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[PATCH 00/14] Implement acp_common_hw_ops support for all platforms
by Venkata Prasad Potturu 10 Mar '25

10 Mar '25

This patch series is to implement acp_common_hw_ops support and refactor interrupt handler registration and interrupt handling. Venkata Prasad Potturu (14): ASoC: amd: acp: Remove redundant acp70 chip->name ASoC: amd: acp: Implement acp_common_hw_ops support for acp platforms ASoC: amd: acp: Refactor dmic-codec platform device creation ASoC: amd: acp: Refactor acp platform device creation ASoC: amd: acp: Refactor acp machine select ASoC: amd: acp: Add new interrupt handle callbacks in acp_common_hw_ops ASoC: amd: acp: Remove redundant acp_dev_data structure ASoC: amd: acp: Move spin_lock and list initialization to acp-pci driver ASoC: amd: acp: Remove white line ASoC: amd: acp: Refactor acp70 platform resource structure ASoC: amd: acp: Refactor acp63 platform resource structure ASoC: amd: acp: Refactor rembrant platform resource structure ASoC: amd: acp: Refactor renoir platform resource structure ASoC: amd: acp: Fix for enabling DMIC on acp platforms via _DSD entry sound/soc/amd/acp/acp-i2s.c | 185 +++++++++--------- sound/soc/amd/acp/acp-legacy-common.c | 254 ++++++++++++++++++------ sound/soc/amd/acp/acp-pci.c | 210 ++++++++++++-------- sound/soc/amd/acp/acp-pdm.c | 53 +++-- sound/soc/amd/acp/acp-platform.c | 145 ++++---------- sound/soc/amd/acp/acp-rembrandt.c | 111 +++-------- sound/soc/amd/acp/acp-renoir.c | 110 ++--------- sound/soc/amd/acp/acp63.c | 94 +++------ sound/soc/amd/acp/acp70.c | 84 +++----- sound/soc/amd/acp/amd.h | 270 +++++++++++++++++++++----- sound/soc/amd/acp/chip_offset_byte.h | 12 +- 11 files changed, 792 insertions(+), 736 deletions(-) -- 2.39.2

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[PATCH alsa-ucm-conf] sof-soundwire: Split cs42l43 dmic initialisation
by Charles Keepax 10 Mar '25

10 Mar '25

Split the initialisation of cs42l43 into a headset part and a dmic part. This will prevent the dmic mute controls getting attached to the mute LED when the codec dmics are not being used. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43-dmic/init.conf | 23 +++++++++++++++++++++++ ucm2/codecs/cs42l43/init.conf | 11 +---------- ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 3 files changed, 25 insertions(+), 11 deletions(-) create mode 100644 ucm2/codecs/cs42l43-dmic/init.conf diff --git a/ucm2/codecs/cs42l43-dmic/init.conf b/ucm2/codecs/cs42l43-dmic/init.conf new file mode 100644 index 0000000..441d082 --- /dev/null +++ b/ucm2/codecs/cs42l43-dmic/init.conf @@ -0,0 +1,23 @@ +# cs42l43 specific control settings + +LibraryConfig.remap.Config { + ctl.default.map { + "name='cs42l43 Microphone Capture Switch'" { + "name='cs42l43 Decimator 3 Switch'".vindex.0 0 + "name='cs42l43 Decimator 4 Switch'".vindex.1 0 + } + "name='cs42l43 Microphone Capture Volume'" { + "name='cs42l43 Decimator 3 Volume'".vindex.0 0 + "name='cs42l43 Decimator 4 Volume'".vindex.1 0 + } + } +} + +FixedBootSequence [ + cset "name='cs42l43 Decimator 3 Switch' 0" + cset "name='cs42l43 Decimator 4 Switch' 0" + exec "-/sbin/modprobe snd_ctl_led" + sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 3 Switch" + sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 4 Switch" +] + diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0fbb517..8e41a0f 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,14 +2,6 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 Microphone Capture Switch'" { - "name='cs42l43 Decimator 3 Switch'".vindex.0 0 - "name='cs42l43 Decimator 4 Switch'".vindex.1 0 - } - "name='cs42l43 Microphone Capture Volume'" { - "name='cs42l43 Decimator 3 Volume'".vindex.0 0 - "name='cs42l43 Decimator 4 Volume'".vindex.1 0 - } "name='cs42l43 Headset Microphone Capture Switch'" { "name='cs42l43 Decimator 1 Switch'".vindex.0 0 } @@ -20,8 +12,7 @@ LibraryConfig.remap.Config { } FixedBootSequence [ + cset "name='cs42l43 Decimator 1 Switch' 0" exec "-/sbin/modprobe snd_ctl_led" sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 1 Switch" - sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 3 Switch" - sysw "-/class/sound/ctl-led/mic/card${CardNumber}/attach:cs42l43 Decimator 4 Switch" ] diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 96d2966..57e5610 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -114,7 +114,7 @@ If.hs_init { If.mic_init { Condition { Type RegexMatch - Regex "(rt712-dmic|rt713-dmic|rt715(-sdca)?)" + Regex "(cs42l43-dmic|rt712-dmic|rt713-dmic|rt715(-sdca)?)" String "${var:MicCodec1}" } True.Include.mic_init.File "/codecs/${var:MicCodec1}/init.conf" -- 2.39.5

