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[PATCH] ALSA: usb-audio: Remove the repeated declaration
by Shaokun Zhang 30 May '21

30 May '21

Function 'snd_usb_endpoint_suspend' is declared twice, so remove the repeated declaration. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Shaokun Zhang <zhangshaokun(a)hisilicon.com> --- sound/usb/endpoint.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index eea4ca49876d..a668f675b52b 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -19,7 +19,6 @@ void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); int snd_usb_endpoint_configure(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep, -- 2.7.4

2 1

alsa-ucm-conf 1.2.4 fails with platform-skl_hda_dsp_generic
by GitHub issues - opened 30 May '21

30 May '21

alsa-project/alsa-ucm-conf issue #93 was opened from bjacquin: Hi, I am experiencing the same issue as reported in issue #66, however I am still able to reproduce this with alsa-lib and alsa-utils 1.2.4 installed. * 1.2.3 `alsa-info` output is attached in [alsa-info.1.2.3.txt](https://github.com/alsa-project/alsa-ucm-conf/files/65…. `pulseaudio` standard logs show: ``` May 30 01:33:34 lady-voodoo.local pulseaudio[23962]: E: [pulseaudio] alsa-ucm.c: Failed to get the verb HiFi May 30 01:33:34 lady-voodoo.local pulseaudio[23962]: E: [pulseaudio] alsa-ucm.c: No UCM verb is valid for sof-hda-dsp ``` `pactl list short cards` list the following cards as available: ``` 0 alsa_card.usb-046d_HD_Pro_Webcam_C920_BD731F4F-02 module-alsa-card.c 1 alsa_card.usb-Sennheiser_Sennheiser_SP_30_A002340202303344-00 module-alsa-card.c 2 alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_genericmodule-alsa-card.c ``` `pulseaudio -vvv` output is attached in [pulseaudio-1.2.3.log](https://github.com/alsa-project/alsa-ucm-conf/files/6…. `pacmd ls` output is attached in [pacmd-ls-1.2.3.txt](https://github.com/alsa-project/alsa-ucm-conf/files/656… * 1.2.4 `alsa-info` output is attached in [alsa-info.1.2.4.txt](https://github.com/alsa-project/alsa-ucm-conf/files/65…. `pulseaudio` standard logs show: ``` May 30 01:14:40 lady-voodoo.local /bin/bash[672]: HISTORY: PID=672 UID=1000 pulseaudio -k May 30 01:14:40 lady-voodoo.local systemd[481]: pulseaudio.service: Succeeded. May 30 01:14:40 lady-voodoo.local systemd[481]: pulseaudio.service: Consumed 7.644s CPU time. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device HDMI3, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device HDMI2, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device HDMI1, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic2, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic1, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. May 30 01:14:41 lady-voodoo.local pulseaudio[18219]: E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="pci-0000_00_1f.3-platform-skl_hda_dsp_generic" card_name="alsa_card.pci-0000_00_1f.3-platform-skl_hda_dsp_generic" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. ``` `pactl list short cards` list the following cards as available: ``` 0 alsa_card.usb-046d_HD_Pro_Webcam_C920_BD731F4F-02 module-alsa-card.c 1 alsa_card.usb-Sennheiser_Sennheiser_SP_30_A002340202303344-00 module-alsa-card.c ``` `pulseaudio -vvv` output is attached in [pulseaudio-1.2.4.log](https://github.com/alsa-project/alsa-ucm-conf/files/6…. `pacmd ls` output is attached in [pacmd-ls-1.2.4.txt](https://github.com/alsa-project/alsa-ucm-conf/files/656…. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/93 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

snd_pcm_avail() returned a value that is exceptionally large "ALC293"
by GitHub issues - edited 29 May '21

29 May '21

alsa-project/alsa-lib issue #141 was edited from Jeronimo17: HDA Intel PCH Realtek ALC293 It is rare that it occurs, it has only happened to me playing, after this the internal keyboard stops responding and I have to put one for USB, and the shutdown of the laptop is impossible without removing power forced ``` may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_avail() returned a value that is exceptionally large: 440960 bytes (2499 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47704880 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810712 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_delay() returned a value that is exceptionally large: -247148 bytes (-1401 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47748960 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810784 ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/141 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_avail() returned a value that is exceptionally large "ALC293"
by GitHub issues - edited 29 May '21

29 May '21

alsa-project/alsa-lib issue #141 was edited from Jeronimo17: HDA Intel PCH Realtek ALC293 It is rare that it occurs, it has only happened to me playing, after this the internal keyboard stops responding and I have to put one for USB, and the shutdown of the laptop is impossible without removing power forced ` may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_avail() returned a value that is exceptionally large: 440960 bytes (2499 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47704880 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810712 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_delay() returned a value that is exceptionally large: -247148 bytes (-1401 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47748960 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810784 ` Issue URL : https://github.com/alsa-project/alsa-lib/issues/141 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_avail() returned a value that is exceptionally large "ALC293"
by GitHub issues - edited 29 May '21

29 May '21

alsa-project/alsa-lib issue #141 was edited from Jeronimo17: HDA Intel PCH Realtek ALC293 It is rare that it occurs, it has only happened to me playing, after this the internal keyboard stops responding and I have to put one for USB, and the shutdown of the laptop is impossible without removing power forced `` may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_avail() returned a value that is exceptionally large: 440960 bytes (2499 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47704880 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810712 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_delay() returned a value that is exceptionally large: -247148 bytes (-1401 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47748960 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810784 `` Issue URL : https://github.com/alsa-project/alsa-lib/issues/141 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_avail() returned a value that is exceptionally large "ALC293"
by GitHub issues - opened 29 May '21

29 May '21

alsa-project/alsa-lib issue #141 was opened from Jeronimo17: HDA Intel PCH Realtek ALC293 It is rare that it occurs, it has only happened to me playing, after this the internal keyboard stops responding and I have to put one for USB, and the shutdown of the laptop is impossible without removing power forced `may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_avail() returned a value that is exceptionally large: 440960 bytes (2499 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47704880 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810712 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_delay() returned a value that is exceptionally large: -247148 bytes (-1401 ms). may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: snd_pcm_dump(): may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Soft volume PCM may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Control: PCM Playback Volume may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: min_dB: -51 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: max_dB: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: resolution: 256 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Slave: Hardware PCM card 0 'HDA Intel PCH' device 0 subdevice 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: Its setup is: may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stream : PLAYBACK may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: access : MMAP_INTERLEAVED may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: format : S16_LE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: subformat : STD may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: channels : 2 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: rate : 44100 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: exact rate : 44100 (44100/1) may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: msbits : 16 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: buffer_size : 4408 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_size : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_time : 24988 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_mode : ENABLE may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: tstamp_type : MONOTONIC may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_step : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: avail_min : 1102 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: period_event : 1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: start_threshold : -1 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: stop_threshold : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_threshold: 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: silence_size : 0 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: boundary : 4962966789362286592 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: appl_ptr : 47748960 may 29 18:44:35 jeronimo17-GIGABYTE-AORUS-7-KB pulseaudio[1370]: hw_ptr : 47810784 ` Issue URL : https://github.com/alsa-project/alsa-lib/issues/141 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.5?
by GitHub issues - edited 29 May '21

29 May '21

alsa-project/alsa-lib issue #140 was edited from kloczek: Looks like 1.2.5 has been released but I don't see related changes and git tags not only here but in other alsa git repos as well. Issue URL : https://github.com/alsa-project/alsa-lib/issues/140 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

1.2.5?
by GitHub issues - opened 29 May '21

29 May '21

alsa-project/alsa-lib issue #140 was opened from kloczek: Looks like 1.2.5 has been released but I dond see related changes and git tags not only here but in other alsa git repos as well. Issue URL : https://github.com/alsa-project/alsa-lib/issues/140 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v5 0/3] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 28 May '21

28 May '21

This series squashes all the ASoC machine drivers into a single one, this change was suggested by Jon Hunter. It also sets driver_name and components string of each card, allowing userspace alsa-lib to find UCMs at predictable path. Changelog: v5: - The v4 removed the customization of components string for Nexus 7, but I missed to remove the "components" hook which is unused now, it's removed in v5 for consistency. - Slightly improved naming of the common 12MHz MCLK rate function to make it more consistent with the rest of the driver functions. v4: - Moved out mclk_rate callback that is currently used only by WM8903 machine driver from the common driver. This was suggested by Jon Hunter. - Dropped patch which was setting custom components string for Nexus 7. Jaroslav Kysela wants it to be specified in a device-tree, but the components string doesn't have a firm specification for today. It's better to drop this change for now since it's optional anyways. - Fixed compilation error that was reported by kernel robot for v3. - Jaroslav Kysela merged alsa-ucm-conf PR [1] which added UCMs for Nexus 7 and Acer A500. The UCMs are fully working using a combination of updated kernel + alsa-ucm-conf master + alsa-lib master, meaning that they will work with the next releases of kernel and ALSA userspace upstream packages. - Added ack from Jaroslav Kysela to the "Specify components string for each card" patch that he gave to v3. v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (3): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 ---------- sound/soc/tegra/tegra_asoc_machine.c | 712 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 45 ++ sound/soc/tegra/tegra_max98090.c | 276 ----------- sound/soc/tegra/tegra_rt5640.c | 222 --------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 351 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------- sound/soc/tegra/trimslice.c | 172 ------- 13 files changed, 850 insertions(+), 2103 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

1 4

[PATCH 00/10] ASoC: rsnd: tidyup and fixup rare MIXer issue
by Kuninori Morimoto 28 May '21

28 May '21

Hi Mark These are tidyup patch, and rare MIXer issue fixup patch for Renesas Sound driver. 1 - 5 : tidyup for rsnd driver 6 : add debugfs support 7 : fixup rare MIXer issue 8 - 10: tidyup for rsnd driver Kuninori Morimoto (10): 1) ASoC: rsnd: ignore runtime NULL case at rsnd_runtime_channel_original_with_params() 2) ASoC: rsnd: attach SSIU when SSI was DMA mode 3) ASoC: rsnd: check BUIF error everytime 4) ASoC: rsnd: indicate unknown error at rsnd_dai_call() 5) ASoC: rsnd: incidate irq error message 6) ASoC: rsnd: add debugfs support 7) ASoC: rsnd: protect mod->status 8) ASoC: rsnd: implement BUSIF related code in ssiu.c 9) ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear() 10) ASoC: rsnd: tidyup rsnd_ssiu_busif_err_irq_ctrl() sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 35 ++++++--- sound/soc/sh/rcar/cmd.c | 14 ++++ sound/soc/sh/rcar/core.c | 65 +++++++++------- sound/soc/sh/rcar/ctu.c | 14 ++++ sound/soc/sh/rcar/debugfs.c | 96 ++++++++++++++++++++++++ sound/soc/sh/rcar/dma.c | 31 ++++++-- sound/soc/sh/rcar/dvc.c | 14 ++++ sound/soc/sh/rcar/gen.c | 9 +++ sound/soc/sh/rcar/mix.c | 14 ++++ sound/soc/sh/rcar/rsnd.h | 39 ++++++++-- sound/soc/sh/rcar/src.c | 26 ++++++- sound/soc/sh/rcar/ssi.c | 146 ++++++++---------------------------- sound/soc/sh/rcar/ssiu.c | 134 +++++++++++++++++++++++++++++---- 14 files changed, 457 insertions(+), 182 deletions(-) create mode 100644 sound/soc/sh/rcar/debugfs.c -- 2.25.1

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[PATCH v2] ASoC: rt5659: Fix the lost powers for the HDA header
by Jack Yu 28 May '21

28 May '21

The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after the DAPM widgets were added, and these powers were set by the JD settings "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, these powers were ignored to prevent them controlled by DAPM. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 87f5709..4a50b16 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 1.9.1

2 1

[PATCH v2 00/11] ASoC: rsnd: tidyup and fixup rare MIXer issue
by Kuninori Morimoto 28 May '21

28 May '21

Hi Mark These are v2 of tidyup patch, and rare MIXer issue fixup patch for Renesas Sound driver. 1 - 6 : tidyup for rsnd driver 7 : add debugfs support 8 : fixup rare MIXer issue 9 - 11: tidyup for rsnd driver v1 -> v2 - add adg.c fixup patch (= [01/11]) - tidyup compile error when CONFIG_DEBUG_FS was not selected (= [07/11]) Link: https://lore.kernel.org/r/87mtsjof3w.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (11): 1 ASoC: rsnd: call unregister for null_hw when removed 2 ASoC: rsnd: ignore runtime NULL case at rsnd_runtime_channel_original_with_params() 3 ASoC: rsnd: attach SSIU when SSI was DMA mode 4 ASoC: rsnd: check BUIF error everytime 5 ASoC: rsnd: indicate unknown error at rsnd_dai_call() 6 ASoC: rsnd: incidate irq error message 7 ASoC: rsnd: add debugfs support 8 ASoC: rsnd: protect mod->status 9 ASoC: rsnd: implement BUSIF related code in ssiu.c 10 ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear() 11 ASoC: rsnd: tidyup rsnd_ssiu_busif_err_irq_ctrl() sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 44 +++++++---- sound/soc/sh/rcar/cmd.c | 14 ++++ sound/soc/sh/rcar/core.c | 65 +++++++++------- sound/soc/sh/rcar/ctu.c | 14 ++++ sound/soc/sh/rcar/debugfs.c | 96 +++++++++++++++++++++++ sound/soc/sh/rcar/dma.c | 31 ++++++-- sound/soc/sh/rcar/dvc.c | 14 ++++ sound/soc/sh/rcar/gen.c | 9 +++ sound/soc/sh/rcar/mix.c | 14 ++++ sound/soc/sh/rcar/rsnd.h | 40 ++++++++-- sound/soc/sh/rcar/src.c | 26 ++++++- sound/soc/sh/rcar/ssi.c | 148 +++++++++--------------------------- sound/soc/sh/rcar/ssiu.c | 134 ++++++++++++++++++++++++++++---- 14 files changed, 465 insertions(+), 186 deletions(-) create mode 100644 sound/soc/sh/rcar/debugfs.c -- 2.25.1

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[PATCH -next] ALSA: core: use DEVICE_ATTR_*() macro
by YueHaibing 28 May '21

28 May '21

Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR, which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/init.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 70114fd26956..228faf9369dc 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -665,17 +665,15 @@ void snd_card_set_id(struct snd_card *card, const char *nid) } EXPORT_SYMBOL(snd_card_set_id); -static ssize_t -card_id_show_attr(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t id_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); return scnprintf(buf, PAGE_SIZE, "%s\n", card->id); } -static ssize_t -card_id_store_attr(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t id_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); char buf1[sizeof(card->id)]; @@ -703,17 +701,16 @@ card_id_store_attr(struct device *dev, struct device_attribute *attr, return count; } -static DEVICE_ATTR(id, 0644, card_id_show_attr, card_id_store_attr); +static DEVICE_ATTR_RW(id); -static ssize_t -card_number_show_attr(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t number_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct snd_card *card = container_of(dev, struct snd_card, card_dev); return scnprintf(buf, PAGE_SIZE, "%i\n", card->number); } -static DEVICE_ATTR(number, 0444, card_number_show_attr, NULL); +static DEVICE_ATTR_RO(number); static struct attribute *card_dev_attrs[] = { &dev_attr_id.attr, -- 2.17.1

2 1

[PATCH 0/3] alsa-gobject: misc fixes
by GitHub pull_request - edited 28 May '21

28 May '21

alsa-project/alsa-gobject pull request #59 was edited from takaswie: This patchset fixes some bugs below: * Ubuntu 19.10/i386 is EOL and no release is available for i386 architecture in Ubuntu project * Update Fedora version for CI environment * Wrong command is in install section of README ``` Takashi Sakamoto (3): correct install section in README fix workflow according to EOF of i386 ubuntu distribution update Fedora release for CI environment .github/workflows/build.yml | 83 +++++++++++++++++++------------------ README.rst | 2 +- 2 files changed, 43 insertions(+), 42 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/59 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/59.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/3] alsa-gobject: misc fixes
by GitHub pull_request - edited 28 May '21

28 May '21

alsa-project/alsa-gobject pull request #59 was edited from takaswie: This patchset fixes some bugs below: * Ubuntu 19.10/i386 is EOL and no release is available for i386 architecture in Ubuntu project * Wrong command is in install section of README ``` Takashi Sakamoto (2): correct install section in README fix workflow according to EOF of i386 ubuntu distribution .github/workflows/build.yml | 81 +++++++++++++++++++------------------ README.rst | 2 +- 2 files changed, 42 insertions(+), 41 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/59 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/59.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/2] alsa-gobject: misc fixes
by GitHub pull_request - opened 28 May '21

28 May '21

alsa-project/alsa-gobject pull request #59 was opened from takaswie: This patchset fixes some bugs below: * Ubuntu 19.10/i386 is EOL and no release is available for i386 architecture in Ubuntu project * Wrong command is in install section of README ``` Takashi Sakamoto (2): correct install section in README fix workflow according to EOF of i386 ubuntu distribution .github/workflows/build.yml | 81 +++++++++++++++++++------------------ README.rst | 2 +- 2 files changed, 42 insertions(+), 41 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/59 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/59.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH 0/3] ALSA: firewire-lib: preparation for media clock recovery
by Takashi Sakamoto 28 May '21

28 May '21

Hi, ALSA IEC 61883-1/6 packet streaming engine have been using pre-computed parameters ideal for nominal sampling transfer frequency (STF), to transfer packets to device since it was added 2011. As a result of user experience for a decade, it is clear that the sequence is not suitable to some actual devices. In design of audio and music unit in IEEE 1394 bus, feedback of effective sampling transfer frequency is delivered by packets transferred from device. Reuse of the sequence of parameter in the packets results in pseudo media clock recovery to transfer packet. In this patchset, an option is added to AMDTP domain so that each of AMDTP stream for outgoing packet has association to AMDTP stream for incoming packet, to replay the sequence of parameter. Some devices require receiving packet on-the-fly to transfer packets, so another option is added. These options are just added, not used yet in this time. In future work, they are going to be used by drivers in ALSA firewire stack. Takashi Sakamoto (3): ALSA: firewire-lib: add replay target to cache sequence of packet ALSA: firewire-lib: replay sequence of incoming packets for outgoing packets ALSA: firewire-lib: transfer rx packets on-the-fly when replaying sound/firewire/amdtp-stream.c | 252 ++++++++++++++++++-- sound/firewire/amdtp-stream.h | 17 +- sound/firewire/bebob/bebob_stream.c | 2 +- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/digi00x/digi00x-stream.c | 2 +- sound/firewire/fireface/ff-stream.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/motu-stream.c | 2 +- sound/firewire/oxfw/oxfw-stream.c | 2 +- sound/firewire/tascam/tascam-stream.c | 2 +- 10 files changed, 259 insertions(+), 26 deletions(-) -- 2.27.0

2 4

[PATCH] ALSA: firewire-lib: support NO_PERIOD_WAKEUP in ALSA PCM runtime
by Takashi Sakamoto 28 May '21

28 May '21

Drivers of ALSA firewire stack can process packets for IT/IR context in process context when the process operates ALSA PCM character device by calling ioctl(2) with some requests. The ioctl requests are: * SNDRV_PCM_IOCTL_HWSYNC * SNDRV_PCM_IOCTL_SYNC_PTR * SNDRV_PCM_IOCTL_REWIND * SNDRV_PCM_IOCTL_FORWARD * SNDRV_PCM_IOCTL_WRITEI_FRAMES * SNDRV_PCM_IOCTL_READI_FRAMES * SNDRV_PCM_IOCTL_WRITEN_FRAMES * SNDRV_PCM_IOCTL_READN_FRAMES This means that general application can process PCM frames apart from hardware IRQ invocation, even if they are programmed by either IRQ-based scheduling model or Timer-based scheduling model. This commit add support for Timer-based scheduling model by allowing PCM runtime to suppress both process wakeup per period and scheduling hardware IRQ. SNDRV_PCM_INFO_BATCH is obsoleted since ALSA IEC 61883-1/6 packet streaming engine can report the number of transferred PCM frames within PCM period boundary. The granularity equals to SYT_INTERVAL in blocking transmission. In non-blocking transmission, it doesn't equal to SYT_INTERVAL but doesn't exceed. This patch is tested with PulseAudio, and --sched-model option of axfer with fix against the issue reported at: * https://lore.kernel.org/alsa-devel/687f9871-7484-1370-04d1-9c968e86f72b@lin… Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 68ffbc33f692..8e72e7f4c082 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -191,14 +191,13 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, unsigned int maximum_usec_per_period; int err; - hw->info = SNDRV_PCM_INFO_BATCH | - SNDRV_PCM_INFO_BLOCK_TRANSFER | + hw->info = SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_JOINT_DUPLEX | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID; + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP; - /* SNDRV_PCM_INFO_BATCH */ hw->periods_min = 2; hw->periods_max = UINT_MAX; @@ -503,7 +502,12 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - queue_work(system_highpri_wq, &s->period_work); + + // The program in user process should periodically check the status of intermediate + // buffer associated to PCM substream to process PCM frames in the buffer, instead + // of receiving notification of period elapsed by poll wait. + if (!pcm->runtime->no_period_wakeup) + queue_work(system_highpri_wq, &s->period_work); } } @@ -949,6 +953,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ unsigned int event_count = s->ctx_data.rx.event_count; unsigned int pkt_header_length; unsigned int packets; + bool need_hw_irq; int i; if (s->packet_index < 0) @@ -968,6 +973,16 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ else pkt_header_length = 0; + if (s == d->irq_target) { + // At NO_PERIOD_WAKEUP mode, the packets for all IT/IR contexts are processed by + // the tasks of user process operating ALSA PCM character device by calling ioctl(2) + // with some requests, instead of scheduled hardware IRQ of an IT context. + struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); + need_hw_irq = !pcm || !pcm->runtime->no_period_wakeup; + } else { + need_hw_irq = false; + } + for (i = 0; i < packets; ++i) { const struct pkt_desc *desc = s->pkt_descs + i; struct { @@ -984,7 +999,7 @@ static void process_rx_packets(struct fw_iso_context *context, u32 tstamp, size_ event_count += desc->data_blocks; if (event_count >= events_per_period) { event_count -= events_per_period; - sched_irq = true; + sched_irq = need_hw_irq; } } -- 2.27.0

2 1

No sound with Realtek ALC3306
by GitHub issues - edited 27 May '21

27 May '21

alsa-project/alsa-lib issue #139 was edited from dantahoua: Realtek ALC3306 (Lenovo 13s Gen2 ITL i7), but seems common problem with any computer using this codec. -Headphone, HDMI, BT working great. -Speaker not working (but computer seeing audio output and sending audio, meter even showing audio in GUI). Seems like something (amp?) is not turned, like missing something between speaker and codec... Works perfect in Windows 10. Tested lots different distro and kernel... Computer report ALC287 but it is not! It's ALC3306... Here is the alsa infos: [Infos](http://alsa-project.org/db/?f=1f3e847b60bd38085e39e26d992ea8680dd908… Issue URL : https://github.com/alsa-project/alsa-lib/issues/139 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No sound with Realtek ALC3306
by GitHub issues - edited 27 May '21

