July 2023 - Alsa-devel - mailman.alsa-project.org

How do I set up multiple codecs on one I2S - without TDM?
by John Watts 01 Aug '23

01 Aug '23

Hello there! I have an interesting problem that I'm trying to solve. It will require writing adding driver support but right now I'm struggling to understand how I should go about it all. As a background, this is with the Allwinner D1 or T113. It has a 16 channel I2S controller and each channel can be configured to use a specific pin and TDM time slice. Even channels are low LRCLK, odd are high LRCLK. So for my situation I want to have six channels: - Channel 0: Timeslot 0, Pin 0, LRCLK 0 -> ADC 1 - Channel 1: Timeslot 0, Pin 0, LRCLK 1 -> ADC 1 - Channel 2: Timeslot 0, Pin 1, LRCLK 0 -> ADC 2 - Channel 3: Timeslot 0, Pin 1, LRCLK 1 -> ADC 2 - Channel 4: Timeslot 0, Pin 2, LRCLK 0 -> ADC 3 - Channel 5: Timeslot 0, Pin 2, LRCLK 1 -> ADC 3 These ADCs are WM8782s, completely TDM unaware. If they were TDM aware then I could run even more ADCs by having multiple separate TDMs going on each pin. I have tried to look online for I2S controllers that do the same thing but haven't had much luck. The current driver for this controller only supports TDM. So I will have to add support for this. So my questions are: - How do I express this in terms of ALSA and sound card concepts? - How do I express this in the device tree? My immediate thought is that this is a situation where I have a card with 3 dai-links between the I2S controller and each separate codec. I have written this tree as a test: wm8782_1: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_2: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_3: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "Testcard In"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_cpu>; simple-audio-card,frame-master = <&sound_cpu>; simple-audio-card,mclk-fs = <128>; #address-cells = <1>; #size-cells = <0>; simple-audio-card,dai-link@0 { reg = <0>; sound_cpu: cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_1>; }; }; simple-audio-card,dai-link@1 { reg = <1>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_2>; }; }; simple-audio-card,dai-link@2 { reg = <2>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_3>; }; }; }; But I get this kernel error: kobject_add_internal failed for 2034000.i2s-wm8782 with -EEXIST, don't try to register things with the same name asoc-simple-card: probe of sound failed with error -12 The WM8782 doesn't support TDM or multiplexing so I'm not exactly sure if this is correct behaviour or my mistake configuring. I tried adding TDM nodes which didn't help. I also tried having a single link with multiple codecs on sound-dai which did create a sound card but it was unclear to me if this was ignoring the other codecs I had listed or not. I also can't test this at this point as the sunxi codec driver assumes that the number of channels is equal to TDM slots, which isn't true any more. Any pointers in the right direction would be a huge help. Surely someone else has had this problem. Thanks, John.

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[PATCH] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 01 Aug '23

01 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- sound/soc/codecs/rt5665.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..de36cad8a559 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4756,6 +4756,13 @@ static void rt5665_calibrate_handler(struct work_struct *work) rt5665_calibrate(rt5665); } +static void rt5665_i2c_disable_regulators(void *data) +{ + struct rt5665_priv *rt5665 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); +} + static int rt5665_i2c_probe(struct i2c_client *i2c) { struct rt5665_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -4786,6 +4793,11 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5665_i2c_disable_regulators, + rt5665); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); if (ret != 0) { -- 2.41.0

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[PATCH 00/23] ASoC: Intel: machine driver updates for 6.6
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

