | 10 +- sound/soc/codecs/sta529.c | 3 +- sound/soc/codecs/tas2562.c | 4 +- sound/soc/codecs/tas2781-i2c.c | 104 +- sound/soc/codecs/tas2783-sdw.c | 4 +- sound/soc/codecs/tas5086.c | 4 +- sound/soc/codecs/tas571x.c | 7 +- sound/soc/codecs/tas5720.c | 4 +- sound/soc/codecs/tas5805m.c | 6 +- sound/soc/codecs/tas6424.c | 4 +- sound/soc/codecs/tfa989x.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 8 +- sound/soc/codecs/tlv320adcx140.c | 6 +- sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/tlv320aic31xx.c | 14 +- sound/soc/codecs/tlv320aic32x4.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/tlv320dac33.c | 9 +- sound/soc/codecs/tscs42xx.c | 6 +- sound/soc/codecs/tscs454.c | 6 +- sound/soc/codecs/twl4030.c | 14 +- sound/soc/codecs/twl6040.c | 15 +- sound/soc/codecs/uda1334.c | 4 +- sound/soc/codecs/uda1380.c | 3 +- sound/soc/codecs/wcd9335.c | 34 +- sound/soc/codecs/wcd934x.c | 44 +- sound/soc/codecs/wcd937x.c | 20 +- sound/soc/codecs/wcd938x.c | 27 +- sound/soc/codecs/wcd939x-sdw.c | 8 +- sound/soc/codecs/wcd939x.c | 24 +- sound/soc/codecs/wm0010.c | 5 +- sound/soc/codecs/wm2000.c | 8 +- sound/soc/codecs/wm5100.c | 4 +- sound/soc/codecs/wm5102.c | 12 +- sound/soc/codecs/wm5110.c | 16 +- sound/soc/codecs/wm8350.c | 7 +- sound/soc/codecs/wm8400.c | 14 +- sound/soc/codecs/wm8510.c | 3 +- sound/soc/codecs/wm8523.c | 3 +- sound/soc/codecs/wm8580.c | 8 +- sound/soc/codecs/wm8711.c | 3 +- sound/soc/codecs/wm8728.c | 3 +- sound/soc/codecs/wm8731.c | 9 +- sound/soc/codecs/wm8737.c | 6 +- sound/soc/codecs/wm8750.c | 3 +- sound/soc/codecs/wm8753.c | 7 +- sound/soc/codecs/wm8770.c | 3 +- sound/soc/codecs/wm8776.c | 3 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8900.c | 12 +- sound/soc/codecs/wm8903.c | 10 +- sound/soc/codecs/wm8904.c | 23 +- sound/soc/codecs/wm8940.c | 6 +- sound/soc/codecs/wm8955.c | 10 +- sound/soc/codecs/wm8958-dsp2.c | 32 +- sound/soc/codecs/wm8960.c | 16 +- sound/soc/codecs/wm8961.c | 5 +- sound/soc/codecs/wm8962.c | 30 +- sound/soc/codecs/wm8971.c | 3 +- sound/soc/codecs/wm8974.c | 3 +- sound/soc/codecs/wm8978.c | 9 +- sound/soc/codecs/wm8983.c | 7 +- sound/soc/codecs/wm8985.c | 9 +- sound/soc/codecs/wm8988.c | 3 +- sound/soc/codecs/wm8990.c | 9 +- sound/soc/codecs/wm8991.c | 5 +- sound/soc/codecs/wm8993.c | 13 +- sound/soc/codecs/wm8994.c | 46 +- sound/soc/codecs/wm8995.c | 7 +- sound/soc/codecs/wm8996.c | 13 +- sound/soc/codecs/wm8997.c | 4 +- sound/soc/codecs/wm8998.c | 8 +- sound/soc/codecs/wm9081.c | 7 +- sound/soc/codecs/wm9090.c | 5 +- sound/soc/codecs/wm9712.c | 7 +- sound/soc/codecs/wm9713.c | 7 +- sound/soc/codecs/wm_adsp.c | 21 +- sound/soc/codecs/wm_hubs.c | 10 +- sound/soc/codecs/wsa881x.c | 9 +- sound/soc/codecs/wsa883x.c | 19 +- sound/soc/codecs/wsa884x.c | 8 +- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_aud2htx.h | 3 +- sound/soc/fsl/fsl_micfil.c | 136 +- sound/soc/fsl/fsl_spdif.c | 16 +- sound/soc/fsl/fsl_xcvr.c | 86 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/intel/atom/sst-atom-controls.c | 15 +- sound/soc/intel/atom/sst/sst_acpi.c | 3 +- sound/soc/intel/avs/boards/da7219.c | 3 +- sound/soc/intel/avs/boards/es8336.c | 5 +- sound/soc/intel/avs/boards/nau8825.c | 3 +- sound/soc/intel/avs/boards/rt274.c | 6 +- sound/soc/intel/avs/boards/rt5514.c | 3 +- sound/soc/intel/avs/boards/rt5640.c | 3 +- sound/soc/intel/avs/control.c | 7 +- sound/soc/intel/avs/debugfs.c | 9 +- sound/soc/intel/avs/path.c | 13 +- sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/avs/topology.c | 113 +- sound/soc/intel/avs/topology.h | 8 + sound/soc/intel/boards/bdw-rt5677.c | 9 +- sound/soc/intel/boards/bytcht_cx2072x.c | 3 +- sound/soc/intel/boards/bytcht_es8316.c | 9 +- sound/soc/intel/boards/bytcr_rt5640.c | 25 +- sound/soc/intel/boards/bytcr_rt5651.c | 18 +- sound/soc/intel/boards/bytcr_wm5102.c | 14 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 5 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 12 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 +- sound/soc/intel/boards/hda_dsp_common.c | 15 +- sound/soc/intel/boards/sof_board_helpers.c | 10 +- sound/soc/intel/boards/sof_cirrus_common.c | 5 +- sound/soc/intel/boards/sof_da7219.c | 5 +- sound/soc/intel/boards/sof_es8336.c | 14 +- sound/soc/intel/boards/sof_maxim_common.c | 22 +- sound/soc/intel/boards/sof_nau8825.c | 2 +- sound/soc/intel/boards/sof_nuvoton_common.c | 5 +- sound/soc/intel/boards/sof_pcm512x.c | 5 +- sound/soc/intel/boards/sof_realtek_common.c | 34 +- sound/soc/intel/boards/sof_rt5682.c | 17 +- sound/soc/intel/boards/sof_sdw.c | 80 +- sound/soc/intel/catpt/device.c | 26 +- sound/soc/intel/catpt/loader.c | 18 +- sound/soc/intel/catpt/pcm.c | 31 +- sound/soc/intel/catpt/sysfs.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 79 + sound/soc/intel/common/sof-function-topology-lib.c | 8 +- sound/soc/mediatek/Kconfig | 31 + sound/soc/mediatek/Makefile | 1 + .../soc/mediatek/common/mtk-afe-platform-driver.c | 7 +- sound/soc/mediatek/common/mtk-btcvsd.c | 24 +- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 5 +- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 4 +- sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 4 +- sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 12 +- sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 4 +- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 4 +- sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 4 +- sound/soc/mediatek/mt8186/mt8186-misc-control.c | 12 +- sound/soc/mediatek/mt8186/mt8186-mt6366-common.c | 2 +- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 19 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 10 +- sound/soc/mediatek/mt8188/mt8188-dai-adda.c | 4 +- sound/soc/mediatek/mt8188/mt8188-dai-dmic.c | 2 +- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 3 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 +- sound/soc/mediatek/mt8189/Makefile | 18 + sound/soc/mediatek/mt8189/mt8189-afe-clk.c | 750 ++ sound/soc/mediatek/mt8189/mt8189-afe-clk.h | 76 + sound/soc/mediatek/mt8189/mt8189-afe-common.h | 240 + sound/soc/mediatek/mt8189/mt8189-afe-pcm.c | 2615 +++++ sound/soc/mediatek/mt8189/mt8189-dai-adda.c | 1228 +++ sound/soc/mediatek/mt8189/mt8189-dai-i2s.c | 1463 +++ sound/soc/mediatek/mt8189/mt8189-dai-pcm.c | 332 + sound/soc/mediatek/mt8189/mt8189-dai-tdm.c | 672 ++ sound/soc/mediatek/mt8189/mt8189-interconnection.h | 97 + sound/soc/mediatek/mt8189/mt8189-nau8825.c | 1178 ++ sound/soc/mediatek/mt8189/mt8189-reg.h | 10773 +++++++++++++++++++ sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 12 +- sound/soc/mediatek/mt8192/mt8192-dai-i2s.c | 4 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 6 +- sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 21 +- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 3 +- sound/soc/meson/aiu-acodec-ctrl.c | 6 +- sound/soc/meson/aiu-codec-ctrl.c | 6 +- sound/soc/meson/axg-spdifout.c | 4 +- sound/soc/meson/axg-tdm-interface.c | 4 +- sound/soc/meson/g12a-toacodec.c | 6 +- sound/soc/meson/g12a-tohdmitx.c | 12 +- sound/soc/meson/t9015.c | 4 +- sound/soc/mxs/mxs-saif.c | 123 +- sound/soc/pxa/spitz.c | 9 +- sound/soc/qcom/qdsp6/audioreach.c | 234 +- sound/soc/qcom/qdsp6/q6adm.c | 179 +- sound/soc/qcom/qdsp6/q6afe.c | 49 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 25 +- sound/soc/qcom/qdsp6/q6apm.c | 57 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 87 +- sound/soc/qcom/qdsp6/q6asm.c | 203 +- sound/soc/qcom/qdsp6/q6asm.h | 1 + sound/soc/qcom/qdsp6/q6prm.c | 27 +- sound/soc/qcom/qdsp6/q6routing.c | 6 +- sound/soc/qcom/qdsp6/q6usb.c | 3 +- sound/soc/qcom/qdsp6/topology.c | 12 +- sound/soc/qcom/sc7180.c | 10 +- sound/soc/qcom/sc7280.c | 67 +- sound/soc/qcom/sc8280xp.c | 34 +- sound/soc/qcom/sdm845.c | 53 +- sound/soc/qcom/sdw.c | 128 +- sound/soc/qcom/sdw.h | 7 +- sound/soc/qcom/sm8250.c | 34 +- sound/soc/qcom/x1e80100.c | 33 +- sound/soc/renesas/fsi.c | 8 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 2 +- sound/soc/rockchip/rockchip_sai.c | 8 +- sound/soc/samsung/aries_wm8994.c | 6 +- sound/soc/samsung/bells.c | 6 +- sound/soc/samsung/littlemill.c | 8 +- sound/soc/samsung/lowland.c | 3 +- sound/soc/samsung/midas_wm1811.c | 8 +- sound/soc/samsung/smdk_wm8994.c | 30 +- sound/soc/samsung/speyside.c | 24 +- sound/soc/samsung/tm2_wm5110.c | 7 +- sound/soc/samsung/tobermory.c | 6 +- sound/soc/sdca/Kconfig | 28 + sound/soc/sdca/Makefile | 10 +- sound/soc/sdca/sdca_asoc.c | 89 +- sound/soc/sdca/sdca_class.c | 304 + sound/soc/sdca/sdca_class.h | 37 + sound/soc/sdca/sdca_class_function.c | 460 + sound/soc/sdca/sdca_device.c | 20 + sound/soc/sdca/sdca_fdl.c | 504 + sound/soc/sdca/sdca_function_device.c | 117 + sound/soc/sdca/sdca_function_device.h | 15 + sound/soc/sdca/sdca_functions.c | 310 +- sound/soc/sdca/sdca_hid.c | 59 +- sound/soc/sdca/sdca_interrupts.c | 280 +- sound/soc/sdca/sdca_regmap.c | 98 +- sound/soc/sdca/sdca_ump.c | 262 + sound/soc/sdw_utils/Makefile | 1 + sound/soc/sdw_utils/soc_sdw_bridge_cs35l56.c | 5 +- sound/soc/sdw_utils/soc_sdw_cs42l42.c | 3 +- sound/soc/sdw_utils/soc_sdw_cs42l43.c | 9 +- sound/soc/sdw_utils/soc_sdw_cs42l45.c | 80 + sound/soc/sdw_utils/soc_sdw_cs_amp.c | 3 +- sound/soc/sdw_utils/soc_sdw_dmic.c | 5 +- sound/soc/sdw_utils/soc_sdw_maxim.c | 8 +- sound/soc/sdw_utils/soc_sdw_rt5682.c | 3 +- sound/soc/sdw_utils/soc_sdw_rt700.c | 3 +- sound/soc/sdw_utils/soc_sdw_rt711.c | 3 +- sound/soc/sdw_utils/soc_sdw_rt_amp.c | 5 +- sound/soc/sdw_utils/soc_sdw_rt_mf_sdca.c | 3 +- sound/soc/sdw_utils/soc_sdw_rt_sdca_jack_common.c | 11 +- sound/soc/sdw_utils/soc_sdw_ti_amp.c | 3 +- sound/soc/sdw_utils/soc_sdw_utils.c | 158 +- sound/soc/soc-core.c | 42 +- sound/soc/soc-dai.c | 5 +- sound/soc/soc-dapm.c | 196 +- sound/soc/soc-jack.c | 2 +- sound/soc/soc-ops.c | 62 +- sound/soc/soc-pcm.c | 38 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/fw-file-profile.c | 13 +- sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/imx/imx9.c | 36 +- sound/soc/sof/intel/Kconfig | 1 + sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 4 +- sound/soc/sof/intel/hda-ipc.c | 2 +- sound/soc/sof/intel/hda.c | 213 +- sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/lnl.c | 1 + sound/soc/sof/intel/mtl.c | 2 + sound/soc/sof/intel/ptl.c | 2 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/intel/skl.c | 1 + sound/soc/sof/intel/tgl.c | 4 + sound/soc/sof/pcm.c | 18 + sound/soc/sof/sof-client-probes-ipc4.c | 2 +- sound/soc/sof/sof-client-probes.c | 11 +- sound/soc/sof/topology.c | 7 +- sound/soc/spacemit/Kconfig | 15 + sound/soc/spacemit/Makefile | 5 + sound/soc/spacemit/k1_i2s.c | 461 + sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_sai.c | 14 +- sound/soc/stm/stm32_sai_sub.c | 57 +- sound/soc/sunxi/sun4i-codec.c | 3 +- sound/soc/sunxi/sun4i-spdif.c | 26 +- sound/soc/sunxi/sun50i-codec-analog.c | 2 +- sound/soc/sunxi/sun8i-codec-analog.c | 14 +- sound/soc/sunxi/sun8i-codec.c | 4 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra186_asrc.c | 24 +- sound/soc/tegra/tegra186_dspk.c | 24 +- sound/soc/tegra/tegra210_admaif.c | 16 +- sound/soc/tegra/tegra210_adx.c | 4 +- sound/soc/tegra/tegra210_ahub.c | 6 +- sound/soc/tegra/tegra210_amx.c | 4 +- sound/soc/tegra/tegra210_dmic.c | 24 +- sound/soc/tegra/tegra210_i2s.c | 32 +- sound/soc/tegra/tegra210_mbdrc.c | 24 +- sound/soc/tegra/tegra210_mixer.c | 4 +- sound/soc/tegra/tegra210_mvc.c | 12 +- sound/soc/tegra/tegra210_ope.c | 4 +- sound/soc/tegra/tegra210_peq.c | 8 +- sound/soc/tegra/tegra210_sfc.c | 16 +- sound/soc/tegra/tegra_asoc_machine.c | 11 +- sound/soc/tegra/tegra_wm8903.c | 3 +- sound/soc/ti/ams-delta.c | 8 +- sound/soc/ti/davinci-evm.c | 11 +- sound/soc/ti/j721e-evm.c | 2 +- sound/soc/ti/n810.c | 12 +- sound/soc/ti/omap-abe-twl6040.c | 2 +- sound/soc/ti/omap-twl4030.c | 2 +- sound/soc/ti/omap3pandora.c | 36 +- sound/soc/ti/rx51.c | 17 +- sound/soc/uniphier/aio-cpu.c | 4 +- sound/soc/uniphier/evea.c | 12 +- sound/soc/ux500/mop500_ab8500.c | 2 +- 571 files changed, 32195 insertions(+), 5513 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/adi,max98363.yaml delete mode 100644 Documentation/devicetree/bindings/sound/adi,ssm2602.txt delete mode 100644 Documentation/devicetree/bindings/sound/adi,ssm3515.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4265.txt delete mode 100644 Documentation/devicetree/bindings/sound/cs4341.txt delete mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt delete mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8189-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8189-nau8825.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8540.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8810.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nxp,tfa9879.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nxp,uda1342.yaml delete mode 100644 Documentation/devicetree/bindings/sound/pcm1789.txt delete mode 100644 Documentation/devicetree/bindings/sound/pcm179x.txt delete mode 100644 Documentation/devicetree/bindings/sound/pcm186x.txt delete mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt create mode 100644 Documentation/devicetree/bindings/sound/spacemit,k1-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm1862.yaml create mode 100644 Documentation/devicetree/bindings/sound/trivial-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8510.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8523.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8580.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8711.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8728.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8737.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8750.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8753.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8776.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8961.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8974.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8770.txt create mode 100644 drivers/gpio/gpio-shared-proxy.c create mode 100644 drivers/gpio/gpiolib-shared.c create mode 100644 drivers/gpio/gpiolib-shared.h create mode 100644 include/sound/sdca_fdl.h create mode 100644 include/sound/sdca_ump.h create mode 100644 sound/soc/codecs/cs530x-spi.c create mode 100644 sound/soc/mediatek/mt8189/Makefile create mode 100644 sound/soc/mediatek/mt8189/mt8189-afe-clk.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-afe-clk.h create mode 100644 sound/soc/mediatek/mt8189/mt8189-afe-common.h create mode 100644 sound/soc/mediatek/mt8189/mt8189-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-dai-adda.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-interconnection.h create mode 100644 sound/soc/mediatek/mt8189/mt8189-nau8825.c create mode 100644 sound/soc/mediatek/mt8189/mt8189-reg.h create mode 100644 sound/soc/sdca/sdca_class.c create mode 100644 sound/soc/sdca/sdca_class.h create mode 100644 sound/soc/sdca/sdca_class_function.c create mode 100644 sound/soc/sdca/sdca_fdl.c create mode 100644 sound/soc/sdca/sdca_function_device.c create mode 100644 sound/soc/sdca/sdca_function_device.h create mode 100644 sound/soc/sdca/sdca_ump.c create mode 100644 sound/soc/sdw_utils/soc_sdw_cs42l45.c create mode 100644 sound/soc/spacemit/Kconfig create mode 100644 sound/soc/spacemit/Makefile create mode 100644 sound/soc/spacemit/k1_i2s.c

