August 2025 - Alsa-devel - mailman.alsa-project.org

ucm2: Qualcomm: add Radxa Dragon Q6A
by GitHub pull_request - opened 10 Aug '25

10 Aug '25

alsa-project/alsa-ucm-conf pull request #601 was opened from strongtz: This is an upcoming SBC (single board computer) based on QCS6490. Full specs: - Headphone jack using WCD9380 codec - Onboard DisplayPort to HDMI bridge using Radxa RA620 - MI2S0 available via the 40-Pin GPIO header (unused by default) - AMICs using WCD9380 codec via a dedicated MIC header (unused by default) The kernel device tree part is not yet but going to be upstreamed soon. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/601.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[HDA][HP EliteDesk 800 G5][8086:a348] – HDMI audio missing after BIOS update
by Hermanns Cyrille 09 Aug '25

09 Aug '25

Hello ALSA team, On an HP EliteDesk 800 G5 Desktop Mini with BIOS R21 Ver. 02.22.00, audio controller Intel Cannon Lake PCH cAVS [8086:a348], HDMI audio outputs are no longer detected after this BIOS update. System: - Linux Mint 22.1 Cinnamon 6.4.8 - Kernel: 6.8.0-71-generic - Codecs: Conexant CX20632 (analog), Intel Kabylake HDMI (digital) Issue: After boot, HDMI outputs appear as “Not connected”. HDMI audio works only if I use hdajackretask (Intel Kabylake HDMI → Show unconnected pins → set to HDMI/DisplayPort) and select “Install boot override”. ALSA report: http://alsa-project.org/db/?f=a1ee220062b23ad5c13c6d38cde3f1d046ce6260 Request: Would it be possible to add a quirk for this model so HDMI pins are enabled automatically at boot? Thank you for your help.

