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sof-soundwire: cs42l43: Switch mixer based on output volume
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-ucm-conf pull request #540 was opened from mstrozek: Hardware volume control in cs42l43 can go up to +31.5dB which can cause audio degradation. For best results, limit hardware volume to 0dB. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4622 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/540 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/540.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.15-merge-window
by Mark Brown 02 Apr '25

02 Apr '25

The following changes since commit 9ef52d529bb75071e03cf85078f724d69c4abe89: ASoC: SDCA: Correct handling of selected mode DisCo property (2025-03-21 15:45:25 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.15-merge-window for you to fetch changes up to 93d34608fd162f725172e780b1c60cc93a920719: ASoC: imx-card: Add NULL check in imx_card_probe() (2025-04-01 17:01:10 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.15 A relatively large set of fixes that came in since the release, mostly for Qualcomm platforms. The biggest block of fixes is the set from Srini which fixes various quality and glitching issues on AudioReach systems. ---------------------------------------------------------------- Alexey Klimov (1): ASoC: qdsp6: q6asm-dai: fix q6asm_dai_compr_set_params error path Christophe JAILLET (1): ASoC: codecs: rt5665: Fix some error handling paths in rt5665_probe() Dan Carpenter (1): ASoC: sma1307: Fix error handling in sma1307_setting_loaded() Henry Martin (1): ASoC: imx-card: Add NULL check in imx_card_probe() Krzysztof Kozlowski (2): ASoC: codecs: wsa883x: Correct VI sense channel mask ASoC: codecs: wsa884x: Correct VI sense channel mask Mark Brown (2): ASoC: codecs: wsa88xx: Correct VI sense channel mask ASoC: q6apm: fix under runs and fragment sizes Peter Ujfalusi (1): ASoC: SOF: hda/ptl: Move mic privacy change notification sending to a work Richard Fitzgerald (1): firmware: cs_dsp: Ensure cs_dsp_load[_coeff]() returns 0 on success Srinivas Kandagatla (5): ASoC: q6apm-dai: schedule all available frames to avoid dsp under-runs ASoC: q6apm: add q6apm_get_hw_pointer helper ASoC: qdsp6: q6apm-dai: set 10 ms period and buffer alignment. ASoC: qdsp6: q6apm-dai: fix capture pipeline overruns. ASoC: q6apm-dai: make use of q6apm_get_hw_pointer drivers/firmware/cirrus/cs_dsp.c | 2 ++ sound/soc/codecs/rt5665.c | 24 +++------------- sound/soc/codecs/sma1307.c | 11 ++------ sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- sound/soc/fsl/imx-card.c | 4 +++ sound/soc/qcom/qdsp6/q6apm-dai.c | 60 +++++++++++++++++++++------------------- sound/soc/qcom/qdsp6/q6apm.c | 18 +++++++++++- sound/soc/qcom/qdsp6/q6apm.h | 3 ++ sound/soc/qcom/qdsp6/q6asm-dai.c | 19 +++++++++---- sound/soc/sof/intel/hda-dsp.c | 8 ++++++ sound/soc/sof/intel/hda.c | 4 +++ sound/soc/sof/intel/hda.h | 8 ++++++ sound/soc/sof/intel/ptl.c | 33 +++++++++++++++++++--- 14 files changed, 128 insertions(+), 70 deletions(-)

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Support mute led function for HP platform
by Kailang 02 Apr '25

02 Apr '25

Hi Takashi, Attach was fixed mute led no function with HP platform. Many Thanks. BR, Kailang

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aplay: add support for G.711 A_LAW enconding in AU file format
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-utils pull request #297 was opened from chleroy: At the time being, aplay only supports G.711 µ-law, 8-bits PCM and 16-bits PCM encodings for the AU format while the AU format supports many more. In Europe the most G.711 used encoding is the A-law, which is also part of the encodings supported by the AU file format. Add support for that encoding in aplay. Request URL : https://github.com/alsa-project/alsa-utils/pull/297 Patch URL : https://github.com/alsa-project/alsa-utils/pull/297.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH alsa-ucm-conf 1/2] sof-soundwire: cs42l43: Drop headset mic from mic mute LED
by Maciej Strozek 01 Apr '25

01 Apr '25

As per conversation in [1], let headsets be dropped from LED bindings. Also remove turning off the mics on device disable since they are now not needed. [1]: https://github.com/alsa-project/alsa-ucm-conf/pull/512 Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43/init.conf | 6 ------ ucm2/sof-soundwire/cs42l43-dmic.conf | 2 -- ucm2/sof-soundwire/cs42l43.conf | 1 - 3 files changed, 9 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 0e987c21..c0fa863c 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -10,9 +10,3 @@ LibraryConfig.remap.Config { } } } - -BootSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" -] - -Macro [{ SetLED { LED="mic" Action="attach" CtlId="cs42l43 Decimator 1 Switch" } }] diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index 70ce40ff..b68a70d1 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -13,8 +13,6 @@ SectionDevice."Mic" { ] DisableSequence [ - cset "name='cs42l43 Decimator 3 Switch' 0" - cset "name='cs42l43 Decimator 4 Switch' 0" cset "name='cs42l43 DP1TX1 Input' 'None'" cset "name='cs42l43 DP1TX2 Input' 'None'" ] diff --git a/ucm2/sof-soundwire/cs42l43.conf b/ucm2/sof-soundwire/cs42l43.conf index 4b4e1187..51832126 100644 --- a/ucm2/sof-soundwire/cs42l43.conf +++ b/ucm2/sof-soundwire/cs42l43.conf @@ -33,7 +33,6 @@ SectionDevice."Headset" { ] DisableSequence [ - cset "name='cs42l43 Decimator 1 Switch' 0" cset "name='cs42l43 DP2TX1 Input' 'None'" cset "name='cs42l43 DP2TX2 Input' 'None'" ] -- 2.39.5

