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ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - opened 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was opened from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated to two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

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Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` I had this monitor connected before and alsacap reported "flipped" results - not like now that both cards are limited to 48000. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

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Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

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Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - opened 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was opened from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

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Add Roland QUAD-CAPTURE USB 2.0 support
by GitHub pull_request - opened 31 Dec '24

31 Dec '24

alsa-project/alsa-ucm-conf pull request #494 was opened from bjornmp: Add Roland QUAD-CAPTURE USB 2.0 support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib-1.2.13 is breaking dssi-1.1.1 because of missing header
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-lib issue #431 was opened from kobboi: What was working with 1.2.12 no longer isn't: ``` /usr/include/alsa/ump_msg.h:1119:65: error: unknown type name ‘size_t’ 1119 | int snd_ump_msg_sysex_expand(const uint32_t *ump, uint8_t *buf, size_t maxlen, | ^~~~~~ /usr/include/alsa/ump_msg.h:13:1: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? 12 | #include <stdint.h> +++ |+#include <stddef.h> 13 | /usr/include/alsa/ump_msg.h:1120:30: error: unknown type name ‘size_t’ 1120 | size_t *filled); | ^~~~~~ /usr/include/alsa/ump_msg.h:1120:30: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? ``` Can this be addressed? Happy holidays! Issue URL : https://github.com/alsa-project/alsa-lib/issues/431 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 30 Dec '24

30 Dec '24

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 910305c11e8a..db50e677c9ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23170,25 +23170,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23198,9 +23215,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

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UCM2 macro breaks on updates for Motu 828
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-ucm-conf issue #493 was opened from beatboxa: It appears that every time there is an update, my UCM profile breaks. So that when I boot my machine, I have no sound (unless I manually map things in a pro audio application). This appears to be due to an issue in the SplitPCM macro--specifically, some parsing. Here is the thread describing the creation of the UCM profile I am using, including a patch that I had to apply in order to get things working: #416 And specifically, here is the patch I apply in order to fix it: https://github.com/alsa-project/alsa-ucm-conf/commit/b68aa52acdd2763fedad5e… It involves adding a single hyphen character to the file: ucm2/common/pcm/split.conf Because it is a single character, I have been manually changing the file. It's hard for me to follow that thread--but is this fixed and I am on older versions, or is this still an open item? Or is there a way to automate this so that my audio doesn't break on every update? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/493 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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