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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

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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

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