27 May '21

alsa-project/alsa-lib issue #139 was edited from dantahoua: Realtek ALC3306 (Lenovo 13s Gen2 ITL i7), but seems common problem with any computer using this codec. -Headphone, HDMI, BT working great. -Speaker not working (but computer seeing audio output and sending audio, meter even showing audio in GUI). Seems like something (amp?) is not turned, like missing something between speaker and codec... Works perfect in Windows 10. Tested lots different distro and kernel... Here is the alsa infos: [Infos](http://alsa-project.org/db/?f=1f3e847b60bd38085e39e26d992ea8680dd908… Issue URL : https://github.com/alsa-project/alsa-lib/issues/139 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

No sound with Realtek ALC3306
by GitHub issues - opened 27 May '21

27 May '21

alsa-project/alsa-lib issue #139 was opened from dantahoua: Realtek ALC3306 (Lenovo 13s Gen2 ITL i7), but seems common problem with any computer using this codec. -Headphone, HDMI, BT working great. -Speaker not working (but computer seeing audio output and sending audio, meter even showing audio in GUI). Seems like something (amp?) is not turned, like missing something between speaker and codec... Works perfect in Windows 10. Tested lots different distro and kernel... Issue URL : https://github.com/alsa-project/alsa-lib/issues/139 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4 0/3] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 27 May '21

27 May '21

This series squashes all the ASoC machine drivers into a single one, this change was suggested by Jon Hunter. It also sets driver_name and components string of each card, allowing userspace alsa-lib to find UCMs at predictable path. Changelog: v4: - Moved out mclk_rate callback that is currently used only by WM8903 machine driver from the common driver. This was suggested by Jon Hunter. - Dropped patch which was setting custom components string for Nexus 7. Jaroslav Kysela wants it to be specified in a device-tree, but the components string doesn't have a firm specification for today. It's better to drop this change for now since it's optional anyways. - Fixed compilation error that was reported by kernel robot for v3. - Jaroslav Kysela merged alsa-ucm-conf PR [1] which added UCMs for Nexus 7 and Acer A500. The UCMs are fully working using a combination of updated kernel + alsa-ucm-conf master + alsa-lib master, meaning that they will work with the next releases of kernel and ALSA userspace upstream packages. - Added ack from Jaroslav Kysela to the "Specify components string for each card" patch that he gave to v3. v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (3): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 ---------- sound/soc/tegra/tegra_asoc_machine.c | 714 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 46 ++ sound/soc/tegra/tegra_max98090.c | 276 ----------- sound/soc/tegra/tegra_rt5640.c | 222 --------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 351 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------- sound/soc/tegra/trimslice.c | 172 ------- 13 files changed, 853 insertions(+), 2103 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

1 3

Internal microphone does not work with 5.12 (Tiger Lake, X1 Nano)
by Nico Schottelius 27 May '21

27 May '21

Hello, it seems that the internal microphone does not work anymore with 5.12.6/5.12.7. I am somewhat sure it used to work within the 5.11.x series and that sound did not work at all on 5.10. I have uploaded the hardware information on http://alsa-project.org/db/?f=f1a2b32db122695b2cf736db27cb6ed2fa8a9042 The microphone of a plugged in headset however *does* work. Is this a known bug/regression and is there anything I can do to debug it? Best regards, Nico -- Sustainable and modern Infrastructures by ungleich.ch

4 8

[PATCH 0/3] ASoC: rsnd: add D3 support
by Kuninori Morimoto 27 May '21

27 May '21

Hi Mark, Rob These adds R-Car D3 support for rsnd driver. [1/3] is tidyup patch for dt-bindings (not only for D3). [2/3], [3/3] are for R-Car D3. Kuninori Morimoto (3): ASoC: dt-bindings: renesas: rsnd: tidyup properties ASoC: rsnd: tidyup loop on rsnd_adg_clk_query() ASoC: rsnd: add null CLOCKIN support .../bindings/sound/renesas,rsnd.yaml | 10 ++++- sound/soc/sh/rcar/adg.c | 37 ++++++++++++++++--- 2 files changed, 41 insertions(+), 6 deletions(-) -- 2.25.1

3 10

[PATCH v8 1/2] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 27 May '21

27 May '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for Dell privacy driver for the Dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. Once the audio or camera privacy mode activated, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled, micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is Ctrl+F9. Currently design only emits SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & HW(hardware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout timer is started) 2) WMI event is emitted from BIOS to kernel 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. If EC not notified, HW mic mute will also be activated when timeout used up, it is just later than active ack Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v7 -> v8: * addressed feedback from Hans * use sysfs_emit_at in the sysfs attributes * Update Kconfig * improve led_classdev_unregister logic in dell-laptop v6 -> v7: * addressed feedback from Hans * addressed feedback from Pierre * optimize some debug format with dev_dbg() v5 -> v6: * remove platform driver,combined privacy acpi driver into single wmi driver file * optimize sysfs interface with string added to be more clearly reading * remove unused function and clear header file v4 -> v5: * addressed feedback from Randy Dunlap * addressed feedback from Pierre-Louis Bossart * rebase to latest 5.12 rc4 upstream kernel * fix some space alignment problem v3 -> v4: * fix format for Kconfig * add sysfs document * add flow comments to the privacy wmi/acpi driver * addressed feedback from Barnabás Pőcze[Thanks very much] * export privacy_valid to make the global state simpler to query * fix one issue which will block the dell-laptop driver to load when privacy driver invalid * addressed feedback from Pierre-Louis Bossart,remove the EC ID match v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly. * fix some code style. * add KEY_END to keymap array. * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 13 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 394 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 25 ++ drivers/platform/x86/dell/dell-wmi.c | 9 +- 7 files changed, 507 insertions(+), 4 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..7f9e18705861 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,55 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported hardware privacy devices are: +Attributes: + Microphone Mute: + Identifies the local microphone can be muted by hardware, no applications + is available to capture system mic sound + + Camera Shutter: + Identifies camera shutter controlled by hardware, which is a micromechanical + shutter assembly that is built onto the camera module to block capturing images + from outside the laptop + + supported: + The privacy device is supported by this system + + unsupported: + The privacy device is not supported on this system + + For example to check which privacy devices are supported: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type + [Microphone Mute] [supported] + [Camera Shutter] [supported] + [ePrivacy Screen] [unsupported] + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities +Attributes: + muted: + Identifies the privacy device is turned off and cannot send stream to OS applications + + unmuted: + Identifies the privacy device is turned on ,audio or camera driver can get + stream from mic and camera module to OS applications + + For example to check all supported current privacy device states: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state + [Microphone] [unmuted] + [Camera Shutter] [unmuted] diff --git a/drivers/platform/x86/dell/Kconfig b/drivers/platform/x86/dell/Kconfig index e0a55337f51a..be570ccbbce0 100644 --- a/drivers/platform/x86/dell/Kconfig +++ b/drivers/platform/x86/dell/Kconfig @@ -204,4 +204,18 @@ config DELL_WMI_SYSMAN To compile this driver as a module, choose M here: the module will be called dell-wmi-sysman. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + depends on DELL_WMI + help + This driver provides integration with the "Dell Hardware Privacy" feature + of Dell laptops. + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. endif # X86_PLATFORM_DRIVERS_DELL diff --git a/drivers/platform/x86/dell/Makefile b/drivers/platform/x86/dell/Makefile index d720a3e42ae3..7da0c33dfcca 100644 --- a/drivers/platform/x86/dell/Makefile +++ b/drivers/platform/x86/dell/Makefile @@ -19,3 +19,4 @@ obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy-wmi.o diff --git a/drivers/platform/x86/dell/dell-laptop.c b/drivers/platform/x86/dell/dell-laptop.c index 70edc5bb3a14..529547fb0cb1 100644 --- a/drivers/platform/x86/dell/dell-laptop.c +++ b/drivers/platform/x86/dell/dell-laptop.c @@ -31,6 +31,8 @@ #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" + struct quirk_entry { bool touchpad_led; bool kbd_led_not_present; @@ -90,6 +92,7 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool micmute_led_registered; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); @@ -2205,11 +2208,13 @@ static int __init dell_init(void) dell_laptop_register_notifier(&dell_laptop_notifier); if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && - dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { + dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE) && + !dell_privacy_present()) { micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); if (ret < 0) goto fail_led; + micmute_led_registered = true; } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2262,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (micmute_led_registered) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2284,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (micmute_led_registered) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.c b/drivers/platform/x86/dell/dell-privacy-wmi.c new file mode 100644 index 000000000000..a32d0fbd99da --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.c @@ -0,0 +1,394 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bitops.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define DELL_PRIVACY_AUDIO_EVENT 0x1 +#define DELL_PRIVACY_CAMERA_EVENT 0x2 +#define led_to_priv(c) container_of(c, struct privacy_wmi_data, cdev) + +/* + * The wmi_list is used to store the privacy_priv struct with mutex protecting + */ +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + struct led_classdev cdev; + u32 features_present; + u32 last_status; +}; + +/* DELL Privacy Type */ +enum dell_hardware_privacy_type { + DELL_PRIVACY_TYPE_AUDIO = 0, + DELL_PRIVACY_TYPE_CAMERA, + DELL_PRIVACY_TYPE_SCREEN, + DELL_PRIVACY_TYPE_MAX, +}; + +static const char * const privacy_types[DELL_PRIVACY_TYPE_MAX] = { + [DELL_PRIVACY_TYPE_AUDIO] = "Microphone", + [DELL_PRIVACY_TYPE_CAMERA] = "Camera Shutter", + [DELL_PRIVACY_TYPE_SCREEN] = "ePrivacy Screen", +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +/* + * global privacy state for other modules to query if the privacy driver loaded successfully + * or no privacy WMI device was detected at initial probe phase + */ +bool dell_privacy_present(void) +{ + struct privacy_wmi_data *priv; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + mutex_unlock(&list_mutex); + + return priv && (priv->features_present & BIT(DELL_PRIVACY_TYPE_AUDIO)); +} +EXPORT_SYMBOL_GPL(dell_privacy_present); + +/* + * The flow of privacy event: + * 1) User presses key. HW does stuff with this key (timeout is started) + * 2) WMI event is emitted from BIOS + * 3) WMI event is received by dell-privacy + * 4) KEY_MICMUTE emitted from dell-privacy + * 5) Userland picks up key and modifies kcontrol for SW mute + * 6) Codec kernel driver catches and calls ledtrig_audio_set defined by + * dell-privacy-acpi driver. Codec driver will call like this to switch micmute led state. + * ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led ? LED_ON :LED_OFF); + * 7) If "LED" is set to on dell-privacy notifies EC,and timeout is cancelled, + * HW mic mute activated. + */ +bool dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + bool ret = false; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) + goto error; + + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16) | code); + if (!key) { + dev_warn(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + dev_dbg(&priv->wdev->dev, "Key with type 0x%04x and code 0x%04x pressed\n", type, code); + + switch (code) { + case DELL_PRIVACY_AUDIO_EVENT: /* Mic mute */ + case DELL_PRIVACY_CAMERA_EVENT: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + ret = true; + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x\n", type, code); + } + +error: + mutex_unlock(&list_mutex); + return ret; +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t dell_privacy_supported_type_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + enum dell_hardware_privacy_type type; + u32 privacy_list; + int len = 0; + + privacy_list = priv->features_present; + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_list & BIT(type)) + len += sysfs_emit_at(buf, len, "[%s] [supported]\n", privacy_types[type]); + else + len += sysfs_emit_at(buf, len, "[%s] [unsupported]\n", privacy_types[type]); + } + + return len; +} + +static ssize_t dell_privacy_current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + u32 privacy_supported = priv->features_present; + enum dell_hardware_privacy_type type; + u32 privacy_state = priv->last_status; + int len = 0; + + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_supported & BIT(type)) { + if (privacy_state & BIT(type)) + len += sysfs_emit_at(buf, len, "[%s] [unmuted]\n", privacy_types[type]); + else + len += sysfs_emit_at(buf, len, "[%s] [muted]\n", privacy_types[type]); + } + } + + return len; +} + +static DEVICE_ATTR_RO(dell_privacy_supported_type); +static DEVICE_ATTR_RO(dell_privacy_current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_dell_privacy_supported_type.attr, + &dev_attr_dell_privacy_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 - None; 0x1 - Microphone; 0x2 - Camera; 0x4 - ePrivacy Screen + * + * [WmiDataId(2), read] uint32 CurrentState; + * 0 - Off; 1 - On; Bit0 - Microphone; Bit1 - Camera; Bit2 - ePrivacyScreen + * }; + */ +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + dev_err(&wdev->dev, "dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + return -EIO; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_wmi_data *priv = led_to_priv(led_cdev); + static char *acpi_method = (char *)"ECAK"; + acpi_status status; + acpi_handle handle; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + + if (!acpi_has_method(handle, acpi_method)) + return -EIO; + + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(&priv->wdev->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + + return 0; +} + +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + return devm_led_classdev_register(dev, &priv->cdev); +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) + pr_debug("Unable to detect available Dell privacy devices!\n"); + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012), + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) + return -ENOMEM; + + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[i] = dell_wmi_keymap_type_0012[i]; + keymap[i].code |= (0x0012 << 16); + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + kfree(keymap); + if (ret) + return ret; + + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + + ret = input_register_device(priv->input_dev); + if (ret) + return ret; + + ret = get_current_status(priv->wdev); + if (ret) + return ret; + + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + return ret; + + if (priv->features_present & BIT(DELL_PRIVACY_TYPE_AUDIO)) { + ret = dell_privacy_leds_setup(&priv->wdev->dev); + if (ret) + return ret; + } + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + return 0; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +module_wmi_driver(dell_privacy_wmi_driver); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.h b/drivers/platform/x86/dell/dell-privacy-wmi.h new file mode 100644 index 000000000000..54004eed7213 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.h @@ -0,0 +1,25 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +bool dell_privacy_present(void); +bool dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +static inline bool dell_privacy_present(void) +{ + return -ENODEV; +} + +static inline bool dell_privacy_process_event(int type, int code, int status) +{ + return false; +} +#endif /* CONFIG_DELL_PRIVACY */ +#endif diff --git a/drivers/platform/x86/dell/dell-wmi.c b/drivers/platform/x86/dell/dell-wmi.c index bbdb3e860892..20367a580fa0 100644 --- a/drivers/platform/x86/dell/dell-wmi.c +++ b/drivers/platform/x86/dell/dell-wmi.c @@ -27,6 +27,7 @@ #include <acpi/video.h> #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -427,7 +428,6 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); @@ -439,6 +439,13 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: + if ((len > 4) && dell_privacy_process_event(buffer_entry[1], buffer_entry[3], + buffer_entry[4])) + /* dell_privacy_process_event has handled the event */; + else if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

3 6

[PATCH][next] ALSA: hda/ca0132: Make a const array static, makes object smaller
by Colin King 27 May '21

27 May '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const array dsp_dma_stream_ids the stack but instead make it static. Makes the object code smaller by 21 bytes. Before: text data bss dec hex filename 189012 70376 192 259580 3f5fc ./sound/pci/hda/patch_ca0132.o After: text data bss dec hex filename 188927 70440 192 259559 3f5e7 ./sound/pci/hda/patch_ca0132.o (gcc version 10.3.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_ca0132.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 49b4fdd2feab..b66e7bdbf483 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7598,7 +7598,7 @@ static void ca0132_alt_free_active_dma_channels(struct hda_codec *codec) */ static void ca0132_alt_start_dsp_audio_streams(struct hda_codec *codec) { - const unsigned int dsp_dma_stream_ids[] = { 0x0c, 0x03, 0x04 }; + static const unsigned int dsp_dma_stream_ids[] = { 0x0c, 0x03, 0x04 }; struct ca0132_spec *spec = codec->spec; unsigned int i, tmp; -- 2.31.1

2 1

[PATCH 1/2] ALSA: drivers: opl3: fix useless self-comparison
by Pierre-Louis Bossart 27 May '21

27 May '21

Sparse throws the following warning: sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always evaluates to false This is likely a 16+ year old confusion between vp2 and vp. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/drivers/opl3/opl3_midi.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index eb23c55323ae..e1b69c65c3c8 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -180,8 +180,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, if (vp2->state == SNDRV_OPL3_ST_ON_2OP) { /* kill two voices, EXPENSIVE */ bp++; - voice_time = (voice_time > vp->time) ? - voice_time : vp->time; + voice_time = max(voice_time, vp2->time); } } else { /* allocate 2op voice */ -- 2.25.1

2 3

[PATCH] sound/oss/dmasound: Remove superfluous "break"
by zuoqilin1＠163.com 27 May '21

27 May '21

From: zuoqilin <zuoqilin(a)yulong.com> Remove superfluous "break", as there is a "return" before them. Signed-off-by: zuoqilin <zuoqilin(a)yulong.com> --- sound/oss/dmasound/dmasound_core.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 22cef0c..0c95828 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1229,31 +1229,22 @@ static char *get_afmt_string(int afmt) switch(afmt) { case AFMT_MU_LAW: return "mu-law"; - break; case AFMT_A_LAW: return "A-law"; - break; case AFMT_U8: return "unsigned 8 bit"; - break; case AFMT_S8: return "signed 8 bit"; - break; case AFMT_S16_BE: return "signed 16 bit BE"; - break; case AFMT_U16_BE: return "unsigned 16 bit BE"; - break; case AFMT_S16_LE: return "signed 16 bit LE"; - break; case AFMT_U16_LE: return "unsigned 16 bit LE"; - break; case 0: return "format not set" ; - break ; default: break ; } -- 1.9.1

2 1

Soft jack injection for USB audio?
by Kai-Heng Feng 27 May '21

27 May '21

I have the need to use soft jack injection to debug userpsace, but currently USB audio doesn't have that ability. The problem I am facing is that the USB audio doesn't use snd_jack_* to control the jack, so we can add the support for USB audio separately. It's not hard to add the support if we open code soft injection in USB audio's build_connector_control(), but would it be possible to use snd_jack_* in USB audio? Kai-Heng

2 1

[PATCH v2 0/7] ASoC: adds new .get_fmt support
by Kuninori Morimoto 27 May '21

27 May '21

Hi Mark These are v2 of .get_fmt support. This is a little bit challenging patch-set. The idea/code is almost same as v1. We need to set dai_link->dai_fmt to select CPU/Codec settings, and it is selected by Sound Card Driver, today. Because of it, Sound Card user need to know both CPU / Codec available dai_fmt, and needs to select it. For example simple-card / audio-graph case, it is selected by "format" and "bitclock/frame-master/inversion" on DT. But, it can be automatically selected if both CPU and Codec drivers indicate it to ALSA SoC Framework, somehow. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .get_dai_fmt callback. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had callbacks. In other words, we can select all dai_fmt via Sound Card driver same as before. Select dai_fmt 100% automatically is very difficult and will be very complex, but select automatically some fields only is very easy, I guess. This patch-set is based on such assumption. v1 -> v2 - Add more detail explanation on git-log, code, comment. - Possible to be Clock/Frame provider is depends on driver's situation. Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (7): ASoC: soc-core: move snd_soc_runtime_set_dai_fmt() to upside ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() ASoC: ak4613: add .get_fmt support ASoC: pcm3168a: add .get_fmt support ASoC: rsnd: add .get_fmt support ASoC: fsi: add .get_fmt support ASoC: hdmi-codec: add .get_fmt support include/sound/soc-dai.h | 45 +++++++ sound/soc/codecs/ak4613.c | 14 ++ sound/soc/codecs/hdmi-codec.c | 22 ++++ sound/soc/codecs/pcm3168a.c | 19 +++ sound/soc/sh/fsi.c | 19 +++ sound/soc/sh/rcar/adg.c | 8 ++ sound/soc/sh/rcar/core.c | 24 +++- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/soc-core.c | 234 +++++++++++++++++++++++++--------- sound/soc/soc-dai.c | 18 +++ sound/soc/soc-utils.c | 32 +++++ 11 files changed, 372 insertions(+), 64 deletions(-) -- 2.25.1

2 11

Working to make ALSA work on a gemini lake tablet
by Sergio Callegari 26 May '21

26 May '21

Hi, I am trying to understand the issues that ALSA has on a recent gemini lake tablet (the CHUWI HI 10 X) for which otherwise linux support is in rather good shape. In case I am successful, I'd be glad to try to set up a guide for users encountering the same or a similar setup. On this machine, even on very recent kernels, it is impossible to have the soundcard recognized (only HDMI audio outputs are shown for aplay -l). The tablet advertises Intel SST and to the best of my understanding that implies that in addition to the basic Intel HDA support there should be some codec chip that needs support too (this used to be the bytcrrt5651 in a previous version of the tablet not based on gemini lake). I also see that in many similar pieces of hardware, regular (non HDMI) audio support can only be activated by setting some GPIO outputs (e.g., https://github.com/danielotero/linux-on-hi10/issues/8). My biggest element of concern at this point is that I am not able to identify the CODEC part, nor to understand if some specific UCM file or firmware file is needed. I wonder if anybody could point me to some hints at how to properly identify the hardware and the issue. I have windows pre-installed on the system but even from there I have so far been unable to identify the audio chipset properly. Thanks for the attention and even more thanks for any help, Best, Sergio

1 0

[PATCH -next] ASoC: cs42l56: use DEVICE_ATTR_WO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l56.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l56.c b/sound/soc/codecs/cs42l56.c index 7cdffdf6b8cf..3cf8a0b4478c 100644 --- a/sound/soc/codecs/cs42l56.c +++ b/sound/soc/codecs/cs42l56.c @@ -1021,9 +1021,8 @@ static int cs42l56_beep_event(struct input_dev *dev, unsigned int type, return 0; } -static ssize_t cs42l56_beep_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t beep_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct cs42l56_private *cs42l56 = dev_get_drvdata(dev); long int time; @@ -1038,7 +1037,7 @@ static ssize_t cs42l56_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l56_beep_set); +static DEVICE_ATTR_WO(beep); static void cs42l56_init_beep(struct snd_soc_component *component) { -- 2.17.1

3 2

[PATCH -next] ASoC: cs42l52: use DEVICE_ATTR_WO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs42l52.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 88547e2cd53d..80161151b3f2 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -957,9 +957,8 @@ static int cs42l52_beep_event(struct input_dev *dev, unsigned int type, return 0; } -static ssize_t cs42l52_beep_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t beep_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct cs42l52_private *cs42l52 = dev_get_drvdata(dev); long int time; @@ -974,7 +973,7 @@ static ssize_t cs42l52_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, cs42l52_beep_set); +static DEVICE_ATTR_WO(beep); static void cs42l52_init_beep(struct snd_soc_component *component) { -- 2.17.1

3 2

[PATCH -next] ASoC: cs43130: Use DEVICE_ATTR_RO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/cs43130.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index de7fa3e4bfbb..44b20c1ef851 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1672,14 +1672,14 @@ static int cs43130_show_dc(struct device *dev, char *buf, u8 ch) cs43130->hpload_dc[ch]); } -static ssize_t cs43130_show_dc_l(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_dc_l_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_dc(dev, buf, HP_LEFT); } -static ssize_t cs43130_show_dc_r(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_dc_r_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_dc(dev, buf, HP_RIGHT); } @@ -1719,22 +1719,22 @@ static int cs43130_show_ac(struct device *dev, char *buf, u8 ch) } } -static ssize_t cs43130_show_ac_l(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_ac_l_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_ac(dev, buf, HP_LEFT); } -static ssize_t cs43130_show_ac_r(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t hpload_ac_r_show(struct device *dev, + struct device_attribute *attr, char *buf) { return cs43130_show_ac(dev, buf, HP_RIGHT); } -static DEVICE_ATTR(hpload_dc_l, 0444, cs43130_show_dc_l, NULL); -static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); -static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); -static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static DEVICE_ATTR_RO(hpload_dc_l); +static DEVICE_ATTR_RO(hpload_dc_r); +static DEVICE_ATTR_RO(hpload_ac_l); +static DEVICE_ATTR_RO(hpload_ac_r); static struct attribute *hpload_attrs[] = { &dev_attr_hpload_dc_l.attr, -- 2.17.1