2 1

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` HDSPMixer 1.11 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes with ABSOLUTELY NO WARRANTY HDSPMixer is free software, see the file COPYING for details Looking for RME cards: Card 0: RME AIO Pro S/N 24102214 at 0x43d00000 irq 16 RME AIO found! Card 1: HDA Intel PCH at 0x44710000 irq 214 Card 2: HDA NVidia at 0x43080000 irq 17 1 RME cards card found. Restoring last presets used Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/snaut/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) bt #0 handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 #1 0x00007ffff7c5c791 in send_handlers () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:935 #2 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1620 #3 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1384 #4 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #5 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #6 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #7 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #8 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #9 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #10 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #11 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #12 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #13 0x00007ffff7d017b2 in sync_done () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1389 #14 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #15 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #16 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #17 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #18 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #19 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #20 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #21 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #22 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #23 0x00007ffff7d03561 in Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1426 #24 Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1406 #25 0x00007ffff7ca6da7 in Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:605 #26 Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:600 #27 0x00007ffff7d03d14 in Fl_Wayland_Screen_Driver::get_system_colors () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1708 #28 0x00007ffff7cec929 in Fl_Window::show () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_arg.cxx:290 #29 0x000055555557b8aa in Fl_Double_Window::show () at /usr/include/FL/Fl_Double_Window.H:36 #30 main () at /home/kenoby/dev/alsa-tools/hdspmixer/src/hdspmixer.cxx:120 ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/snaut/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) bt #0 handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 #1 0x00007ffff7c5c791 in send_handlers () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:935 #2 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1620 #3 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1384 #4 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #5 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #6 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #7 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #8 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #9 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #10 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #11 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #12 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #13 0x00007ffff7d017b2 in sync_done () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1389 #14 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #15 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #16 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #17 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #18 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #19 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #20 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #21 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #22 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #23 0x00007ffff7d03561 in Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1426 #24 Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1406 #25 0x00007ffff7ca6da7 in Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:605 #26 Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:600 #27 0x00007ffff7d03d14 in Fl_Wayland_Screen_Driver::get_system_colors () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1708 #28 0x00007ffff7cec929 in Fl_Window::show () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_arg.cxx:290 #29 0x000055555557b8aa in Fl_Double_Window::show () at /usr/include/FL/Fl_Double_Window.H:36 #30 main () at /home/kenoby/dev/alsa-tools/hdspmixer/src/hdspmixer.cxx:120 ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) bt #0 handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 #1 0x00007ffff7c5c791 in send_handlers () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:935 #2 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1620 #3 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1384 #4 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #5 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #6 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #7 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #8 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #9 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #10 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #11 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #12 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #13 0x00007ffff7d017b2 in sync_done () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1389 #14 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #15 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #16 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #17 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #18 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #19 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #20 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #21 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #22 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #23 0x00007ffff7d03561 in Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1426 #24 Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1406 #25 0x00007ffff7ca6da7 in Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:605 #26 Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:600 #27 0x00007ffff7d03d14 in Fl_Wayland_Screen_Driver::get_system_colors () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1708 #28 0x00007ffff7cec929 in Fl_Window::show () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_arg.cxx:290 #29 0x000055555557b8aa in Fl_Double_Window::show () at /usr/include/FL/Fl_Double_Window.H:36 #30 main () at /home/kenoby/dev/alsa-tools/hdspmixer/src/hdspmixer.cxx:120 ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` 278 if (fl_win->label()) { (gdb) bt #0 handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 #1 0x00007ffff7c5c791 in send_handlers () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:935 #2 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1620 #3 Fl::handle_ () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1384 #4 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #5 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #6 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #7 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #8 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #9 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #10 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #11 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #12 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #13 0x00007ffff7d017b2 in sync_done () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1389 #14 0x00007ffff6dfb2ee in ffi_call_unix64 () at ../src/x86/unix64.S:104 #15 0x00007ffff6df79f8 in ffi_call_int () at ../src/x86/ffi64.c:676 #16 0x00007ffff6dfa6ce in ffi_call () at ../src/x86/ffi64.c:713 #17 0x00007ffff732d47a in wl_closure_invoke () at ../wayland-1.24.0/src/connection.c:1241 #18 0x00007ffff732e2e9 in dispatch_event () at ../wayland-1.24.0/src/wayland-client.c:1707 #19 0x00007ffff732e6bb in dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:1853 #20 wl_display_dispatch_queue_pending () at ../wayland-1.24.0/src/wayland-client.c:2190 #21 0x00007ffff7332776 in wl_display_dispatch_queue_timeout () at ../wayland-1.24.0/src/wayland-client.c:2096 #22 0x00007ffff7332860 in wl_display_dispatch_queue () at ../wayland-1.24.0/src/wayland-client.c:2163 #23 0x00007ffff7d03561 in Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1426 #24 Fl_Wayland_Screen_Driver::open_display_platform () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1406 #25 0x00007ffff7ca6da7 in Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:605 #26 Fl_Screen_Driver::open_display () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:600 #27 0x00007ffff7d03d14 in Fl_Wayland_Screen_Driver::get_system_colors () at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1708 #28 0x00007ffff7cec929 in Fl_Window::show () at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_arg.cxx:290 #29 0x000055555557b8aa in Fl_Double_Window::show () at /usr/include/FL/Fl_Double_Window.H:36 #30 main () at /home/kenoby/dev/alsa-tools/hdspmixer/src/hdspmixer.cxx:120 ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/snaut/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: ``` Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/kenoby/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) ``` Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - edited 01 Dec '25