Lots of small cleanups coming from Bard Liao and Charles Keepax for SoundWire platforms, and minor additions for RVPs and Chromebooks. Balamurugan C (1): ASoC: Intel: soc-acpi: Add entry for rt711-sdca-sdw at link 0 in RPL match table Bard Liao (2): ASoC: Intel: sof_sdw: rename link_id to be_id ASoC: Intel: sof_sdw: break earlier when a adr link contains different codecs Chao Song (1): ASoC: Intel: sof_sdw: add cs35l56 codec info Charles Keepax (13): ASoC: intel: sof_sdw: Use consistent variable naming for links ASoC: intel: sof_sdw: Rename codec_idx to codec_dlc_index ASoC: intel: sof_sdw: Remove some extra line breaks ASoC: intel: sof_sdw: Use a module device table ASoC: intel: sof_sdw: Simplify find_codec_info_acpi ASoC: intel: sof_sdw: Constify parameter to find_codec_part_info ASoC: intel: sof_sdw: Minor tidy up of mc_probe ASoC: intel: sof_sdw: Remove redundant parameters in dai creation ASoC: intel: sof_sdw: Move amp_num initialisation to mc_probe ASoC: intel: sof_sdw: Allow direct specification of CODEC name ASoC: intel: sof_sdw: Clean up DAI link counting ASoC: intel: sof_sdw: Merge codec_conf_alloc into dailink_info ASoC: intel: sof_sdw: Move group_generated logic Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: reorder SoundWire codecs in Kconfig ASoC: Intel: sof_sdw: allow mockup amplifier to provide feedback ASoC: Intel: sof_sdw: add support for SKU 0AFE Terry Cheong (1): ASoC: Intel: sof_rt5682: add RPL support for MAX98357A speaker Uday M Bhat (1): ASoC: Intel: sof_sdw: Update BT offload config for soundwire config apoorv (1): ASoC: Intel: sof_rt5682 add support for HDMI_In capture sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_rt5682.c | 55 +++ sound/soc/intel/boards/sof_sdw.c | 404 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 7 + sound/soc/intel/boards/sof_sdw_cs_amp.c | 47 ++ .../intel/common/soc-acpi-intel-adl-match.c | 7 + .../intel/common/soc-acpi-intel-rpl-match.c | 27 ++ 8 files changed, 342 insertions(+), 212 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_cs_amp.c -- 2.39.2

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[PATCH v4 0/1] Intel: sof_rt5682: remove quirk flag
by Brent Lu 01 Aug '23

01 Aug '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable V3 Changes: - replace pr_err() with dev_err() V4 Changes: - merge two patches to avoid build break - remove redundant assignment in max_98390_hw_params() - refactor MAX_98390_DEVX_NAME macros Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag Brent Lu (1): ASoC: Intel: maxim-common: get codec number from ACPI table sound/soc/intel/boards/sof_maxim_common.c | 170 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 111 insertions(+), 117 deletions(-) -- 2.34.1

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[PATCH 0/8] ASoC/SOF/Intel/AMD: cleanups for GCC11 -fanalyzer checks
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

GCC11 provides an '-fanalyzer' static analysis option which does not provide too many false-positives. This patch cleans-up known problematic code paths to help enable this capability in CI. We've used this for about a month already. Pierre-Louis Bossart (8): ASoC: SOF: sof-client-probes-ipc4: add checks to prevent static analysis warnings ASoC: SOF: ipc3: add checks to prevent static analysis warnings ASoC: SOF: topology: simplify code to prevent static analysis warnings ASoC: SOF: imx: remove error checks on NULL ipc ASoC: SOF: mediatek: remove error checks on NULL ipc ASoC: Intel: bdw_rt286: add checks to avoid static analysis warnings ASoC: Intel: atom: remove static analysis false positive ASoC: amd: acp5x-mach:add checks to avoid static analysis warnings include/linux/firmware/imx/dsp.h | 6 ------ include/linux/firmware/mediatek/mtk-adsp-ipc.h | 6 ------ sound/soc/amd/vangogh/acp5x-mach.c | 3 +++ sound/soc/intel/atom/sst/sst_stream.c | 13 +++++++------ sound/soc/intel/boards/bdw_rt286.c | 6 ++++++ sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-client-probes-ipc4.c | 9 +++++++++ sound/soc/sof/topology.c | 5 +++-- 8 files changed, 29 insertions(+), 21 deletions(-) -- 2.39.2

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Audio output heavily distorted on Xen PV dom0 if not recording (Intel ADL)
by Marek Marczykowski-Górecki 01 Aug '23