3 2

[PATCH] ASoC: cs35l3x: Use neutral language in amp drivers
by David Rhodes 26 May '21

26 May '21

Revise variable names and comments in cs35l35 and cs35l36 amp drivers. Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l35.c | 16 ++++++++-------- sound/soc/codecs/cs35l35.h | 2 +- sound/soc/codecs/cs35l36.c | 16 ++++++++-------- 3 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/cs35l35.c b/sound/soc/codecs/cs35l35.c index f20ed838b958..f4e2dc0ed5cc 100644 --- a/sound/soc/codecs/cs35l35.c +++ b/sound/soc/codecs/cs35l35.c @@ -367,16 +367,16 @@ static int cs35l35_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_component *component = codec_dai->component; struct cs35l35_private *cs35l35 = snd_soc_component_get_drvdata(component); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 1 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = false; + cs35l35->clock_consumer = false; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: regmap_update_bits(cs35l35->regmap, CS35L35_CLK_CTL1, CS35L35_MS_MASK, 0 << CS35L35_MS_SHIFT); - cs35l35->slave_mode = true; + cs35l35->clock_consumer = true; break; default: return -EINVAL; @@ -555,8 +555,8 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, } sp_sclks = ((cs35l35->sclk / srate) / 4) - 1; - /* Only certain ratios are supported in I2S Slave Mode */ - if (cs35l35->slave_mode) { + /* Only certain ratios supported when device is a clock consumer */ + if (cs35l35->clock_consumer) { switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_48FS: @@ -567,7 +567,7 @@ static int cs35l35_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } } else { - /* Only certain ratios supported in I2S MASTER Mode */ + /* Only certain ratios supported when device is a clock provider */ switch (sp_sclks) { case CS35L35_SP_SCLKS_32FS: case CS35L35_SP_SCLKS_64FS: diff --git a/sound/soc/codecs/cs35l35.h b/sound/soc/codecs/cs35l35.h index ffb154cd962c..2e69fef318e3 100644 --- a/sound/soc/codecs/cs35l35.h +++ b/sound/soc/codecs/cs35l35.h @@ -282,7 +282,7 @@ struct cs35l35_private { int sclk; bool pdm_mode; bool i2s_mode; - bool slave_mode; + bool clock_consumer; /* GPIO for /RST */ struct gpio_desc *reset_gpio; struct completion pdn_done; diff --git a/sound/soc/codecs/cs35l36.c b/sound/soc/codecs/cs35l36.c index a038bcec2d17..895ceed12cd9 100644 --- a/sound/soc/codecs/cs35l36.c +++ b/sound/soc/codecs/cs35l36.c @@ -756,14 +756,14 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, { struct cs35l36_private *cs35l36 = snd_soc_component_get_drvdata(component_dai->component); - unsigned int asp_fmt, lrclk_fmt, sclk_fmt, slave_mode, clk_frc; + unsigned int asp_fmt, lrclk_fmt, sclk_fmt, clock_provider, clk_frc; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - slave_mode = 1; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + clock_provider = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: - slave_mode = 0; + case SND_SOC_DAIFMT_CBC_CFC: + clock_provider = 0; break; default: return -EINVAL; @@ -771,10 +771,10 @@ static int cs35l36_set_dai_fmt(struct snd_soc_dai *component_dai, regmap_update_bits(cs35l36->regmap, CS35L36_ASP_TX_PIN_CTRL, CS35L36_SCLK_MSTR_MASK, - slave_mode << CS35L36_SCLK_MSTR_SHIFT); + clock_provider << CS35L36_SCLK_MSTR_SHIFT); regmap_update_bits(cs35l36->regmap, CS35L36_ASP_RATE_CTRL, CS35L36_LRCLK_MSTR_MASK, - slave_mode << CS35L36_LRCLK_MSTR_SHIFT); + clock_provider << CS35L36_LRCLK_MSTR_SHIFT); switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { case SND_SOC_DAIFMT_CONT: -- 2.25.1

3 2

[PATCH -next] ASoC: wm8962: Use DEVICE_ATTR_WO macro
by YueHaibing 26 May '21

26 May '21

Use DEVICE_ATTR_WO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/wm8962.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34080f497584..ba16bdf9e478 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3219,9 +3219,8 @@ static int wm8962_beep_event(struct input_dev *dev, unsigned int type, return 0; } -static ssize_t wm8962_beep_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t beep_store(struct device *dev, struct device_attribute *attr, + const char *buf, size_t count) { struct wm8962_priv *wm8962 = dev_get_drvdata(dev); long int time; @@ -3236,7 +3235,7 @@ static ssize_t wm8962_beep_set(struct device *dev, return count; } -static DEVICE_ATTR(beep, 0200, NULL, wm8962_beep_set); +static DEVICE_ATTR_WO(beep); static void wm8962_init_beep(struct snd_soc_component *component) { -- 2.17.1

3 2

[PATCH -next] ASoC: samsung: midas_wm1811: Fix build error when CONFIG_GPIOLIB is not set
by Wei Yongjun 26 May '21

26 May '21

Fix build error when CONFIG_GPIOLIB is not set: sound/soc/samsung/midas_wm1811.c: In function 'midas_fm_set': sound/soc/samsung/midas_wm1811.c:205:3: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 205 | gpiod_set_value_cansleep(priv->gpio_fm_sel, 1); | ^~~~~~~~~~~~~~~~~~~~~~~~ | gpio_set_value_cansleep sound/soc/samsung/midas_wm1811.c: In function 'midas_probe': sound/soc/samsung/midas_wm1811.c:445:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_clk_get_optional'? [-Werror=implicit-function-declaration] 445 | priv->gpio_fm_sel = devm_gpiod_get_optional(dev, "fm-sel", GPIOD_OUT_HIGH); | ^~~~~~~~~~~~~~~~~~~~~~~ | devm_clk_get_optional sound/soc/samsung/midas_wm1811.c:445:61: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 445 | priv->gpio_fm_sel = devm_gpiod_get_optional(dev, "fm-sel", GPIOD_OUT_HIGH); | ^~~~~~~~~~~~~~ | GPIOF_INIT_HIGH Fixes: fd0ea9cd9698 ("ASoC: samsung: Add sound support for Midas boards") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/samsung/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index a2221ebb1b6a..22be597b1dd9 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -228,6 +228,7 @@ config SND_SOC_SAMSUNG_ARIES_WM8994 config SND_SOC_SAMSUNG_MIDAS_WM1811 tristate "SoC I2S Audio support for Midas boards" depends on SND_SOC_SAMSUNG + depends on GPIOLIB select SND_SAMSUNG_I2S select SND_SOC_WM8994 help

2 1

Sparse errors
by Pierre-Louis Bossart 26 May '21

26 May '21

Hi Takashi, Sparse reports a lot of new issues in our last checks with more options: export ARCH=x86_64 CF="-Wsparse-error -Wsparse-all -Wno-bitwise-pointer -Wno-pointer-arith -Wno-typesign -Wnoshadow -Wno-sizeof-bool" make -k sound/ C=2 most are linked to the __user and pcm_format_t restricted types, but I found the simpler ones below which are useless comparisons. I can send a patch for the last but not sure how to address the first two. Thanks for your feedback -Pierre sound/core/info.c:95:38: error: self-comparison always evaluates to false if (pos < 0 || (long) pos != pos || (ssize_t) count < 0) return false; not sure what the second comparison is meant to check? sound/drivers/opl3/opl3_midi.c:183:60: error: self-comparison always evaluates to false This indeed makes no sense. the voice_time and vp->time are not changed in the loop, the test is either redundant or something else is missing. sound/pci/lx6464es/lx_core.c:677:34: error: self-comparison always evaluates to false That seems like dead code indeed: u32 channels = runtime->channels; if (runtime->channels != channels) dev_err(chip->card->dev, "channel count mismatch: %d vs %d", runtime->channels, channels);

3 5

Re: [PATCH] ASoC: rt5659: Fix the lost powers for the HDA header
by Mark Brown 26 May '21

26 May '21

On Wed, May 26, 2021 at 11:40:58AM +0000, Oder Chiou wrote: > Oder Chiou <oder_chiou(a)realtek.com> 於 2021年5月26日週三下午7:27寫道： > > > > On Wed, May 26, 2021 at 10:10:15AM +0000, Jack Yu wrote: > > > The patch fixes the lost powers for the HDA header. > > > > What are the issues and how does this patch fix them? > The power of "LDO2", "MICBIAS1" and "Mic Det Power" were powered off after > the DAPM widgets were added, and these powers were set by the JD settings > "RT5659_JD_HDA_HEADER" in the probe function. In the codec probe function, > these powers were ignored to prevent them controlled by DAPM, thanks. OK, this should be covered in the changelog.

1 0

[PATCH] ASoC: rt5659: Fix the lost powers for the HDA header
by Jack Yu 26 May '21

26 May '21

The patch fixes the lost powers for the HDA header. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5659.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 87f5709..4a50b16 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -2433,13 +2433,18 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { +static const struct snd_soc_dapm_widget rt5659_particular_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("LDO2", RT5659_PWR_ANLG_3, RT5659_PWR_LDO2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, - NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, + 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5659_PWR_VOL, RT5659_PWR_MIC_DET_BIT, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget rt5659_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5659_PWR_ANLG_3, RT5659_PWR_PLL_BIT, 0, + NULL, 0), SND_SOC_DAPM_SUPPLY("Mono Vref", RT5659_PWR_ANLG_1, RT5659_PWR_VREF3_BIT, 0, NULL, 0), @@ -2464,8 +2469,6 @@ static int set_dmic_power(struct snd_soc_dapm_widget *w, RT5659_ADC_MONO_R_ASRC_SFT, 0, NULL, 0), /* Input Side */ - SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5659_PWR_ANLG_2, RT5659_PWR_MB1_BIT, - 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5659_PWR_ANLG_2, RT5659_PWR_MB2_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICBIAS3", RT5659_PWR_ANLG_2, RT5659_PWR_MB3_BIT, @@ -3660,10 +3663,23 @@ static int rt5659_set_bias_level(struct snd_soc_component *component, static int rt5659_probe(struct snd_soc_component *component) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); struct rt5659_priv *rt5659 = snd_soc_component_get_drvdata(component); rt5659->component = component; + switch (rt5659->pdata.jd_src) { + case RT5659_JD_HDA_HEADER: + break; + + default: + snd_soc_dapm_new_controls(dapm, + rt5659_particular_dapm_widgets, + ARRAY_SIZE(rt5659_particular_dapm_widgets)); + break; + } + return 0; } -- 1.9.1

2 1

[PATCH AUTOSEL 5.10 30/62] ASoC: rt5645: add error checking to rt5645_probe function
by Sasha Levin 26 May '21

26 May '21

From: Phillip Potter <phil(a)philpotter.co.uk> [ Upstream commit 5e70b8e22b64eed13d5bbebcb5911dae65bf8c6b ] Check for return value from various snd_soc_dapm_* calls, as many of them can return errors and this should be handled. Also, reintroduce the allocation failure check for rt5645->eq_param as well. Make all areas where return values are checked lead to the end of the function in the case of an error. Finally, introduce a comment explaining how resources here are actually eventually cleaned up by the caller. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Phillip Potter <phil(a)philpotter.co.uk> Link: https://lore.kernel.org/r/20210503115736.2104747-56-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 48 +++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ed4b59ba63f3..a5665e992ecb 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3376,30 +3376,44 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (ret < 0) + goto exit; + if (rt5645->v_id < 3) { - snd_soc_dapm_add_routes(dapm, + ret = snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); + if (ret < 0) + goto exit; } break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); + if (ret < 0) + goto exit; break; } @@ -3407,9 +3421,17 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) + goto exit; } if (rt5645->pdata.long_name) @@ -3419,7 +3441,15 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - return 0; + if (!rt5645->eq_param) + ret = -ENOMEM; +exit: + /* + * If there was an error above, everything will be cleaned up by the + * caller if we return an error here. This will be done with a later + * call to rt5645_remove(). + */ + return ret; } static void rt5645_remove(struct snd_soc_component *component) -- 2.30.2

3 3

topology/pcm.c: remove duplicated AC97 hw format
by GitHub pull_request - opened 26 May '21

26 May '21

alsa-project/alsa-lib pull request #138 was opened from aiChaoSONG: Remove the second occurrence of AC97 hardware format from audio hardware format array. Signed-off-by: Chao Song <chao.song(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-lib/pull/138 Patch URL : https://github.com/alsa-project/alsa-lib/pull/138.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 5.10 29/62] Revert "ASoC: rt5645: fix a NULL pointer dereference"
by Sasha Levin 26 May '21

26 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1e0ce84215dbfd6065872e5d3755352da34f198b ] This reverts commit 51dd97d1df5fb9ac58b9b358e63e67b530f6ae21. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. Lots of things seem to be still allocated here and must be properly cleaned up if an error happens here. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-55-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 420003d062c7..ed4b59ba63f3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3419,9 +3419,6 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - if (!rt5645->eq_param) - return -ENOMEM; - return 0; } -- 2.30.2

2 2

[PATCH AUTOSEL 5.12 32/63] ASoC: cs43130: handle errors in cs43130_probe() properly
by Sasha Levin 26 May '21

26 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 2da441a6491d93eff8ffff523837fd621dc80389 ] cs43130_probe() does not do any valid error checking of things it initializes, OR what it does, it does not unwind properly if there are errors. Fix this up by moving the sysfs files to an attribute group so the driver core will correctly add/remove them all at once and handle errors with them, and correctly check for creating a new workqueue and unwinding if that fails. Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-58-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index c2b6f0ae6d57..80cd3ea0c157 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static struct attribute *hpload_attrs[] = { + &dev_attr_hpload_dc_l.attr, + &dev_attr_hpload_dc_r.attr, + &dev_attr_hpload_ac_l.attr, + &dev_attr_hpload_ac_r.attr, +}; +ATTRIBUTE_GROUPS(hpload); + static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, @@ -2302,23 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component) cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = device_create_file(component->dev, &dev_attr_hpload_dc_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_dc_r); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_r); - if (ret < 0) + ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) { + sysfs_remove_groups(&component->dev->kobj, hpload_groups); + return -ENOMEM; + } INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

3 3

[PATCH v3 0/4] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 25 May '21

25 May '21

During review of the RT5631 machine driver, Jon Hunter suggested that it will be better to squash all the current ASoC machine drivers into a single one. This suggestion is implemented by this patchset. The RT5631 support will come later with the ASUS Transformer changes. This series needs to be approved by Jaroslav Kysela before it can be merged. Changelog: v3: - Added components string as was suggested by Jaroslav Kysela to v2. - Renamed MCLK rate function that is used by max98090 and other codecs to make it look more generic. Added option for specifying CLK ID per device. This all was suggested by Jon Hunter to v2. v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (4): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers ASoC: tegra: Specify components string for each card ASoC: tegra: Specify components string for Nexus 7 sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 --------- sound/soc/tegra/tegra_asoc_machine.c | 761 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 47 ++ sound/soc/tegra/tegra_max98090.c | 276 ---------- sound/soc/tegra/tegra_rt5640.c | 222 -------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 358 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------ 13 files changed, 894 insertions(+), 2117 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

4 13

[PATCH v3] ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
by Peter Ujfalusi 25 May '21

25 May '21

Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces bug plock/pop noise over line out (green jack on the back) which can be fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine. Convert the existing entry for the same SSID to chain to apply this fixup as well. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, Changes since v2: - rebased on sound.git for-linus branch Changes since v1 - instead of deny_list use chained fixup to fix the plock/pop noise Changes since RFC: - Fixed up the PCI ID ever since I have this workstation (three weeks) the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 Missing details from alsa-info output: # lspci -nnk | grep -A2 Audio 00:1f.3 Audio device [0403]: Intel Corporation Comet Lake PCH cAVS [8086:06c8] DeviceName: Onboard - Sound Subsystem: Lenovo Device [17aa:1048] CPU: i9-10900k The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? [1] https://lore.kernel.org/alsa-devel/20210524152533.7479-1-peter.ujfalusi@lin… Thanks you, Peter Ujfalusi sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6571c3713732..90bf0d3a830a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6567,6 +6567,7 @@ enum { ALC295_FIXUP_HP_OMEN, ALC285_FIXUP_HP_SPECTRE_X360, ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, + ALC623_FIXUP_LENOVO_THINKSTATION_P340, }; static const struct hda_fixup alc269_fixups[] = { @@ -8139,6 +8140,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, }, + [ALC623_FIXUP_LENOVO_THINKSTATION_P340] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC283_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8457,7 +8464,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0xc019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xc022, "Clevo NH77[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), - SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -8724,6 +8731,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, + {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.31.1

2 1

[PATCH AUTOSEL 5.12 30/63] ASoC: rt5645: add error checking to rt5645_probe function
by Sasha Levin 25 May '21

25 May '21

From: Phillip Potter <phil(a)philpotter.co.uk> [ Upstream commit 5e70b8e22b64eed13d5bbebcb5911dae65bf8c6b ] Check for return value from various snd_soc_dapm_* calls, as many of them can return errors and this should be handled. Also, reintroduce the allocation failure check for rt5645->eq_param as well. Make all areas where return values are checked lead to the end of the function in the case of an error. Finally, introduce a comment explaining how resources here are actually eventually cleaned up by the caller. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Phillip Potter <phil(a)philpotter.co.uk> Link: https://lore.kernel.org/r/20210503115736.2104747-56-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 48 +++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index ab06133a85da..85ab33bdcdec 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3364,30 +3364,44 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (ret < 0) + goto exit; + if (rt5645->v_id < 3) { - snd_soc_dapm_add_routes(dapm, + ret = snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); + if (ret < 0) + goto exit; } break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); + if (ret < 0) + goto exit; break; } @@ -3395,9 +3409,17 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) + goto exit; } if (rt5645->pdata.long_name) @@ -3407,7 +3429,15 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - return 0; + if (!rt5645->eq_param) + ret = -ENOMEM; +exit: + /* + * If there was an error above, everything will be cleaned up by the + * caller if we return an error here. This will be done with a later + * call to rt5645_remove(). + */ + return ret; } static void rt5645_remove(struct snd_soc_component *component) -- 2.30.2

3 2

[PATCH 1/4] ASoC: cs42l42: Fix 1536000 Bit Clock instability
by Lucas Tanure 25 May '21

25 May '21

The 16 Bits, 2 channels, 48K sample rate use case needs to configure a safer pll_divout during the start of PLL After 800us from the start of PLL the correct pll_divout can be set Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 47 +++++++++++++++++++++++++------------- sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 33 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index b084acd1e86b..94788a55fa3b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -589,6 +589,7 @@ struct cs42l42_pll_params { u8 pll_divout; u32 mclk_int; u8 pll_cal_ratio; + u8 n; }; /* @@ -596,21 +597,21 @@ struct cs42l42_pll_params { * Table 4-5 from the Datasheet */ static const struct cs42l42_pll_params pll_ratio_table[] = { - { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125 }, - { 2822400, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128 }, - { 3000000, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128 }, - { 3072000, 0, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125 }, - { 4000000, 0, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96 }, - { 4096000, 0, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94 }, - { 5644800, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128 }, - { 6000000, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128 }, - { 6144000, 0, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125 }, - { 11289600, 0, 0, 0, 0, 0, 0, 0, 11289600, 0 }, - { 12000000, 0, 0, 0, 0, 0, 0, 0, 12000000, 0 }, - { 12288000, 0, 0, 0, 0, 0, 0, 0, 12288000, 0 }, - { 22579200, 1, 0, 0, 0, 0, 0, 0, 22579200, 0 }, - { 24000000, 1, 0, 0, 0, 0, 0, 0, 24000000, 0 }, - { 24576000, 1, 0, 0, 0, 0, 0, 0, 24576000, 0 } + { 1536000, 0, 1, 0x00, 0x7D, 0x000000, 0x03, 0x10, 12000000, 125, 2}, + { 2822400, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 3000000, 0, 1, 0x00, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 3072000, 0, 1, 0x00, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 4000000, 0, 1, 0x00, 0x30, 0x800000, 0x03, 0x10, 12000000, 96, 1}, + { 4096000, 0, 1, 0x00, 0x2E, 0xE00000, 0x03, 0x10, 12000000, 94, 1}, + { 5644800, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 11289600, 128, 1}, + { 6000000, 0, 1, 0x01, 0x40, 0x000000, 0x03, 0x10, 12000000, 128, 1}, + { 6144000, 0, 1, 0x01, 0x3E, 0x800000, 0x03, 0x10, 12000000, 125, 1}, + { 11289600, 0, 0, 0, 0, 0, 0, 0, 11289600, 0, 1}, + { 12000000, 0, 0, 0, 0, 0, 0, 0, 12000000, 0, 1}, + { 12288000, 0, 0, 0, 0, 0, 0, 0, 12288000, 0, 1}, + { 22579200, 1, 0, 0, 0, 0, 0, 0, 22579200, 0, 1}, + { 24000000, 1, 0, 0, 0, 0, 0, 0, 24000000, 0, 1}, + { 24576000, 1, 0, 0, 0, 0, 0, 0, 24576000, 0, 1} }; static int cs42l42_pll_config(struct snd_soc_component *component) @@ -746,8 +747,12 @@ static int cs42l42_pll_config(struct snd_soc_component *component) snd_soc_component_update_bits(component, CS42L42_PLL_CTL3, CS42L42_PLL_DIVOUT_MASK, - pll_ratio_table[i].pll_divout + (pll_ratio_table[i].pll_divout * pll_ratio_table[i].n) << CS42L42_PLL_DIVOUT_SHIFT); + if (pll_ratio_table[i].n != 1) + cs42l42->pll_divout = pll_ratio_table[i].pll_divout; + else + cs42l42->pll_divout = 0; snd_soc_component_update_bits(component, CS42L42_PLL_CAL_RATIO, CS42L42_PLL_CAL_RATIO_MASK, @@ -902,6 +907,16 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); + + if (cs42l42->pll_divout) { + usleep_range(CS42L42_PLL_DIVOUT_TIME_US, + CS42L42_PLL_DIVOUT_TIME_US * 2); + snd_soc_component_update_bits(component, CS42L42_PLL_CTL3, + CS42L42_PLL_DIVOUT_MASK, + cs42l42->pll_divout << + CS42L42_PLL_DIVOUT_SHIFT); + } + ret = regmap_read_poll_timeout(cs42l42->regmap, CS42L42_PLL_LOCK_STATUS, regval, diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 24f7be228d5f..7bf05ff05f74 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -755,6 +755,7 @@ #define CS42L42_NUM_SUPPLIES 5 #define CS42L42_BOOT_TIME_US 3000 +#define CS42L42_PLL_DIVOUT_TIME_US 800 #define CS42L42_CLOCK_SWITCH_DELAY_US 150 #define CS42L42_PLL_LOCK_POLL_US 250 #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 @@ -777,6 +778,7 @@ struct cs42l42_private { int bclk; u32 sclk; u32 srate; + u8 pll_divout; u8 plug_state; u8 hs_type; u8 ts_inv; -- 2.31.1

