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

alsa-info.8: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 21 Feb '25

21 Feb '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") troff:<stdin>:59: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.12-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-6 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250125-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250125-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

FiiO K7 USB DAC no Inputs
by GitHub issues - opened 21 Feb '25

21 Feb '25

alsa-project/alsa-lib issue #439 was opened from Starfoxfs: I have problems with Fiio K7 USB DAC in Linux Mint Cinnamon 22.1 (6.8.0-56-generic) Alsa Version: 1.0.25+dfsg-0ubuntu7 Pipewire: 1.0.5-1ubuntu2 Wireplumber: 0.4.17-1ubuntu4 The USB Dac has this Ports: 2x Line IN (Cinch) 2x Line OUT or Pre OUT(Cinch) 1x Digital IN TOSLINK 1x COAX/RCA IN (Yellow Cinch) The Problem is: Alsa didnt find any of this IN Ports. There are 4 OUTPUTS: Analog Stereo works fine with Alsa Digital IEC958 (thats wrong Device has no Digital Output) Digital Surround 51 (thats wrong Device has no Digital Output) Pro Audio works fine with Alsa Would be nice to have the Inputs. Heres my Alsa-Info Output: [alsa-info.txt](https://github.com/user-attachments/files/18907991/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/439 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 21 Feb '25

21 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

2 1

[PATCH] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2
by Dmitry Panchenko | d-Systems 20 Feb '25

20 Feb '25

From: Dmitry Panchenko <dmitry(a)d-systems.ee> Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This device does not not work without setting sample-rate. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a97efb7b131e..09210fb4ac60 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1868,6 +1868,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; -- 2.43.0

2 1

[PATCH v2 1/4] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Feb '25

20 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - No further update v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 5

Problems with async handler
by GitHub issues - edited 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was edited from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 434. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Problems with async handler
by GitHub issues - opened 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was opened from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 441. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support
by Dmitry Panchenko | d-Systems 19 Feb '25

19 Feb '25

Hello, Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB interfaces. Both have a MIDI controller, 10 playback and 12 capture channels. I've submitted Pioneer DJM-900NXS2 support patches back in 2020, but someone has removed set_format_quirk function for DJM-900NXS2 model. The playback and capture do not work, quirks.c needs to be patched. The patch is attached, please add. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> -- Kind regards, Dmitry Panchenko

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related commands output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 inxi -A Audio: Device-1: Intel Alder Lake PCH-P High Definition Audio driver: sof-audio-pci-intel-tgl Device-2: NVIDIA GA104 High Definition Audio driver: snd_hda_intel API: ALSA v: k6.13.2-arch1-1 status: kernel-api Server-1: PipeWire v: 1.2.7 status: active Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd\n [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: ` sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - opened 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was opened from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RESEND 0/7] Initial Support for CS40L26
by Fred Treven 19 Feb '25

19 Feb '25

Introduce driver for Cirrus Logic Device CS40L26: A boosted haptics driver with integrated DSP and waveform memory with advanced closed loop algorithms and LRA protection. The core CS40L26 driver is in MFD and touches the Input Force Feedback subsystem for haptics and the ASoC subsystem for audio to haptics streaming. This patchset includes changes to the CS DSP firmware driver which fixes two bugs and introduces support for multiple coefficient files. Fred Treven (7): firmware: cs_dsp: Fix error checking in wseq_write() firmware: cs_dsp: Check for valid num_regs in cs_dsp_wseq_multi_write() firmware: cs_dsp: Add ability to load multiple coefficient files dt-bindings: mfd: cirrus,cs40l26: Support for CS40L26 mfd: cs40l26: Add support for CS40L26 core driver ASoC: cs40l26: Support I2S streaming to CS40L26 Input: cs40l26 - Add support for CS40L26 haptic driver .../bindings/mfd/cirrus,cs40l26.yaml | 81 + MAINTAINERS | 4 +- drivers/firmware/cirrus/cs_dsp.c | 70 +- drivers/input/misc/Kconfig | 10 + drivers/input/misc/Makefile | 1 + drivers/input/misc/cs40l26-vibra.c | 669 ++++++++ drivers/mfd/Kconfig | 29 + drivers/mfd/Makefile | 4 + drivers/mfd/cs40l26-core.c | 1412 +++++++++++++++++ drivers/mfd/cs40l26-i2c.c | 63 + drivers/mfd/cs40l26-spi.c | 63 + include/linux/firmware/cirrus/cs_dsp.h | 14 + include/linux/mfd/cs40l26.h | 341 ++++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs40l26-codec.c | 523 ++++++ 16 files changed, 3281 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/cirrus,cs40l26.yaml create mode 100644 drivers/input/misc/cs40l26-vibra.c create mode 100644 drivers/mfd/cs40l26-core.c create mode 100644 drivers/mfd/cs40l26-i2c.c create mode 100644 drivers/mfd/cs40l26-spi.c create mode 100644 include/linux/mfd/cs40l26.h create mode 100644 sound/soc/codecs/cs40l26-codec.c -- 2.34.1