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Channel count in Focusrite Scarlett Gen 4 (1235:8218) SplitPCM configuration
by GitHub issues - edited 09 Mar '25

09 Mar '25

alsa-project/alsa-ucm-conf issue #515 was edited from pv: The configuration https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/Fo… specifies `HWChannels 2` for the Mic1 and Mic2 devices. This gets translated to `CaptureChannels 2` for `alsaucm open '<<<SplitPCM=1>>>hw:2'`. However, we've got a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4597#note_2815906 that looks like snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) for the CapturePCM "_ucm0001.hw:Gen,0" fails with error. The PCM can be opened with 4 channels, though. Without `SplitPCM=1`, it also probably works but I'm not sure what channel count the `dshare` plugin is using. The issue also affects the `Direct` profile, which also specifies `CaptureChannels 2`. Is the `HWChannels 2` for Mic1/2 input direction right? Shall `CaptureChannels 2` in UCM device be interpreted as that the device can be opened with exactly 2 channels, or that it can be opened with some number of channels >= 2? <details> ``` $ alsaucm open '<<<SplitPCM=1>>>hw:2' dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:Gen" PlaybackCTL "_ucm0001.hw:Gen" PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 0 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:Gen" } } Device.Mic2 { Comment "Input 2 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 1 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 100 PlaybackCTL "_ucm0001.hw:Gen" } } } Verb.Direct { Comment "Direct Scarlett Solo 4th Gen" Device.Direct { Comment "Direct Scarlett Solo 4th Gen" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:Gen" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 1000 } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/515 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Channel count in Focusrite Scarlett Gen 4 (1235:8218) SplitPCM configuration
by GitHub issues - opened 09 Mar '25

09 Mar '25

alsa-project/alsa-ucm-conf issue #515 was opened from pv: The configuration https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/USB-Audio/Fo… specifies `HWChannels 2` for the Mic1 and Mic2 devices. This gets translated to `CaptureChannels 2` for `alsaucm open '<<<SplitPCM=1>>>hw:2'`. However, we've got a report https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4597#note_2815906 that looks like snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 2) for the CapturePCM "_ucm0001.hw:Gen,0" fails with error. The PCM can be opened with 4 channels, though. Without `SplitPCM=1`, it also probably works but I'm not sure what channel count the `dshare` plugin is using. Is the `HWChannels 2` for Mic1/2 input direction right? Shall `CaptureChannels 2` in UCM device be interpreted as that the device can be opened with exactly 2 channels, or that it can be opened with some number of channels >= 2? <details> ``` $ alsaucm open '<<<SplitPCM=1>>>hw:2' dump text Verb.HiFi { Comment Default Device.Line1 { Comment "Headphones / Line 1-2" Values { CaptureCTL "_ucm0001.hw:Gen" PlaybackCTL "_ucm0001.hw:Gen" PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 200 } } Device.Mic1 { Comment "Input 1 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 0 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 200 PlaybackCTL "_ucm0001.hw:Gen" } } Device.Mic2 { Comment "Input 2 Mic/Inst/Line" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannel0 1 CaptureChannelPos0 MONO CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen,0" CapturePriority 100 PlaybackCTL "_ucm0001.hw:Gen" } } } Verb.Direct { Comment "Direct Scarlett Solo 4th Gen" Device.Direct { Comment "Direct Scarlett Solo 4th Gen" Values { CaptureCTL "_ucm0001.hw:Gen" CaptureChannels 2 CapturePCM "_ucm0001.hw:Gen" CapturePriority 1000 PlaybackCTL "_ucm0001.hw:Gen" PlaybackChannels 2 PlaybackPCM "_ucm0001.hw:Gen" PlaybackPriority 1000 } } } ``` </details> Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/515 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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RE: [PATCH v5 2/2] ASoC: dt-bindings: Add Everest ES8389 audio CODEC
by Zhang Yi 09 Mar '25