3 8

[PATCH v2] ALSA: hda/realtek: Chain in pop reduction fixup for ThinkStation P340
by Peter Ujfalusi 25 May '21

25 May '21

Lenovo ThinkStation P340 uses ALC623 codec (SSID 17aa:1048) and it produces bug plock/pop noise over line out (green jack on the back) which can be fixed by applying ALC269_FIXUP_NO_SHUTUP tot he machine. Convert the existing entry for the same SSID to chain to apply this fixup as well. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, Changes since v1 - instead of deny_list use chained fixup to fix the plock/pop noise Changes since RFC: - Fixed up the PCI ID ever since I have this workstation (three weeks) the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 Missing details from alsa-info output: # lspci -nnk | grep -A2 Audio 00:1f.3 Audio device [0403]: Intel Corporation Comet Lake PCH cAVS [8086:06c8] DeviceName: Onboard - Sound Subsystem: Lenovo Device [17aa:1048] CPU: i9-10900k The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? [1] https://lore.kernel.org/alsa-devel/20210524152533.7479-1-peter.ujfalusi@lin… Thanks you, Peter Ujfalusi sound/pci/hda/patch_realtek.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3e269de84079..e2e0ee7b137d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6543,6 +6543,7 @@ enum { ALC295_FIXUP_ASUS_DACS, ALC295_FIXUP_HP_OMEN, ALC285_FIXUP_HP_SPECTRE_X360, + ALC623_FIXUP_LENOVO_THINKSTATION_P340, }; static const struct hda_fixup alc269_fixups[] = { @@ -8109,6 +8110,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, }, + [ALC623_FIXUP_LENOVO_THINKSTATION_P340] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + .chained = true, + .chain_id = ALC283_FIXUP_HEADSET_MIC, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8412,7 +8419,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL53RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x17aa, 0x1036, "Lenovo P520", ALC233_FIXUP_LENOVO_MULTI_CODECS), - SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x17aa, 0x1048, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE), @@ -8677,6 +8684,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.31.1

3 2

[PATCH] ALSA: hda: Add Lenovo ThinkStation P340 to power_save_denylist
by Peter Ujfalusi 25 May '21

25 May '21

On playback start there is a huge plock/pop noise via the green jack used in Line out mode. The only way I was able to fix it to set the power_save to 0 for the snd_hda_intel. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, Changes since RFC: - Fixed up the PCI ID ever since I have this workstation (three weeks) the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 Missing details from alsa-info output: # lspci -nnk | grep -A2 Audio 00:1f.3 Audio device [0403]: Intel Corporation Comet Lake PCH cAVS [8086:06c8] DeviceName: Onboard - Sound Subsystem: Lenovo Device [17aa:1048] CPU: i9-10900k The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? Thanks you, Peter Ujfalusi sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..9cacd1cb211b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2213,6 +2213,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Loud plock/pop noise via the green jack on playback start */ + SND_PCI_QUIRK(0x17aa, 0x1048, "Lenovo ThinkStation P340", 0), {} }; #endif /* CONFIG_PM */ -- 2.31.1

3 6

[PATCH] ASoC: Intel: boards: Select SND_SOC_HDAC_HDMI for Soundwire machine driver
by Richard Fitzgerald 25 May '21

25 May '21

Add select of SND_SOC_HDAC_HDMI for SND_SOC_INTEL_SOUNDWIRE_SOF_MACH. Without this the build results in missing references to hdac_hdmi_jack_port_init and hdac_hdmi_jack_init. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/intel/boards/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index eef5f4ac87c5..6a1639b14b67 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -601,6 +601,7 @@ config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH select SND_SOC_DMIC select SND_SOC_INTEL_HDA_DSP_COMMON select SND_SOC_INTEL_SOF_MAXIM_COMMON + select SND_SOC_HDAC_HDMI help Add support for Intel SoundWire-based platforms connected to MAX98373, RT700, RT711, RT1308 and RT715 -- 2.20.1

3 3

[PATCH 00/11] drm/vc4: hdmi: Enable Channel Mapping, IEC958, HBR Passthrough using hdmi-codec
by Maxime Ripard 25 May '21

25 May '21

Hi, hdmi-codec allows to have a lot of HDMI-audio related infrastructure in place, it's missing a few controls to be able to provide HBR passthrough. This series adds more infrastructure for the drivers, and leverages it in the vc4 HDMI controller driver. One thing that felt a bit weird is that even though https://www.kernel.org/doc/html/latest/sound/kernel-api/writing-an-alsa-dri… mentions that the iec958 mask control should be a mixer control and the default control should be a PCM one, it feels a bit weird to have two different control type for two controls so similar, and other drivers are pretty inconsistent with this. Should we update the documentation? Thanks! Maxime Dom Cobley (5): drm/vc4: hdmi: Set HD_CTL_WHOLSMP and HD_CTL_CHALIGN_SET drm/vc4: hdmi: Set HDMI_MAI_FMT drm/vc4: hdmi: Set VC4_HDMI_MAI_CONFIG_FORMAT_REVERSE drm/vc4: hdmi: Remove firmware logic for MAI threshold setting ARM: dts: bcm2711: Tune DMA parameters for HDMI audio Maxime Ripard (6): snd: iec958: split status creation and fill ASoC: hdmi-codec: Rework to support more controls ASoC: hdmi-codec: Add iec958 controls ASoC: hdmi-codec: Add a prepare hook drm/vc4: hdmi: Register HDMI codec drm/vc4: hdmi: Remove redundant variables arch/arm/boot/dts/bcm2711.dtsi | 4 +- drivers/gpu/drm/vc4/Kconfig | 1 + drivers/gpu/drm/vc4/vc4_hdmi.c | 322 ++++++++++++++------------------- drivers/gpu/drm/vc4/vc4_hdmi.h | 5 +- drivers/gpu/drm/vc4/vc4_regs.h | 30 +++ include/sound/hdmi-codec.h | 12 +- include/sound/pcm_iec958.h | 8 + sound/core/pcm_iec958.c | 131 +++++++++----- sound/soc/codecs/hdmi-codec.c | 219 +++++++++++++++++----- 9 files changed, 456 insertions(+), 276 deletions(-) -- 2.31.1

2 17

[PATCH -next] ALSA: control_led - use DEVICE_ATTR_*() macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_*() helper instead of plain DEVICE_ATTR, which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/control_led.c | 34 +++++++++++++++++++--------------- 1 file changed, 19 insertions(+), 15 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index 25f57c14f294..a5e751f26d46 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -375,7 +375,7 @@ static void snd_ctl_led_disconnect(struct snd_card *card) * sysfs */ -static ssize_t show_mode(struct device *dev, +static ssize_t mode_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -390,7 +390,8 @@ static ssize_t show_mode(struct device *dev, return sprintf(buf, "%s\n", str); } -static ssize_t store_mode(struct device *dev, struct device_attribute *attr, +static ssize_t mode_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -419,7 +420,7 @@ static ssize_t store_mode(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t show_brightness(struct device *dev, +static ssize_t brightness_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_ctl_led *led = container_of(dev, struct snd_ctl_led, dev); @@ -427,8 +428,8 @@ static ssize_t show_brightness(struct device *dev, return sprintf(buf, "%u\n", ledtrig_audio_get(led->trigger_type)); } -static DEVICE_ATTR(mode, 0644, show_mode, store_mode); -static DEVICE_ATTR(brightness, 0444, show_brightness, NULL); +static DEVICE_ATTR_RW(mode); +static DEVICE_ATTR_RO(brightness); static struct attribute *snd_ctl_led_dev_attrs[] = { &dev_attr_mode.attr, @@ -562,22 +563,25 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si return count; } -static ssize_t parse_attach(struct device *dev, struct device_attribute *attr, +static ssize_t attach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, true); } -static ssize_t parse_detach(struct device *dev, struct device_attribute *attr, +static ssize_t detach_store(struct device *dev, + struct device_attribute *attr, const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); return set_led_id(led_card, buf, count, false); } -static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t reset_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); int err; @@ -590,8 +594,8 @@ static ssize_t ctl_reset(struct device *dev, struct device_attribute *attr, return count; } -static ssize_t ctl_list(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t list_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct snd_ctl_led_card *led_card = container_of(dev, struct snd_ctl_led_card, dev); struct snd_card *card; @@ -624,10 +628,10 @@ static ssize_t ctl_list(struct device *dev, return buf2 - buf; } -static DEVICE_ATTR(attach, 0200, NULL, parse_attach); -static DEVICE_ATTR(detach, 0200, NULL, parse_detach); -static DEVICE_ATTR(reset, 0200, NULL, ctl_reset); -static DEVICE_ATTR(list, 0444, ctl_list, NULL); +static DEVICE_ATTR_WO(attach); +static DEVICE_ATTR_WO(detach); +static DEVICE_ATTR_WO(reset); +static DEVICE_ATTR_RO(list); static struct attribute *snd_ctl_led_card_attrs[] = { &dev_attr_attach.attr, -- 2.17.1

3 2

[PATCH] ALSA: line6: Improve poor error handling in line6_init_cap_control
by Hyeonggon Yoo 25 May '21

25 May '21

line6_init_cap_control doesn't free resources properly when allocations like kmalloc, usb_alloc_urb fails. It can cause memory leak when some allocations succeed, and then an error occurs. This patch makes line6_init_cap_control to properly free the resources, freeing previously allocated resources when there's an error. Signed-off-by: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> --- sound/usb/line6/driver.c | 36 +++++++++++++++++++++++++++--------- 1 file changed, 27 insertions(+), 9 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 9602929b7de9..6991cb855023 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -688,34 +688,52 @@ static int line6_init_cap_control(struct usb_line6 *line6) /* initialize USB buffers: */ line6->buffer_listen = kmalloc(LINE6_BUFSIZE_LISTEN, GFP_KERNEL); - if (!line6->buffer_listen) - return -ENOMEM; + if (!line6->buffer_listen) { + ret = -ENOMEM; + goto fail_ret; + } line6->urb_listen = usb_alloc_urb(0, GFP_KERNEL); - if (!line6->urb_listen) - return -ENOMEM; + if (!line6->urb_listen) { + ret = -ENOMEM; + goto fail1; + } if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) { line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); - if (!line6->buffer_message) - return -ENOMEM; + if (!line6->buffer_message) { + ret = -ENOMEM; + goto fail2; + } ret = line6_init_midi(line6); if (ret < 0) - return ret; + goto fail3; } else { ret = line6_hwdep_init(line6); if (ret < 0) - return ret; + goto fail2; } ret = line6_start_listen(line6); if (ret < 0) { dev_err(line6->ifcdev, "cannot start listening: %d\n", ret); - return ret; + if (line6->properties->capabilities & LINE6_CAP_CONTROL_MIDI) + goto fail3; + else + goto fail2; } return 0; + +fail3: + kfree(line6->buffer_message); +fail2: + usb_free_urb(line6->urb_listen); +fail1: + kfree(line6->buffer_listen); +fail_ret: + return ret; } static void line6_startup_work(struct work_struct *work) -- 2.25.1

2 3

[GIT PULL] ASoC fixes for v5.13-rc3
by Mark Brown 25 May '21

25 May '21

The following changes since commit 25c4a9b614f101bb9f3e687960815db7dc439c0f: ASoC: simple-card: Fix breakage on kontron-sl28-var3-ads2 (2021-04-23 18:13:32 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.13-rc3 for you to fetch changes up to af2702549d68519ac78228e915d9b2c199056787: ASoC: qcom: lpass-cpu: Use optional clk APIs (2021-05-21 13:12:29 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.13 A collection of fixes that have come in since the merge window, mainly device specific things. The fixes to the generic cards from Morimoto-san are handling regressions that were introduced in the merge window on at least the Kontron sl28-var3-ads2. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency Bixuan Cui (1): ASoC: codecs: lpass-tx-macro: add missing MODULE_DEVICE_TABLE Charles Keepax (5): ASoC: cs53l30: Add missing regmap use_single config ASoC: cs42l73: Add missing regmap use_single config ASoC: cs35l34: Add missing regmap use_single config ASoC: cs35l32: Add missing regmap use_single config ASoC: cs42l52: Minor tidy up of error paths Dan Carpenter (1): ASoC: cs35l33: fix an error code in probe() Hans de Goede (2): ASoC: Intel: bytcr_rt5640: Add quirk for the Glavey TM800A550L tablet ASoC: Intel: bytcr_rt5640: Add quirk for the Lenovo Miix 3-830 tablet Jerome Brunet (2): ASoC: stm32: do not request a new clock consummer reference ASoC: da7219: do not request a new clock consummer reference Kuninori Morimoto (5): ASoC: simple-card: add simple_parse_node() ASoC: simple-card: add simple_link_init() ASoC: audio-graph: tidyup graph_dai_link_of_dpcm() ASoC: audio-graph: tidyup graph_parse_node() ASoC: soc-dai.h: Align the word of comment for SND_SOC_DAIFMT_CBC_CFC Marco Felsch (1): ASoC: max98088: fix ni clock divider calculation Mark Brown (2): Merge series "ASoC: Revert clk_hw_get_clk() cleanup" from Jerome Brunet <jbrunet(a)baylibre.com>: Merge series "ASoC: simple-card / audio-graph re-cleanup" from Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>: Ranjani Sridharan (1): ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware Richard Fitzgerald (1): ASoC: cs42l42: Regmap must use_single_read/write Shengjiu Wang (1): ASoC: ak5558: Correct the dai name for ak5552 Shuming Fan (1): ASoC: rt711-sdca: fix the function number of SDCA control for feature unit 0x1E Srinivas Kandagatla (1): ASoC: codecs: lpass-rx-macro: add missing MODULE_DEVICE_TABLE Stephen Boyd (1): ASoC: qcom: lpass-cpu: Use optional clk APIs Vijendar Mukunda (1): ASoC: amd: fix for pcm_read() error Zou Wei (1): ASoC: sti-sas: add missing MODULE_DEVICE_TABLE include/sound/soc-dai.h | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 10 -- sound/soc/amd/raven/acp3x.h | 1 + sound/soc/amd/raven/pci-acp3x.c | 15 +++ sound/soc/codecs/ak5558.c | 2 +- sound/soc/codecs/cs35l32.c | 3 + sound/soc/codecs/cs35l33.c | 1 + sound/soc/codecs/cs35l34.c | 3 + sound/soc/codecs/cs42l42.c | 3 + sound/soc/codecs/cs42l56.c | 7 +- sound/soc/codecs/cs42l73.c | 3 + sound/soc/codecs/cs53l30.c | 3 + sound/soc/codecs/da7219.c | 5 +- sound/soc/codecs/lpass-rx-macro.c | 1 + sound/soc/codecs/lpass-tx-macro.c | 1 + sound/soc/codecs/max98088.c | 13 ++- sound/soc/codecs/rt711-sdca.c | 4 +- sound/soc/codecs/sti-sas.c | 1 + sound/soc/fsl/Kconfig | 1 + sound/soc/generic/audio-graph-card.c | 57 ++++++------ sound/soc/generic/simple-card.c | 168 +++++++++++++++++----------------- sound/soc/intel/boards/bytcr_rt5640.c | 25 +++++ sound/soc/qcom/lpass-cpu.c | 12 +-- sound/soc/sof/intel/hda-dai.c | 5 + sound/soc/stm/stm32_sai_sub.c | 5 +- 25 files changed, 202 insertions(+), 149 deletions(-)

2 1

[PATCH] ASoC: meson: use dev_err_probe
by Jerome Brunet 25 May '21

25 May '21

Use dev_err_probe() helper function to handle probe deferral. It removes the open coded test for -EPROBE_DEFER but more importantly, it sets the deferral reason in debugfs which is great for debugging. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/aiu.c | 17 +++++++---------- sound/soc/meson/axg-fifo.c | 11 +++++------ sound/soc/meson/axg-pdm.c | 9 +++------ sound/soc/meson/axg-spdifin.c | 6 ++---- sound/soc/meson/axg-spdifout.c | 6 ++---- sound/soc/meson/axg-tdm-formatter.c | 18 ++++++------------ sound/soc/meson/axg-tdm-interface.c | 10 ++++------ sound/soc/meson/meson-card-utils.c | 4 ++-- sound/soc/meson/t9015.c | 7 +++---- 9 files changed, 34 insertions(+), 54 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index ba15d5762b0b..37036adf14ce 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -219,31 +219,29 @@ static int aiu_clk_get(struct device *dev) aiu->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(aiu->pclk)) { - if (PTR_ERR(aiu->pclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu pclk\n"); + dev_err_probe(dev, PTR_ERR(aiu->pclk), + "Can't get the aiu pclk\n"); return PTR_ERR(aiu->pclk); } aiu->spdif_mclk = devm_clk_get(dev, "spdif_mclk"); if (IS_ERR(aiu->spdif_mclk)) { - if (PTR_ERR(aiu->spdif_mclk) != -EPROBE_DEFER) - dev_err(dev, "Can't get the aiu spdif master clock\n"); + dev_err_probe(dev, PTR_ERR(aiu->spdif_mclk), + "Can't get the aiu spdif master clock\n"); return PTR_ERR(aiu->spdif_mclk); } ret = aiu_clk_bulk_get(dev, aiu_i2s_ids, ARRAY_SIZE(aiu_i2s_ids), &aiu->i2s); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the i2s clocks\n"); + dev_err_probe(dev, ret, "Can't get the i2s clocks\n"); return ret; } ret = aiu_clk_bulk_get(dev, aiu_spdif_ids, ARRAY_SIZE(aiu_spdif_ids), &aiu->spdif); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Can't get the spdif clocks\n"); + dev_err_probe(dev, ret, "Can't get the spdif clocks\n"); return ret; } @@ -282,8 +280,7 @@ static int aiu_probe(struct platform_device *pdev) ret = device_reset(dev); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(dev, "Failed to reset device\n"); + dev_err_probe(dev, ret, "Failed to reset device\n"); return ret; } diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index b9af2d513e09..d670a32d56ab 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -352,17 +352,16 @@ int axg_fifo_probe(struct platform_device *pdev) fifo->pclk = devm_clk_get(dev, NULL); if (IS_ERR(fifo->pclk)) { - if (PTR_ERR(fifo->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %ld\n", - PTR_ERR(fifo->pclk)); + dev_err_probe(dev, PTR_ERR(fifo->pclk), + "failed to get pclk: %ld\n", PTR_ERR(fifo->pclk)); return PTR_ERR(fifo->pclk); } fifo->arb = devm_reset_control_get_exclusive(dev, NULL); if (IS_ERR(fifo->arb)) { - if (PTR_ERR(fifo->arb) != -EPROBE_DEFER) - dev_err(dev, "failed to get arb reset: %ld\n", - PTR_ERR(fifo->arb)); + dev_err_probe(dev, PTR_ERR(fifo->arb), + "failed to get arb reset: %ld\n", + PTR_ERR(fifo->arb)); return PTR_ERR(fifo->arb); } diff --git a/sound/soc/meson/axg-pdm.c b/sound/soc/meson/axg-pdm.c index bfd37d49a73e..f630311b5b29 100644 --- a/sound/soc/meson/axg-pdm.c +++ b/sound/soc/meson/axg-pdm.c @@ -613,24 +613,21 @@ static int axg_pdm_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->dclk = devm_clk_get(dev, "dclk"); if (IS_ERR(priv->dclk)) { ret = PTR_ERR(priv->dclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } priv->sysclk = devm_clk_get(dev, "sysclk"); if (IS_ERR(priv->sysclk)) { ret = PTR_ERR(priv->sysclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get dclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get dclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifin.c b/sound/soc/meson/axg-spdifin.c index d0d09f945b48..8f587c3bb5d7 100644 --- a/sound/soc/meson/axg-spdifin.c +++ b/sound/soc/meson/axg-spdifin.c @@ -481,16 +481,14 @@ static int axg_spdifin_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->refclk = devm_clk_get(dev, "refclk"); if (IS_ERR(priv->refclk)) { ret = PTR_ERR(priv->refclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-spdifout.c b/sound/soc/meson/axg-spdifout.c index e769a5ee6e27..58e02f6898b7 100644 --- a/sound/soc/meson/axg-spdifout.c +++ b/sound/soc/meson/axg-spdifout.c @@ -424,16 +424,14 @@ static int axg_spdifout_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { ret = PTR_ERR(priv->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } priv->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(priv->mclk)) { ret = PTR_ERR(priv->mclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-formatter.c b/sound/soc/meson/axg-tdm-formatter.c index cab7fa2851aa..108f11e14006 100644 --- a/sound/soc/meson/axg-tdm-formatter.c +++ b/sound/soc/meson/axg-tdm-formatter.c @@ -284,8 +284,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(formatter->pclk)) { ret = PTR_ERR(formatter->pclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get pclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get pclk: %d\n", ret); return ret; } @@ -293,8 +292,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(formatter->sclk)) { ret = PTR_ERR(formatter->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -302,8 +300,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(formatter->lrclk)) { ret = PTR_ERR(formatter->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -311,8 +308,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->sclk_sel = devm_clk_get(dev, "sclk_sel"); if (IS_ERR(formatter->sclk_sel)) { ret = PTR_ERR(formatter->sclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk_sel: %d\n", ret); return ret; } @@ -320,8 +316,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->lrclk_sel = devm_clk_get(dev, "lrclk_sel"); if (IS_ERR(formatter->lrclk_sel)) { ret = PTR_ERR(formatter->lrclk_sel); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk_sel: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk_sel: %d\n", ret); return ret; } @@ -329,8 +324,7 @@ int axg_tdm_formatter_probe(struct platform_device *pdev) formatter->reset = devm_reset_control_get_optional_exclusive(dev, NULL); if (IS_ERR(formatter->reset)) { ret = PTR_ERR(formatter->reset); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get reset: %d\n", ret); + dev_err_probe(dev, ret, "failed to get reset: %d\n", ret); return ret; } diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 87cac440b369..a2c7f658f6b4 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -519,8 +519,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->sclk = devm_clk_get(dev, "sclk"); if (IS_ERR(iface->sclk)) { ret = PTR_ERR(iface->sclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get sclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get sclk: %d\n", ret); return ret; } @@ -528,8 +527,7 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) iface->lrclk = devm_clk_get(dev, "lrclk"); if (IS_ERR(iface->lrclk)) { ret = PTR_ERR(iface->lrclk); - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get lrclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get lrclk: %d\n", ret); return ret; } @@ -545,8 +543,8 @@ static int axg_tdm_iface_probe(struct platform_device *pdev) if (ret == -ENOENT) { iface->mclk = NULL; } else { - if (ret != -EPROBE_DEFER) - dev_err(dev, "failed to get mclk: %d\n", ret); + dev_err_probe(dev, ret, "failed to get mclk: %d\n", + ret); return ret; } } diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c index 300ac8be46ef..ac062c03cf33 100644 --- a/sound/soc/meson/meson-card-utils.c +++ b/sound/soc/meson/meson-card-utils.c @@ -86,8 +86,8 @@ int meson_card_parse_dai(struct snd_soc_card *card, ret = of_parse_phandle_with_args(node, "sound-dai", "#sound-dai-cells", 0, &args); if (ret) { - if (ret != -EPROBE_DEFER) - dev_err(card->dev, "can't parse dai %d\n", ret); + dev_err_probe(card->dev, ret, + "can't parse sound-dai at %pOFn\n", node); return ret; } *dai_of_node = args.np; diff --git a/sound/soc/meson/t9015.c b/sound/soc/meson/t9015.c index 4c1349dd1e06..7ff16f51dbd4 100644 --- a/sound/soc/meson/t9015.c +++ b/sound/soc/meson/t9015.c @@ -259,15 +259,14 @@ static int t9015_probe(struct platform_device *pdev) priv->pclk = devm_clk_get(dev, "pclk"); if (IS_ERR(priv->pclk)) { - if (PTR_ERR(priv->pclk) != -EPROBE_DEFER) - dev_err(dev, "failed to get core clock\n"); + dev_err_probe(dev, PTR_ERR(priv->pclk), + "failed to get core clock\n"); return PTR_ERR(priv->pclk); } priv->avdd = devm_regulator_get(dev, "AVDD"); if (IS_ERR(priv->avdd)) { - if (PTR_ERR(priv->avdd) != -EPROBE_DEFER) - dev_err(dev, "failed to AVDD\n"); + dev_err_probe(dev, PTR_ERR(priv->avdd), "failed to AVDD\n"); return PTR_ERR(priv->avdd); } -- 2.32.0.rc0