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Problems with "snd_pcm_hw_params_set_rate_near" and "snd_pcm_hw_params_set_period_size_near" in x86_64 assembly.
by GitHub issues - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-lib issue #471 was opened from sc1ntilla: ### **PREAMBLE:** I have never worked with any sound system before, so I'm almost sure that it's just me doing something wrong, but I can't see what, so I'm asking for help. I'm doing this project for fun, and to learn a lot of stuff, so feel free to give your opinions / advice if you want to. ### **WHAT HAPPENS:** When calling `snd_pcm_hw_params_set_rate_near` and `snd_pcm_hw_params_set_period_size_near` memory protection violation is issued. I find that strange, because `snd_pcm_hw_params_set_buffer_size_near` works perfectly fine, and it's similiar to these two. ### **WHAT I TRIED:** - Making sure that stack after call is on address divisible by 96 (by adding 8 to stack pointer), - Making sure that `*val` argument is on address divisible by 96, - Misaligning stack by 8, 16, 24 and 32 before calling these functions (by subtracting from stack pointer), - Providing 0 to `*val` argument, to rule out that writing to output crashes, - Providing valid output address to `*dir` argument to rule out that 0 is causing crashes, - Calling these functions in different order, - Trying to call `snd_pcm_set_params` instead, but it says _"Channels count (0) not available for PLAYBACK: Invalid argument"_ however, I'm sure that I provided a valid number (2). I even hardcoded it. But to no avail. Is it my misunderstanding of how alsa-lib works? Very likely. Is my assembly code faulty? Quite possible. I just can't see where the problem lies. If you see error somewhere, please let me know. ### **ADDITIONAL INFORMATION:** I have multiple audio outputs: monitors with built-in speakers, standalone speakers, and analog jack headphones. I hardcoded `hw:2,0` as device name to open. Sometimes it results in `snd_pcm_open` returning _"File descriptor in bad state"_. When I tried using `sysdefault` it often returns _"ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.14/work/alsa-lib-1.2.14/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave"_ Below is .data section in which globals are stored, and source of function that will be used to play audio buffers. .data section: ``` .section .data .align 32 test2: .8byte 0 test: .8byte 0 SND_PCM_NONBLOCK: .8byte 0x1 SND_PCM_ASYNC: .8byte 0x2 SND_PCM_STREAM_PLAYBACK: .8byte 0 SND_PCM_FORMAT_U8: .8byte 1 SND_PCM_FORMAT_U16_LE: .8byte 4 SND_PCM_FORMAT_U24_LE: .8byte 8 SND_PCM_FORMAT_U32_LE: .8byte 12 SND_PCM_FORMAT_S8: .8byte 0 SND_PCM_FORMAT_S16_LE: .8byte 2 SND_PCM_FORMAT_S24_LE: .8byte 6 SND_PCM_FORMAT_S32_LE: .8byte 10 SND_PCM_ACCESS_MMAP_INTERLEAVED: .8byte 0 SND_PCM_STATE_PREPARED: .8byte 2 AUDIO_BUFFER_STATE_IDLE: .8byte 0 AUDIO_BUFFER_STATE_WRITING: .8byte 1 AUDIO_BUFFER_STATE_PLAYING: .8byte 2 t1: .8byte 0 t2: .8byte 0 alsa: alsa.pcm_handle: .8byte 0 alsa.hw_params: .8byte 0 alsa.last_error: .8byte 0 alsa.initialized: .8byte 0 alsa.device_name: .ascii "hw:2,0\0" audio_buffers.count_max: .8byte 16 amd64_audio_last_error: .8byte 0 ``` function: ``` amd64_audio_play: /* rdi: (input) address to audio data: * (+0/2) audio format (1: PCM, 2: IEEE754) * (+2/2) amount of channels * (+4/4) rate in hertz * (+8/4) bytes per sec (frequency * bytes per block) * (+12/2) bytes per block (number of channels * bits per sample / 8) * (+14/2) bits per sample * (+16/8) total size of samples data * (+24/8) address to samples data */ /* Align stack on 96byte boundary: */ push %r15 mov %rsp, %r15 and $-32, %rsp mov %rsp, %rax mov $96, %r14 xor %rdx, %rdx div %r14 sub %rdx, %rsp sub $96, %rsp mov %r15, (%rsp) /* Store registers: */ sub $192, %rsp call amd64_util_store_gpr_regs /* Create stack space: */ sub $192, %rsp mov %rdi, (%rsp) /* stack layout: * (+0/8) rdi (address to audio data) * (+8/8) buffer size (altered by *buffer_size_near) * (+16/8) period size (set when setting period size) * (+24/8) rate in hertz * (+32/8) temp */ xor %r15, %r15 mov 16(%rdi), %r15 mov %r15, 8(%rsp) xor %r15, %r15 mov 4(%rdi), %r15d mov %r15d, 24(%rsp) movq $0, 32(%rsp) /* Open pcm if needed: */ cmpq $0, (alsa.pcm_handle) jne .amd64_audio_play.pcm_is_open lea alsa.pcm_handle, %rdi mov %rdi, (%rdi) lea alsa.device_name, %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx mov (SND_PCM_NONBLOCK), %r10 call snd_pcm_open cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_open .amd64_audio_play.pcm_is_open: /* Allocate hw params: */ cmpq $0, (alsa.hw_params) jne .amd64_audio_play.hw_params_are_allocated lea alsa.hw_params, %rdi mov %rdi, (%rdi) call snd_pcm_hw_params_malloc cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_malloc .amd64_audio_play.hw_params_are_allocated: /* Get all possible combinations into hw params: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_any cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_any /* Set access: */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov (SND_PCM_ACCESS_MMAP_INTERLEAVED), %rdx call snd_pcm_hw_params_set_access cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_access /* Set format: */ xor %rdi, %rdi mov (%rsp), %rdi mov 14(%rdi), %r15w mov (%rdi), %r14w cmp $1, %r14w jne .amd64_audio_play.error.format_unsupported cmpw $32, %r15w je .amd64_audio_play.format32 cmpw $24, %r15w je .amd64_audio_play.format24 cmpw $16, %r15w je .amd64_audio_play.format16 cmpw $8, %r15w je .amd64_audio_play.format8 jmp .amd64_audio_play.error.format_unsupported .amd64_audio_play.format32: mov (SND_PCM_FORMAT_S32_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format24: mov (SND_PCM_FORMAT_S24_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format16: mov (SND_PCM_FORMAT_S16_LE), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format8: mov (SND_PCM_FORMAT_S8), %rdx jmp .amd64_audio_play.format_end .amd64_audio_play.format_end: mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi call snd_pcm_hw_params_set_format cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_format /* Set channels: */ xor %rdx, %rdx mov (%rsp), %r15 mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov 2(%r15), %dx call snd_pcm_hw_params_set_channels cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_channels /* Set buffer size near: */ /* Returns 0, meaning it succeeded. No crashes generated. */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $8, %rdx mov $0, %r10 call snd_pcm_hw_params_set_buffer_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near /* Set rate near: */ /* CRASHES */ mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $24, %rdx mov $0, %r10 call snd_pcm_hw_params_set_rate_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near /* Set period size near: */ /* CRASHES */ mov 8(%rsp), %rax mov $4, %r15 xor %rdx, %rdx div %r15 mov %rax, 16(%rsp) mov (alsa.pcm_handle), %rdi mov (alsa.hw_params), %rsi mov %rsp, %rdx add $16, %rdx mov $0, %r10 call snd_pcm_hw_params_set_period_size_near cmp $0, %rax jne .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near /* Free hw params: */ cmpq $0, (alsa.hw_params) je .amd64_audio_play.hw_params_not_allocated mov (alsa.hw_params), %rdi call snd_pcm_hw_params_free movq $0, (alsa.hw_params) .amd64_audio_play.hw_params_not_allocated: /* Register errors: */ xor %rax, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_open: mov %rax, (amd64_audio_last_error) mov $-1, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_malloc: mov %rax, (amd64_audio_last_error) mov $-2, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_any: mov %rax, (amd64_audio_last_error) mov $-3, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_access: mov %rax, (amd64_audio_last_error) mov $-4, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.format_unsupported: mov $-5, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_format: mov %rax, (amd64_audio_last_error) mov $-6, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_channels: mov %rax, (amd64_audio_last_error) mov $-7, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_buffer_size_near: mov %rax, (amd64_audio_last_error) mov $-8, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_period_size_near: mov %rax, (amd64_audio_last_error) mov $-9, %rax jmp .amd64_audio_play.exit .amd64_audio_play.error.snd_pcm_hw_params_set_rate_near: mov %rax, (amd64_audio_last_error) mov $-10, %rax jmp .amd64_audio_play.exit /* Exit: */ .amd64_audio_play.exit: /* Print error to stdout: */ mov %rax, %rdi call amd64_util_print_dec call amd64_util_print_newline cmpq $0, (amd64_audio_last_error) je .amd64_audio_play.no_error mov (amd64_audio_last_error), %rdi call snd_strerror mov %rax, %rdi call amd64_util_print2 call amd64_util_print_newline .amd64_audio_play.no_error: /* Restore stack: */ add $192, %rsp /* Restore registers without rax: */ mov %rax, 128(%rsp) call amd64_util_restore_gpr_regs mov 128(%rsp), %rax add $192, %rsp /* Restore stack alignment: */ pop %rsp pop %r15 ret ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/471 Repository URL: https://github.com/alsa-project/alsa-lib