09 Mar '25

> > 1 file changed, 50 insertions(+) > > create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml > > > > diff --git a/Documentation/devicetree/bindings/sound/everest,es8389.yaml b/Documentation/devicetree/bindings/sound/everest,es8389.yaml > > > You got feedback how to create CC list. Two times. > > You still create it incorrectly and Cc address which does not exist in > the kernel and is a proof you work on wrong tree, which I asked to correct. > > What's more, there is still no changelog and you never responded to > review comments. > > Can you slow down with resubmits and rethink the received feedback? I > expect that feedback to be responded to, so I am not wasting time here > repeating the same every time. I'm sorry for the trouble I've caused you. And I have responded to the comments at v3 and v4.

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PORT: GTK2 to GTK3
by GitHub pull_request - opened 07 Mar '25

07 Mar '25

alsa-project/alsa-tools pull request #27 was opened from genbtc: I've undertaken a task of porting alsa-tools programs from GTK2 to GTK3. These patches start with a base of 5eaadd15bda4cf0424ee2470a6d304a0c2ce10cb (master) Date: Mon Jan 27 (latest pull) Results: Compilation completes successfully, Linked to GTK3, but I do not have the hardware cards to test run them. All work has been done by me, and I request a code review on the GDK 2 Pixmap to GDK 3 Pixbuf + Cairo surfaces. Like it needs new logic and new plan. not simply change statement A to B (whatever I did to get it to compile) I tried to comply with the existing style, and There was also a fair bit of inconsequential whitespace removed like (\n\t\n) or trailing space chars as I discovered them. Looking forward to upstreaming these changes somehow if you would like to work on this. - genBTC @ gentoo Request URL : https://github.com/alsa-project/alsa-tools/pull/27 Patch URL : https://github.com/alsa-project/alsa-tools/pull/27.patch Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH v5 0/2] ASoC: codecs: add support for ES8389
by Zhang Yi 07 Mar '25

07 Mar '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.yaml | 50 + sound/soc/codecs/Kconfig | 6 +- sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 966 ++++++++++++++++++ sound/soc/codecs/es8389.h | 140 +++ 5 files changed, 1163 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/everest,es8389.yaml create mode 100644 sound/soc/codecs/es8389.c create mode 100644 sound/soc/codecs/es8389.h -- 2.17.1

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RE: [PATCH v4 1/2] ASoC: codecs: add support for ES8389
by Zhang Yi 07 Mar '25

07 Mar '25

> > > No, the machine driver should be configuring different TDM slots for > > > each device - that's the whole point of the API. > > > We are using multiple codecs as a single device.So we can't use set_tdm_slot to configure > > different slots for multiple codecs under one device. > > What do you mean by using it "as a single device"? Multiple CODECs on > the same link is the main use case for set_tdm_slot(). Thanks for the advice.The method you mentioned could work. I will modify my driver.

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regresion: snd_hdsp
by GitHub issues - edited 07 Mar '25