2 2

[PATCH -next] ALSA: pcm: use DEVICE_ATTR_RO macro
by YueHaibing 25 May '21

25 May '21

Use DEVICE_ATTR_RO() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/core/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm.c b/sound/core/pcm.c index b163164a83ec..6fd3677685d7 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1004,7 +1004,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->pstr->substream_opened--; } -static ssize_t show_pcm_class(struct device *dev, +static ssize_t pcm_class_show(struct device *dev, struct device_attribute *attr, char *buf) { struct snd_pcm_str *pstr = container_of(dev, struct snd_pcm_str, dev); @@ -1024,7 +1024,7 @@ static ssize_t show_pcm_class(struct device *dev, return sprintf(buf, "%s\n", str); } -static DEVICE_ATTR(pcm_class, 0444, show_pcm_class, NULL); +static DEVICE_ATTR_RO(pcm_class); static struct attribute *pcm_dev_attrs[] = { &dev_attr_pcm_class.attr, NULL -- 2.17.1

2 1

[PATCH] ALSA: ac97: fix PM reference leak in ac97_bus_remove()
by Yufen Yu 25 May '21

25 May '21

pm_runtime_get_sync will increment pm usage counter even it failed. Forgetting to putting operation will result in reference leak here. Fix it by replacing it with pm_runtime_resume_and_get to keep usage counter balanced. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yufen Yu <yuyufen(a)huawei.com> --- sound/ac97/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index d9077e91382b..6ddf646cda65 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -520,7 +520,7 @@ static int ac97_bus_remove(struct device *dev) struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver); int ret; - ret = pm_runtime_get_sync(dev); + ret = pm_runtime_resume_and_get(dev); if (ret < 0) return ret; -- 2.25.4

2 1

[PATCH v2 0/6] ALSA: Prep work for PCI rescan support
by Takashi Iwai 25 May '21

25 May '21

Hi, this is a revised patch set as a prep work for the future-planned PCI rescan support(*). The essential part is the patch to track in-flight tasks, and the rest are cleanups and fixes in the core code. With those changes, the driver can perform a sort of software suspend on the device without the hardware intervention, which is required during the PCI BAR movement, for example. v1->v2: * Move / drop unnecessary snd_power_wait() calls * Rephrase changelogs and comments, reorder patches Takashi (*) https://lore.kernel.org/alsa-devel/e25017c6-e5e4-7a24-e793-14a2e70a434e@amd… === Takashi Iwai (6): ALSA: core: Use READ_ONCE() / WRITE_ONCE() for power state change ALSA: control: Track in-flight control read/write/tlv accesses ALSA: control: Drop superfluous snd_power_wait() calls ALSA: control: Minor optimization for SNDRV_CTL_IOCTL_POWER_STATE ALSA: Drop superfluous argument from snd_power_wait() ALSA: pcm: Block the release until the system resume finishes include/sound/core.h | 50 ++++++++++++++++++++++++++++++++++--- sound/core/control.c | 38 +++++++++++++--------------- sound/core/control_compat.c | 14 +++-------- sound/core/init.c | 46 ++++++++++++++++++++++++++++------ sound/core/pcm_native.c | 6 ++++- sound/soc/soc-core.c | 2 +- sound/usb/usx2y/us122l.c | 2 +- 7 files changed, 113 insertions(+), 45 deletions(-) -- 2.26.2

3 9

[PATCH v2 0/4] ALSA: firewire-lib: drop initial NODATA packets or empty packets
by Takashi Sakamoto 25 May '21

25 May '21

Hi, This patchset is take 2 of my previous one; * https://lore.kernel.org/alsa-devel/20210523124114.272134-1-o-takashi@sakamo… The devices based on BeBoB ASICs or the devices in Tascam FireWire series transfer a batch of NODATA packet or empty packet in initial step of streaming. To avoid processing them, current implementation uses an option to skip processing content of tx packet during some initial cycles. However, the hard-coded number is not enough useful. In 1st patch, ALSA IEC 61883-1/6 packet streaming engine becomes to drop the initial packets. As a result, The tx_init_skip_cycles argument of amdtp_domain_start() function changes its meaning. In the following patches, ALSA bebob driver is refactored. Changes from v1: * Fix -Wunused-but-set-variable warning reported by 0day-ci * Add 2nd patch to obsolete unused member of structure Takashi Sakamoto (4): ALSA: firewire-lib: drop initial NODATA or empty packet ALSA: firewire-lib: obsolete callbacked member ALSA: bebob: cancel switching connection order ALSA: bebob: distinguish M-Audio ProFire Lightbridge quirk sound/firewire/amdtp-stream.c | 145 +++++++++++++++++++--------- sound/firewire/amdtp-stream.h | 6 +- sound/firewire/bebob/bebob.c | 10 +- sound/firewire/bebob/bebob.h | 1 + sound/firewire/bebob/bebob_stream.c | 25 ++--- 5 files changed, 122 insertions(+), 65 deletions(-) -- 2.27.0

2 5

[PATCH v2 0/2] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 25 May '21

25 May '21

During review of the RT5631 machine driver, Jon Hunter suggested that it will be better to squash all the current ASoC machine drivers into a single one. This suggestion is implemented by this patchset. The RT5631 support will come later with the ASUS Transformer changes. This series needs to be approved by Jaroslav Kysela before it can be merged. Changelog: v2: - Dropped use of of_device_compatible_match(), like it was suggested by Rob Herring in a review comment to v1. - Added patch that sets card's driver_name of as Tegra ASoC drivers. In a comment to v1 Jaroslav Kysela suggested that the Tegra drivers don't set the card name properly and he was right. I opened pull request with the new Tegra UCMs and updated lookup paths for older UCMs [1]. [1] https://github.com/alsa-project/alsa-ucm-conf/pull/92 Dmitry Osipenko (2): ASoC: tegra: Set driver_name=tegra for all machine drivers ASoC: tegra: Unify ASoC machine drivers sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 ---------- sound/soc/tegra/tegra_asoc_machine.c | 732 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 45 ++ sound/soc/tegra/tegra_max98090.c | 276 ---------- sound/soc/tegra/tegra_rt5640.c | 222 -------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 357 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------- 13 files changed, 862 insertions(+), 2117 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

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RE: Re: [PATCH] ALSA: pcm: Need to check whether runtime is valid or not
by EUNWOO KIM 25 May '21

25 May '21

> > > Sender : Takashi Iwai <tiwai(a)suse.de> > > > Date : 2021-05-12 16:08 (GMT+9) > > > Title : Re: [PATCH] ALSA: pcm: Need to check whether runtime is v alid or not > > > > > > On Wed, 12 May 2021 06:43:23 +0200, > > > Chanho Park wrote: > > > > > > > > From: eunwoo kim <ew.kim(a)samsung.com> > > > > > > > > Since snd_pcm_ioctl_sw_params_compat has no runtime variable ch eck, > > > > if application call the ioctl after close, it can make kernel c rash. > > > > So, snd_pcm_ioctl_sw_params_compat needs to check the runtime v ariable > > > > at the beginning of the function. > > > > > > > In principle, you cannot call ioctl for an already closed file. > > > > Or do you mean other code path? > > > > > > Yes, other code path such as application layer or alsa framwork l ayer. > > > > But how can it go over the 32bit compat ioctl layer...? > > It's always tied with a file object, so the runtime object is > > assured. > > > > > > Yes, I think so too. but in fact, some case can call 32bit comp at ioctl. > > > > > In our case, suspend call is releasing all running pcm device. ot her side some application want to start music. > > > it make critical section between suspend and pcm_open from applic ation. > > > this is error and we have to solve this problem without this patc h. > > > but I think kernel don't make kernel crash even if application or system architecture have problem. > > > > I guess you're scratching a wrong surface. > > > > > could you tell me more detail? why do you think that. > > > In 64bit native case, it didn't make kernel crash on same test ca se. because native always check runtime in ioctl. > > > Try to put WARN_ON() there. If you can catch the real case, it'd b e > > worth to merge. Otherwise, it's just a wrong place to look at. > > I already found root cause in our issue case. And I am fixing this is sue via changing sequence. > > I think that you want to know issue case because why this patch need. > So I explain more about our case. > 1. application playback sound in android. > 2. call pcm_open in tinyalsa. > 3. call file open and hw param in pcm_open > 4. go to suspend > 5. freezing all task before calling sw param in pcm_open. > 6. kernel layer try to go suspend > 7. release all running substream, because all HW IPs must to go idle for power save mode. > -> detach runtime from substream. > 8. go to resume. > 9. call sw param in pcm_open. > 10-1. native case : has runtime check in ioctl -> return error -> aud io hal layer recovery hw connect. > 10-2. 32bit compat case : has not runtime check -> runtime is not val id -> make kernel crash. > > - if AAOS can support wakelock, we can postpone suspend until finishe d pcm_open. but AAOS didn't support wakelock. > I'm afraid that the approach is wrong. It breaks the fundamental PCM > state change rule. Yes, that is wrong approach I know. So I am changing that. > I guess it's a downstream driver that does that? If so, tweaking > something superfluous in the upstream code is unacceptable. > BTW, if you want further discussions, please don't drop Cc to > alsa-devel ML. Safety is very important in automotive. so it must not to make kernel crash even if someone make wrong code. because it can connect to accident. If 32bit compat don't need checking-runtime, why do pcm_native check runtime? thanks, Takashi [update?userid=ew.kim&do=bWFpbElEPTIwMjEwNTE5MjMxNjE1ZXBjbXMycDE4Y2NlMT Q1YzJhOWMwZDk5ZmViZjMxNDBjMTY3ZDIyNiZyZWNpcGllbnRBZGRyZXNzPWFsc2EtZGV2Z WxAYWxzYS1wcm9qZWN0Lm9yZw__]

3 3

[RFC] ALSA: hda: Add Lenovo ThinkStation P340 to power_save_denylist
by Peter Ujfalusi 25 May '21

25 May '21

On playback start there is a huge plock/pop noise via the green jack used in Line out mode. The only way I was able to fix it to set the power_save to 0 for the snd_hda_intel. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi Takashi, ever since I have this workstation the plock/pop noise on playback start bugged me. So far the only solution I have found is to disable the power_save which I find acceptable on a desktop compared to the pops. The alsa-info.sh output is can be found here: http://alsa-project.org/db/?f=414b1b236fc73db386ad4f938bc3b304f6b30b05 The machine have ALC623 codec and the jacks are correctly discovered and they do what they supposed to be. When I have the headset connected to the front and audio is routed there the line out (green jack) from the back still prodices the plock/pop. Is there a known quirk for similar issue or a better way to handle it? Thanks you, Peter Ujfalusi sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 79ade335c8a0..e16d93d8291f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2213,6 +2213,8 @@ static const struct snd_pci_quirk power_save_denylist[] = { SND_PCI_QUIRK(0x17aa, 0x36a7, "Lenovo C50 All in one", 0), /* https://bugs.launchpad.net/bugs/1821663 */ SND_PCI_QUIRK(0x1631, 0xe017, "Packard Bell NEC IMEDIA 5204", 0), + /* Loud pluck/pop noise via the green jack on playback start */ + SND_PCI_QUIRK(0x8086, 0x06c8, "Lenovo ThinkStation P340", 0), {} }; #endif /* CONFIG_PM */ -- 2.31.1

2 1

[PATCH 00/10] Documentation build warning fixes
by Mauro Carvalho Chehab 25 May '21

25 May '21

Hi Jon, This small series contain a series of fixes for the documentation. it is against your docs-next branch. Three of the patches fix duplicated symbols at the ABI documents. There are still some ABI warnings from IIO, but all but one were already fixed at linux-next. So, hopefully, after having everything merged, the ABI warnings will be solved. Mauro Carvalho Chehab (10): docs: update sysfs-platform_profile.rst reference docs: vcpu-requests.rst: fix reference for atomic ops docs: translations/zh_CN: fix a typo at 8.Conclusion.rst docs: sched-bwc.rst: fix a typo on a doc name docs: update pin-control.rst references docs: virt: api.rst: fix a pointer to SGX documentation docs: ABI: iommu: remove duplicated definition for sysfs-kernel-iommu_groups docs: ABI: sysfs-class-backlight: unify ambient light zone nodes docs: ABI: sysfs-class-led-trigger-pattern: remove repeat duplication iio: documentation: fix a typo Documentation/ABI/testing/sysfs-bus-iio | 4 +- .../ABI/testing/sysfs-class-backlight | 100 ++++++++++++++++++ .../ABI/testing/sysfs-class-backlight-adp5520 | 31 ------ .../ABI/testing/sysfs-class-backlight-adp8860 | 37 ------- .../sysfs-class-backlight-driver-adp8870 | 32 ------ .../testing/sysfs-class-led-driver-el15203000 | 9 -- .../testing/sysfs-class-led-trigger-pattern | 3 + .../ABI/testing/sysfs-kernel-iommu_groups | 12 +-- Documentation/scheduler/sched-bwc.rst | 2 +- .../zh_CN/process/8.Conclusion.rst | 2 +- Documentation/virt/kvm/api.rst | 2 +- Documentation/virt/kvm/vcpu-requests.rst | 2 +- include/linux/device.h | 2 +- include/linux/mfd/madera/pdata.h | 2 +- include/linux/pinctrl/pinconf-generic.h | 2 +- include/linux/platform_profile.h | 2 +- 16 files changed, 117 insertions(+), 127 deletions(-) delete mode 100644 Documentation/ABI/testing/sysfs-class-backlight-adp5520 delete mode 100644 Documentation/ABI/testing/sysfs-class-backlight-adp8860 delete mode 100644 Documentation/ABI/testing/sysfs-class-backlight-driver-adp8870 delete mode 100644 Documentation/ABI/testing/sysfs-class-led-driver-el15203000 -- 2.31.1

5 5

[RFC PATCH 0/6] soc-pcm: Add separate snd_pcm_runtime for BEs
by Codrin Ciubotariu 24 May '21

24 May '21

This patchset adds a different snd_pcm_runtime in the BE's substream, replacing the FE's snd_pcm_runtime. With a different structure, the BE HW capabilities and constraints will no longer merge with the FE ones. This allows for error detection if the be_hw_params_fixup() applies HW parameters not supported by the BE DAIs. Also, it calculates values needed for mem-to-dev/dev-to-mem DMA transfers, such as buffer size and period size, if needed. The first 4 patches are preparatory patches, that just group and export functions used to allocate and initialize the snd_pcm_runtime. Also, the functions that set and apply the HW constraints are exported. The 5th patch does (almost) everything need to create the new snd_pcm_runtime for BEs, which includes allocation, initializing the HW capabilities, HW constraints and HW parameters. The BE HW parameters are no longer copied from the FE. They are recalculated, based on HW capabilities, constraints and the be_hw_params_fixup() callback. The 6th and last patch basically adds support for the PCM generic dmaengine to be used as a platform driver for BE DAI links. It allocates a buffer, needed by the DMA transfers that do not support dev-to-dev transfers between FE and BE DAIs. This is a superset of https://mailman.alsa-project.org/pipermail/alsa-devel/2021-March/182630.html which only handles the BE HW constraints. This patchset aims to be more complete, defining a a snd_pcm_runtime between each FE and BE and can be used between any DAI link connection. I am sure I am not handling all the needed members of snd_pcm_runtime (such as handling struct snd_pcm_mmap_status *status), but I would like to have your feedback regarding this idea. Codrin Ciubotariu (6): ALSA: core: pcm: Create helpers to allocate/free struct snd_pcm_runtime ALSA: pcm: Export constraints initialization functions ALSA: pcm: Check for substream->ops before substream->ops->mmap ALSA: pcm: Create function for snd_pcm_runtime initialization ASoC: soc-pcm: Create new snd_pcm_runtime for BE DAIs ASoC: dmaengine: Allocate buffer if substream is unmanaged include/sound/pcm.h | 7 ++ sound/core/pcm.c | 82 ++++++++++------- sound/core/pcm_native.c | 116 +++++++++++++----------- sound/soc/soc-generic-dmaengine-pcm.c | 18 ++++ sound/soc/soc-pcm.c | 126 ++++++++++++++++++-------- 5 files changed, 229 insertions(+), 120 deletions(-) -- 2.27.0

3 12

[PATCH v7 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 24 May '21

24 May '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v6: - Removed unnecessary maxItems from qcom,tx/rx-device property as suggested by RobH. Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 146 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 298 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 705 ++++ 9 files changed, 5343 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 12

[PATCH AUTOSEL 4.4 05/16] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 0c7fe1418447..815639c04c52 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -109,7 +109,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 06/19] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index e75bfc511e3e..f7a7995588f2 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -109,7 +109,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 06/21] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 1eb8b61a185b..fb59e5179075 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -109,7 +109,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 12/25] ASoC: cs43130: handle errors in cs43130_probe() properly
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 2da441a6491d93eff8ffff523837fd621dc80389 ] cs43130_probe() does not do any valid error checking of things it initializes, OR what it does, it does not unwind properly if there are errors. Fix this up by moving the sysfs files to an attribute group so the driver core will correctly add/remove them all at once and handle errors with them, and correctly check for creating a new workqueue and unwinding if that fails. Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-58-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 80dc42197154..cf29dec28b5e 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1738,6 +1738,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static struct attribute *hpload_attrs[] = { + &dev_attr_hpload_dc_l.attr, + &dev_attr_hpload_dc_r.attr, + &dev_attr_hpload_ac_l.attr, + &dev_attr_hpload_ac_r.attr, +}; +ATTRIBUTE_GROUPS(hpload); + static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, @@ -2305,23 +2313,15 @@ static int cs43130_probe(struct snd_soc_component *component) cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = device_create_file(component->dev, &dev_attr_hpload_dc_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_dc_r); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_r); - if (ret < 0) + ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) { + sysfs_remove_groups(&component->dev->kobj, hpload_groups); + return -ENOMEM; + } INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 11/25] ASoC: rt5645: add error checking to rt5645_probe function
by Sasha Levin 24 May '21

24 May '21

From: Phillip Potter <phil(a)philpotter.co.uk> [ Upstream commit 5e70b8e22b64eed13d5bbebcb5911dae65bf8c6b ] Check for return value from various snd_soc_dapm_* calls, as many of them can return errors and this should be handled. Also, reintroduce the allocation failure check for rt5645->eq_param as well. Make all areas where return values are checked lead to the end of the function in the case of an error. Finally, introduce a comment explaining how resources here are actually eventually cleaned up by the caller. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Phillip Potter <phil(a)philpotter.co.uk> Link: https://lore.kernel.org/r/20210503115736.2104747-56-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 48 +++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9185bd7c5a6d..d8b941bb9f8a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3413,30 +3413,44 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (ret < 0) + goto exit; + if (rt5645->v_id < 3) { - snd_soc_dapm_add_routes(dapm, + ret = snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); + if (ret < 0) + goto exit; } break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); + if (ret < 0) + goto exit; break; } @@ -3444,9 +3458,17 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) + goto exit; } if (rt5645->pdata.long_name) @@ -3456,7 +3478,15 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - return 0; + if (!rt5645->eq_param) + ret = -ENOMEM; +exit: + /* + * If there was an error above, everything will be cleaned up by the + * caller if we return an error here. This will be done with a later + * call to rt5645_remove(). + */ + return ret; } static void rt5645_remove(struct snd_soc_component *component) -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 06/25] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 1eb8b61a185b..fb59e5179075 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -109,7 +109,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 29/52] ASoC: cs43130: handle errors in cs43130_probe() properly
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 2da441a6491d93eff8ffff523837fd621dc80389 ] cs43130_probe() does not do any valid error checking of things it initializes, OR what it does, it does not unwind properly if there are errors. Fix this up by moving the sysfs files to an attribute group so the driver core will correctly add/remove them all at once and handle errors with them, and correctly check for creating a new workqueue and unwinding if that fails. Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-58-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index bb46e993c353..8f70dee95878 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static struct attribute *hpload_attrs[] = { + &dev_attr_hpload_dc_l.attr, + &dev_attr_hpload_dc_r.attr, + &dev_attr_hpload_ac_l.attr, + &dev_attr_hpload_ac_r.attr, +}; +ATTRIBUTE_GROUPS(hpload); + static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, @@ -2302,23 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component) cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = device_create_file(component->dev, &dev_attr_hpload_dc_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_dc_r); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_r); - if (ret < 0) + ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) { + sysfs_remove_groups(&component->dev->kobj, hpload_groups); + return -ENOMEM; + } INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 28/52] Revert "ASoC: cs43130: fix a NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit fdda0dd2686ecd1f2e616c9e0366ea71b40c485d ] This reverts commit a2be42f18d409213bb7e7a736e3ef6ba005115bb. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch here is not correct, sysfs files that were created are not unwound. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-57-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 7fb34422a2a4..bb46e993c353 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2319,8 +2319,6 @@ static int cs43130_probe(struct snd_soc_component *component) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); - if (!cs43130->wq) - return -ENOMEM; INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 27/52] ASoC: rt5645: add error checking to rt5645_probe function
by Sasha Levin 24 May '21