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Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "GenericName" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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Update hdajackretask.desktop
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was edited from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (closes #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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Update hdajackretask.desktop
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-tools pull request #39 was opened from trebmuh: - adds a FR comment - change the "genericname" to "Comment" - adds a Icon= field (close #38 ) Request URL : https://github.com/alsa-project/alsa-tools/pull/39 Patch URL : https://github.com/alsa-project/alsa-tools/pull/39.patch Repository URL: https://github.com/alsa-project/alsa-tools

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ucm2: Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm MONACO-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Qualcomm: Add MONACO-EVK HiFi config
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #597 was edited from mohsRafi: Add UCM2 configs for the Qualcomm QCS9075-IQ8-EVK Board to handle: - I2S Speaker Amplifier - I2S Mic Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/597.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Rename qcs6490-rb3gen2 and qcs9075-iq-evk ucm2 conf
by GitHub pull_request - edited 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #596 was edited from mohsRafi: Rename the ucm2 conf for Qualcomm qcs6490-rb3gen2 and qcs9075-iq-evk qcs6490-rb3gen2-snd-card.conf -> QCS6490-RB3Gen2.conf qcs9075-iq-evk-snd-card.conf -> LEMANS-EVK.conf Removed snd-card tags from conf files Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/596.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2: sof-soundwire: add rt721 ucm support
by GitHub pull_request - opened 08 Aug '25

08 Aug '25

alsa-project/alsa-ucm-conf pull request #600 was opened from jack-cy-yu: Add support for rt721's UCM. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/600.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.17-merge-window
by Mark Brown 06 Aug '25