6 17

[PATCH] ASoC: amd: ps: fix inconsistent indenting warning in check_and_handle_sdw_dma_irq()
by Vijendar Mukunda 18 Feb '25

18 Feb '25

Fix below inconsistent indenting smatch warning. smatch warnings: sound/soc/amd/ps/pci-ps.c:68 check_and_handle_sdw_dma_irq() warn: inconsistent indenting Fixes: 4516be370ced ("ASoC: amd: ps: refactor soundwire dma interrupt handling") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202502132134.BlkNw1Iq-lkp@intel.com/ Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 2ff8e67c19bd..221c65ff03c9 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -129,7 +129,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); - adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; + adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; sdw_dma_irq_flag = 1; } if (ext_intr_stat1 & ACP63_P1_AUDIO1_TX_THRESHOLD) { -- 2.34.1

2 1

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - edited 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was edited from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id `0db0:cd0e`) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - opened 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was opened from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id 0db0:cd0e) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 17 Feb '25

17 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 0f1e585635bb..f2eda8ac66e8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -170,6 +177,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -195,6 +203,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 3

[PATCH alsa-lib] seq: Fix bogus return of snd_seq_client_info_get_ump_conversion()
by Takashi Iwai 17 Feb '25

17 Feb '25

snd_seq_client_info_get_ump_conversion() should have returned the proper bit of group_filter bit field, but it just did return midi_version field -- a stupid copy & paste error. Let's fix it. Fixes: 2aefb5c41cc0 ("seq: Add UMP support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/seq/seq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/seq/seq.c b/src/seq/seq.c index d3af93f58579..38cb1afd872f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1870,7 +1870,7 @@ int snd_seq_client_info_get_ump_groupless_enabled(const snd_seq_client_info_t *i int snd_seq_client_info_get_ump_conversion(const snd_seq_client_info_t *info) { assert(info); - return info->midi_version; + return !(info->group_filter & SNDRV_SEQ_FILTER_NO_CONVERT); } /** -- 2.43.0

1 0

Ask snd_seq_set_client_midi_version
by Correo Alternativo 17 Feb '25

17 Feb '25

Hi, I'm trying out MIDI 2.0 and found a problem, when using snd_seq_set_client_midi_version With SND_SEQ_CLIENT_LEGACY_MIDI I get the following sequence of events when connecting a midi device (using the method snd_seq_event_input or snd_seq_ump_event_input) (timidity command:timidity --interface=A --buffer-fragments=2,8 --output-mode=s1l --sampling-freq=44100): SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START But with SND_SEQ_CLIENT_UMP_MIDI_1_0 and SND_SEQ_CLIENT_UMP_MIDI_2_0 I don't get the SND_SEQ_EVENT_PORT_START events, I only get the following: SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM I don't know if I'm asking this in the right place, what mistake am I making? Just by modifying that it stops working.

3 3

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18814815/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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