24 May '21

From: Phillip Potter <phil(a)philpotter.co.uk> [ Upstream commit 5e70b8e22b64eed13d5bbebcb5911dae65bf8c6b ] Check for return value from various snd_soc_dapm_* calls, as many of them can return errors and this should be handled. Also, reintroduce the allocation failure check for rt5645->eq_param as well. Make all areas where return values are checked lead to the end of the function in the case of an error. Finally, introduce a comment explaining how resources here are actually eventually cleaned up by the caller. Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Phillip Potter <phil(a)philpotter.co.uk> Link: https://lore.kernel.org/r/20210503115736.2104747-56-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 48 +++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 44e88d0dbb7a..e9e1993e8d74 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3376,30 +3376,44 @@ static int rt5645_probe(struct snd_soc_component *component) { struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); struct rt5645_priv *rt5645 = snd_soc_component_get_drvdata(component); + int ret = 0; rt5645->component = component; switch (rt5645->codec_type) { case CODEC_TYPE_RT5645: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5645_specific_dapm_widgets, ARRAY_SIZE(rt5645_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5645_specific_dapm_routes, ARRAY_SIZE(rt5645_specific_dapm_routes)); + if (ret < 0) + goto exit; + if (rt5645->v_id < 3) { - snd_soc_dapm_add_routes(dapm, + ret = snd_soc_dapm_add_routes(dapm, rt5645_old_dapm_routes, ARRAY_SIZE(rt5645_old_dapm_routes)); + if (ret < 0) + goto exit; } break; case CODEC_TYPE_RT5650: - snd_soc_dapm_new_controls(dapm, + ret = snd_soc_dapm_new_controls(dapm, rt5650_specific_dapm_widgets, ARRAY_SIZE(rt5650_specific_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_add_routes(dapm, rt5650_specific_dapm_routes, ARRAY_SIZE(rt5650_specific_dapm_routes)); + if (ret < 0) + goto exit; break; } @@ -3407,9 +3421,17 @@ static int rt5645_probe(struct snd_soc_component *component) /* for JD function */ if (rt5645->pdata.jd_mode) { - snd_soc_dapm_force_enable_pin(dapm, "JD Power"); - snd_soc_dapm_force_enable_pin(dapm, "LDO2"); - snd_soc_dapm_sync(dapm); + ret = snd_soc_dapm_force_enable_pin(dapm, "JD Power"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_force_enable_pin(dapm, "LDO2"); + if (ret < 0) + goto exit; + + ret = snd_soc_dapm_sync(dapm); + if (ret < 0) + goto exit; } if (rt5645->pdata.long_name) @@ -3419,7 +3441,15 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - return 0; + if (!rt5645->eq_param) + ret = -ENOMEM; +exit: + /* + * If there was an error above, everything will be cleaned up by the + * caller if we return an error here. This will be done with a later + * call to rt5645_remove(). + */ + return ret; } static void rt5645_remove(struct snd_soc_component *component) -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 26/52] Revert "ASoC: rt5645: fix a NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1e0ce84215dbfd6065872e5d3755352da34f198b ] This reverts commit 51dd97d1df5fb9ac58b9b358e63e67b530f6ae21. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. Lots of things seem to be still allocated here and must be properly cleaned up if an error happens here. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-55-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index c83f7f5da96b..44e88d0dbb7a 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3419,9 +3419,6 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - if (!rt5645->eq_param) - return -ENOMEM; - return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 15/52] Revert "ALSA: usx2y: Fix potential NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4667a6fc1777ce071504bab570d3599107f4790f ] This reverts commit a2c6433ee5a35a8de6d563f6512a26f87835ea0f. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch was incorrect, and would leak memory if the error path the patch added was hit. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-37-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usb_stream.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 091c071b270a..6bba17bf689a 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -91,12 +91,7 @@ static int init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, for (u = 0; u < USB_STREAM_NURBS; ++u) { sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->inurb[u]) - return -ENOMEM; - sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->outurb[u]) - return -ENOMEM; } if (init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe) || -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 14/52] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index d67eae3988bd..77bedfe33eea 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -94,7 +94,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 13/52] Revert "ALSA: gus: add a check of the status of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1dacca7fa1ebea47d38d20cd2df37094805d2649 ] This reverts commit 0f25e000cb4398081748e54f62a902098aa79ec1. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original commit did nothing if there was an error, except to print out a message, which is pointless. So remove the commit as it gives a "false sense of doing something". Cc: Kangjie Lu <kjlu(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-33-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_main.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index af6b4d89d695..39911a637e80 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -77,17 +77,8 @@ static const struct snd_kcontrol_new snd_gus_joystick_control = { static void snd_gus_init_control(struct snd_gus_card *gus) { - int ret; - - if (!gus->ace_flag) { - ret = - snd_ctl_add(gus->card, - snd_ctl_new1(&snd_gus_joystick_control, - gus)); - if (ret) - snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n", - ret); - } + if (!gus->ace_flag) + snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus)); } /* -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 04/52] Revert "ALSA: sb: fix a missing check of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4b059ce1f4b368208c2310925f49be77f15e527b ] This reverts commit beae77170c60aa786f3e4599c18ead2854d8694d. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It is safe to ignore this error as the mixer element is optional, and the driver is very legacy. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-8-gregkh@linuxfoundation.o… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb16_main.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 0768bbf8fd71..679f9f48370f 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -864,14 +864,10 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { - err = snd_ctl_add(card, snd_ctl_new1( - &snd_sb16_dma_control, chip)); - if (err) - return err; - } else { + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) + snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); + else pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - } snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 32/62] ASoC: cs43130: handle errors in cs43130_probe() properly
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 2da441a6491d93eff8ffff523837fd621dc80389 ] cs43130_probe() does not do any valid error checking of things it initializes, OR what it does, it does not unwind properly if there are errors. Fix this up by moving the sysfs files to an attribute group so the driver core will correctly add/remove them all at once and handle errors with them, and correctly check for creating a new workqueue and unwinding if that fails. Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-58-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index bb46e993c353..8f70dee95878 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1735,6 +1735,14 @@ static DEVICE_ATTR(hpload_dc_r, 0444, cs43130_show_dc_r, NULL); static DEVICE_ATTR(hpload_ac_l, 0444, cs43130_show_ac_l, NULL); static DEVICE_ATTR(hpload_ac_r, 0444, cs43130_show_ac_r, NULL); +static struct attribute *hpload_attrs[] = { + &dev_attr_hpload_dc_l.attr, + &dev_attr_hpload_dc_r.attr, + &dev_attr_hpload_ac_l.attr, + &dev_attr_hpload_ac_r.attr, +}; +ATTRIBUTE_GROUPS(hpload); + static struct reg_sequence hp_en_cal_seq[] = { {CS43130_INT_MASK_4, CS43130_INT_MASK_ALL}, {CS43130_HP_MEAS_LOAD_1, 0}, @@ -2302,23 +2310,15 @@ static int cs43130_probe(struct snd_soc_component *component) cs43130->hpload_done = false; if (cs43130->dc_meas) { - ret = device_create_file(component->dev, &dev_attr_hpload_dc_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_dc_r); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_l); - if (ret < 0) - return ret; - - ret = device_create_file(component->dev, &dev_attr_hpload_ac_r); - if (ret < 0) + ret = sysfs_create_groups(&component->dev->kobj, hpload_groups); + if (ret) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); + if (!cs43130->wq) { + sysfs_remove_groups(&component->dev->kobj, hpload_groups); + return -ENOMEM; + } INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 31/62] Revert "ASoC: cs43130: fix a NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit fdda0dd2686ecd1f2e616c9e0366ea71b40c485d ] This reverts commit a2be42f18d409213bb7e7a736e3ef6ba005115bb. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch here is not correct, sysfs files that were created are not unwound. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-57-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 7fb34422a2a4..bb46e993c353 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2319,8 +2319,6 @@ static int cs43130_probe(struct snd_soc_component *component) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); - if (!cs43130->wq) - return -ENOMEM; INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 18/62] Revert "ALSA: usx2y: Fix potential NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4667a6fc1777ce071504bab570d3599107f4790f ] This reverts commit a2c6433ee5a35a8de6d563f6512a26f87835ea0f. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch was incorrect, and would leak memory if the error path the patch added was hit. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-37-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usb_stream.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 091c071b270a..6bba17bf689a 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -91,12 +91,7 @@ static int init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, for (u = 0; u < USB_STREAM_NURBS; ++u) { sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->inurb[u]) - return -ENOMEM; - sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->outurb[u]) - return -ENOMEM; } if (init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe) || -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 17/62] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 438109f167d6..fc1ba3d43265 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -94,7 +94,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 16/62] Revert "ALSA: gus: add a check of the status of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1dacca7fa1ebea47d38d20cd2df37094805d2649 ] This reverts commit 0f25e000cb4398081748e54f62a902098aa79ec1. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original commit did nothing if there was an error, except to print out a message, which is pointless. So remove the commit as it gives a "false sense of doing something". Cc: Kangjie Lu <kjlu(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-33-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_main.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index afc088f0377c..b7518122a10d 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -77,17 +77,8 @@ static const struct snd_kcontrol_new snd_gus_joystick_control = { static void snd_gus_init_control(struct snd_gus_card *gus) { - int ret; - - if (!gus->ace_flag) { - ret = - snd_ctl_add(gus->card, - snd_ctl_new1(&snd_gus_joystick_control, - gus)); - if (ret) - snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n", - ret); - } + if (!gus->ace_flag) + snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus)); } /* -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 05/62] Revert "ALSA: sb: fix a missing check of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4b059ce1f4b368208c2310925f49be77f15e527b ] This reverts commit beae77170c60aa786f3e4599c18ead2854d8694d. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It is safe to ignore this error as the mixer element is optional, and the driver is very legacy. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-8-gregkh@linuxfoundation.o… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb16_main.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 38dc1fde25f3..aa4870531023 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -846,14 +846,10 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { - err = snd_ctl_add(card, snd_ctl_new1( - &snd_sb16_dma_control, chip)); - if (err) - return err; - } else { + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) + snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); + else pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - } snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, 64*1024, 128*1024); -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 51/63] ALSA: dice: disable double_pcm_frames mode for M-Audio Profire 610, 2626 and Avid M-Box 3 Pro
by Sasha Levin 24 May '21

24 May '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 9f079c1bdc9087842dc5ac9d81b1d7f2578e81ce ] ALSA dice driver detects jumbo payload at high sampling transfer frequency for below models: * Avid M-Box 3 Pro * M-Audio Profire 610 * M-Audio Profire 2626 Although many DICE-based devices have a quirk at high sampling transfer frequency to multiplex double number of PCM frames into data block than the number in IEC 61883-1/6, the above devices are just compliant to IEC 61883-1/6. This commit disables the mode of double_pcm_frames for the models. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210518012510.37126-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/dice/dice-pcm.c | 4 ++-- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/dice/dice.c | 24 ++++++++++++++++++++++++ sound/firewire/dice/dice.h | 3 ++- 4 files changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index af8a90ee40f3..a69ca1111b03 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -218,7 +218,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (frames_per_period > 0) { // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { frames_per_period *= 2; frames_per_buffer *= 2; } @@ -273,7 +273,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, mutex_lock(&dice->mutex); // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { events_per_period /= 2; events_per_buffer /= 2; } diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 1a14c083e8ce..c4dfe76500c2 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -181,7 +181,7 @@ static int keep_resources(struct snd_dice *dice, struct amdtp_stream *stream, // as 'Dual Wire'. // For this quirk, blocking mode is required and PCM buffer size should // be aligned to SYT_INTERVAL. - double_pcm_frames = rate > 96000; + double_pcm_frames = (rate > 96000 && !dice->disable_double_pcm_frames); if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 107a81691f0e..239d164b0eea 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -21,6 +21,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. #define OUI_PRESONUS 0x000a92 #define OUI_HARMAN 0x000fd7 +#define OUI_AVID 0x00a07e #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 @@ -222,6 +223,14 @@ static int dice_probe(struct fw_unit *unit, (snd_dice_detect_formats_t)entry->driver_data; } + // Below models are compliant to IEC 61883-1/6 and have no quirk at high sampling transfer + // frequency. + // * Avid M-Box 3 Pro + // * M-Audio Profire 610 + // * M-Audio Profire 2626 + if (entry->vendor_id == OUI_MAUDIO || entry->vendor_id == OUI_AVID) + dice->disable_double_pcm_frames = true; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); init_completion(&dice->clock_accepted); @@ -278,7 +287,22 @@ static void dice_bus_reset(struct fw_unit *unit) #define DICE_INTERFACE 0x000001 +#define DICE_DEV_ENTRY_TYPICAL(vendor, model, data) \ + { \ + .match_flags = IEEE1394_MATCH_VENDOR_ID | \ + IEEE1394_MATCH_MODEL_ID | \ + IEEE1394_MATCH_SPECIFIER_ID | \ + IEEE1394_MATCH_VERSION, \ + .vendor_id = (vendor), \ + .model_id = (model), \ + .specifier_id = (vendor), \ + .version = DICE_INTERFACE, \ + .driver_data = (kernel_ulong_t)(data), \ + } + static const struct ieee1394_device_id dice_id_table[] = { + // Avid M-Box 3 Pro. To match in probe function. + DICE_DEV_ENTRY_TYPICAL(OUI_AVID, 0x000004, snd_dice_detect_extension_formats), /* M-Audio Profire 2626 has a different value in version field. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index adc6f7c84460..3c967d1b3605 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -109,7 +109,8 @@ struct snd_dice { struct fw_iso_resources rx_resources[MAX_STREAMS]; struct amdtp_stream tx_stream[MAX_STREAMS]; struct amdtp_stream rx_stream[MAX_STREAMS]; - bool global_enabled; + bool global_enabled:1; + bool disable_double_pcm_frames:1; struct completion clock_accepted; unsigned int substreams_counter; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 31/63] Revert "ASoC: cs43130: fix a NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit fdda0dd2686ecd1f2e616c9e0366ea71b40c485d ] This reverts commit a2be42f18d409213bb7e7a736e3ef6ba005115bb. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch here is not correct, sysfs files that were created are not unwound. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-57-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs43130.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index 80bc7c10ed75..c2b6f0ae6d57 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -2319,8 +2319,6 @@ static int cs43130_probe(struct snd_soc_component *component) return ret; cs43130->wq = create_singlethread_workqueue("cs43130_hp"); - if (!cs43130->wq) - return -ENOMEM; INIT_WORK(&cs43130->work, cs43130_imp_meas); } -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 29/63] Revert "ASoC: rt5645: fix a NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1e0ce84215dbfd6065872e5d3755352da34f198b ] This reverts commit 51dd97d1df5fb9ac58b9b358e63e67b530f6ae21. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. Lots of things seem to be still allocated here and must be properly cleaned up if an error happens here. Cc: Kangjie Lu <kjlu(a)umn.edu> Cc: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20210503115736.2104747-55-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5645.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 63a7e052eaa0..ab06133a85da 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3407,9 +3407,6 @@ static int rt5645_probe(struct snd_soc_component *component) RT5645_HWEQ_NUM, sizeof(struct rt5645_eq_param_s), GFP_KERNEL); - if (!rt5645->eq_param) - return -ENOMEM; - return 0; } -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 18/63] Revert "ALSA: usx2y: Fix potential NULL pointer dereference"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4667a6fc1777ce071504bab570d3599107f4790f ] This reverts commit a2c6433ee5a35a8de6d563f6512a26f87835ea0f. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original patch was incorrect, and would leak memory if the error path the patch added was hit. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-37-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/usx2y/usb_stream.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 091c071b270a..6bba17bf689a 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -91,12 +91,7 @@ static int init_urbs(struct usb_stream_kernel *sk, unsigned use_packsize, for (u = 0; u < USB_STREAM_NURBS; ++u) { sk->inurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->inurb[u]) - return -ENOMEM; - sk->outurb[u] = usb_alloc_urb(sk->n_o_ps, GFP_KERNEL); - if (!sk->outurb[u]) - return -ENOMEM; } if (init_pipe_urbs(sk, use_packsize, sk->inurb, indata, dev, in_pipe) || -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 17/63] ALSA: sb8: Add a comment note regarding an unused pointer
by Sasha Levin 24 May '21

24 May '21

From: Atul Gopinathan <atulgopinathan(a)gmail.com> [ Upstream commit a28591f61b60fac820c6de59826ffa710e5e314e ] The field "fm_res" of "struct snd_sb8" is never used/dereferenced throughout the sb8.c code. Therefore there is no need for any null value check after the "request_region()". Add a comment note to make developers know about this and prevent any "NULL check" patches on this part of code. Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Atul Gopinathan <atulgopinathan(a)gmail.com> Link: https://lore.kernel.org/r/20210503115736.2104747-36-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb8.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 6c9d534ce8b6..e17b58437e66 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -93,7 +93,11 @@ static int snd_sb8_probe(struct device *pdev, unsigned int dev) acard = card->private_data; card->private_free = snd_sb8_free; - /* block the 0x388 port to avoid PnP conflicts */ + /* + * Block the 0x388 port to avoid PnP conflicts. + * No need to check this value after request_region, + * as we never do anything with it. + */ acard->fm_res = request_region(0x388, 4, "SoundBlaster FM"); if (!acard->fm_res) { err = -EBUSY; -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 16/63] Revert "ALSA: gus: add a check of the status of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 1dacca7fa1ebea47d38d20cd2df37094805d2649 ] This reverts commit 0f25e000cb4398081748e54f62a902098aa79ec1. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It will be fixed up "correctly" in a later kernel change. The original commit did nothing if there was an error, except to print out a message, which is pointless. So remove the commit as it gives a "false sense of doing something". Cc: Kangjie Lu <kjlu(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-33-gregkh@linuxfoundation.… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/gus/gus_main.c | 13 ++----------- 1 file changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index afc088f0377c..b7518122a10d 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -77,17 +77,8 @@ static const struct snd_kcontrol_new snd_gus_joystick_control = { static void snd_gus_init_control(struct snd_gus_card *gus) { - int ret; - - if (!gus->ace_flag) { - ret = - snd_ctl_add(gus->card, - snd_ctl_new1(&snd_gus_joystick_control, - gus)); - if (ret) - snd_printk(KERN_ERR "gus: snd_ctl_add failed: %d\n", - ret); - } + if (!gus->ace_flag) + snd_ctl_add(gus->card, snd_ctl_new1(&snd_gus_joystick_control, gus)); } /* -- 2.30.2

1 0

[PATCH AUTOSEL 5.12 05/63] Revert "ALSA: sb: fix a missing check of snd_ctl_add"
by Sasha Levin 24 May '21

24 May '21

From: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> [ Upstream commit 4b059ce1f4b368208c2310925f49be77f15e527b ] This reverts commit beae77170c60aa786f3e4599c18ead2854d8694d. Because of recent interactions with developers from @umn.edu, all commits from them have been recently re-reviewed to ensure if they were correct or not. Upon review, this commit was found to be incorrect for the reasons below, so it must be reverted. It is safe to ignore this error as the mixer element is optional, and the driver is very legacy. Cc: Aditya Pakki <pakki001(a)umn.edu> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20210503115736.2104747-8-gregkh@linuxfoundation.o… Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb16_main.c | 10 +++------- 1 file changed, 3 insertions(+), 7 deletions(-) diff --git a/sound/isa/sb/sb16_main.c b/sound/isa/sb/sb16_main.c index 38dc1fde25f3..aa4870531023 100644 --- a/sound/isa/sb/sb16_main.c +++ b/sound/isa/sb/sb16_main.c @@ -846,14 +846,10 @@ int snd_sb16dsp_pcm(struct snd_sb *chip, int device) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb16_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb16_capture_ops); - if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) { - err = snd_ctl_add(card, snd_ctl_new1( - &snd_sb16_dma_control, chip)); - if (err) - return err; - } else { + if (chip->dma16 >= 0 && chip->dma8 != chip->dma16) + snd_ctl_add(card, snd_ctl_new1(&snd_sb16_dma_control, chip)); + else pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX; - } snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, 64*1024, 128*1024); -- 2.30.2

1 0

[PATCH -next] ASoC: imx-card: Make some symbols static
by Wei Yongjun 24 May '21

24 May '21

The sparse tool complains as follows: sound/soc/fsl/imx-card.c:121:27: warning: symbol 'ak4458_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:138:31: warning: symbol 'ak4458_tdm_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:149:27: warning: symbol 'ak4497_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:166:27: warning: symbol 'ak5558_fs_mul' was not declared. Should it be static? sound/soc/fsl/imx-card.c:180:31: warning: symbol 'ak5558_tdm_fs_mul' was not declared. Should it be static? Those symbols are not used outside of imx-card.c, so marks them static. Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/fsl/imx-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index ab424735bbfe..58fd0639a069 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -118,7 +118,7 @@ struct imx_card_data { u32 asrc_format; }; -struct imx_akcodec_fs_mul ak4458_fs_mul[] = { +static struct imx_akcodec_fs_mul ak4458_fs_mul[] = { /* Normal, < 32kHz */ { .rmin = 8000, .rmax = 24000, .wmin = 1024, .wmax = 1024, }, /* Normal, 32kHz */ @@ -135,7 +135,7 @@ struct imx_akcodec_fs_mul ak4458_fs_mul[] = { { .rmin = 705600, .rmax = 768000, .wmin = 16, .wmax = 64, }, }; -struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { +static struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { /* * Table 13 - Audio Interface Format * For TDM mode, MCLK should is set to @@ -146,7 +146,7 @@ struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { { .min = 512, .max = 512, .mul = 1024 }, /* TDM512 */ }; -struct imx_akcodec_fs_mul ak4497_fs_mul[] = { +static struct imx_akcodec_fs_mul ak4497_fs_mul[] = { /** * Table 7 - mapping multiplier and speed mode * Tables 8 & 9 - mapping speed mode and LRCK fs @@ -163,7 +163,7 @@ struct imx_akcodec_fs_mul ak4497_fs_mul[] = { * Auto MCLK selection based on LRCK for Normal Mode * (Table 4 from datasheet) */ -struct imx_akcodec_fs_mul ak5558_fs_mul[] = { +static struct imx_akcodec_fs_mul ak5558_fs_mul[] = { { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, @@ -177,7 +177,7 @@ struct imx_akcodec_fs_mul ak5558_fs_mul[] = { * because of SAI we also add the restriction: MCLK >= 2 * BCLK * (Table 9 from datasheet) */ -struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { +static struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { { .min = 128, .max = 128, .mul = 256 }, { .min = 256, .max = 256, .mul = 512 }, { .min = 512, .max = 512, .mul = 1024 },

1 0

[PATCH v3 0/7] PinePhone BT audio bringup
by Samuel Holland 24 May '21

24 May '21

This series uses the additional DAIs added to the sun8i-codec driver to add hardware routing for BT SCO (headset) audio on the PinePhone. The BT audio connection is represented by the "dummy" bt-sco codec. The connection to the Quectel EG-25G modem via AIF2 works as well, but I do not include it here because there is no appropriate codec driver in tree. We have been using an out-of-tree "dummy" codec driver for the modem similar to bt-sco, and I'm not sure if such a driver would be desired upstream. Changes from v2: - Also accept #sound-dai-cells in the binding. Since dt-core.yaml already sets the type of this property, it is not possible to use oneOf, nor make a specific value deprecated. Changes from v1: - Fixed DT binding example to follow new binding Arnaud Ferraris (1): arm64: dts: allwinner: pinephone: Set audio card name Samuel Holland (6): ASoC: dt-bindings: sun8i-codec: Increase #sound-dai-cells ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Add pinmux nodes for AIF2/AIF3 arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: pinephone: Add support for Bluetooth audio .../sound/allwinner,sun8i-a33-codec.yaml | 8 +++- arch/arm/boot/dts/sun8i-a33.dtsi | 4 +- .../dts/allwinner/sun50i-a64-pinephone.dtsi | 25 +++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 37 ++++++++++++++----- 4 files changed, 60 insertions(+), 14 deletions(-) -- 2.26.3

4 13

[PATCH] ASoC: core: Fix Null-point-dereference in fmt_single_name()
by Kefeng Wang 24 May '21

