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

[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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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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