07 Mar '25

alsa-project/alsa-firmware issue #11 was edited from juanpc2018: ![Image](https://github.com/user-attachments/assets/e57192a7-ad1c-44f9-b51d-7806efed9a32) ``` $ pulseaudio W: [pulseaudio] module.c: module-combine is deprecated: Please use module-combine-sink instead of module-combine! W: [pulseaudio] module-combine.c: We will now load module-combine-sink. Please make sure to remove module-combine from your configuration. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hdsp'. Please report this issue to the ALSA developers. E: [alsa-sink-RME Hammerfall HDSP 9632] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` when i play any video or sound file, most of the time PulseAudio takes between 7 to 15 seconds to fill the buffer in proper order, audio is played in wrong order, chopped, like [stutter](https://www.youtube.com/watch?v=9ZaDn5wne8Y&t=609s).[2](https://ww… i have tested /etc/pulse/deamon.conf s24le to s24be problem persists. forced sample rate to 48Khz, APIC ACPI Tables v2 & v3 in Bios, Audacity 3.6. ALSA hw:0 also Crash. but Jack Audio works ok. i remember hdsp9632 worked ok on U/Kubuntu 10.04 & 10.10 Now 20.04.4 LTS has problems. im using Liquorix 6.3 Kernel. ``` $ sudo apt show alsa-firmware* Package: alsa-firmware-loaders Version: 1.1.7-1ubuntu1 Priority: extra Section: multiverse/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 211 kB Depends: fxload, udev, libasound2 (>= 1.0.24.1), libc6 (>= 2.4) Homepage: https://www.alsa-project.org/ Download-Size: 25,0 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/multiverse amd64 Packages Description: ALSA software loaders for specific hardware $ sudo apt show alsa-tools-gui Package: alsa-tools-gui Version: 1.1.7-1ubuntu1 Priority: extra Section: universe/sound Source: alsa-tools Origin: Ubuntu Maintainer: Ubuntu Developers <ubuntu-devel-discuss(a)lists.ubuntu.com> Original-Maintainer: Debian ALSA Maintainers <pkg-alsa-devel(a)lists.alioth.debian.org> Bugs: https://bugs.launchpad.net/ubuntu/+filebug Installed-Size: 1.002 kB Depends: libasound2 (>= 1.0.24.1), libc6 (>= 2.29), libfltk1.1 (>= 1.1.6), libglib2.0-0 (>= 2.35.9), libgtk-3-0 (>= 3.0.0), libgtk2.0-0 (>= 2.8.0), libstdc++6 (>= 5.2) Homepage: https://www.alsa-project.org/ Task: ubuntustudio-audio Download-Size: 220 kB APT-Manual-Installed: yes APT-Sources: http://archive.ubuntu.com/ubuntu focal/universe amd64 Packages Description: GUI based ALSA utilities for specific hardware ``` ---------------------------------- found another way to trigger the bug.... if i play a video on YT, the first 7 seconds are corrupted, but if in the middle of the video i pause, and wait 7 seconds, and continue to play, the corrupted 7 seconds happen again. Issue URL : https://github.com/alsa-project/alsa-firmware/issues/11 Repository URL: https://github.com/alsa-project/alsa-firmware

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HDSPconf missing setting...
by GitHub issues - edited 06 Mar '25

06 Mar '25

alsa-project/alsa-tools issue #26 was edited from juanpc2018: $ alsamixer v1.2.2 ![Image](https://github.com/user-attachments/assets/6b2860d4-73a0-4526-9a97-13a7c5d6785a) ![Image](https://github.com/user-attachments/assets/ec8253d0-627a-4dda-a45e-1a2080d20563) HDSPconf 1.4 from alsa-tools-gui ![Image](https://github.com/user-attachments/assets/d2d1dba5-8da6-43ae-a594-cfd89bdd7e1f) ![Image](https://github.com/user-attachments/assets/17d070d3-067e-4d1b-adf9-131f843a634c) i dont know why is called DDS Sample Offset but it should be like HDSPmixer ![Image](https://github.com/user-attachments/assets/3ac4ca5c-df61-4cb3-8e8d-2af8fc44019c) DDS "Sample Offset" is similar to a PLL with a more aggressive rejection of the external clock signal. IF external clock coming from s/pdif is better vs. RME DDS 200pico second jitter clock, worsens the clock. like a blend control: 0% preference for internal DDS clock = 100% preference for external clock, 100% preference for internal DDS clock = 0% preference for external clock. at: [100% DDS](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=27s) DDS is the Truck, still follows ext.clock, as a recommendation, Not like an [Orchestra](https://www.youtube.com/watch?v=HqCIYWZ5b7k&t=0s)_[Director](htt… 0% DDS, ext.clock becomes like a Whip, internal like a Horse / Slave / [Circus Lion](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=105s) DDS is the Lion. Clock [Jitter](https://www.youtube.com/watch?v=zJLw2SF_L-g&t=152s) Issue URL : https://github.com/alsa-project/alsa-tools/issues/26 Repository URL: https://github.com/alsa-project/alsa-tools

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