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Difference in behaviour between physical devices and virtual endpoints
by GitHub issues - opened 31 Mar '25

31 Mar '25

alsa-project/alsa-lib issue #447 was opened from reuk: As a disclaimer, there's a good chance this is user error/misunderstanding rather than a bug, but I'd appreciate any insights to help me understand the behaviour I'm seeing. # Environment - Fedora 41 x86_64 - alsa-lib-devel-1.2.13-3 # Steps - Run `aseqdump -u 1` in a terminal - Connect ProtoZOA hardware via USB - Launch QjackCtl -> Graph, then connect the ProtoZOA's MIDI 2.0 output port (0) to `aseqdump`'s input port. <img width="535" alt="Image" src="https://github.com/user-attachments/assets/17bf6a74-f83d-41b4-afc6-05413a7a…" /> - Observe that `aseqdump` reports a new subscription from `24:0 -> 128:0` (the client IDs may vary, the important thing is that the subscription is between to ports with ID `0`). - Move one of the controllers on the ProtoZOA. Observe that `aseqdump` reports channel voice events on group 0 with source `24:1`. <img width="502" alt="Image" src="https://github.com/user-attachments/assets/2dfc5929-123f-4e33-abbb-ea0c2635…" /> - Connect `aseqdump` to port `1:Group 1 (FB1, FB2)`, and interact with the ProtoZOA. Observe that `aseqdump` still reports channel voice events on group 0 with source `24:1`. - Now, build and run the [seq-ump-example](https://github.com/alsa-project/alsa-lib/blob/master/test/…. - Connect the ProtoZOA to the ACMESynth input, and the ACMESynth output to `aseqdump` as shown. Observe that `aseqdump` does **not** report channel voice events when interacting with the ProtoZOA. <img width="790" alt="Image" src="https://github.com/user-attachments/assets/71349cd6-14b8-4df9-8ee6-ca1091c2…" /> - Finally, update the connections as shown, so that `aseqdump` is connected to ACMESynth's `1:Group 1 (Filter 1)` output. Observe that `aseqdump` now reports channel voice events with source `128:1` when interacting with the ProtoZOA. <img width="772" alt="Image" src="https://github.com/user-attachments/assets/57f4e767-b2d6-4156-8514-578851d5…" /> # TL;DR When we subscribe to a USB `MIDI 2.0` port, we receive messages from all of the endpoint's ports. Messages are dispatched to the appropriate port based on the message group. When we subscribe to a virtual UMP endpoint's `MIDI 2.0` port, this appears to *only* subscribe to that single port, and not to the rest of the ports on the endpoint, so that messages addressed to specific groups get lost. # Questions - Is this difference in behaviour between hardware and virtual endpoints intentional? - What is the "correct" way to subscribe to all UMP events from an endpoint? Should I subscribe just to the `MIDI 2.0` endpoint, or to all ports on the endpoint, or something else? Issue URL : https://github.com/alsa-project/alsa-lib/issues/447 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH alsa-ucm-conf v1] sof-soundwire: Add LED support for cs35l56 amplifiers
by Stefan Binding 31 Mar '25