01 Dec '25

alsa-project/alsa-tools pull request #42 was edited from cheldt: Fixes following error: Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/snaut/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 Related [Issue](https://github.com/alsa-project/alsa-tools/issues/43) Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

HDSPmixer coredumps on Arch Linux
by GitHub issues - opened 30 Nov '25

30 Nov '25

alsa-project/alsa-tools issue #43 was opened from rubin55: This is recent, I suspect after fltk upgrade to 1.4.x november the 25th. Here is the bt: ``` HDSPMixer 1.11 - Copyright (C) 2003 Thomas Charbonnel <thomas(a)undata.org> This program comes with ABSOLUTELY NO WARRANTY HDSPMixer is free software, see the file COPYING for details Looking for RME cards: Card 0: HDA Intel PCH at 0x4802120000 irq 201 Card 1: RME AIO Pro S/N 24093311 at 0x82500000 irq 16 RME AIO found! Card 2: Logi 4K Stream Edition at usb-0000:00:14.0-5, super speed Card 3: HDA NVidia at 0x82080000 irq 17 1 RME cards card found. Initializing default presets Program received signal SIGSEGV, Segmentation fault. Downloading 47.21 K source file /usr/src/debug/alsa-tools/alsa-tools-1.2.14/hdspmixer/src/HDSPMixerWindow.cxx handler_cb (event=24) at /usr/src/debug/alsa-tools/alsa-tools-1.2.14/hdspmixer/src/HDSPMixerWindow.cxx:278 278 if (fl_win->label()) { (gdb) bt #0 handler_cb (event=24) at /usr/src/debug/alsa-tools/alsa-tools-1.2.14/hdspmixer/src/HDSPMixerWindow.cxx:278 #1 0x00007ffff7c69009 in send_handlers (e=24) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:935 #2 Fl::handle_ (e=<optimized out>, window=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1620 #3 Fl::handle_ (e=<optimized out>, window=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl.cxx:1384 #4 0x00007ffff6b9cac6 in ffi_call_unix64 () at ../src/x86/unix64.S:104 #5 0x00007ffff6b9976b in ffi_call_int (cif=cif@entry=0x7fffffffc9e0, fn=fn@entry=0x7ffff7cfcd40 <output_done(void*, wl_output*)>, rvalue=<optimized out>, rvalue@entry=0x0, avalue=avalue@entry=0x7fffffffcab0, closure=closure@entry=0x0) at ../src/x86/ffi64.c:676 #6 0x00007ffff6b9c06e in ffi_call (cif=cif@entry=0x7fffffffc9e0, fn=0x7ffff7cfcd40 <output_done(void*, wl_output*)>, rvalue=rvalue@entry=0x0, avalue=avalue@entry=0x7fffffffcab0) at ../src/x86/ffi64.c:713 #7 0x00007ffff763c48d in wl_closure_invoke (closure=closure@entry=0x55555596e0f0, target=<optimized out>, target@entry=0x555555971060, opcode=opcode@entry=2, data=<optimized out>, flags=1) at ../wayland-1.24.0/src/connection.c:1241 #8 0x00007ffff763d2e9 in dispatch_event (display=display@entry=0x5555555a9b70, queue=queue@entry=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:1707 #9 0x00007ffff763d6f3 in dispatch_queue (display=0x5555555a9b70, queue=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:1853 #10 wl_display_dispatch_queue_pending (display=display@entry=0x5555555a9b70, queue=queue@entry=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:2190 #11 0x00007ffff7640f34 in wl_display_dispatch_queue_timeout (display=0x5555555a9b70, queue=0x5555555a9c68, timeout=timeout@entry=0x0) at ../wayland-1.24.0/src/wayland-client.c:2096 #12 0x00007ffff7641010 in wl_display_dispatch_queue (display=<optimized out>, queue=<optimized out>) at ../wayland-1.24.0/src/wayland-client.c:2163 #13 0x00007ffff7cf9aba in sync_done (data=<optimized out>, cb=<optimized out>, time=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1389 #14 0x00007ffff6b9cac6 in ffi_call_unix64 () at ../src/x86/unix64.S:104 #15 0x00007ffff6b9976b in ffi_call_int (cif=cif@entry=0x7fffffffced0, fn=fn@entry=0x7ffff7cf9a60 <sync_done(void*, wl_callback*, uint32_t)>, rvalue=<optimized out>, rvalue@entry=0x0, avalue=avalue@entry=0x7fffffffcfa0, closure=closure@entry=0x0) at ../src/x86/ffi64.c:676 #16 0x00007ffff6b9c06e in ffi_call (cif=cif@entry=0x7fffffffced0, fn=0x7ffff7cf9a60 <sync_done(void*, wl_callback*, uint32_t)>, rvalue=rvalue@entry=0x0, avalue=avalue@entry=0x7fffffffcfa0) at ../src/x86/ffi64.c:713 #17 0x00007ffff763c48d in wl_closure_invoke (closure=closure@entry=0x555555970da0, target=<optimized out>, target@entry=0x55555596e1d0, opcode=opcode@entry=0, data=<optimized out>, flags=1) at ../wayland-1.24.0/src/connection.c:1241 #18 0x00007ffff763d2e9 in dispatch_event (display=display@entry=0x5555555a9b70, queue=queue@entry=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:1707 #19 0x00007ffff763d6f3 in dispatch_queue (display=0x5555555a9b70, queue=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:1853 #20 wl_display_dispatch_queue_pending (display=display@entry=0x5555555a9b70, queue=queue@entry=0x5555555a9c68) at ../wayland-1.24.0/src/wayland-client.c:2190 #21 0x00007ffff7640f34 in wl_display_dispatch_queue_timeout (display=0x5555555a9b70, queue=0x5555555a9c68, timeout=timeout@entry=0x0) at ../wayland-1.24.0/src/wayland-client.c:2096 --Type <RET> for more, q to quit, c to continue without paging-- #22 0x00007ffff7641010 in wl_display_dispatch_queue (display=<optimized out>, queue=<optimized out>) at ../wayland-1.24.0/src/wayland-client.c:2163 #23 0x00007ffff7cfa169 in Fl_Wayland_Screen_Driver::open_display_platform (this=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1426 #24 Fl_Wayland_Screen_Driver::open_display_platform (this=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1406 #25 0x00007ffff7cacafa in Fl_Screen_Driver::open_display (this=0x55555596df60) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:605 #26 Fl_Screen_Driver::open_display (this=0x55555596df60) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_Screen_Driver.cxx:600 #27 0x00007ffff7cfd942 in Fl_Wayland_Screen_Driver::get_system_colors (this=<optimized out>) at /usr/src/debug/fltk/fltk-release-1.4.4/src/drivers/Wayland/Fl_Wayland_Screen_Driver.cxx:1708 #28 0x00007ffff7ce301e in Fl_Window::show (this=0x5555555c75a0, argc=1, argv=0x7fffffffd558) at /usr/src/debug/fltk/fltk-release-1.4.4/src/Fl_arg.cxx:290 #29 0x000055555555aa17 in Fl_Double_Window::show (this=<optimized out>, a=1, b=0x7fffffffd558) at /usr/include/FL/Fl_Double_Window.H:36 #30 main (argc=1, argv=0x7fffffffd558) at /usr/src/debug/alsa-tools/alsa-tools-1.2.14/hdspmixer/src/hdspmixer.cxx:120 ``` Issue URL : https://github.com/alsa-project/alsa-tools/issues/43 Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Creative ZXR channels and volume messed up
by GitHub issues - edited 30 Nov '25

30 Nov '25

alsa-project/alsa-lib issue #484 was edited from CommonDirt: posted this over @ Pipewire, but i'm guessing this is an ALSA issue https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4996 something's changed with the CA0132 firmware, because previously in alsamixer the 10 band EQ was not exposed <img width="899" height="439" alt="Image" src="https://github.com/user-attachments/assets/755ecd00-b59a-47fd-b5b0-6fa5456f…" /> Issue URL : https://github.com/alsa-project/alsa-lib/issues/484 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Creative ZXR channels and volume messed up
by GitHub issues - opened 30 Nov '25

30 Nov '25

alsa-project/alsa-lib issue #484 was opened from CommonDirt: posted this over @ Pipewire, but i'm guessing this is an ALSA issue https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4996 Issue URL : https://github.com/alsa-project/alsa-lib/issues/484 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

hdspmixer: fix null pointer dereference in HDSPMixerWindow:handler_cb
by GitHub pull_request - opened 30 Nov '25

30 Nov '25

alsa-project/alsa-tools pull request #42 was opened from cheldt: Fixes following error: Program received signal SIGSEGV, Segmentation fault. handler_cb () at /home/snaut/dev/alsa-tools/hdspmixer/src/HDSPMixerWindow.cxx:278 Request URL : https://github.com/alsa-project/alsa-tools/pull/42 Patch URL : https://github.com/alsa-project/alsa-tools/pull/42.patch Repository URL: https://github.com/alsa-project/alsa-tools

1 0

Roland OCTA-CAPTURE (0582:0120) - White Noise/Static on Playback
by Niklas Aldervall 29 Nov '25

