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ALC4080 - general discussion (driver support)
by GitHub issues - opened 03 Apr '25

03 Apr '25

alsa-project/alsa-ucm-conf issue #541 was opened from perexg: UCM cannot add basic functionalities when the driver does not expose them. I already explained situation in [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218913). **For issues like** 1. jack functionality assignment is unstable (jack functionality is not fixed - e.g. changes on warm reboot from windows) 2. low microphone volume (which cannot be controlled using native ALSA mixer controls) 3. sound is activated only after some magic plug/unplug sequence **Further steps** Try to analyze the audio driver using direct ALSA API. See [DEBUG.md](https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/DE…. Very nice report is in #528. Please, use it as a reference what to do. Reverse engineering of the Windows driver. Capture USB communication and save it to a file which can be replayed by [wireshark] (https://www.wireshark.org). Try to write an explanation including timing (what happened in given time slots). Driver reports should go to [kernel bugzilla](https://bugzilla.kernel.org). Add a comment with a link here. You may also push hardware vendors to create USB device which fully follows the USB audio specification, so the ALSA driver will work out-of-box (without "quirks"). **Documentation links** The documentation is not complete. Realtek support is generally very limited to provide full specification to their codecs / audio chips those years. They are providing details only to hardware vendors with a contract. If you know more, please, let us know. - [ALC4080 block scheme](https://bugzilla.kernel.org/attachment.cgi?id=306370) - [ALC4080 datasheet](https://bugzilla.kernel.org/attachment.cgi?id=307224) **Some details** ALC4080 is USB codec, thus it exposes functionality through the USB bus. The block scheme is nice to know internals, but look to the MCU paragraph in the datasheet: ``` 2.1.2. Micro Controller Unit - On-chip high-performance and low-power MCU - Ultra-low power consumption when MCU is at idle state - MCU controls connection to USB bus for re-enumeration without hot-plug - Internal programmable memory support for customized audio function - Watchdog control for MCU reset and interrupt - Configurable VID (Vendor ID), PID (Product ID), and serial number string ``` This means, that the internal firmware for this MCU unit can change the behaviour of audio hardware. Usually, the firmware is vendor specific. The USB audio hardware should comply with the USB Audio specification (refer https://www.usb.org/). The control units parsed by the ALSA driver can be obtained using 'cat /proc/asound/cardX/usbmixer' command (replace X with the correct ALSA card number). Those units are mapped to the standard mixer controls. My guess is that the firmware is not complaint with the USB audio specification and some functionality is controlled using specific USB handshake (data blocks). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/541 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Delete alsalisp code
by GitHub pull_request - opened 02 Apr '25

02 Apr '25

alsa-project/alsa-lib pull request #448 was opened from fragglet: Install of the alsalisp binary has been disabled since 2006 (in commit 8d382ccd), and building of it was disabled by default in 2018 (in commit 32ceab21), so it is reasonable to assume that nobody is using it. Use within the alsa-lib project is limited to an aliases file that looks like it is intended as an example, plus some very small .alisp files associated with the SiS SI7018 PCI sound card which has not been manufactured in years. These too have not been installed since 2018 when commit 32ceab21 disabled building of the alsalisp binary. In preparing this change, I searched the Github issue tracker for "lisp", "alisp" and "alsalisp", and found no complaints about the above changes. I also did a Github code search for projects that might be including the `alisp.h` header and found none. Therefore I think this code can be safely deleted and nobody is likely to object. Request URL : https://github.com/alsa-project/alsa-lib/pull/448 Patch URL : https://github.com/alsa-project/alsa-lib/pull/448.patch Repository URL: https://github.com/alsa-project/alsa-lib

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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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[PATCH v2] ASoC: amd: use new ACP dev names for DAI links
by Brady Norander 30 Mar '25

30 Mar '25

On AMD SoC platforms with an ACP2x gpu ip block (such as stoneyridge), the amdgpu driver will create several platform devices for the ACP ASoC driver to communicate with the ACP hardware block on the gpu. These platform devices include dma for audio and one or multiple i2s interfaces. The amdgpu driver has always created these platform devices with automatic ids. The ASoC machine drives hardcode the platform device name. This creates an issue where if the ACP platform devices are not the first to be created, the ids can be different to what the machine drivers expect, causing them to not find the ACP platform devices and failing to load. Switch to using static ids for these ACP platform devices so that the names never change. Depends on patch: drm/amdgpu: use static ids for ACP platform devs [1] [1] https://lore.kernel.org/all/20250325210517.2097188-1-bradynorander@gmail.co… Signed-off-by: Brady Norander <bradynorander(a)gmail.com> --- v2: rewrite commit message to better describe the issue --- sound/soc/amd/acp-da7219-max98357a.c | 8 ++++---- sound/soc/amd/acp-es8336.c | 4 ++-- sound/soc/amd/acp-rt5645.c | 6 +++--- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 02b04f355ca6..42aa009c4e13 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -517,11 +517,11 @@ static const struct snd_soc_ops cz_rt5682_dmic1_cap_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(designware3, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.3"))); SND_SOC_DAILINK_DEF(dlgs, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-DLGS7219:00", "da7219-hifi"))); @@ -533,7 +533,7 @@ SND_SOC_DAILINK_DEF(adau, DAILINK_COMP_ARRAY(COMP_CODEC("ADAU7002:00", "adau7002-hifi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_7219_98357[] = { { diff --git a/sound/soc/amd/acp-es8336.c b/sound/soc/amd/acp-es8336.c index 0193b3eae7a6..b16dde0e2987 100644 --- a/sound/soc/amd/acp-es8336.c +++ b/sound/soc/amd/acp-es8336.c @@ -137,11 +137,11 @@ static const struct snd_soc_ops st_es8336_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ESSX8336:00", "ES8316 HiFi"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.1.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link st_dai_es8336[] = { { diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index 72ddad24dbda..11d373169380 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -108,15 +108,15 @@ static const struct snd_soc_ops cz_aif1_ops = { }; SND_SOC_DAILINK_DEF(designware1, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.1"))); SND_SOC_DAILINK_DEF(designware2, - DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2.auto"))); + DAILINK_COMP_ARRAY(COMP_CPU("designware-i2s.2"))); SND_SOC_DAILINK_DEF(codec, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); SND_SOC_DAILINK_DEF(platform, - DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0.auto"))); + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp_audio_dma.0"))); static struct snd_soc_dai_link cz_dai_rt5650[] = { { -- 2.49.0

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