01 Aug '23

Hi, I have a rather interesting issue on HP EliteBook 1040, with Intel ADL HDA. Sometimes (often) audio output is heavily distorted, but then opening pavucontrol or just launching `parecord /dev/null` fixes the issue instantly. But, when the mixer or recording is closed, then few seconds later issue comes back. You can hear this at https://cloud.marmarek.net/s/46prqZnwrqDkBdD Initially recording was running, then about 10s stopped and you can hear the issue few seconds later. Then I starter recording again, and the issue is gone. This is how it should sound: https://youtube.com/watch?v=4Tr0otuiQuU I do not see any kernel messages when issue happens. The issue happens on Linux 6.1.26, 6.2.10, and 6.3.2. I haven't tested other versions. It happens only when running under Xen (in PV dom0). The very same Linux started directly works fine. Full dmesg of both native and Xen boots is at https://gist.github.com/marmarek/548ee79282446344a172109bcd943930 When running Linux 6.3.2 there, I noticed that just after boot first playback is okay, but if I start and stop recording it breaks. And then after some time of idle it fixes itself again. I have not observed this behavior on 6.2.10 (it was broken initially too), but I could be also doing some things differently. I have few other ADL-based systems (but not HP), and they do not show this issue, only this HP laptop is affected. Example dmesg from another ADL-based system: https://gist.github.com/marmarek/38c5ba3ec58f9300fbc1842ddc9fa4f5 I'm not sure if that's the relevant difference, but comparing dmesg from working and not working systems, I see those on boot for the problematic case: snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: enabling device (0000 -> 0002) xen: registering gsi 16 triggering 0 polarity 1 Already setup the GSI :16 sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) sof-audio-pci-intel-tgl 0000:00:1f.3: use msi interrupt mode sof-audio-pci-intel-tgl 0000:00:1f.3: hda codecs found, mask 5 sof-audio-pci-intel-tgl 0000:00:1f.3: using HDA machine driver skl_hda_dsp_generic now sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: unknown sof_ext_man header type 3 size 0x30 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:0 skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Parent card not yet available, widget card binding deferred cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.0 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.1 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) Any ideas? -- Best Regards, Marek Marczykowski-Górecki Invisible Things Lab

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[PATCH 0/9] ALSA: hda/cs35l56: Various bugfixes
by Richard Fitzgerald 01 Aug '23

01 Aug '23

A collection of various bugfixes to the cs35l56 driver. Richard Fitzgerald (9): ALSA: hda/cs35l56: Complete firmware reboot before calling cs_dsp_run() ALSA: hda/cs35l56: Do not mark cache dirty after REINIT ALSA: hda/cs35l56: Call cs_dsp_power_down() before reloading firmware ALSA: hda/cs35l56: Always power-up and start cs_dsp ALSA: hda/cs35l56: Call cs_dsp_power_down() before calling cs_dsp_remove() ALSA: hda/cs35l56: cs_dsp_power_down() on cs35l56_hda_fw_load() error path ALSA: hda/cs35l56: Do not download firmware over existing RAM firmware ALSA: hda/cs35l56: Fail if .bin not found and firmware not patched ALSA: hda/cs35l56: Reject I2C alias addresses sound/pci/hda/cs35l56_hda.c | 91 ++++++++++++++++++++++++++----------- 1 file changed, 65 insertions(+), 26 deletions(-) -- 2.30.2

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[PATCH 0/7] usb: Add USB MIDI 2.0 Gadget Function Driver
by Takashi Iwai 01 Aug '23

01 Aug '23

Hi, This is a patch set for adding a USB gadget function driver for USB MIDI 2.0 interface. The MIDI 2.0 support in the host side was already enabled since 6.5 kernel in snd-usb-audio driver. Now, this provides a similar gadget functionality using the existing ALSA UMP helper functions. The gadget driver provides both MIDI 2.0 and MIDI 1.0 fallback interfaces with one or more loopback UMP rawmidi devices. The configfs implementation is flexible enough to allow users to set up multiple UMP Endpoints and Function Blocks. The first patch adds the bare f_midi2 driver without configfs support, followed by the configfs support, MIDI 1.0 interface support and a few other extensions and documentation. Takashi === Takashi Iwai (7): usb: gadget: Add support for USB MIDI 2.0 function driver usb: gadget: midi2: Add configfs support usb: gadget: midi2: Dynamically create MIDI 1.0 altset descriptors usb: gadget: midi2: MIDI 1.0 interface (altset 0) support usb: gadget: midi2: Add testing documentation usb: gadget: midi2: Add "Operation Mode" control usb: gadget: midi2: More flexible MIDI 1.0 configuration .../ABI/testing/configfs-usb-gadget-midi2 | 54 + Documentation/usb/gadget-testing.rst | 154 + drivers/usb/gadget/Kconfig | 18 + drivers/usb/gadget/function/Makefile | 2 + drivers/usb/gadget/function/f_midi2.c | 2883 +++++++++++++++++ drivers/usb/gadget/function/u_midi2.h | 81 + 6 files changed, 3192 insertions(+) create mode 100644 Documentation/ABI/testing/configfs-usb-gadget-midi2 create mode 100644 drivers/usb/gadget/function/f_midi2.c create mode 100644 drivers/usb/gadget/function/u_midi2.h -- 2.35.3