29 Nov '25

# ALSA Bug Report: Roland OCTA-CAPTURE (0582:0120) - White Noise/Static on Playback ## Summary Roland OCTA-CAPTURE USB audio interface produces white noise/static instead of clean audio on Linux, despite being detected and initialized by snd_usb_audio driver. ## Device Information - **Model**: Roland OCTA-CAPTURE (UA-1010) - **USB ID**: 0582:0120 - **Manufacturer**: Roland Corporation - **Device Class**: Vendor Specific (0xFF) - NOT USB Audio Class compliant - **Serial**: ISRL4B236F81 ## System Information - **Kernel**: 6.17.9-2-cachyos - **Distribution**: Arch Linux (CachyOS) - **Audio Server**: PipeWire 1.2.7 - **ALSA Driver**: snd_usb_audio (built as module) ## Symptoms 1. Device is correctly detected and recognized by ALSA 2. ALSA shows proper capabilities: S32_LE, 10ch playback, 12ch capture, 44100Hz 3. Audio streams can be sent to device without errors 4. **OUTPUT**: Only white noise/static is produced instead of clean audio 5. Issue occurs with both direct ALSA (speaker-test) and PipeWire 6. Issue persists across all available profiles (multichannel, pro-audio) ## Device Detection (lsusb) ``` Bus 001 Device 008: ID 0582:0120 Roland Corp. OCTA-CAPTURE bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 idVendor 0x0582 Roland Corp. idProduct 0x0120 OCTA-CAPTURE ``` ## ALSA Stream Configuration ``` Roland OCTA-CAPTURE at usb-0000:00:14.0-3, high speed : USB Audio Playback: Status: Stop Interface 0 Altset 1 Format: S32_LE Channels: 10 Endpoint: 0x05 (5 OUT) (ASYNC) Rates: 44100 Data packet interval: 125 us Bits: 0 Sync Endpoint: 0x85 (5 IN) Sync EP Interface: 1 Sync EP Altset: 1 Implicit Feedback Mode: Yes Capture: Status: Stop Interface 1 Altset 1 Format: S32_LE Channels: 12 Endpoint: 0x85 (5 IN) (ASYNC) Rates: 44100 Data packet interval: 125 us Bits: 0 Sync Endpoint: 0x05 (5 OUT) Sync EP Interface: 0 Sync EP Altset: 1 Implicit Feedback Mode: Yes ``` ## Hardware Configuration - **Digital Input**: OFF (changed from AUTO - no effect) - **Sample Rate**: 44.1kHz (hardware configured) - **Output**: Connected to OUT 1/2 (analog) - **Routing**: Factory reset performed - no effect ## Workarounds Attempted (All Failed) 1. **PipeWire profiles**: Tried pro-audio, multichannel-duplex 2. **Module parameters** (all individually and in combination): - `lowlatency=0` - Disable low-latency mode - `autoclock=0` - Force manual clock - `implicit_fb=0` - Disable implicit feedback - `device_setup=0,1` - Different device modes - `quirk_flags=0x8,0x20` - Various quirk flags 3. **Hardware configuration**: - DIGITAL = OFF (vs AUTO) - Factory reset via hardware buttons 4. **Different audio paths**: - Direct ALSA (hw:1,0) - PipeWire - All produce same static ## Test Command Used ```bash speaker-test -D hw:1,0 -c 10 -t sine -f 440 -r 44100 -F S32_LE -l 3 ``` **Result**: White noise/static instead of clean 440Hz tone ## Kernel Support Status - Device added to implicit feedback skip list in kernel 5.11.9 (March 2021) - Patch: `IMPLICIT_FB_SKIP_DEV(0x0582, 0x0120)` - No additional patches found for kernels 6.x ## Related Devices - Roland Quad-Capture (0582:0118) - Similar architecture, better Linux support - Roland U-8 (older device) - Has third-party Linux tools (u8ctl) ## Known Issues from Community - LinuxMusicians forum: "loud white noise sounds generated along with audio" - Multiple users report same symptom since 2013 - No documented solution exists ## Hypothesis The OCTA-CAPTURE uses Roland's proprietary USB protocol (vendor-specific class 0xFF). The device likely requires: 1. Proprietary initialization sequence not performed by generic snd_usb_audio 2. SysEx MIDI configuration for internal DSP/routing/clock 3. Vendor-specific control messages for proper operation Without Roland's cooperation or reverse engineering of Windows driver, proper support may be impossible. ## Request 1. Is there additional debugging we can enable to trace USB communication? 2. Could vendor-specific initialization be missing? 3. Would Roland device reverse engineering resources help? 4. Should this device be blacklisted if it cannot work properly? ## Additional Resources - Device manual: https://static.roland.com/assets/media/pdf/OCTA-CAPTURE_v1.6_OM.pdf - Kernel patch (2021): https://patchwork.kernel.org/project/alsa-devel/patch/CAOsVg8psuOkQRoccDs7A… - LinuxMusicians thread: https://linuxmusicians.com/viewtopic.php?t=17423 ## Contact Best Regards Niklas Aldervall

2 1

[PATCH v5] platform/x86: serial-multi-instantiate: Add IRQ_RESOURCE_OPT for IRQ missing projects
by Baojun Xu 28 Nov '25

28 Nov '25

The tas2781-hda supports multi-projects. In some projects, GpioInt() was dropped due to no IRQ connection. See the example code below: Device (SPKR) { Name (_ADR, One) Name (_HID, "TXNW2781") Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x0038, ...) I2cSerialBusV2 (0x0039, ...) // GpioInt (Edge, ...) { 0x0000 } //"GpioInt (...) {}" was commented out due to no IRQ connection. }) Return (RBUF) } } But in smi_i2c_probe(), smi_spi_probe() (serial-multi-instantiate.c), if looking for IRQ by smi_get_irq() fails, it will return an error, will not add new device, and cause smi_probe() to fail: [ 2.356546] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: IRQ index 0 not found [ 2.356561] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: Error requesting irq at index 0 So, we need to add an exception case for these situations. BTW, this patch will take effect on both I2C and SPI devices. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v5: - Change the description for this patch, remove cover letter. v4: - Change the description for this patch. v3: - Add IRQ_RESOURCE_OPT for IRQ missing cases. v2: - Remove error ignore, change to AUTO compatible with NONE. --- drivers/platform/x86/serial-multi-instantiate.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index db030b0f176a..1a369334f9cb 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -22,6 +22,7 @@ #define IRQ_RESOURCE_GPIO 1 #define IRQ_RESOURCE_APIC 2 #define IRQ_RESOURCE_AUTO 3 +#define IRQ_RESOURCE_OPT BIT(2) enum smi_bus_type { SMI_I2C, @@ -64,6 +65,10 @@ static int smi_get_irq(struct platform_device *pdev, struct acpi_device *adev, dev_dbg(&pdev->dev, "Using platform irq\n"); break; } + if (inst->flags & IRQ_RESOURCE_OPT) { + dev_dbg(&pdev->dev, "No irq\n"); + return 0; + } break; case IRQ_RESOURCE_GPIO: ret = acpi_dev_gpio_irq_get(adev, inst->irq_idx); @@ -386,10 +391,10 @@ static const struct smi_node cs35l57_hda = { static const struct smi_node tas2781_hda = { .instances = { - { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, - { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, - { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, - { "tas2781-hda", IRQ_RESOURCE_AUTO, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO | IRQ_RESOURCE_OPT, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO | IRQ_RESOURCE_OPT, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO | IRQ_RESOURCE_OPT, 0 }, + { "tas2781-hda", IRQ_RESOURCE_AUTO | IRQ_RESOURCE_OPT, 0 }, {} }, .bus_type = SMI_AUTO_DETECT, -- 2.25.1

4 3

[GIT PULL] ASoC fixes for v6.18-rc7
by Mark Brown 27 Nov '25

27 Nov '25

The following changes since commit 6a23ae0a96a600d1d12557add110e0bb6e32730c: Linux 6.18-rc6 (2025-11-16 14:25:38 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc7 for you to fetch changes up to 73b97d46dde64fa184d47865d4a532d818c3a007: ASoC: fsl_xcvr: clear the channel status control memory (2025-11-26 13:24:19 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A small pile of driver specific fixes that came in during the past few weeks, none of them especially major. ---------------------------------------------------------------- Baojun Xu (1): ASoC: tas2781: Correct the wrong chip ID for reset variable check Biju Das (2): ASoC: renesas: rz-ssi: Fix channel swap issue in full duplex mode ASoC: renesas: rz-ssi: Fix rz_ssi_priv::hw_params_cache::sample_width Ma Ke (2): ASoC: codecs: wcd937x: Fix error handling in wcd937x codec driver ASoC: codecs: Fix error handling in pm4125 audio codec driver Shenghao Ding (1): ASoC: tas2781: correct the wrong period Shengjiu Wang (1): ASoC: fsl_xcvr: clear the channel status control memory Srinivas Kandagatla (1): ASoC: codecs: wcd934x: add explicit soundwire depenency sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/pm4125.c | 21 ++++++++++-- sound/soc/codecs/tas2781-comlib-i2c.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/codecs/wcd937x.c | 43 ++++++++++++++++++----- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/renesas/rz-ssi.c | 64 +++++++++++++++++++++++++---------- 7 files changed, 103 insertions(+), 32 deletions(-)

2 2

[PATCH v2] platform/x86: serial-multi-instantiate: IRQ_RESOURCE_AUTO should be compatible with IRQ_RESOURCE_NONE
by Baojun Xu 27 Nov '25

27 Nov '25

The tas2781-hda supports multi-projects, in some projects, no irq is required, so the IRQ_RESOURCE_AUTO should be compatible with IRQ_RESOURCE_NONE. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v2: - Remove error ignore, change to AUTO compatible with NONE. --- drivers/platform/x86/serial-multi-instantiate.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index db030b0f176a..eb3fffbc29ad 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -64,7 +64,8 @@ static int smi_get_irq(struct platform_device *pdev, struct acpi_device *adev, dev_dbg(&pdev->dev, "Using platform irq\n"); break; } - break; + dev_dbg(&pdev->dev, "No irq\n"); + return 0; case IRQ_RESOURCE_GPIO: ret = acpi_dev_gpio_irq_get(adev, inst->irq_idx); break; -- 2.25.1

3 2

Lenovo Yoga 7 2-in-1 16AKP10, Realtek ALC287 codec – Bottom speakers not responding to volume control in linux
by Titas Valantinas 27 Nov '25

27 Nov '25

different outputs of aplay, arecord, codec, dmesg, lspci commands. Hardware: - Laptop: Lenovo Yoga 7 2-in-1 16AKP10 - Audio: Intel HDA (Realtek ALC287 codec, quad speaker setup: 2 top + 2 bottom) Software: - Distribution: [CachyOS] - Kernel: [6.17.0-rc5-2-cachyos-rc and 6.16.6-3 (same result in both kernel versions)] - ALSA version: [Advanced Linux Sound Architecture Driver Version k6.17.0-rc5-2-cachyos-rc. (got with command cat /proc/asound/version)] - Desktop environment: [KDE Wayland] Problem: When adjusting system volume, only the top speakers change their volume The bottom speakers keep playing at full volume regardless of system settings. (expect when volume goes to 0 or mutes then all speakers mute) Steps tried: - Tested with PipeWire - Checked `alsamixer`: only one speaker group responds to volume changes. - In Windows 11, all four speakers respond correctly → confirms hardware works. - Updated BIOS, firmware, and kernel → issue persists. Expected result: - All four speakers (top + bottom) should respond to system volume control. Actual result: - Only the top speakers change volume. Bottom speakers remain loud at all times. alsa-info.sh result: http://alsa-project.org/db/?f=1e9f3782e184d05a0ef932305d6ac04bacbd51ad sudo dmidecode -s system-product-name sudo dmidecode -s system-version sudo dmidecode -s system-manufacturer results: 83JU Yoga 7 2-in-1 16AKP10 LENOVO dmesg | grep -iE 'snd|hda' result: [ 0.199062] ACPI: \_SB_.PCI0.GPPA.HDAU.PWRS: New power resource [ 5.701995] snd_acp_pci 0000:04:00.5: enabling device (0000 -> 0002) [ 5.704462] snd_hda_intel 0000:04:00.1: enabling device (0000 -> 0002) [ 5.704483] snd_hda_intel 0000:04:00.1: Handle vga_switcheroo audio client [ 5.704544] snd_hda_intel 0000:04:00.6: enabling device (0000 -> 0002) [ 5.722776] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 5.743110] snd_hda_codec_alc269 hdaudioC1D0: ALC287: picked fixup for PCI SSID 17aa:38b5 [ 5.743523] snd_hda_codec_alc269 hdaudioC1D0: autoconfig for ALC287: line_outs=2 (0x17/0x14/0x0/0x0/0x0) type:speaker [ 5.743526] snd_hda_codec_alc269 hdaudioC1D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.743527] snd_hda_codec_alc269 hdaudioC1D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.743528] snd_hda_codec_alc269 hdaudioC1D0: mono: mono_out=0x0 [ 5.743528] snd_hda_codec_alc269 hdaudioC1D0: inputs: [ 5.743529] snd_hda_codec_alc269 hdaudioC1D0: Mic=0x19 If you need more logs, I can provide them. Thank you in advance for the help.