06 Aug '25

The following changes since commit e95122a32e777309412e30dc638dbc88b9036811: ASoC: codecs: Add acpi_match_table for aw88399 driver (2025-07-25 13:44:29 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.17-merge-window for you to fetch changes up to 614d416dd8aee2675fb591c598308a901a660db8: ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies (2025-08-06 12:02:32 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.17 This is a relatively small set of fixes and device quirks that came in during the merge window, the AMD changes adding support for ACP 7.2 systems are all just adding IDs for the devices rather than any substantial code - the actual code is the same as for prior versions of the platform. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: SOF: Intel: hda-sdw-bpt: fix SND_SOF_SOF_HDA_SDW_BPT dependencies Baojun Xu (1): ASoC: tas2781: Fix the wrong step for TLV on tas2781 Cezary Rojewski (1): ASoC: Intel: avs: Fix uninitialized pointer error in probe() Mark Brown (1): Add audio support for acp7.2 platform Muhammad Usama Anjum (1): ASoC: SOF: amd: acp-loader: Use GFP_KERNEL for DMA allocations in resume context Peter Jakubek (1): ASoC: Intel: sof_sdw: Add quirk for Alienware Area 51 (2025) 0CCC SKU Shengjiu Wang (1): ASoC: wm8962: Clear master mode when enter runtime suspend Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform include/sound/tas2781-tlv.h | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/codecs/wm8962.c | 11 +++++++++++ sound/soc/intel/avs/core.c | 3 ++- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ sound/soc/sof/amd/acp-loader.c | 6 +++--- sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + sound/soc/sof/intel/Kconfig | 3 ++- 15 files changed, 53 insertions(+), 6 deletions(-)

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[PATCH 0/4] Add audio support for acp7.2 platform
by Venkata Prasad Potturu 06 Aug '25

06 Aug '25

This patch series is to add legacy and sof audio support for acp7.2 platform. Venkata Prasad Potturu (4): ASoC: SOF: amd: Add sof audio support for acp7.2 platform ASoC: amd: ps: Add SoundWire pci and dma driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire legacy machine driver support for acp7.2 platform ASoC: amd: acp: Add SoundWire SOF machine driver support for acp7.2 platform sound/soc/amd/acp/acp-sdw-legacy-mach.c | 3 +++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 1 + sound/soc/amd/acp/soc_amd_sdw_common.h | 2 ++ sound/soc/amd/ps/acp63.h | 1 + sound/soc/amd/ps/pci-ps.c | 4 ++++ sound/soc/amd/ps/ps-sdw-dma.c | 5 +++++ sound/soc/sof/amd/acp.c | 8 ++++++++ sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-acp70.c | 1 + 9 files changed, 26 insertions(+) -- 2.43.0

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[PATCH v1] ASoC: tas2781: Fix the wrong step for TLV on tas2781
by Baojun Xu 06 Aug '25

06 Aug '25

The step for TLV on tas2781, should be 50 (-0.5dB). Fixes: 678f38eba1f2 ("ASoC: tas2781: Add Header file for tas2781 driver") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index d87263e43fdb..ef9b9f19d212 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -15,7 +15,7 @@ #ifndef __TAS2781_TLV_H__ #define __TAS2781_TLV_H__ -static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); #endif -- 2.43.0

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Re: ALSAfirewire broken / Pipewire 90ms delay
by Goran Kovac 06 Aug '25

06 Aug '25

While this is mostly a non issue for playback only, recording in studio environment is unfortunately unusable. Would be really nice if this can be solved. My RME 800 can go really low with buffer size but its always +90ms.

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Re: ASoC HDMI errors with sof-hda-dsp on Intel platform (no PCM for intel-hdmi-hifiX)
by Damian Tometzki 06 Aug '25

06 Aug '25

Hi all, I'm experiencing ASoC errors on an Intel-based system using the `sof-hda-dsp` driver. After boot, the kernel logs repeated errors related to HDMI outputs: [ 90.933290] snd_hda_codec_generic ehdaudio0D2: snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at snd_soc_dai_startup() on intel-hdmi-hifi1 ... [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and this breaks HDMI audio initialization. **System info:** - Kernel: current Linus build 2938ae2fa7cd - Distro: Fedora 42 - HDMI monitor with audio capability isn´t connected Is there a known fix or workaround? Thanks in advance, Damian On Sat, Aug 2, 2025 at 12:46 PM Damian Tometzki <damian(a)riscv-rocks.de> wrote: > > Hi all, > > I'm experiencing ASoC errors on an Intel-based system using the > `sof-hda-dsp` driver. After boot, the kernel logs repeated errors > related to HDMI outputs: > > [ 90.933290] snd_hda_codec_generic ehdaudio0D2: > snd_soc_find_pcm_from_dai: didn't find PCM for DAI intel-hdmi-hifi1 > [ 90.933295] snd_hda_codec_generic ehdaudio0D2: ASoC error (-22): at > snd_soc_dai_startup() on intel-hdmi-hifi1 > ... > [ 91.016259] HDMI3: ASoC error (-22): at dpcm_fe_dai_startup() on HDMI3 > > These errors are repeated for HDMI1, HDMI2, and HDMI3. It appears that > `snd_soc_find_pcm_from_dai()` fails to resolve the DAI to a PCM, and > this breaks HDMI audio initialization. > > **System info:** > - Kernel: current Linus build 2938ae2fa7cd > - Distro: Fedora 42 > - HDMI monitor with audio capability isn´t connected > > Is there a known fix or workaround? > Thanks in advance, > Damian