31 Mar '25

Systems may have up to 8 CS35L56 amplifiers, and each amplifier should be attached to the speaker LED. A mapped Playback switch is required to ensure all amps are muted and unmute appropriately. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- ucm2/codecs/cs35l56-bridge/init.conf | 11 ++++++++ ucm2/codecs/cs35l56/init.conf | 36 +++++++++++++++++++++++++++ ucm2/sof-soundwire/cs35l56.conf | 32 ++---------------------- ucm2/sof-soundwire/sof-soundwire.conf | 4 +-- 4 files changed, 51 insertions(+), 32 deletions(-) create mode 100644 ucm2/codecs/cs35l56-bridge/init.conf create mode 100644 ucm2/codecs/cs35l56/init.conf diff --git a/ucm2/codecs/cs35l56-bridge/init.conf b/ucm2/codecs/cs35l56-bridge/init.conf new file mode 100644 index 000000000000..407bb0937559 --- /dev/null +++ b/ucm2/codecs/cs35l56-bridge/init.conf @@ -0,0 +1,11 @@ +# cs35l56 specific control settings + +FixedBootSequence [ + cset "name='AMPL Speaker Switch' 0" + cset "name='AMPR Speaker Switch' 0" +] + +Macro [ + { SetLED { LED="speaker" Action="attach" CtlId="AMPL Speaker Switch" } } + { SetLED { LED="speaker" Action="attach" CtlId="AMPR Speaker Switch" } } +] \ No newline at end of file diff --git a/ucm2/codecs/cs35l56/init.conf b/ucm2/codecs/cs35l56/init.conf new file mode 100644 index 000000000000..dca14c709fda --- /dev/null +++ b/ucm2/codecs/cs35l56/init.conf @@ -0,0 +1,36 @@ +# cs35l56 specific control settings + +# +# Arguments: +# ForAmps - regex filter for var:SpeakerAmps +# Amp - amplifier number 1-8 +# +DefineMacro.cs35l56spkInit.If.0 { + Condition { + Type RegexMatch + Regex "${var:__ForAmps}" + String "${var:SpeakerAmps}" + } + True { + LibraryConfig.remap.Config.ctl.default.map { + "name='cs35l56 Speaker Playback Switch'" { + "name='AMP${var:__Amp} Speaker Switch'".vindex."${eval:($__Amp-1)}" 0 + } + } + FixedBootSequence [ + cset "name='AMP${var:__Amp} Speaker Switch' 0" + ] + Macro [{ SetLED { LED="speaker" Action="attach" CtlId="AMP${var:__Amp} Speaker Switch" } }] + } +} + +Macro [ + { cs35l56spkInit { ForAmps "[1-8]" Amp 1 } } + { cs35l56spkInit { ForAmps "[2-8]" Amp 2 } } + { cs35l56spkInit { ForAmps "[3-8]" Amp 3 } } + { cs35l56spkInit { ForAmps "[4-8]" Amp 4 } } + { cs35l56spkInit { ForAmps "[5-8]" Amp 5 } } + { cs35l56spkInit { ForAmps "[6-8]" Amp 6 } } + { cs35l56spkInit { ForAmps "[78]" Amp 7 } } + { cs35l56spkInit { ForAmps "8" Amp 8 } } +] diff --git a/ucm2/sof-soundwire/cs35l56.conf b/ucm2/sof-soundwire/cs35l56.conf index f74b99afbb55..53e4d6dd8eda 100644 --- a/ucm2/sof-soundwire/cs35l56.conf +++ b/ucm2/sof-soundwire/cs35l56.conf @@ -1,40 +1,12 @@ # Use case Configuration for sof-soundwire card -# -# Arguments: -# ForAmps - regex filter for var:SpeakerAmps -# Amp - amplifier number 1-8 -# -DefineMacro.cs42l43spk.If.0 { - Condition { - Type RegexMatch - Regex "${var:__ForAmps}" - String "${var:SpeakerAmps}" - } - True { - EnableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 1" - ] - DisableSequence [ - cset "name='AMP${var:__Amp} Speaker Switch' 0" - ] - } -} - SectionDevice."Speaker" { Comment "Speaker" - Macro.num1.cs42l43spk { ForAmps "[1-8]" Amp 1 } - Macro.num2.cs42l43spk { ForAmps "[2-8]" Amp 2 } - Macro.num3.cs42l43spk { ForAmps "[3-8]" Amp 3 } - Macro.num4.cs42l43spk { ForAmps "[4-8]" Amp 4 } - Macro.num5.cs42l43spk { ForAmps "[5-8]" Amp 5 } - Macro.num6.cs42l43spk { ForAmps "[6-8]" Amp 6 } - Macro.num7.cs42l43spk { ForAmps "[78]" Amp 7 } - Macro.num8.cs42l43spk { ForAmps "8" Amp 8 } - Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId},2" + PlaybackMixer "default:${CardId}" + PlaybackMixerElem "cs35l56 Speaker" } } diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 1cddfbf4cfbf..8f9bc82c2d3b 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -102,10 +102,10 @@ If.multi_headset { } } -If.spk_init_rt1318 { +If.spk_init { Condition { Type RegexMatch - Regex "(rt1318(-1)?)" + Regex "(rt1318(-1)?|cs35l56(-bridge)?)" String "${var:SpeakerCodec1}" } True.Include.spk_init.File "/codecs/${var:SpeakerCodec1}/init.conf" -- 2.43.0

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USB-Audio: Add focusrite scarlett 18i20 lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #539 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. IO includes XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/539.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - edited 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was edited from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 1st gen, 2nd gen, 3rd gen, and 4th gen. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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USB-Audio: Add Focusrite Scarlett 18i20 Lineup
by GitHub pull_request - opened 30 Mar '25

30 Mar '25

alsa-project/alsa-ucm-conf pull request #538 was opened from ScarletRav3n: Other solutions exist to add full mixing and routing with these devices. This PR adds basic functionality in an out-of-the-box style for 2nd gen, 3rd gen, and 4th gen. 1st gen is not supported with this iteration of PR. I/O includes all XLR, Line, S/PDIF, ADAT, and Headphone ports. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/538.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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