1 0

USB-Audio: Add jack controls for HP Thunderbolt Dock G2
by GitHub pull_request - opened 27 Nov '25

27 Nov '25

alsa-project/alsa-ucm-conf pull request #650 was opened from tatokis: Similar to the Dell WD15/WD19, this allows for jack detection (headphones/headset) on the HP Thunderbolt Dock G2. Link: https://lore.kernel.org/linux-sound/20251126003805.2705503-1-tasos@tasossah… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/650 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/650.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 0/1] platform/x86: serial-multi-instantiate: ACPI example code
by Baojun Xu 26 Nov '25

26 Nov '25

The tas2781-hda supports multi-projects, In some projects, GpioInt() was dropped due to no IRQ connection. See the example code below: But in smi_i2c_probe(), smi_spi_probe() (serial-multi-instantiate.c), if looking for IRQ by smi_get_irq() fails, it will return an error, will not add new device, and cause smi_probe() to fail: [ 2.356546] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: IRQ index 0 not found [ 2.356561] Serial bus multi instantiate pseudo device driver TXNW2781:00: error -ENXIO: Error requesting irq at index 0 So need to add an exception case for these situations. BTW, this patch will take effect on both I2C and SPI devices. Device (SPKR) { Name (_ADR, One) Name (_HID, "TXNW2781") Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x0038, ...) I2cSerialBusV2 (0x0039, ...) // GpioInt (Edge, ...) { 0x0000 } //"GpioInt (...) {}" was commented out due to no IRQ connection. }) Return (RBUF) } } Baojun Xu (1): platform/x86: serial-multi-instantiate: Add IRQ_RESOURCE_OPT for IRQ missing projects drivers/platform/x86/serial-multi-instantiate.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) -- 2.25.1

2 2

[PATCH v3 0/1] platform/x86: serial-multi-instantiate: ACPI example code
by Baojun Xu 26 Nov '25

26 Nov '25

The tas2781-hda supports multi-projects, In some projects, GpioInt was dropped due to no IRQ connection. See the example code below: But in smi_i2c_probe(), smi_spi_probe() (serial-multi-instantiate.c), if looking for IRQ by smi_get_irq() fails, it will return an error, will not add new device, and cause smi_probe to fail. So need to add an exception case for these situations. BTW, this patch will take effect on both I2C and SPI devices. Device (SPKR) { Name (_ADR, One) Name (_HID, "TXNW2781") Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { I2cSerialBusV2 (0x0038, ...) I2cSerialBusV2 (0x0039, ...) // GpioInt (Edge, ...) { 0x0000 } //"GpioInt (...) {}" was commented out due to no IRQ connection. }) Return (RBUF) } } Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (1): platform/x86: serial-multi-instantiate: Add IRQ_RESOURCE_OPT for IRQ missing projects drivers/platform/x86/serial-multi-instantiate.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) -- 2.25.1

3 3

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

25 Nov '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

3 2

[PATCH v1] ASoC: tas2781: correct the wrong period
by Shenghao Ding 24 Nov '25

24 Nov '25

A wrong preiod at the end of the sentence was reported by one of my customers. Their thorough code review is greatly appreciated. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 0a86964dfcfd..d1c76ab0144d 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -1400,7 +1400,7 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) /* * Alloc kcontrol via devm_kzalloc(), which don't manually - * free the kcontrol。 + * free the kcontrol. */ cali_ctrls = devm_kcalloc(priv->dev, nctrls, sizeof(cali_ctrls[0]), GFP_KERNEL); -- 2.43.0

2 1

[PATCH v1] ASoC: tas2781: Correct the wrong chip ID for reset variable check
by Baojun Xu 24 Nov '25

24 Nov '25

The new variable of reset was added for TAS58XX on TAS5825 first. And TAS5802/5815... was added later, so this reset variable check should be changed to lowest chip of TAS58XX. Fixes: 53a3c6e22283 ("ASoC: tas2781: Support more newly-released amplifiers tas58xx in the driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/soc/codecs/tas2781-comlib-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-comlib-i2c.c b/sound/soc/codecs/tas2781-comlib-i2c.c index b3fd7350143b..e24d56a14cfd 100644 --- a/sound/soc/codecs/tas2781-comlib-i2c.c +++ b/sound/soc/codecs/tas2781-comlib-i2c.c @@ -320,7 +320,7 @@ void tasdevice_reset(struct tasdevice_priv *tas_dev) for (i = 0; i < tas_dev->ndev; i++) { ret = tasdevice_dev_write(tas_dev, i, TASDEVICE_REG_SWRESET, - tas_dev->chip_id >= TAS5825 ? + tas_dev->chip_id >= TAS5802 ? TAS5825_REG_SWRESET_RESET : TASDEVICE_REG_SWRESET_RESET); if (ret < 0) -- 2.25.1

1 1

configure: Allow systemd service installation without systemd.pc
by GitHub pull_request - opened 24 Nov '25

24 Nov '25

alsa-project/alsa-utils pull request #309 was opened from fossdd: If the systemdsystemunitdir got passed anyway as a variable, we don't need the systemd.pc dependency anymore and allow building without a systemd build-dependency. Relevant in Alpine Linux, where we allowed systemd services to be (sub-)packaged (e.g. for downstreams like postmarketOS), but don't have systemd pacakged in Alpine itself, yet. Request URL : https://github.com/alsa-project/alsa-utils/pull/309 Patch URL : https://github.com/alsa-project/alsa-utils/pull/309.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] ALSA: hda/tas2781: Ignore irq for I2C device
by Baojun Xu 24 Nov '25

24 Nov '25

In some projects (HP Lampass), no GPIO was assigned for irq, it would be better to ignore it. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- drivers/platform/x86/serial-multi-instantiate.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index db030b0f176a..86cecf3ed916 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -214,8 +214,6 @@ static int smi_i2c_probe(struct platform_device *pdev, struct smi *smi, board_info.dev_name = name; ret = smi_get_irq(pdev, adev, &inst_array[i]); - if (ret < 0) - goto error; board_info.irq = ret; smi->i2c_devs[i] = i2c_acpi_new_device(dev, i, &board_info); -- 2.25.1

1 0

rt713: add mic led support
by GitHub pull_request - opened 24 Nov '25

24 Nov '25

alsa-project/alsa-ucm-conf pull request #649 was opened from shumingfan: rt713: add mic led support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/649 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/649.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Scarlett Solo gen 3, Direct Scarlett profile is missing, "Input 1 Mic" option is missing too.
by GitHub issues - opened 23 Nov '25

23 Nov '25

alsa-project/alsa-ucm-conf issue #648 was opened from scaledteam: Hello! I just discovered that Alsa UCM might be cause of my issues, but i don't know how to confirm and diagnose that. So i'm asking for help. When i was using Arch Linux, it allowed me to select mono microphone input. It's important feature, because many communication programs just average microphone and instrument input, resulting in loosing half of the volume, which cannot be compensated with 200% gain in pavucontrol. Now i switched to Linux Mint and i lost that feature. But ucm files in /usr/share/alsa seems identical to archlinux. Just to be sure i tried Manjaro liveusb and i found all of this options, no config required. There is alsa info just in case: [alsa-info.txt](https://github.com/user-attachments/files/23697024/alsa-info… Can someone help me with this? Linux mint: <img width="700" height="421" alt="Image" src="https://github.com/user-attachments/assets/c555d380-bb2b-4fcf-b835-d0c92ce5…" /> Manjaro: ![Image](https://github.com/user-attachments/assets/e2ecae99-0fb0-4203-82f2-b9c4c97a733f) ![Image](https://github.com/user-attachments/assets/5c277b7a-f19c-4e0b-a786-b51eb51ee5f9) Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/648 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add MSI MPG B850I Edge TI Wifi motherboard support for ALC4080
by GitHub issues - opened 23 Nov '25

23 Nov '25

alsa-project/alsa-ucm-conf issue #647 was opened from YourNumNums: The MSI MPG B850I Edge TI Wifi motherboard only plays sound out of the rear i/o panel auxiliary port. I do not see any way to activate the rear optical output or the front auxiliary port. Here is my [alsa-info](https://alsa-project.org/db/?f=67f95e30cfe710cefa38544c4b5dd6c1b…, and the device ID is below: ``` $ lsusb | grep Audio Bus 001 Device 004: ID 0db0:d4fa Micro Star International USB Audio ``` This looks like a similar issue to [issue 172](https://github.com/alsa-project/alsa-ucm-conf/issues/172), except this time `d4fa` needs to be added to the 0db0 regex for ucm2/USB-Audio/USB-Audio.conf for ALC4080 support. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/647 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

BUGREP No hardware speaker volume control on ALC287 (Lenovo Yoga, Ubuntu 25.10): master/sink volume has no effect
by Sergii O. Uriupin 22 Nov '25