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amixer: always use line buffering for (s)events
by GitHub pull_request - opened 06 Aug '25

06 Aug '25

alsa-project/alsa-utils pull request #306 was opened from elenril: Line buffering is the obviously correct mode for (s)events, but block buffering would typically be used when piping its output to another program. Request URL : https://github.com/alsa-project/alsa-utils/pull/306 Patch URL : https://github.com/alsa-project/alsa-utils/pull/306.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v1] ALSA: hda/tas2781: Support L"SmartAmpCalibrationData" to save calibrated data
by Shenghao Ding 06 Aug '25

06 Aug '25

Some devices save the calibrated data into L"CALI_DATA", and others into L"SmartAmpCalibrationData". Driver code will support both. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/hda/codecs/side-codecs/tas2781_hda.c | 47 +++++++++++++++------- sound/hda/codecs/side-codecs/tas2781_hda.h | 2 +- 2 files changed, 33 insertions(+), 16 deletions(-) diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.c b/sound/hda/codecs/side-codecs/tas2781_hda.c index 34217ce9f28e..f46d2e06c64f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda.c @@ -18,6 +18,8 @@ #include "tas2781_hda.h" +#define CALIBRATION_DATA_AREA_NUM 2 + const efi_guid_t tasdev_fct_efi_guid[] = { /* DELL */ EFI_GUID(0xcc92382d, 0x6337, 0x41cb, 0xa8, 0x8b, 0x8e, 0xce, 0x74, @@ -160,36 +162,51 @@ int tas2781_save_calibration(struct tas2781_hda *hda) * manufactory. */ efi_guid_t efi_guid = tasdev_fct_efi_guid[LENOVO]; - static efi_char16_t efi_name[] = TASDEVICE_CALIBRATION_DATA_NAME; + /* + * Some devices save the calibrated data into L"CALI_DATA", + * and others into L"SmartAmpCalibrationData". + */ + static efi_char16_t *efi_name[CALIBRATION_DATA_AREA_NUM] = { + L"CALI_DATA", + L"SmartAmpCalibrationData", + }; struct tasdevice_priv *p = hda->priv; struct calidata *cali_data = &p->cali_data; unsigned long total_sz = 0; unsigned int attr, size; unsigned char *data; efi_status_t status; + int i; if (hda->catlog_id < LENOVO) efi_guid = tasdev_fct_efi_guid[hda->catlog_id]; cali_data->cali_dat_sz_per_dev = 20; size = p->ndev * (cali_data->cali_dat_sz_per_dev + 1); - /* Get real size of UEFI variable */ - status = efi.get_variable(efi_name, &efi_guid, &attr, &total_sz, NULL); - cali_data->total_sz = total_sz > size ? total_sz : size; - if (status == EFI_BUFFER_TOO_SMALL) { - /* Allocate data buffer of data_size bytes */ - data = p->cali_data.data = devm_kzalloc(p->dev, - p->cali_data.total_sz, GFP_KERNEL); - if (!data) { - p->cali_data.total_sz = 0; - return -ENOMEM; + for (i = 0; i < CALIBRATION_DATA_AREA_NUM; i++) { + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name[i], &efi_guid, &attr, + &total_sz, NULL); + cali_data->total_sz = total_sz > size ? total_sz : size; + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + data = cali_data->data = devm_kzalloc(p->dev, + cali_data->total_sz, GFP_KERNEL); + if (!data) { + status = -ENOMEM; + continue; + } + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name[i], &efi_guid, + &attr, &cali_data->total_sz, data); } - /* Get variable contents into buffer */ - status = efi.get_variable(efi_name, &efi_guid, &attr, - &p->cali_data.total_sz, data); + /* Check whether get the calibrated data */ + if (status == EFI_SUCCESS) + break; } + if (status != EFI_SUCCESS) { - p->cali_data.total_sz = 0; + cali_data->total_sz = 0; return status; } diff --git a/sound/hda/codecs/side-codecs/tas2781_hda.h b/sound/hda/codecs/side-codecs/tas2781_hda.h index 575a701c8dfb..66188909a0bb 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda.h +++ b/sound/hda/codecs/side-codecs/tas2781_hda.h @@ -11,7 +11,7 @@ /* Flag of calibration registers address. */ #define TASDEV_UEFI_CALI_REG_ADDR_FLG BIT(7) -#define TASDEVICE_CALIBRATION_DATA_NAME L"CALI_DATA" + #define TASDEV_CALIB_N 5 /* -- 2.43.0