24 May '21

Check the return value of devm_kstrdup() in case of Null-point-dereference. Fixes: 45dd9943fce0 ("ASoC: core: remove artificial component and DAI name constraint") Cc: Dmitry Baryshkov <dmitry.baryshkov(a)linaro.org> Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Kefeng Wang <wangkefeng.wang(a)huawei.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c0904acb935..a76974ccfce1 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2225,6 +2225,8 @@ static char *fmt_single_name(struct device *dev, int *id) return NULL; name = devm_kstrdup(dev, devname, GFP_KERNEL); + if (!name) + return NULL; /* are we a "%s.%d" name (platform and SPI components) */ found = strstr(name, dev->driver->name); -- 2.26.2

2 1

[RFC PATCH 1/2] ASoC: dt-bindings: codecs: Add bindings for nxp, tfa989x
by Stephan Gerhold 24 May '21

24 May '21

NXP/Goodix TFA989X (TFA1) amplifiers are controlled via an I2C bus. Add simple device tree bindings that describe how to set them up in the device tree. Right now only nxp,tfa9895 is supported but this will be extended to at least nxp,tfa9897 in the near future. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- .../bindings/sound/nxp,tfa989x.yaml | 54 +++++++++++++++++++ 1 file changed, 54 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml new file mode 100644 index 000000000000..45db5776550c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -0,0 +1,54 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,tfa989x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP/Goodix TFA989X (TFA1) Audio Amplifiers + +maintainers: + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - nxp,tfa9895 + + reg: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: + Used as prefix for sink/source names of the component. Must be a + unique string among multiple instances of the same component. + +required: + - compatible + - reg + - '#sound-dai-cells' + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@34 { + compatible = "nxp,tfa9895"; + reg = <0x34>; + sound-name-prefix = "Speaker Left"; + #sound-dai-cells = <0>; + }; + audio-codec@36 { + compatible = "nxp,tfa9895"; + reg = <0x36>; + sound-name-prefix = "Speaker Right"; + #sound-dai-cells = <0>; + }; + }; -- 2.31.1

3 3

[PATCH -next] ASoC: tlv320aic26: use DEVICE_ATTR_RW macro
by YueHaibing 24 May '21

24 May '21

Use DEVICE_ATTR_RW() helper instead of plain DEVICE_ATTR(), which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tlv320aic26.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index c7baef8948d4..077415a57225 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -261,8 +261,8 @@ static const struct snd_kcontrol_new aic26_snd_controls[] = { * SPI device portion of driver: sysfs files for debugging */ -static ssize_t aic26_keyclick_show(struct device *dev, - struct device_attribute *attr, char *buf) +static ssize_t keyclick_show(struct device *dev, + struct device_attribute *attr, char *buf) { struct aic26 *aic26 = dev_get_drvdata(dev); int val, amp, freq, len; @@ -276,9 +276,9 @@ static ssize_t aic26_keyclick_show(struct device *dev, } /* Any write to the keyclick attribute will trigger the keyclick event */ -static ssize_t aic26_keyclick_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t keyclick_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct aic26 *aic26 = dev_get_drvdata(dev); @@ -288,7 +288,7 @@ static ssize_t aic26_keyclick_set(struct device *dev, return count; } -static DEVICE_ATTR(keyclick, 0644, aic26_keyclick_show, aic26_keyclick_set); +static DEVICE_ATTR_RW(keyclick); /* --------------------------------------------------------------------- * SoC CODEC portion of driver: probe and release routines -- 2.17.1

1 0

usb: u_audio: Notifying gadget audio device about starting/stopping capture/playback on the host
by Pavel Hofman 24 May '21

24 May '21

Hi, I am trying to notify the user-space code running on USB audio gadget that the USB host has started/stopped capture/playback. For stopping action, I am enclosing a patch snippet. I used snd_pcm_stop code when alsa SPDIF receivers detect a change in the stream https://github.com/torvalds/linux/blob/master/sound/i2c/other/ak4117.c#L505 When testing the code, aplay/arecord ends, sox just reports alsa device warning and does not stop. IMO that is the desired outcome - the snd_pcm_read/write methods returned errors when capture/playback on the host side stopped. For starting the capture/playback on the host, perhaps some snd_ctl_notify on some dedicated alsa control could be used, notifying the user space application that it can open the gadget audio device? The patch below does not compile on upstream kernel as it uses changes not included in upstream yet (explicit feedback EP, samplerate switching), but I am just discussing principles and options now. Thank you very much for your opinion and recommendations. With regards, Pavel. diff --git a/drivers/usb/gadget/function/u_audio.c b/drivers/usb/gadget/function/u_audio.c index 45367d650c5a..c6cdb844fec1 100644 --- a/drivers/usb/gadget/function/u_audio.c +++ b/drivers/usb/gadget/function/u_audio.c @@ -565,6 +565,7 @@ int u_audio_start_capture(struct g_audio *audio_dev) struct uac_params *params = &audio_dev->params; int req_len, i; + dev_dbg(dev, "starting capture with rate %d\n", params->c_srate_active); ep = audio_dev->out_ep; prm = &uac->c_prm; config_ep_by_speed(gadget, &audio_dev->func, ep); @@ -635,6 +636,23 @@ EXPORT_SYMBOL_GPL(u_audio_start_capture); void u_audio_stop_capture(struct g_audio *audio_dev) { struct snd_uac_chip *uac = audio_dev->uac; + unsigned long _flags; + struct snd_pcm_substream *substream; + struct uac_rtd_params *prm; + + dev_dbg(uac->card->dev, "stopping capture\n"); + prm = &uac->c_prm; + if (prm) { + substream = prm->ss; + if (substream) { + dev_dbg(uac->card->dev, "stopping capture substream\n"); + snd_pcm_stream_lock_irqsave(substream, _flags); + if (snd_pcm_running(substream)) { + snd_pcm_stop(substream, SNDRV_PCM_STATE_DRAINING); + } + snd_pcm_stream_unlock_irqrestore(substream, _flags); + } + } if (audio_dev->in_ep_fback) free_ep_fback(&uac->c_prm, audio_dev->in_ep_fback); @@ -655,7 +673,7 @@ int u_audio_start_playback(struct g_audio *audio_dev) const struct usb_endpoint_descriptor *ep_desc; int req_len, i; - dev_dbg(dev, "start playback with rate %d\n", params->p_srate_active); + dev_dbg(dev, "starting playback with rate %d\n", params->p_srate_active); ep = audio_dev->in_ep; prm = &uac->p_prm; config_ep_by_speed(gadget, &audio_dev->func, ep); @@ -715,6 +733,23 @@ EXPORT_SYMBOL_GPL(u_audio_start_playback); void u_audio_stop_playback(struct g_audio *audio_dev) { struct snd_uac_chip *uac = audio_dev->uac; + unsigned long _flags; + struct snd_pcm_substream *substream; + struct uac_rtd_params *prm; + + dev_dbg(uac->card->dev, "stopping playback\n"); + prm = &uac->p_prm; + if (prm) { + substream = prm->ss; + if (substream) { + dev_dbg(uac->card->dev, "stopping playback substream\n"); + snd_pcm_stream_lock_irqsave(substream, _flags); + if (snd_pcm_running(substream)) { + snd_pcm_stop(substream, SNDRV_PCM_STATE_DRAINING); + } + snd_pcm_stream_unlock_irqrestore(substream, _flags); + } + } free_ep(&uac->p_prm, audio_dev->in_ep); }

1 1

Add UCMs for Google Asus Nexus 7 and Acer Iconia A500
by GitHub pull_request - reopened 24 May '21

24 May '21

alsa-project/alsa-ucm-conf pull request #92 was reopened from digetx: @perexg As was discussed previouly at https://www.spinics.net/lists/alsa-devel/msg126382.html I'm opening PR with the UCMs for Nexus 7 and A500. Both these devices are supported by the mainline Linux kernel. You're right that Tegra ASoC driver doesn't set card's name properly. In particular I found that we need to set the card's `driver_name` to `tegra`, then UCM get a proper lookup path. I rebased UCMs on top of the recent alsa-ucm-conf master and switched Nexus 7 UCM to use generic RT5640 UCM. Everything works great once the card name of the kernel driver is corrected. I'll send the kernel patches ASAP for review. Nexus 7 alsa-info: https://alsa-project.org/db/?f=0a29f1dab39ec5b33e01215dee42dbfd1dbbf08e A500 alsa-info: https://alsa-project.org/db/?f=bde4f7938a39b1d26904b414969fc811e726242f The alsa-info is taken using the updated kernel ASoC driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Srinivasa Rao Mandadapu 24 May '21

24 May '21

This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops prepare call for introducing some delay before capture start and clock enable. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since V1: -- Enableed BCLK and LRCLK in dai ops prepare API instead of startup API -- Added comments sound/soc/qcom/lpass-cpu.c | 48 ++++++++++++++++++++++++++++++++++++-- 1 file changed, 46 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..c5bb3f16d25f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -93,9 +93,18 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); - clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + /* To ensure BCLK/LRCLK disabled even in device node validation. + Will not impact if disabled in trigger suspend */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); + else + regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_DISABLE); + + clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -275,6 +284,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* To ensure lpass BCLK/LRCLK is enabled during + device resume. Will not impact if enabled in prepare */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); @@ -296,6 +307,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* To ensure lpass BCLK/LRCLK is disabled during + device suspend */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_DISABLE); @@ -308,19 +321,50 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret); clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); - + break; + default: break; } return ret; } +static int lpass_cpu_daiops_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret = -EINVAL; + /* To ensure lpass BCLK/LRCLK is enabled bit before + playback/capture data flow starts */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ret = regmap_fields_write(i2sctl->spken, id, LPAIF_I2SCTL_SPKEN_ENABLE); + else + ret = regmap_fields_write(i2sctl->micen, id, LPAIF_I2SCTL_MICEN_ENABLE); + + if (ret) + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", + ret); + + ret = clk_enable(drvdata->mi2s_bit_clk[id]); + + if (ret) { + dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); + clk_disable(drvdata->mi2s_osr_clk[id]); + return ret; + } + return 0; +} + + const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, .hw_params = lpass_cpu_daiops_hw_params, .trigger = lpass_cpu_daiops_trigger, + .prepare = lpass_cpu_daiops_prepare, }; EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_dai_ops); -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 1

Dmix plugin does not update state when sound card is removed.
by GitHub issues - opened 23 May '21

23 May '21

alsa-project/alsa-lib issue #137 was opened from MHoyer12345678: Playing sound using snd_pcm_writei in a dmix setup might result in an infinite loop when sound card is removed. **Setup:** - dmix setup - pcm dmix device opened non blocking - snd_pcm_writei used with poll (POLLERR is not evaluated) **Trigger:** - usb sound card is removed (e.g. by "echo 0 > /sys/bus/usb/x-x/authorized") **Observation:** - snd_pcm_writei returns -EAGAIN on each call - snd_pcm_state report state RUNNING **Expected:** - snd_pcm_writei returns -ENODEV (as happening in case a pure hw configuration is used without dmix plugin) - snd_pcm_state report state DISCONNECTED (as happening in case a pure hw configuration is used without dmix plugin) **Root Cause:** - snd_pcm_writei calls snd_pcm_write_areas - snd_pcm_write_areas requests state of dmix pcm device (state = __snd_pcm_state(pcm);) - dmix implementation of snd_pcm_state requests state from slave hw device (state = snd_pcm_state(dmix->spcm);) - hw implementation of snd_pcm_state executes an ioctl (query_status -> sync_ptr1) - the ioctl call fails with -ENODEV (after the device has been removed) - pcm_hw returns -ENODEV as state - dmix implementation of snd_pcm_state ends in default section of the switch block - this leads to the unchanged dmix->state returned - this way, the -ENODEV from the slave device is finally ignored in dmix plugin **Potential Fix:** - Add the following three lines to snd_pcm_hw_state in pcm_hw.c after query_status_data ``` //disconnected HW is reported via a failed IOCTL return -ENODEV if (err == -ENODEV) return SND_PCM_STATE_DISCONNECTED; ``` - The fix takes -ENODEV as trigger to return state DISCONNECTED Hope this helps ... Issue URL : https://github.com/alsa-project/alsa-lib/issues/137 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/3] ALSA: firewire-lib: drop initial NODATA packets or empty packets
by Takashi Sakamoto 23 May '21

23 May '21

Hi, This patchset is for convenience of ALSA bebob and firewire-tascam drivers. The devices based on BeBoB ASICs or the devices in Tascam FireWire series transfer a batch of NODATA packet or empty packet in initial step of streaming. To avoid processing them, current implementation uses an option to skip processing content of tx packet during some initial cycles. However, the hard-coded number is not enough useful. In 1st patch, ALSA IEC 61883-1/6 packet streaming engine becomes to drop the initial packets. As a result, The tx_init_skip_cycles argument of amdtp_domain_start() function changes its meaning. In the following patches, ALSA bebob driver is refactored. Takashi Sakamoto (3): ALSA: firewire-lib: drop initial tx packets with NODATA or empty ALSA: bebob: cancel switching connection order ALSA: bebob: distinguish M-Audio ProFire Lightbridge quirk sound/firewire/amdtp-stream.c | 130 +++++++++++++++++++++------- sound/firewire/amdtp-stream.h | 3 + sound/firewire/bebob/bebob.c | 10 ++- sound/firewire/bebob/bebob.h | 1 + sound/firewire/bebob/bebob_stream.c | 25 ++---- 5 files changed, 119 insertions(+), 50 deletions(-) -- 2.27.0

2 4

[PATCH 0/8] iio: accel: bmc150: Add support for yoga's with dual accelerometers with an ACPI HID of DUAL250E
by Hans de Goede 23 May '21

23 May '21

Hi All, Some 360 degree hinges (yoga) style 2-in-1 devices use 2 bmc150 accels to allow the OS to determine the angle between the display and the base of the device, so that the OS can determine if the 2-in-1 is in laptop or in tablet-mode. We already support this setup on devices using a single ACPI node with a HID of "BOSC0200" to describe both accelerometers. This patch set extends this support to also support the same setup but then using a HID of "DUAL250E". While testing this I found some crashes on rmmod, patches 1-2 fix those patches, patch 3 does some refactoring and patch 4 adds support for the "DUAL250E" HID. Unfortunately we need some more special handling though, which the rest of the patches are for. On Windows both accelerometers are read (polled) by a special service and this service calls a DSM (Device Specific Method), which in turn translates the angles to one of laptop/tablet/tent/stand mode and then notifies the EC about the new mode and the EC then enables or disables the builtin keyboard and touchpad based in the mode. When the 2-in-1 is powered-on or resumed folded in tablet mode the EC senses this independent of the DSM by using a HALL effect sensor which senses that the keyboard has been folded away behind the display. At power-on or resume the EC disables the keyboard based on this and the only way to get the keyboard to work after this is to call the DSM to re-enable it (similar to how we also need to call a special DSM in the kxcjk-1013.c accel driver to re-enable the keyboard). Patches 5-7 deal with the DSM mess and patch 8 adds labels to the 2 accelerometers specifying which one is which. Regards, Hans Hans de Goede (8): iio: accel: bmc150: Fix dereferencing the wrong pointer in bmc150_get/set_second_device iio: accel: bmc150: Don't make the remove function of the second accelerometer unregister itself iio: accel: bmc150: Move check for second ACPI device into a separate function iio: accel: bmc150: Add support for dual-accelerometers with a DUAL250E HID iio: accel: bmc150: Move struct bmc150_accel_data definition to bmc150-accel.h iio: accel: bmc150: Remove bmc150_set/get_second_device() accessor functions iio: accel: bmc150: Add support for DUAL250E ACPI DSM for setting the hinge angle iio: accel: bmc150: Set label based on accel-location for ACPI DUAL250E fwnodes drivers/iio/accel/bmc150-accel-core.c | 87 ++---------- drivers/iio/accel/bmc150-accel-i2c.c | 192 +++++++++++++++++++++----- drivers/iio/accel/bmc150-accel.h | 66 ++++++++- 3 files changed, 239 insertions(+), 106 deletions(-) -- 2.31.1

3 28

[sound:for-linus 117/119] sound/usb/mixer_scarlett_gen2.c:2000:5: sparse: sparse: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static?
by kernel test robot 23 May '21

23 May '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 9ebaef0540a981093bce5df15af32354d32391d9 commit: 265d1a90e4fb6d3264d8122fbd10760e5e733be6 [117/119] ALSA: usb-audio: scarlett2: Improve driver startup messages config: x86_64-randconfig-s021-20210522 (attached as .config) compiler: gcc-9 (Debian 9.3.0-22) 9.3.0 reproduce: # apt-get install sparse # sparse version: v0.6.3-341-g8af24329-dirty # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-linus git checkout 265d1a90e4fb6d3264d8122fbd10760e5e733be6 # save the attached .config to linux build tree make W=1 C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' W=1 ARCH=x86_64 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/usb/mixer_scarlett_gen2.c:2000:5: sparse: sparse: symbol 'snd_scarlett_gen2_controls_create' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

3 5

[sound:for-linus 117/119] sound/usb/mixer_scarlett_gen2.c:2000:5: warning: no previous prototype for function 'snd_scarlett_gen2_controls_create'
by kernel test robot 22 May '21

22 May '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 9ebaef0540a981093bce5df15af32354d32391d9 commit: 265d1a90e4fb6d3264d8122fbd10760e5e733be6 [117/119] ALSA: usb-audio: scarlett2: Improve driver startup messages config: x86_64-randconfig-a014-20210522 (attached as .config) compiler: clang version 13.0.0 (https://github.com/llvm/llvm-project e84a9b9bb3051c35dea993cdad7b3d2575638f85) reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # install x86_64 cross compiling tool for clang build # apt-get install binutils-x86-64-linux-gnu # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-linus git checkout 265d1a90e4fb6d3264d8122fbd10760e5e733be6 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross ARCH=x86_64 If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): >> sound/usb/mixer_scarlett_gen2.c:2000:5: warning: no previous prototype for function 'snd_scarlett_gen2_controls_create' [-Wmissing-prototypes] int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, ^ sound/usb/mixer_scarlett_gen2.c:2000:1: note: declare 'static' if the function is not intended to be used outside of this translation unit int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, ^ static 1 warning generated. vim +/snd_scarlett_gen2_controls_create +2000 sound/usb/mixer_scarlett_gen2.c 1999 > 2000 int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, 2001 const struct scarlett2_device_info *info) 2002 { 2003 int err; 2004 2005 /* Initialise private data, routing, sequence number */ 2006 err = scarlett2_init_private(mixer, info); 2007 if (err < 0) 2008 return err; 2009 2010 /* Read volume levels and controls from the interface */ 2011 err = scarlett2_read_configs(mixer); 2012 if (err < 0) 2013 return err; 2014 2015 /* Create the analogue output controls */ 2016 err = scarlett2_add_line_out_ctls(mixer); 2017 if (err < 0) 2018 return err; 2019 2020 /* Create the analogue input controls */ 2021 err = scarlett2_add_line_in_ctls(mixer); 2022 if (err < 0) 2023 return err; 2024 2025 /* Create the input, output, and mixer mux input selections */ 2026 err = scarlett2_add_mux_enums(mixer); 2027 if (err < 0) 2028 return err; 2029 2030 /* Create the matrix mixer controls */ 2031 err = scarlett2_add_mixer_ctls(mixer); 2032 if (err < 0) 2033 return err; 2034 2035 /* Create the level meter controls */ 2036 err = scarlett2_add_meter_ctl(mixer); 2037 if (err < 0) 2038 return err; 2039 2040 /* Set up the interrupt polling if there are hardware buttons */ 2041 if (info->button_count) { 2042 err = scarlett2_mixer_status_create(mixer); 2043 if (err < 0) 2044 return err; 2045 } 2046 2047 return 0; 2048 } 2049 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[sound:for-linus 117/119] sound/usb/mixer_scarlett_gen2.c:2000:5: warning: no previous prototype for 'snd_scarlett_gen2_controls_create'
by kernel test robot 22 May '21

22 May '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: 9ebaef0540a981093bce5df15af32354d32391d9 commit: 265d1a90e4fb6d3264d8122fbd10760e5e733be6 [117/119] ALSA: usb-audio: scarlett2: Improve driver startup messages config: arc-allyesconfig (attached as .config) compiler: arceb-elf-gcc (GCC) 9.3.0 reproduce (this is a W=1 build): wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound for-linus git checkout 265d1a90e4fb6d3264d8122fbd10760e5e733be6 # save the attached .config to linux build tree COMPILER_INSTALL_PATH=$HOME/0day COMPILER=gcc-9.3.0 make.cross ARCH=arc If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All warnings (new ones prefixed by >>): >> sound/usb/mixer_scarlett_gen2.c:2000:5: warning: no previous prototype for 'snd_scarlett_gen2_controls_create' [-Wmissing-prototypes] 2000 | int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/snd_scarlett_gen2_controls_create +2000 sound/usb/mixer_scarlett_gen2.c 1999 > 2000 int snd_scarlett_gen2_controls_create(struct usb_mixer_interface *mixer, 2001 const struct scarlett2_device_info *info) 2002 { 2003 int err; 2004 2005 /* Initialise private data, routing, sequence number */ 2006 err = scarlett2_init_private(mixer, info); 2007 if (err < 0) 2008 return err; 2009 2010 /* Read volume levels and controls from the interface */ 2011 err = scarlett2_read_configs(mixer); 2012 if (err < 0) 2013 return err; 2014 2015 /* Create the analogue output controls */ 2016 err = scarlett2_add_line_out_ctls(mixer); 2017 if (err < 0) 2018 return err; 2019 2020 /* Create the analogue input controls */ 2021 err = scarlett2_add_line_in_ctls(mixer); 2022 if (err < 0) 2023 return err; 2024 2025 /* Create the input, output, and mixer mux input selections */ 2026 err = scarlett2_add_mux_enums(mixer); 2027 if (err < 0) 2028 return err; 2029 2030 /* Create the matrix mixer controls */ 2031 err = scarlett2_add_mixer_ctls(mixer); 2032 if (err < 0) 2033 return err; 2034 2035 /* Create the level meter controls */ 2036 err = scarlett2_add_meter_ctl(mixer); 2037 if (err < 0) 2038 return err; 2039 2040 /* Set up the interrupt polling if there are hardware buttons */ 2041 if (info->button_count) { 2042 err = scarlett2_mixer_status_create(mixer); 2043 if (err < 0) 2044 return err; 2045 } 2046 2047 return 0; 2048 } 2049 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH 0/6] ALSA: firewire-lib: code refactoring for processing rx packets
by Takashi Sakamoto 22 May '21

22 May '21

Hi, This patchset refactors mainly for processing rx packets. Except for devices handled by ALSA bebob and dice drivers, the devices doesn't take care of the value of syt field of CIP header for playback timing. To code it explicitly, the new flag is added in 1st patch. Some of devices based on DICE ASICs supports two pairs of packet streams. In the case, the sequence of rx packet in the streams should be different independently so that each of them deliver timing information from each peer. On the other hand, current implementation pools the sequence in AMDTP domain, then the rx packets are processed with the same pool. It is inconvenient and in 2nd patch the pool is prepared per streams. The other patches are for my future work. Regards Takashi Sakamoto (6): ALSA: firewire-lib: add flag to unaware of syt in CIP header ALSA: firewire-lib: pool sequence of packet in IT context independently ALSA: firewire-lib: code refactoring for generation of packet descriptors ALSA: firewire-lib: code refactoring for generation of syt sequence ALSA: firewire-lib: code refactoring for generation of data block sequence ALSA: firewire-lib: code refactoring for transfer delay sound/firewire/amdtp-stream.c | 267 +++++++++++++-------------- sound/firewire/amdtp-stream.h | 33 ++-- sound/firewire/motu/amdtp-motu.c | 4 +- sound/firewire/tascam/amdtp-tascam.c | 6 +- 4 files changed, 146 insertions(+), 164 deletions(-) -- 2.27.0