22 Nov '25

Hardware: - Laptop: Lenovo Yoga (AMD APU with ALC287) - Audio: Family 17h/19h/1ah HD Audio Controller, codec Realtek ALC287 - OS: Ubuntu 25.10 (kernel 6.17.0-6-generic) - Audio stack: PulseAudio 17.0 (PipeWire audio removed) Problem: - System master/sink volume (1–100%) does not change actual speaker loudness. - Only slight change >100% (as if a high-frequency path kicks in). - Actual audible volume changes only when per-stream volume is adjusted (e.g., `pactl set-sink-input-volume <id> ...`). - In ALSA controls there is no usable “Speaker Playback Volume”, only switches; so PA/desktop slider has no effect on internal speakers. - Reproducible after clean boot, with PulseAudio as the active server (pipewire-audio removed). Data: - ALSA info: <ALSA_INFO_URL> - `amixer -c1 controls`: <paste output> - `amixer -c1 contents`: <paste output> - `pactl info`: <paste output> - `pactl list sinks short`: <paste output> - `pactl list sink-inputs short`: <paste output> Expected: - Master/sink volume should control speaker loudness. Actual: - Master/sink volume does nothing; only per-stream volume changes affect sound. Notes: - UCM profile (HDA/HiFi) points speaker to non-existent “Speaker Playback Volume”; mapping to Bass Speaker volume locally (user UCM override) still yields minimal effect. Needs correct UCM/driver mapping for ALC287 on this platform. s@ubunt:~/soft/Lenovo$ sudo apt install alsa-utils # Full ALSA dump (will upload and give a URL) alsa-info.sh --upload # Mixer controls amixer -c1 controls amixer -c1 contents # PulseAudio state pactl info pactl list sinks short pactl list sink-inputs short [sudo: authenticate] Password: Уже установлен пакет alsa-utils самой новой версии (1.2.14-1ubuntu1). alsa-utils помечен как установленный вручную. Сводка: Обновление: 0, Установка: 0, Удаление: 0, Пропуск обновления: 4 alsa-info.sh: команда не найдена numid=15,iface=CARD,name='Headphone Jack' numid=12,iface=CARD,name='Mic Jack' numid=13,iface=CARD,name='Speaker Front Phantom Jack' numid=14,iface=CARD,name='Speaker Surround Phantom Jack' numid=11,iface=MIXER,name='Master Playback Switch' numid=10,iface=MIXER,name='Master Playback Volume' numid=5,iface=MIXER,name='Headphone Playback Switch' numid=4,iface=MIXER,name='Headphone Playback Volume' numid=19,iface=MIXER,name='Mic ACP LED Capture Switch' numid=9,iface=MIXER,name='Mic Boost Volume' numid=8,iface=MIXER,name='Capture Switch' numid=7,iface=MIXER,name='Capture Volume' numid=6,iface=MIXER,name='Auto-Mute Mode' numid=3,iface=MIXER,name='Bass Speaker Playback Switch' numid=2,iface=MIXER,name='Bass Speaker Playback Volume' numid=1,iface=MIXER,name='Speaker Playback Switch' numid=17,iface=PCM,name='Capture Channel Map' numid=16,iface=PCM,name='Playback Channel Map' numid=15,iface=CARD,name='Headphone Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=12,iface=CARD,name='Mic Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off numid=13,iface=CARD,name='Speaker Front Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=14,iface=CARD,name='Speaker Surround Phantom Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on numid=11,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=10,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=1,min=0,max=87,step=0 : values=87 numid=5,iface=MIXER,name='Headphone Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=off,off numid=4,iface=MIXER,name='Headphone Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=52,52 | dBscale-min=-65.25dB,step=0.75dB,mute=0 numid=19,iface=MIXER,name='Mic ACP LED Capture Switch' ; type=BOOLEAN,access=rw----W-,values=1 : values=off numid=9,iface=MIXER,name='Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=0,0 | dBscale-min=0.00dB,step=10.00dB,mute=0 numid=8,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=7,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=63,step=0 : values=38,38 | dBscale-min=-17.25dB,step=0.75dB,mute=0 numid=6,iface=MIXER,name='Auto-Mute Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Disabled' ; Item #1 'Enabled' : values=0 numid=3,iface=MIXER,name='Bass Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=2,iface=MIXER,name='Bass Speaker Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=87,step=0 : values=87,87 | dBscale-min=-65.25dB,step=0.75dB,mute=0 numid=1,iface=MIXER,name='Speaker Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=17,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=16,iface=PCM,name='Playback Channel Map' ; type=INTEGER,access=r--v-R--,values=4,min=0,max=36,step=0 : values=3,4,0,0 | container | chmap-fixed=FL,FR | chmap-fixed=FL,FR,LFE,LFE Строка сервера: /run/user/1000/pulse/native Версия протокола библиотеки: 35 Версия протокола сервера: 35 Выполняется локально: да Номер клиента: 20 Размер блока памяти: 65472 Имя пользователя: s Имя хоста: ubunt Имя сервера: pulseaudio Версия сервера: 17.0 Спецификация отсчётов по умолчанию: s16le 2-канальный 4410 Схема каналов по умолчанию: front-left,front-right Аудиоприёмник по умолчанию: alsa_output.pci-0000_04_00.6.HiFi__Speaker__sink Источник по умолчанию: alsa_input.pci-0000_04_00.6.HiFi__Mic1__source Cookie: 8342:bbd4 0 alsa_output.pci-0000_04_00.6.HiFi__Speaker__sink module-alsa-card.c s16le 2-канальный 4800 RUNNING 7 0 17 protocol-native.c float32le 2-канальный s@ubunt:~/soft/Lenovo$

1 0

[PATCH] add the SND quirk for HP pavilion aero laptop 13z-be200
by Jacob Zhong 21 Nov '25

21 Nov '25

The laptop uses ALC287 chip (as shown in /proc/asound/card1/codec#0). It seems that every HP pavilion laptop in the table uses the same quirk, so I just copied them. I have verified that the mute LED on my laptop works with this patch. For reference, here's the alsa-info of my laptop: https://alsa-project.org/db/?f=2d5f297087708610bc01816ab12052abdd4a17c0 Signed-off-by: Jacob Zhong <cmpute(a)qq.com> --- sound/hda/codecs/realtek/alc269.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/codecs/realtek/alc269.c b/sound/hda/codecs/realtek/alc269.c index 269b6c1e3b6d..b18777c09d9c 100644 --- a/sound/hda/codecs/realtek/alc269.c +++ b/sound/hda/codecs/realtek/alc269.c @@ -6572,6 +6572,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8bc8, "HP Victus 15-fa1xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8bcd, "HP Omen 16-xd0xxx", ALC245_FIXUP_HP_MUTE_LED_V1_COEFBIT), SND_PCI_QUIRK(0x103c, 0x8bd4, "HP Victus 16-s0xxx (MB 8BD4)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT), + SND_PCI_QUIRK(0x103c, 0x8bd6, "HP Pavilion Aero Laptop 13z-be200", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bdd, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bde, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8bdf, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2), -- 2.43.0

1 1

SplitPCMDevice macro broken with two or more channels
by GitHub issues - opened 21 Nov '25

21 Nov '25

alsa-project/alsa-ucm-conf issue #646 was opened from karcsesz: I have a Behringer XR18 that I am currently trying to write a custom UCM profile for that splits its 18x18 audio interface into separate mono and stereo virtual devices. I am able to create mono devices using the `SplitPCM` and `SplitPCMDevice` macros, however only when none of the channels higher than Channel0 are defined, as the created Capture/PlaybackPCM configuration values have the channel IDs duplicated: ``` $ alsaucm -c3 dump text [snip] Verb."18x18 Stereo" { Comment "9x9 Stereo Duplex" Device."Ch1 In" { Comment "Ch1 from" Values { CaptureCTL "_ucm0001.hw:XR18" CaptureChannels 2 CapturePCM "_ucm0001.behringer_xr18_stereo_in:XR18,0,0,1,1" PlaybackCTL "_ucm0001.hw:XR18" } } Device."Ch2 In" { Comment "Ch2 from" Values { CaptureCTL "_ucm0001.hw:XR18" CaptureChannels 2 CapturePCM "_ucm0001.behringer_xr18_stereo_in:XR18,0,2,3,3" PlaybackCTL "_ucm0001.hw:XR18" } } [snip] ``` `spa-acp-tool` also fails to find the PCM devices due to this issue ``` $ spa-acp-tool -vvv -c3 [snip] I alsa-ucm.c:2567 Probing profile 18x18 Stereo I alsa-ucm.c:2571 Set ucm verb to 18x18 Stereo D alsa-util.c:714 Trying _ucm0001.behringer_xr18_stereo_in:XR18,0,16,17,17 with SND_PCM_NO_AUTO_FORMAT ... I conf.c:5541 Unknown parameter 4 I conf.c:5712 Parse arguments error: No such file or directory I pcm.c:2722 Unknown PCM behringer_xr18_stereo_in:XR18,0,16,17,17 I alsa-util.c:721 Error opening PCM device _ucm0001.behringer_xr18_stereo_in:XR18,0,16,17,17: No such file or directory I alsa-ucm.c:2611 Profile '18x18 Stereo' mapping '18x18 Stereo: Ch9 In: source': input PCM open failed I alsa-ucm.c:2621 Profile 18x18 Stereo not supported ``` Appropriate snippet from the verb configuration: ``` Include.pcm_split.File "/common/pcm/split.conf" Macro [ [snip] { SplitPCM { Name "behringer_xr18_stereo_in" Direction Capture Format S24_3LE Channels 2 HWChannels 18 HWChannelPos0 FL HWChannelPos1 FR HWChannelPos2 FL HWChannelPos3 FR HWChannelPos4 FL HWChannelPos5 FR HWChannelPos6 FL HWChannelPos7 FR HWChannelPos8 FL HWChannelPos9 FR HWChannelPos10 FL HWChannelPos11 FR HWChannelPos12 FL HWChannelPos13 FR HWChannelPos14 FL HWChannelPos15 FR HWChannelPos16 FL HWChannelPos17 FR } } [snip] ] [snip] SectionDevice."Ch1 In" { Comment "Ch1 from" Macro.pcm_split.SplitPCMDevice { Name "behringer_xr18_stereo_in" Direction Capture HwChannels 18 Channels 2 Channel0 0 Channel1 1 ChannelPos0 FL ChannelPos1 FR } } SectionDevice."Ch2 In" { Comment "Ch2 from" Macro.pcm_split.SplitPCMDevice { Name "behringer_xr18_stereo_in" Direction Capture HwChannels 18 Channels 2 Channel0 2 Channel1 3 ChannelPos0 FL ChannelPos1 FR } } [snip] ``` My current hunch is the append within the `SplitPCMDevice_addchn` macro somehow getting called twice (maybe due to [evaluation restart](https://www.alsa-project.org/alsa-doc/alsa-lib/group__ucm__conf.ht…) [alsa-info.txt](https://github.com/user-attachments/files/23665421/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/646 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/8] ASoC: tas2783A: sdw_utils: support ch 3 & 4
by Niranjan H Y 20 Nov '25

20 Nov '25

Currently the machine driver for tas2783A can only support 2 channels. This patch adds support for 2 channel playback with 4 device setup. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/sdw_utils/soc_sdw_ti_amp.c | 4 ++++ sound/soc/sdw_utils/soc_sdw_utils.c | 22 ++++++++++++++++++---- 2 files changed, 22 insertions(+), 4 deletions(-) diff --git a/sound/soc/sdw_utils/soc_sdw_ti_amp.c b/sound/soc/sdw_utils/soc_sdw_ti_amp.c index cbd60faec..488ef2ef4 100644 --- a/sound/soc/sdw_utils/soc_sdw_ti_amp.c +++ b/sound/soc/sdw_utils/soc_sdw_ti_amp.c @@ -58,6 +58,10 @@ int asoc_sdw_ti_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, strscpy(speaker, "Left Spk", sizeof(speaker)); } else if (!strncmp(prefix, "tas2783-2", strlen("tas2783-2"))) { strscpy(speaker, "Right Spk", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-3", strlen("tas2783-3"))) { + strscpy(speaker, "Left Spk2", sizeof(speaker)); + } else if (!strncmp(prefix, "tas2783-4", strlen("tas2783-4"))) { + strscpy(speaker, "Right Spk2", sizeof(speaker)); } else { ret = -EINVAL; dev_err(card->dev, "unhandled prefix %s", prefix); diff --git a/sound/soc/sdw_utils/soc_sdw_utils.c b/sound/soc/sdw_utils/soc_sdw_utils.c index 824fb613c..ec00f13a6 100644 --- a/sound/soc/sdw_utils/soc_sdw_utils.c +++ b/sound/soc/sdw_utils/soc_sdw_utils.c @@ -40,11 +40,25 @@ static const struct snd_soc_dapm_widget lr_spk_widgets[] = { SND_SOC_DAPM_SPK("Right Spk", NULL), }; +static const struct snd_soc_dapm_widget lr_4spk_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), + SND_SOC_DAPM_SPK("Left Spk2", NULL), + SND_SOC_DAPM_SPK("Right Spk2", NULL), +}; + static const struct snd_kcontrol_new lr_spk_controls[] = { SOC_DAPM_PIN_SWITCH("Left Spk"), SOC_DAPM_PIN_SWITCH("Right Spk"), }; +static const struct snd_kcontrol_new lr_4spk_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), + SOC_DAPM_PIN_SWITCH("Left Spk2"), + SOC_DAPM_PIN_SWITCH("Right Spk2"), +}; + static const struct snd_soc_dapm_widget rt700_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), @@ -69,10 +83,10 @@ struct asoc_sdw_codec_info codec_info_list[] = { .dailink = {SOC_SDW_AMP_OUT_DAI_ID, SOC_SDW_AMP_IN_DAI_ID}, .init = asoc_sdw_ti_amp_init, .rtd_init = asoc_sdw_ti_spk_rtd_init, - .controls = lr_spk_controls, - .num_controls = ARRAY_SIZE(lr_spk_controls), - .widgets = lr_spk_widgets, - .num_widgets = ARRAY_SIZE(lr_spk_widgets), + .controls = lr_4spk_controls, + .num_controls = ARRAY_SIZE(lr_4spk_controls), + .widgets = lr_4spk_widgets, + .num_widgets = ARRAY_SIZE(lr_4spk_widgets), }, }, .dai_num = 1, -- 2.25.1