2 1

ALC4082: capturing S/PDIF output causes volume to jump to max - asus rog crosshair x670e hero
by GitHub issues - edited 06 Aug '25

06 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v1] ALSA: hda: Fix the wrong register was used for DVC of TAS2770
by Baojun Xu 05 Aug '25

05 Aug '25

The wrong register was used for digital volume control of TAS2770, The definition was changed, and usage was also updated. Fixes: ab29b3460c5c ("ALSA: hda: Add TAS2770 support") Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2770-tlv.h | 4 ++-- sound/hda/codecs/side-codecs/tas2781_hda_i2c.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/tas2770-tlv.h b/include/sound/tas2770-tlv.h index c0bd495b4a07..5518b34abdde 100644 --- a/include/sound/tas2770-tlv.h +++ b/include/sound/tas2770-tlv.h @@ -14,10 +14,10 @@ #ifndef __TAS2770_TLV_H__ #define __TAS2770_TLV_H__ -#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x17) +#define TAS2770_DVC_LEVEL TASDEVICE_REG(0x0, 0x0, 0x05) #define TAS2770_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) -static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, 1650, 50, 0); +static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_dvc_tlv, -1000, 50, 0); static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2770_amp_tlv, 1100, 50, 0); #endif diff --git a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c index a0b132681804..45ac5e41bd4f 100644 --- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c +++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c @@ -260,7 +260,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = { 0, 0, 20, 0, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_amp_tlv), ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2770_DVC_LEVEL, - 0, 0, 31, 0, tas2781_amp_getvol, + 0, 0, 200, 1, tas2781_amp_getvol, tas2781_amp_putvol, tas2770_dvc_tlv), }; -- 2.43.0

2 1

hdajackretask.desktop missing icon
by GitHub issues - opened 05 Aug '25

05 Aug '25

alsa-project/alsa-tools issue #38 was opened from Dejweed: The hdajackretask.desktop file doesn't have the `Icon=hdajackretask`. Issue URL : https://github.com/alsa-project/alsa-tools/issues/38 Repository URL: https://github.com/alsa-project/alsa-tools

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Subject: [BUG] snd_hda_intel runtime power management unsupported on MSI MPG Z390 GAMING PLUS with Realtek codec (kernel 6.8.0) causing excessive audio power draw
by Thomas van Houten 05 Aug '25

05 Aug '25

Dear Linux kernel and ALSA developers, I am reporting an issue related to runtime power management support for the onboard Realtek audio codec on my desktop motherboard, which results in continuously high audio hardware power usage and increased idle system power draw. *System Information:* * Kernel: 6.8.0-64-generic x86_64 * Distribution: Linux Mint 21.3 Virginia * Motherboard: MSI MPG Z390 GAMING PLUS (MS-7B51) * CPU: Intel Core i5-9400F * Audio Driver: snd_hda_intel * Audio Codec: Realtek (likely ALC1220 or similar) *Issue Summary:* The onboard Realtek audio codec does not enter runtime power management low-power states, causing it to remain at 100% power usage even when idle. This significantly contributes to my system's high idle power consumption (~160W total), despite low CPU and GPU draw. *Detailed Diagnostics:* 1. *PCI Audio Devices:* text |00:1f.3 Audio device [0403]: Intel Corporation Cannon Lake PCH cAVS [8086:a348] (rev 10) 03:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 31 HDMI/DP Audio [1002:ab30] 07:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Navi 21/23 HDMI/DP Audio Controller [1002:ab28] | 2. *Kernel Messages related to snd_hda_intel or audio:* No logs found via|dmesg | grep -i snd_hda_intel|or|dmesg | grep -i audio|. 3. *Power management sysfs settings:* text |cat /sys/module/snd_hda_intel/parameters/power_save 10 cat /sys/class/sound/hwC*/power/runtime_status unsupported unsupported unsupported cat /sys/class/sound/hwC*/power/control auto auto auto | The|runtime_status|reports "unsupported", indicating no runtime PM support is available for these audio devices. 4. *PulseAudio and user-space processes inspection:* Stopped PulseAudio and all audio clients, with no change in power usage or device ownership of audio controls, confirming this is not caused by user-space audio applications. 5. *Powertop Reports:* text |100.0% Device Audio codec hwC2D0: Realtek | Consistently 100% usage from the Realtek audio codec, confirmed by powertop. 6. *Driver parameter set:* |snd_hda_intel|module|power_save|parameter successfully set to 10 seconds (value 10), but no effective power saving observed. *Impact:* The lack of runtime power management support for this audio codec causes continuous high power draw, increasing system idle power consumption significantly, reducing energy efficiency. *Request:* Please investigate whether this is a driver limitation, missing firmware quirk, or configuration issue with|snd_hda_intel|on the MSI MPG Z390 GAMING PLUS motherboard's Realtek codec. Guidance, workarounds, or patches to enable runtime power management would be appreciated. *Additional notes:* * Tested power management settings according to current best practices. * BIOS is up to date as of the latest MSI release. * System behavior consistent on Live USB of the same Linux Mint version—problem not due to local configuration. Thank you for your attention to this matter. I’m happy to provide additional logs or assist with testing patches. Best regards, Thomas van Houten