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[PATCH] ASoC: soc-core.c: use devm_snd_soc_register_dai()
by Kuninori Morimoto 01 Aug '23

01 Aug '23

It is still using snd_soc_{un}register_dai() manually. Let's use devm_snd_soc_register_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 42 ++++++++++-------------------------------- 1 file changed, 10 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5b96c17633a..7284363b9088 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2552,48 +2552,28 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(snd_soc_register_dai); /** - * snd_soc_unregister_dais - Unregister DAIs from the ASoC core - * - * @component: The component for which the DAIs should be unregistered - */ -static void snd_soc_unregister_dais(struct snd_soc_component *component) -{ - struct snd_soc_dai *dai, *_dai; - - for_each_component_dais_safe(component, dai, _dai) - snd_soc_unregister_dai(dai); -} - -/** - * snd_soc_register_dais - Register a DAI with the ASoC core + * devm_snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs */ -static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, - size_t count) +static int devm_snd_soc_register_dais(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv, + size_t count) { struct snd_soc_dai *dai; unsigned int i; - int ret; for (i = 0; i < count; i++) { - dai = snd_soc_register_dai(component, dai_drv + i, count == 1 && - component->driver->legacy_dai_naming); - if (dai == NULL) { - ret = -ENOMEM; - goto err; - } + dai = devm_snd_soc_register_dai(component->dev, + component, dai_drv + i, count == 1 && + component->driver->legacy_dai_naming); + if (!dai) + return -ENOMEM; } return 0; - -err: - snd_soc_unregister_dais(component); - - return ret; } #define ENDIANNESS_MAP(name) \ @@ -2644,8 +2624,6 @@ static void snd_soc_del_component_unlocked(struct snd_soc_component *component) { struct snd_soc_card *card = component->card; - snd_soc_unregister_dais(component); - if (card) snd_soc_unbind_card(card, false); @@ -2696,7 +2674,7 @@ int snd_soc_add_component(struct snd_soc_component *component, } } - ret = snd_soc_register_dais(component, dai_drv, num_dai); + ret = devm_snd_soc_register_dais(component, dai_drv, num_dai); if (ret < 0) { dev_err(component->dev, "ASoC: Failed to register DAIs: %d\n", ret); -- 2.25.1

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[PATCH] ASoC: codecs/hda.c: use devm_snd_soc_register_dai()
by Kuninori Morimoto 01 Aug '23

01 Aug '23

It is still using snd_soc_{un}register_dai() manually. Let's use devm_snd_soc_register_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/hda.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index d57b043d6bfe..551bd0cbfc44 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -99,7 +99,7 @@ static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_compo list_for_each_entry(pcm, &codec->pcm_list_head, list) { struct snd_soc_dai *dai; - dai = snd_soc_register_dai(component, drvs, false); + dai = devm_snd_soc_register_dai(component->dev, component, drvs, false); if (!dai) { dev_err(component->dev, "register dai for %s failed\n", pcm->name); return -EINVAL; @@ -108,7 +108,6 @@ static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_compo ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret < 0) { dev_err(component->dev, "create widgets failed: %d\n", ret); - snd_soc_unregister_dai(dai); return ret; } @@ -135,7 +134,6 @@ static void hda_codec_unregister_dais(struct hda_codec *codec, for_each_pcm_streams(stream) snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); - snd_soc_unregister_dai(dai); break; } } -- 2.25.1

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