2 8

Support Tas2781 and mute led for HP platform
by Kailang 20 Nov '25

20 Nov '25

Hi Takashi, Attach patch was supported Tas2781 and mute led for HP platform. BR, Kailang

2 2

[PATCH] ASoC: wm8994: Prevent access to invalid VU register bits on WM1811
by Sylwester Nawrocki 20 Nov '25

20 Nov '25

The ADC2 and DAC2 are not available on WM1811 device. This patch moves the ADC2, DAC2 VU bitfields to a separate array so we can skip accessing them and avoid unreadable register access on WM1811. This allows to get rid of warnings during boot like: wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5 Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/codecs/wm8994.c | 60 ++++++++++++++++++++++++++++----------- 1 file changed, 44 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..038be667c1a6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -4346,6 +4366,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, -- 2.17.1

4 3

No soundcards recognised by ALSA after upgrading to Linux 6.17.8 from Linux 6.17.1 (Lenovo Yoga Slim 7 14ILL10, cs42l43)
by GitHub issues - opened 20 Nov '25

20 Nov '25

alsa-project/alsa-ucm-conf issue #645 was opened from emfol: After upgrading to Linux 6.17.8 (from Linux 6.17.1), the sound card is no longer recognized. I ran the `alsa-info.sh` script after booting each one of the kernels and the results follow: - 6.17.1: https://alsa-project.org/db/?f=5887d318c6cd7866365a0f134eeaf5721f33b82d - 6.17.8: https://alsa-project.org/db/?f=a5ca95f07b9a1844c102f63e6c0e36cd2e5755ef #### DMESG 6.17.8 ``` Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: enabling device (0000 -> 0002) Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: bound 0000:00:02.0 (ops intel_audio_component_bind_ops [xe]) Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: use msi interrupt mode Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: hda codecs found, mask 4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: NHLT device BT(0) detected, ssp_mask 0x4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: BT link detected in NHLT tables: 0x4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DMICs detected in NHLT tables: 4 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware paths/files for ipc type 1: Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware file: intel/sof-ipc4/lnl/sof-lnl.ri Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware lib path: intel/sof-ipc4-lib/lnl Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology file: intel/sof-ipc4-tplg/sof-lnl-cs42l43-l0-4ch.tplg Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:33:46 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Booted firmware version: 2.13.0.1 ``` #### DMESG 6.17.1 ``` Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: SoundWire enabled on CannonLake+ platform, using SOF driver Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: enabling device (0000 -> 0002) Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: bound 0000:00:02.0 (ops intel_audio_component_bind_ops [xe]) Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: use msi interrupt mode Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: hda codecs found, mask 4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: NHLT device BT(0) detected, ssp_mask 0x4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: BT link detected in NHLT tables: 0x4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: DMICs detected in NHLT tables: 4 Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware paths/files for ipc type 1: Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware file: intel/sof-ipc4/lnl/sof-lnl.ri Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Firmware lib path: intel/sof-ipc4-lib/lnl Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology file: intel/sof-ipc4-tplg/sof-lnl-cs42l43-l0-4ch.tplg Nov 19 22:22:14 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Booted firmware version: 2.13.0.1 Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Topology: ABI 3:29:1 Kernel ABI 3:23:1 Nov 19 22:22:15 ys7i14-em kernel: sof_sdw sof_sdw: ASoC: Parent card not yet available, widget card binding deferred Nov 19 22:22:15 ys7i14-em kernel: sof-audio-pci-intel-lnl 0000:00:1f.3: Loaded firmware library: ADSPFW, version: 2.13.0.1 Nov 19 22:22:21 ys7i14-em kernel: sof_sdw sof_sdw: Setting CS42L43 Speaker volume limit to 128 Nov 19 22:22:21 ys7i14-em kernel: sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 3 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire Jack as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input14 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=5 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input15 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=6 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input16 Nov 19 22:22:21 ys7i14-em kernel: input: sof-soundwire HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:1f.3/sof_sdw/sound/card0/input17 Nov 19 22:33:07 ys7i14-em systemd[1]: sys-devices-pci0000:00-0000:00:1f.3-sof_sdw-sound-card0-controlC0.device: Failed to enqueue SYSTEMD_WANTS job, ignoring: Transaction for sound.target/start is destructive (reboot.target has 'start' job queued, but 'stop' is included in transaction). ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/645 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Update cs42l45 JackControls
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-ucm-conf pull request #644 was opened from mstrozek: Update JackControls to follow machine driver changes [1] [1] https://github.com/thesofproject/linux/pull/5590 Keeping this PR as a draft until the SOF PR is finalized and merged Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/644.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HDA: route speaker volume to Post Mixer Analog
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-ucm-conf pull request #643 was opened from Lyapsus: On certain laptops (e.g., Lenovo Legion Pro 7 16IAX10H), the live speaker gain is exposed through the Post Mixer Analog controls. The default UCM entries point at Speaker/Master, so software volume never touches the woofer amps. Routing the Speaker device to the Post Mixer Analog mixer and volume elements restores proper volume control without affecting other paths Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/643 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/643.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

fix typo in error.h to avoid compile error when gcc <= 2.95
by GitHub pull_request - opened 19 Nov '25

19 Nov '25

alsa-project/alsa-lib pull request #483 was opened from versaloon: a typo in include/error.h Request URL : https://github.com/alsa-project/alsa-lib/pull/483 Patch URL : https://github.com/alsa-project/alsa-lib/pull/483.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 7 14IRH8 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. I first create an issue on pipewire and they directed me here with this outcome: Which mixer controls are used for the volume is controlled by the card UCM profile, so this is probably bug in the UCM profile. Its issue tracker is here: https://github.com/alsa-project/alsa-ucm-conf/issues Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 7 14IRH8 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 7 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 18 Nov '25

18 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME/KDE slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was edited from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot any Linux distribution. I tested on Fedora (GNOME and KDE) and openSUSE Tumbleweed, but it will probably occur on other distros or desktop environments as well. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Lenovo Yoga Pro 14 built-in speakers volume not controlled by GNOME/PipeWire
by GitHub issues - opened 17 Nov '25

17 Nov '25

alsa-project/alsa-ucm-conf issue #642 was opened from 2O77: Device: Lenovo Yoga Pro 14 14IMH9 Sound card: sof-hda-dsp Problem: ALSA Post-Mixer / Pre-Mixer controls work via alsamixer, but GNOME volume slider or PipeWire sink volume does not affect hardware output. Likely an issue in the UCM profile for this device. Steps to reproduce: 1. Boot Fedora (GNOME or KDE) or openSUSE Tumbleweed. 2. Attempt to change volume via GNOME slider or `pactl` / PipeWire controls. 3. Observe that output volume of laptop speakers does not change. 4. Testing with headphones (AirPods, Beats) or external speakers works correctly. Notes: - This issue is specific to the built-in laptop speakers. - Different desktop environments (GNOME, KDE) and distributions (Fedora GNOME, Fedora KDE, openSUSE Tumbleweed) were tested and exhibit the same problem. - ALSA `Post-Mixer` / `Pre-Mixer` controls work correctly when adjusted via `alsamixer`. Expected behavior: - GNOME / PipeWire slider should affect the actual hardware output volume of the laptop speakers. Suggested area: - UCM profile for sof-hda-dsp on this Lenovo model may need adjustment. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/642 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list
by Niranjan H Y 17 Nov '25

17 Nov '25

"struct sdca_control" declares "values" field as integer array. But the memory allocated to it is of char array. This causes crash for sdca_parse_function API. This patch addresses the issue by allocating correct data size. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- v2: - use kcalloc instead of kzalloc --- sound/soc/sdca/sdca_functions.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index b2e3fab9b..49b98fe2d 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -950,7 +950,8 @@ static int find_sdca_entity_control(struct device *dev, struct sdca_entity *enti return ret; } - control->values = devm_kzalloc(dev, hweight64(control->cn_list), GFP_KERNEL); + control->values = devm_kcalloc(dev, hweight64(control->cn_list), + sizeof(int), GFP_KERNEL); if (!control->values) return -ENOMEM; -- 2.25.1

4 3

[PATCH 5.10] ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf()
by Alexandra Diupina 17 Nov '25

17 Nov '25

From: Takashi Iwai <tiwai(a)suse.de> commit 94c1ceb043c1a002de9649bb630c8e8347645982 upstream. snprintf() returns the would-be-filled size when the string overflows the given buffer size, hence using this value may result in the buffer overflow (although it's unrealistic). This patch replaces with a safer version, scnprintf() for papering over such a potential issue. Fixes: 29c8e4398f02 ("ASoC: SOF: Intel: hda: add extended rom status dump to error log") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20220801165420.25978-4-tiwai@suse.de Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Alexandra Diupina <adiupina(a)astralinux.ru> --- Backport fix for CVE-2022-50050 sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index b4cc72483137..1d879c2b81e1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -437,7 +437,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev) for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_SRAM_REG_ROM_STATUS + i * 0x4); - len += snprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); } sof_dev_dbg_or_err(sdev->dev, hda->boot_iteration == HDA_FW_BOOT_ATTEMPTS, -- 2.30.2

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` ``` $pacman -Q | grep alsa alsa-card-profiles 1:1.4.9-1 alsa-lib 1.2.14-2 alsa-plugins 1:1.2.12-5 alsa-topology-conf 1.2.5.1-4 alsa-ucm-conf 1.2.14-2 alsa-utils 1.2.14-1 lib32-alsa-lib 1.2.14-2 lib32-alsa-plugins 1.2.12-1 vlc-plugin-alsa 3.0.21-31 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - edited 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was edited from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 ``` I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd-loopback does not work
by GitHub issues - opened 17 Nov '25

17 Nov '25

alsa-project/alsa-lib issue #482 was opened from Sameguyy: ``` $pactl list short sources 80 alsa_output.platform-snd_aloop.0.analog-stereo.monitor PipeWire s32le 2ch 48000Hz RUNNING 81 alsa_input.platform-snd_aloop.0.analog-stereo PipeWire s32le 2ch 48000Hz RUNNING ``` These are two interfaces that Pipewire automatically pulled in, and the sound goes into 3.0 and doesn't leave from 3.1 That is, I play a sound on hw:3.0 and when I try to read something from hw:3.1 I get silence aplay\arecord records silence $ uname -r 6.17.8-arch1-1 Wayland PipeWire gnome ```$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC892 Analog [ALC892 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC892 Digital [ALC892 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: DACX6 [DAC-X6], device 1: USB Audio [USB Audio #1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 3: HDMI 0 [SAMSUNG] Subdevices: 0/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: NVidia [HDA NVidia], device 9: HDMI 3 [HDMI 3] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: Loopback [Loopback], device 0: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 3: Loopback [Loopback], device 1: Loopback PCM [Loopback PCM] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 I don't understand what I'm doing wrong can sm1 guess Issue URL : https://github.com/alsa-project/alsa-lib/issues/482 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.18-rc5
by Mark Brown 14 Nov '25