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ucm2: Qualcomm: add Lenovo Ideapad 5 (Slim 5x / 2in1) support
by GitHub pull_request - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf pull request #599 was opened from jglathe: Same layout as T14s, 2 speakers, headphone jack, DMIC Tested on Ideapad 5 2in1 14Q8X9. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/599.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - edited 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was edited from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.txt](https://github.com/user-attachments/files/21583734/alsa-info… <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4082: S/PDIF volume redirecting to headphone volume - asus rog crosshair x670e hero
by GitHub issues - opened 04 Aug '25

04 Aug '25

alsa-project/alsa-ucm-conf issue #598 was opened from LuanVSO: hi, i had a problem where opening the kde plasma volume applet would make the volume on my headphone jump to the maximum. after muting the "PCM 2" channel on alsamixer i notice that the volume on the headphone would go muted instead. so i edited the ALC4080-HiFi.conf file and remove every mention of S/PDIF and rebooted my machine. after the change the problem didn't happen anymore. what do you need to fix the mapping problem? i think this happens because the "PCM 2" volume controls is redirecting to headphone output, but it doenst have a slider, so it is either max or muted: <img width="1259" height="967" alt="Image" src="https://github.com/user-attachments/assets/3b266bb7-7e1c-4a7a-b448-affabf06…" /> (also the S/PDIF aways appeared as connected even though i did not have anything connected to it, don't know if it is expected) [alsa-info.sh](https://paste.centos.org/view/c6545d44) <details> <summary>edited ALC4080-HiFi.conf</summary> ``` Define { SpeakerMixer "PCM" SpeakerMindex "0" SpeakerJack "Speaker - Output Jack" SpeakerPCM "hw:${CardId}" HeadphonesName "Front Headphones" HeadphonesMixer "PCM" HeadphonesMindex "1" HeadphonesJack "Headphone - Output Jack" HeadphonesPCM "hw:${CardId},1" Mic1Name "Microphone" Mic1Mixer "Mic" Mic1Mindex "0" Mic1Jack "Mic - Input Jack" Mic1PCM "hw:${CardId},2" Mic2Name "" Mic2Mixer "" Mic2Mindex "0" Mic2Jack "" Mic2PCM "" Line1Name "Line Input" Line1Mixer "Line" Line1Mindex "0" Line1Jack "Line - Input Jack" Line1PCM "hw:${CardId},1" # SpdifName "S/PDIF Output" # SpdifPCM "hw:${CardId},3" # SpdifMixer "PCM" # SpdifMindex "2" } If.speaker_ctl { Condition { Type ControlExists Control "name='Speaker Playback Switch'" } True.Define.SpeakerMixer "Speaker" } If.front_hp_ctl { Condition { Type ControlExists Control "name='Front Headphone Playback Switch'" } True.Define { HeadphonesMixer "Front Headphone" HeadphonesMindex "0" } } #If.spdif_ctl { # Condition { # Type ControlExists # Control "name='IEC958 Playback Switch'" # } # True.Define { # SpdifMixer "IEC958" # SpdifMindex "0" # } #} #If.spdif_nodev { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0db0:36e7)" # } # True.Define.SpdifName "" #} #If.spdif_dev2 { # Condition { # Type RegexMatch # String "${CardComponents}" # Regex "USB(0b05:(1996|1a5[23c])|0db0:1feb)" # } # True.Define.SpdifPCM "hw:${CardId},2" #} If.gigabyte-aorus-ultra { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0414:a014)" } True.Define { Mic1Name "Front Microphone" Mic1PCM "hw:${CardId},0" Mic2Name "Rear Microphone" Mic2Mixer "Mic" Mic2Mindex "1" Mic2Jack "name='Mic - Input Jack',index=1" Mic2PCM "hw:${CardId},1" SpdifName "" Line1Name "" } } If.asus-rog-usb { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1996)" } True.Define { Mic1Name "Front Microphone" Mic2Name "Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.asrock-taichi { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(26ce:0a0[68b])" } True.Define { Line1Name "" Mic1Name "Rear Input" Mic1PCM "hw:${CardId},0" Mic2Name "Front Microphone" Mic2Mixer "Mic" Mic2Jack "Front Mic - Input Jack" Mic2PCM "hw:${CardId},1" } } If.wrx80-sage { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1984)" } True.Define { Mic2Name "Front Microphone" Mic2Mixer "Analog In" Mic2Jack "Analog In - Input Jack" Mic2PCM "hw:${CardId}" } } If.msi-meg-unify { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0db0:82c7)" } True.Define { HeadphonesName "" SpdifName "" } } If.asus-rog-strix { Condition { Type RegexMatch String "${CardComponents}" Regex "USB(0b05:1999)" } True.Define { Line1PCM "hw:${CardId},3" HeadphonesName "" SpdifName "" } } SectionVerb { EnableSequence [ disdevall "" ] Value.TQ "HiFi" } SectionDevice."Speaker" { Comment "Speakers" EnableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} on" ] DisableSequence [ cset "name='${var:SpeakerMixer} Playback Switch',index=${var:SpeakerMindex} off" ] Value { PlaybackChannels 2 PlaybackPriority 200 PlaybackPCM "${var:SpeakerPCM}" JackControl "${var:SpeakerJack}" PlaybackMixerElem "${var:SpeakerMixer},${var:SpeakerMindex}" } Variant."HiFi 5+1".Value.PlaybackChannels 6 Variant."HiFi 7+1".Value.PlaybackChannels 8 } If.headphones { Condition { Type String Empty "${var:HeadphonesName}" } False.SectionDevice."Headphones" { Comment "${var:HeadphonesName}" EnableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} on" ] DisableSequence [ cset "name='${var:HeadphonesMixer} Playback Switch',index=${var:HeadphonesMindex} off" ] Value { PlaybackPriority 300 PlaybackPCM "${var:HeadphonesPCM}" JackControl "${var:HeadphonesJack}" PlaybackMixerElem "${var:HeadphonesMixer},${var:HeadphonesMindex}" } } } #If.spdif { # Condition { # Type String # Empty "${var:SpdifName}" # } # False.SectionDevice."SPDIF" { # Comment "${var:SpdifName}" # # EnableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} on" # ] # # DisableSequence [ # cset "name='${var:SpdifMixer} Playback Switch',index=${var:SpdifMindex} off" # ] # # Value { # PlaybackPriority 100 # PlaybackPCM "${var:SpdifPCM}" # PlaybackMixerElem "${var:SpdifMixer},${var:SpdifMindex}" # } # } #} If.line1 { Condition { Type String Empty "${var:Line1Name}" } False.SectionDevice."Line1" { Comment "${var:Line1Name}" EnableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} on" ] DisableSequence [ cset "name='${var:Line1Mixer} Capture Switch',index=${var:Line1Mindex} off" ] Value { CapturePriority 100 CapturePCM "${var:Line1PCM}" JackControl "${var:Line1Jack}" CaptureMixerElem "${var:Line1Mixer},${var:Line1Mindex}" } } } If.mic1 { Condition { Type String Empty "${var:Mic1Name}" } False.SectionDevice."Mic1" { Comment "${var:Mic1Name}" EnableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} on" ] DisableSequence [ cset "name='${var:Mic1Mixer} Capture Switch',index=${var:Mic1Mindex} off" ] Value { CapturePriority 300 CapturePCM "${var:Mic1PCM}" JackControl "${var:Mic1Jack}" CaptureMixerElem "${var:Mic1Mixer},${var:Mic1Mindex}" } } } If.mic2 { Condition { Type String Empty "${var:Mic2Name}" } False.SectionDevice."Mic2" { Comment "${var:Mic2Name}" EnableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} on" ] DisableSequence [ cset "name='${var:Mic2Mixer} Capture Switch',index=${var:Mic2Mindex} off" ] Value { CapturePriority 400 CapturePCM "${var:Mic2PCM}" JackControl "${var:Mic2Jack}" CaptureMixerElem "${var:Mic2Mixer},${var:Mic2Mindex}" } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/598 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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