14 Nov '25

The following changes since commit 6146a0f1dfae5d37442a9ddcba012add260bceb0: Linux 6.18-rc4 (2025-11-02 11:28:02 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.18-rc5 for you to fetch changes up to 360b3730f8eab6c4467c6cca4cb0e30902174a63: ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() (2025-11-13 00:36:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.18 A small collection of fixes, all driver specific and none especially remarkable unless you have the hardware (many not even then). ---------------------------------------------------------------- Claudiu Beznea (1): ASoC: da7213: Use component driver suspend/resume Haotian Zhang (3): ASoC: cs4271: Fix regulator leak on probe failure ASoC: codecs: va-macro: fix resource leak in probe error path ASoC: rsnd: fix OF node reference leak in rsnd_ssiu_probe() Miaoqian Lin (1): ASoC: sdw_utils: fix device reference leak in is_sdca_endpoint_present() Niranjan H Y (2): ASoC: tas2783A: Fix issues in firmware parsing ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list Richard Fitzgerald (1): ASoC: doc: cs35l56: Update firmware filename description for B0 silicon Shenghao Ding (1): ASoC: tas2781: fix getting the wrong device number Documentation/sound/codecs/cs35l56.rst | 9 ++--- sound/soc/codecs/cs4271.c | 10 +++-- sound/soc/codecs/da7213.c | 69 ++++++++++++++++++++++------------ sound/soc/codecs/da7213.h | 1 + sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/tas2781-i2c.c | 9 ++++- sound/soc/codecs/tas2783-sdw.c | 20 +++++++++- sound/soc/renesas/rcar/ssiu.c | 3 +- sound/soc/sdca/sdca_functions.c | 3 +- sound/soc/sdw_utils/soc_sdw_utils.c | 20 +++++++--- 10 files changed, 99 insertions(+), 47 deletions(-)

2 1

[PATCH alsa-utils] amidi: Ignore inactive MIDI ports as default at listing
by Takashi Iwai 13 Nov '25

13 Nov '25

When listing the devices, currently we show all MIDI ports including inactive ones. But those inactive ports are rarely useful, and it'd be more convenient to filter them out. This patch introduces the filtering of inactive ports at listing devices via amidi -l option. When user needs to scan all MIDI ports including inactive ports, pass the new option -x in addition. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- amidi/amidi.1 | 5 +++++ amidi/amidi.c | 11 ++++++++++- 2 files changed, 15 insertions(+), 1 deletion(-) diff --git a/amidi/amidi.1 b/amidi/amidi.1 index 5bc24ba3ad54..d3d321f16558 100644 --- a/amidi/amidi.1 +++ b/amidi/amidi.1 @@ -52,6 +52,11 @@ Prints the current version. .I \-l, \-\-list\-devices Prints a list of all hardware MIDI ports. +.TP +.I \-x, \-\-list\-inactive +Use together with \fI\-l\fP option. +Print all MIDI ports including inactive ports. + .TP .I \-L, \-\-list\-rawmidis Prints all RawMIDI definitions. diff --git a/amidi/amidi.c b/amidi/amidi.c index 75fb8c0a42a3..9a2aaf7205f0 100644 --- a/amidi/amidi.c +++ b/amidi/amidi.c @@ -57,6 +57,7 @@ static int stop; static int sysex_interval; static snd_rawmidi_t *input, **inputp; static snd_rawmidi_t *output, **outputp; +static int list_all; static void error(const char *format, ...) { @@ -76,6 +77,7 @@ static void usage(void) "-h, --help this help\n" "-V, --version print current version\n" "-l, --list-devices list all hardware ports\n" + "-x, --list-inactive list inactive ports, too\n" "-L, --list-rawmidis list all RawMIDI definitions\n" "-p, --port=name select port by name\n" "-s, --send=file send the contents of a (.syx) file\n" @@ -151,6 +153,9 @@ static void list_device(snd_ctl_t *ctl, int card, int device) card, device, sub, snd_strerror(err)); return; } + if (!list_all && + (snd_rawmidi_info_get_flags(info) & SNDRV_RAWMIDI_INFO_STREAM_INACTIVE)) + continue; name = snd_rawmidi_info_get_name(info); sub_name = snd_rawmidi_info_get_subdevice_name(info); if (sub == 0 && sub_name[0] == '\0') { @@ -471,11 +476,12 @@ static void add_send_hex_data(const char *str) int main(int argc, char *argv[]) { - static const char short_options[] = "hVlLp:s:r:S::dt:aci:T:"; + static const char short_options[] = "hVlxLp:s:r:S::dt:aci:T:"; static const struct option long_options[] = { {"help", 0, NULL, 'h'}, {"version", 0, NULL, 'V'}, {"list-devices", 0, NULL, 'l'}, + {"list-inactive", 0, NULL, 'x'}, {"list-rawmidis", 0, NULL, 'L'}, {"port", 1, NULL, 'p'}, {"send", 1, NULL, 's'}, @@ -508,6 +514,9 @@ int main(int argc, char *argv[]) case 'l': do_device_list = 1; break; + case 'x': + list_all = 1; + break; case 'L': do_rawmidi_list = 1; break; -- 2.51.1

1 0

[BUG] Realtek ALC236 Internal Mic not working on HP 255 G10 (Subsystem ID 103c:8b2f)
by Pape Samba Mbodj Sarr 13 Nov '25

13 Nov '25

Bonjour, J'ai un problème avec le microphone interne (intégré) qui n'est pas détecté par le système (affiché comme "non branché" / "not available") sur mon ordinateur portable HP. J'ai épuisé toutes les options de configuration utilisateur (modèle ALSA, patchs manuels). Ceci semble être un "quirk" (correctif) manquant pour le pilote. **Informations du système :** * **Ordinateur :** HP 255 15.6 inch G10 Notebook PC * **Codec Audio :** Realtek ALC236 * **Subsystem ID :** 103c:8b2f (Hewlett-Packard Company) * **Noyau Linux :** 6.14.0-35-generic (Version très récente) **URL du rapport alsa-info.sh (Codec Dump) :**http://alsa-project.org/db/?f=55da44f60f5deb6073c9edfe2f49c3b1cb03b53e L'entrée microphone (analog-input-mic) est marquée comme "not available" (non disponible) dans pactl. Je soupçonne que le pilote ne parvient pas à initialiser le microphone numérique (DMIC) ou confond l'entrée interne avec le jack externe. Merci d'avance de votre aide pour l'intégration de ce correctif. Cordialement, SARR Mbodj

1 0

aplay: cannot play S24_LE wav file recorded with arecord
by GitHub issues - opened 13 Nov '25

13 Nov '25

alsa-project/alsa-utils issue #308 was opened from yoshihisa-yamamoto-re: My environment is alsa-utils version : 1.2.14 (Debian trixie) I tried to record and play PCM audio as S24_LE format over alsa as follows: ``` arecord -f S24_LE out.wav aplay out.wav ``` However, the audio turned into noise and did not seem to play properly. This problem does **not** seem to occur when using alsa-utils version 1.2.9 (Ubuntu 24.04 LTS). Searching through the commits, I found the closed issue https://github.com/alsa-project/alsa-utils/pull/260. From reading the source, it appears that the header information has been rewritten, so I dumped it to check, and it looks like this: hexdump of wav created with alsa-utils version 1.2.14 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 20 00 64 61 74 61 00 e2 04 00 e6 67 8d ff |.. .data.....g..| 00000030 c8 fd 8e ff 0c 8e 90 ff c6 18 92 ff 0a 9e 93 ff |................| 00000040 ec 1d 95 ff 7e 98 96 ff d2 0d 98 ff fd 7d 99 ff |....~........}..| 00000050 10 e9 9a ff 1c 4f 9c ff 35 b0 9d ff 6b 0c 9f ff |.....O..5...k...| 00000060 cf 63 a0 ff 74 b6 a1 ff 68 04 a3 ff be 4d a4 ff |.c..t...h....M..| 00000070 86 92 a5 ff cf d2 a6 ff a9 0e a8 ff 25 46 a9 ff |............%F..| ``` hexdump of wav created with alsa-utils version 1.2.9 : ``` 00000000 52 49 46 46 24 e2 04 00 57 41 56 45 66 6d 74 20 |RIFF$...WAVEfmt | 00000010 10 00 00 00 01 00 01 00 40 1f 00 00 00 7d 00 00 |........@....}..| 00000020 04 00 18 00 64 61 74 61 00 e2 04 00 17 3d fe ff |....data.....=..| 00000030 3e 3d fe ff 86 41 fe ff cb 47 fe ff 3b 46 fe ff |>=...A...G..;F..| 00000040 10 47 fe ff e3 4d fe ff 66 4f fe ff 8d 50 fe ff |.G...M..fO...P..| 00000050 be 4f fe ff 40 58 fe ff e7 5b fe ff 27 60 fe ff |.O..@X...[..'`..| 00000060 f9 60 fe ff ce 61 fe ff ce 63 fe ff d3 68 fe ff |.`...a...c...h..| 00000070 83 6f fe ff 84 6c fe ff 22 73 fe ff 91 6d fe ff |.o...l.."s...m..| ``` As the change in Commit 9cdcbe0e932b97fac88a6195f87647159a6c7194, the header information has changed BitsPerSample (bit_p_spl) from 24bit to 32bit (0x22-0x23). If I change this header info and treat the 1.2.14 wav as 24-bit per sample as before, the audio seems to play correctly. I don't have the correct information for S24_LE format, so I don't know if the header information is wrong or if I should modify the arecord to match this 32-bit. Issue URL : https://github.com/alsa-project/alsa-utils/issues/308 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ucm2: IO-Boards: Toradex: aquila: add support
by GitHub pull_request - opened 10 Nov '25

10 Nov '25

alsa-project/alsa-ucm-conf pull request #641 was opened from hiagofranco: Add support for Toradex Aquila Development board, using the WM8904 audio codec. This is a carrier board for the Toradex Aquila family, where any Toradex Aquila SoM can be connected to it, therefore this is being added to the IO-Boards instead of a specific hardware vendor. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/641 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/641.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: SDCA: bug fix while parsing mipi-sdca-control-cn-list
by Niranjan H Y 10 Nov '25

10 Nov '25

"struct sdca_control" declares "values" field as integer array. But the memory allocated to it is of char array. This causes crash for sdca_parse_function API. This patch addresses the issue by allocating correct data size. Signed-off-by: Niranjan H Y <niranjan.hy(a)ti.com> --- sound/soc/sdca/sdca_functions.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sdca/sdca_functions.c b/sound/soc/sdca/sdca_functions.c index b2e3fab9b..6975c71a9 100644 --- a/sound/soc/sdca/sdca_functions.c +++ b/sound/soc/sdca/sdca_functions.c @@ -950,7 +950,8 @@ static int find_sdca_entity_control(struct device *dev, struct sdca_entity *enti return ret; } - control->values = devm_kzalloc(dev, hweight64(control->cn_list), GFP_KERNEL); + control->values = devm_kzalloc(dev, hweight64(control->cn_list) * sizeof(int), + GFP_KERNEL); if (!control->values) return -ENOMEM; -- 2.25.1

1 0