2 7

[PATCH] ALSA: hda/realtek: the bass speaker can't output sound on Yoga 9i
by Hui Wang 22 May '21

22 May '21

The Lenovo Yoga 9i has bass speaker, but the bass speaker can't work, that is because there is an i2s amplifier on that speaker, need to run ideapad_s740_coef() to initialize the amplifier. And also needs to apply ALC285_FIXUP_THINKPAD_HEADSET_JACK to rename the speaker's mixer control name, otherwise the PA can't handle them. BugLink: http://bugs.launchpad.net/bugs/1926165 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4eab336dc739..c6b5db831ed0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6552,6 +6552,7 @@ enum { ALC295_FIXUP_ASUS_DACS, ALC295_FIXUP_HP_OMEN, ALC285_FIXUP_HP_SPECTRE_X360, + ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, }; static const struct hda_fixup alc269_fixups[] = { @@ -8118,6 +8119,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1, }, + [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_ideapad_s740_coef, + .chained = true, + .chain_id = ALC285_FIXUP_THINKPAD_HEADSET_JACK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8486,6 +8493,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), + SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), @@ -8701,6 +8709,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

2 1

[PATCH] ALSA: hda/realtek: Headphone volume is controlled by Front mixer
by Hui Wang 22 May '21

22 May '21

On some ASUS and MSI machines, the audio codec is alc1220 and the Headphone is connected to audio mixer 0xf and DAC 0x5, in theory the Headphone volume is controlled by DAC 0x5 (Heapdhone Playback Volume), but somehow it is controlled by DAC 0x2 (Front Playback Volume), maybe this is a defect on the codec alc1220. Because of this issue, the PA couldn't switch the headphone and Lineout correctly, If we apply the quirk CLEVO_P950 to those machines, the Lineout and Headphone will share the audio mixer 0xc and DAC 0x2, and generate Headphone+LO mixer, then PA could handle them when switching between them. BugLink: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1206 Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 23 +++++++++++++++++++++++ 1 file changed, 23 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c6b5db831ed0..3d40d32ef3ba 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2603,6 +2603,28 @@ static const struct hda_model_fixup alc882_fixup_models[] = { {} }; +static const struct snd_hda_pin_quirk alc882_pin_fixup_tbl[] = { + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1043, "ASUS", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01014010}, + {0x15, 0x01011012}, + {0x16, 0x01016011}, + {0x18, 0x01a19040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181304f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01456130}), + SND_HDA_PIN_QUIRK(0x10ec1220, 0x1462, "MS-7C35", ALC1220_FIXUP_CLEVO_P950, + {0x14, 0x01015010}, + {0x15, 0x01011012}, + {0x16, 0x01011011}, + {0x18, 0x01a11040}, + {0x19, 0x02a19050}, + {0x1a, 0x0181104f}, + {0x1b, 0x0221401f}, + {0x1e, 0x01451130}), + {} +}; + /* * BIOS auto configuration */ @@ -2644,6 +2666,7 @@ static int patch_alc882(struct hda_codec *codec) snd_hda_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, alc882_fixups); + snd_hda_pick_pin_fixup(codec, alc882_pin_fixup_tbl, alc882_fixups, true); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: initialize the cur variable
by trix＠redhat.com 22 May '21

22 May '21

From: Tom Rix <trix(a)redhat.com> Static analysis reports this problem clock.c:338:10: warning: The right operand of '==' is a garbage value if (i == cur) ^ ~~~ In __uac_clock_find_source() the goto find_others skips over setting of the 'cur' variable. Since the pin loop variable starts with 1, it is safe to initialize cur to 0. Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/usb/clock.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 772478c725c2e..edc458c50bf2c 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -258,9 +258,10 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, union uac23_clock_source_desc *source; union uac23_clock_selector_desc *selector; union uac23_clock_multiplier_desc *multiplier; - int ret, i, cur, err, pins, clock_id; + int ret, i, err, pins, clock_id; const u8 *sources; int proto = fmt->protocol; + int cur = 0; entity_id &= 0xff; -- 2.26.3

2 1

[PATCH][next] ALSA: firewire-lib: Fix uninitialized variable err issue
by Colin King 22 May '21

22 May '21

From: Colin Ian King <colin.king(a)canonical.com> Currently in the case where the payload_length is less than the cip_header_size the error return variable err is not being set and function parse_ir_ctx_header can return an uninitialized error return value. Fix this by setting err to zero. Addresses-Coverity: ("Uninitialized scalar variable") Fixes: c09010eeb373 ("ALSA: firewire-lib: handle the case that empty isochronous packet payload for CIP") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/firewire/amdtp-stream.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index af5c3629f1ac..242b1147d768 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -663,6 +663,7 @@ static int parse_ir_ctx_header(struct amdtp_stream *s, unsigned int cycle, } else { // Handle the cycle so that empty packet arrives. cip_header = NULL; + err = 0; *data_blocks = 0; *syt = 0; } -- 2.31.1

3 4

Motu M4 - where to report this issue?
by Georges Basile Stavracas Neto 21 May '21

21 May '21

Greetings ALSA community, I recently acquired a Motu M4 interface, and seems to be working well so far. Everyday usage is completely functional. However, I noticed an odd sound artifact that happens from time to time when using DAWs like Ardour and Waveform11. It's a windy, ghostly sound that happens whenever I open any DAWs (the simple act of opening them triggers this). Interestingly, audio apps that use PulseAudio do not suffer from this issue. It's hard to describe it, so I recorded a video reproducing this problem: https://youtu.be/bM8x-YuXLVI A few things I've discovered while trying to understand what's going on: * It only happens with apps that use JACK or ALSA APIs. Using PulseAudio (through PipeWire) does not trigger it. * Seems like this sound is not an artifact of the interface; it appears that this odd sound is sent to the interface through USB. (You can see this in the video.) * It's triggered per track; in the video, for example, the drums and keyboard tracks are affected independetly. I was oriented in #alsa at Freenode that the output of 'alsa-info' might be helpful, so here it is: http://alsa-project.org/db/?f=62f686710cc6c9344a8986707bf3d09248c514d1 (I'm running this on top of a 5.12.5 kernel, which should include this fix: https://bugzilla.kernel.org/show_bug.cgi?id=207023) I'd like to know where should I report this bug (kernel bugzilla?), and if there's any other information I can provide to help fixing it. With respect, Georges

2 1

[PATCH 1/2] ASoC: Intel: common: Add entries for sdw codecs in ADL match table
by Kai Vehmanen 21 May '21

21 May '21

From: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> RT5682 and Max98373 are added with SDW0,SDW2 links respectively. Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- .../intel/common/soc-acpi-intel-adl-match.c | 45 +++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 39ac6d52106f..22c465f1d5d8 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -223,6 +223,30 @@ static const struct snd_soc_acpi_link_adr adl_sdw_rt1316_link2_rt714_link0[] = { {} }; +static const struct snd_soc_acpi_adr_device mx8373_2_adr[] = { + { + .adr = 0x000223019F837300ull, + .num_endpoints = 1, + .endpoints = &spk_l_endpoint, + .name_prefix = "Left" + }, + { + .adr = 0x000227019F837300ull, + .num_endpoints = 1, + .endpoints = &spk_r_endpoint, + .name_prefix = "Right" + } +}; + +static const struct snd_soc_acpi_adr_device rt5682_0_adr[] = { + { + .adr = 0x000021025D568200ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt5682" + } +}; + static const struct snd_soc_acpi_link_adr adl_rvp[] = { { .mask = BIT(0), @@ -232,6 +256,20 @@ static const struct snd_soc_acpi_link_adr adl_rvp[] = { {} }; +static const struct snd_soc_acpi_link_adr adl_chromebook_base[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt5682_0_adr), + .adr_d = rt5682_0_adr, + }, + { + .mask = BIT(2), + .num_adr = ARRAY_SIZE(mx8373_2_adr), + .adr_d = mx8373_2_adr, + }, + {} +}; + static const struct snd_soc_acpi_codecs adl_max98373_amp = { .num_codecs = 1, .codecs = {"MX98373"} @@ -295,6 +333,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[] = { .drv_name = "sof_sdw", .sof_tplg_filename = "sof-adl-rt711.tplg", }, + { + .link_mask = 0x5, /* rt5682 on link0 & 2xmax98373 on link 2 */ + .links = adl_chromebook_base, + .drv_name = "sof_sdw", + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-sdw-max98373-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_sdw_machines); base-commit: aa736700f42fa0813e286ca2f9274ffaa25163b9 -- 2.31.0

2 2

Motu M4 - where to report this issue?
by Georges Basile Stavracas Neto 21 May '21

21 May '21

Greetings ALSA community, I recently acquired a Motu M4 interface, and seems to be working well so far. Everyday usage is completely functional. However, I noticed an odd sound artifact that happens from time to time when using DAWs like Ardour and Waveform11. It's a windy, ghostly sound that happens whenever I open any DAWs (the simple act of opening them triggers this). Interestingly, audio apps that use PulseAudio do not suffer from this issue. It's hard to describe it, so I recorded a video reproducing this problem: https://youtu.be/bM8x-YuXLVI A few things I've discovered while trying to understand what's going on: * It only happens with apps that use JACK or ALSA APIs. Using PulseAudio (through PipeWire) does not trigger it. * Seems like this sound is not an artifact of the interface; it appears that this odd sound is sent to the interface through USB. (You can see this in the video.) * It's triggered per track; in the video, for example, the drums and keyboard tracks are affected independetly. I was oriented in #alsa at Freenode that the output of 'alsa-info' might be helpful, so here it is: http://alsa-project.org/db/?f=62f686710cc6c9344a8986707bf3d09248c514d1 (I'm running this on top of a 5.12.5 kernel, which should include this fix: https://bugzilla.kernel.org/show_bug.cgi?id=207023) I'd like to know where should I report this bug (kernel bugzilla?), and if there's any other information I can provide to help fixing it. With respect, Georges

1 0

[PATCH v1] ASoC: dt-bindings: Convert imx-audmux binding to json schema
by Oleksij Rempel 21 May '21

21 May '21

Convert the imx-audmux binding to DT schema format using json-schema Signed-off-by: Oleksij Rempel <o.rempel(a)pengutronix.de> --- .../devicetree/bindings/sound/imx-audmux.txt | 28 ----- .../devicetree/bindings/sound/imx-audmux.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 28 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.txt create mode 100644 Documentation/devicetree/bindings/sound/imx-audmux.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.txt b/Documentation/devicetree/bindings/sound/imx-audmux.txt deleted file mode 100644 index 2db4dcbee1b9..000000000000 --- a/Documentation/devicetree/bindings/sound/imx-audmux.txt +++ /dev/null @@ -1,28 +0,0 @@ -Freescale Digital Audio Mux (AUDMUX) device - -Required properties: - - - compatible : "fsl,imx21-audmux" for AUDMUX version firstly used - on i.MX21, or "fsl,imx31-audmux" for the version - firstly used on i.MX31. - - - reg : Should contain AUDMUX registers location and length. - -An initial configuration can be setup using child nodes. - -Required properties of optional child nodes: - - - fsl,audmux-port : Integer of the audmux port that is configured by this - child node. - - - fsl,port-config : List of configuration options for the specific port. - For imx31-audmux and above, it is a list of tuples - <ptcr pdcr>. For imx21-audmux it is a list of pcr - values. - -Example: - -audmux@21d8000 { - compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; - reg = <0x021d8000 0x4000>; -}; diff --git a/Documentation/devicetree/bindings/sound/imx-audmux.yaml b/Documentation/devicetree/bindings/sound/imx-audmux.yaml new file mode 100644 index 000000000000..6078ce13bd46 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audmux.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audmux.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Digital Audio Mux device + +maintainers: + - Oleksij Rempel <o.rempel(a)pengutronix.de> + +properties: + compatible: + oneOf: + - items: + - enum: + - fsl,imx27-audmux + - const: fsl,imx21-audmux + - items: + - enum: + - fsl,imx25-audmux + - fsl,imx35-audmux + - fsl,imx50-audmux + - fsl,imx51-audmux + - fsl,imx53-audmux + - fsl,imx6q-audmux + - fsl,imx6sl-audmux + - fsl,imx6sll-audmux + - fsl,imx6sx-audmux + - const: fsl,imx31-audmux + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + clock-names: + items: + - const: audmux + +patternProperties: + "^mux-[0-9a-z]*$": + type: object + properties: + fsl,audmux-port: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Integer of the audmux port that is configured by this child node + + fsl,port-config: + $ref: /schemas/types.yaml#/definitions/uint32-array + description: | + List of configuration options for the specific port. + For imx31-audmux and above, it is a list of tuples ptcr pdcr. + For imx21-audmux it is a list of pcr values. + + required: + - fsl,audmux-port + - fsl,port-config + + additionalProperties: false + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + }; + - | + audmux@10016000 { + compatible = "fsl,imx27-audmux", "fsl,imx21-audmux"; + reg = <0x10016000 0x1000>; + clocks = <&clks 1>; + clock-names = "audmux"; + + mux-ssi0 { + fsl,audmux-port = <0>; + fsl,port-config = <0xcb205000>; + }; + + mux-pins4 { + fsl,audmux-port = <2>; + fsl,port-config = <0x00001000>; + }; + }; + - | + #include <dt-bindings/sound/fsl-imx-audmux.h> + audmux@21d8000 { + compatible = "fsl,imx6q-audmux", "fsl,imx31-audmux"; + reg = <0x021d8000 0x4000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_audmux>; + + mux-ssi1 { + fsl,audmux-port = <0>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN 0 + IMX_AUDMUX_V2_PTCR_TFSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TCSEL(2) 0 + IMX_AUDMUX_V2_PTCR_TFSDIR 0 + IMX_AUDMUX_V2_PTCR_TCLKDIR IMX_AUDMUX_V2_PDCR_RXDSEL(2) + >; + }; + + mux-pins3 { + fsl,audmux-port = <2>; + fsl,port-config = < + IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(0) + 0 IMX_AUDMUX_V2_PDCR_TXRXEN + >; + }; + }; -- 2.29.2

2 2

[PATCH 0/7] ASoC: SOF: code cleanups for 5.14
by Kai Vehmanen 21 May '21

21 May '21

Set of code cleanups for issues found in code review. Jaska Uimonen (1): ASoC: SOF: topology: fix assignment to use le32_to_cpu Keyon Jie (1): ASoC: SOF: ops: print out the polling register Peter Ujfalusi (4): ASoC: SOF: Check desc->ops directly in acpi/pci/of probe functions ASoC: SOF: pci: No need to cast second time to save the desc ASoC: SOF: loader: Use snd_sof_dsp_block_read() instead sof_block_read() ASoC: SOF: Intel: hda: Remove conditions against CONFIG_PCI Pierre-Louis Bossart (1): ASoC: SOF: ops: don't return void value sound/soc/sof/intel/hda.c | 8 +++----- sound/soc/sof/loader.c | 2 +- sound/soc/sof/ops.h | 10 ++++++---- sound/soc/sof/sof-acpi-dev.c | 5 +---- sound/soc/sof/sof-of-dev.c | 5 +---- sound/soc/sof/sof-pci-dev.c | 7 ++----- sound/soc/sof/topology.c | 2 +- 7 files changed, 15 insertions(+), 24 deletions(-) base-commit: aa736700f42fa0813e286ca2f9274ffaa25163b9 -- 2.31.0

2 8

[PATCH] ASoC: qcom: lpass-cpu: Use optional clk APIs
by Stephen Boyd 21 May '21

21 May '21

This driver spits out a warning for me at boot: sc7180-lpass-cpu 62f00000.lpass: asoc_qcom_lpass_cpu_platform_probe() error getting optional null: -2 but it looks like it is all an optional clk. Use the optional clk APIs here so that we don't see this message and everything else is the same. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Fixes: 3e53ac8230c1 ("ASoC: qcom: make osr clock optional") Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- sound/soc/qcom/lpass-cpu.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..28c7497344e3 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -835,18 +835,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) if (dai_id == LPASS_DP_RX) continue; - drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(dev, + drvdata->mi2s_osr_clk[dai_id] = devm_clk_get_optional(dev, variant->dai_osr_clk_names[i]); - if (IS_ERR(drvdata->mi2s_osr_clk[dai_id])) { - dev_warn(dev, - "%s() error getting optional %s: %ld\n", - __func__, - variant->dai_osr_clk_names[i], - PTR_ERR(drvdata->mi2s_osr_clk[dai_id])); - - drvdata->mi2s_osr_clk[dai_id] = NULL; - } - drvdata->mi2s_bit_clk[dai_id] = devm_clk_get(dev, variant->dai_bit_clk_names[i]); if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) { base-commit: 6efb943b8616ec53a5e444193dccf1af9ad627b5 -- https://chromeos.dev

3 2

[PATCH 0/2] ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci
by Geoffrey D. Bennett 21 May '21

21 May '21

This patchset fixes a show-stopper bug with the Scarlett Gen 2 mixer driver not working with the ehci-pci host driver, and improves the user experience with better driver startup messages. Patch 1: Fixes the driver hanging the device by using the correct direction pipe now. Patch 2: Improves the user experience with more specific enabling instructions, positive feedback when it is enabled or if initialisation fails. The previous instructions were incomplete and didn't confirm that the driver was enabled. #Cc: stable(a)vger.kernel.org please (sorry, not sure if I should be sending this there myself or if it gets added by Takashi after review?) Geoffrey D. Bennett (2): ALSA: usb-audio: scarlett2: Fix device hang with ehci-pci ALSA: usb-audio: scarlett2: Improve driver startup messages sound/usb/mixer_quirks.c | 2 +- sound/usb/mixer_scarlett_gen2.c | 81 +++++++++++++++++++++------------ sound/usb/mixer_scarlett_gen2.h | 2 +- 3 files changed, 53 insertions(+), 32 deletions(-) -- 2.31.1

2 3

[PATCH] ALSA: usb-audio: fix control-request direction
by Johan Hovold 21 May '21

21 May '21

The direction of the pipe argument must match the request-type direction bit or control requests may fail depending on the host-controller-driver implementation. Fix the UAC2_CS_CUR request which erroneously used usb_sndctrlpipe(). Fixes: 93db51d06b32 ("ALSA: usb-audio: Check valid altsetting at parsing rates for UAC2/3") Cc: stable(a)vger.kernel.org # 5.10 Signed-off-by: Johan Hovold <johan(a)kernel.org> --- There's a related bug in sound/usb/mixer_scarlett_gen2.c, which Geoffrey reported and said he was preparing a patch for here: https://lore.kernel.org/r/20210520180819.GA95348@m.b4.vu Johan sound/usb/format.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/format.c b/sound/usb/format.c index e6ff317a6785..2287f8c65315 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -436,7 +436,7 @@ static bool check_valid_altsetting_v2v3(struct snd_usb_audio *chip, int iface, if (snd_BUG_ON(altsetting >= 64 - 8)) return false; - err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR, + err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, UAC2_AS_VAL_ALT_SETTINGS << 8, iface, &raw_data, sizeof(raw_data)); -- 2.26.3

2 1

[bug report] ALSA: usb-audio: Handle error for the current selector gracefully
by Dan Carpenter 21 May '21

21 May '21

Hello Takashi Iwai, The patch 481f17c41803: "ALSA: usb-audio: Handle error for the current selector gracefully" from May 18, 2021, leads to the following static checker warning: sound/usb/clock.c:338 __uac_clock_find_source() error: uninitialized symbol 'cur'. sound/usb/clock.c 254 static int __uac_clock_find_source(struct snd_usb_audio *chip, 255 const struct audioformat *fmt, int entity_id, 256 unsigned long *visited, bool validate) 257 { 258 union uac23_clock_source_desc *source; 259 union uac23_clock_selector_desc *selector; 260 union uac23_clock_multiplier_desc *multiplier; 261 int ret, i, cur, err, pins, clock_id; ^^^ 262 const u8 *sources; 263 int proto = fmt->protocol; 264 265 entity_id &= 0xff; 266 267 if (test_and_set_bit(entity_id, visited)) { 268 usb_audio_warn(chip, 269 "%s(): recursive clock topology detected, id %d.\n", 270 __func__, entity_id); 271 return -EINVAL; 272 } 273 274 /* first, see if the ID we're looking for is a clock source already */ 275 source = snd_usb_find_clock_source(chip, entity_id, proto); 276 if (source) { 277 entity_id = GET_VAL(source, proto, bClockID); 278 if (validate && !uac_clock_source_is_valid(chip, fmt, 279 entity_id)) { 280 usb_audio_err(chip, 281 "clock source %d is not valid, cannot use\n", 282 entity_id); 283 return -ENXIO; 284 } 285 return entity_id; 286 } 287 288 selector = snd_usb_find_clock_selector(chip, entity_id, proto); 289 if (selector) { 290 pins = GET_VAL(selector, proto, bNrInPins); 291 clock_id = GET_VAL(selector, proto, bClockID); 292 sources = GET_VAL(selector, proto, baCSourceID); 293 294 if (pins == 1) { 295 ret = 1; 296 goto find_source; 297 } 298 299 /* the entity ID we are looking for is a selector. 300 * find out what it currently selects */ 301 ret = uac_clock_selector_get_val(chip, clock_id); 302 if (ret < 0) { 303 if (!chip->autoclock) 304 return ret; 305 goto find_others; ^^^^^^^^^^^^^^^^^ Assume we hit one of these gotos 306 } 307 308 /* Selector values are one-based */ 309 310 if (ret > pins || ret < 1) { 311 usb_audio_err(chip, 312 "%s(): selector reported illegal value, id %d, ret %d\n", 313 __func__, clock_id, ret); 314 315 if (!chip->autoclock) 316 return -EINVAL; 317 ret = 0; 318 goto find_others; ^^^^^^^^^^^^^^^^^ 319 } 320 321 find_source: 322 cur = ret; 323 ret = __uac_clock_find_source(chip, fmt, 324 sources[ret - 1], 325 visited, validate); 326 if (ret > 0) { 327 err = uac_clock_selector_set_val(chip, entity_id, cur); 328 if (err < 0) 329 return err; 330 } 331 332 if (!validate || ret > 0 || !chip->autoclock) 333 return ret; 334 335 find_others: 336 /* The current clock source is invalid, try others. */ 337 for (i = 1; i <= pins; i++) { 338 if (i == cur) ^^^^^^^^ 339 continue; 340 341 ret = __uac_clock_find_source(chip, fmt, 342 sources[i - 1], 343 regards, dan carpenter

2 1

Broken out-of-tree build
by GitHub issues - opened 20 May '21

20 May '21

alsa-project/alsa-lib issue #136 was opened from concatime: Hi. Due to missing headers, I cannot build alsa-lib if the build directory is outside of the source tree. ``` make[2]: Entering directory '/home/iemaghni/Code/tmp/build/src/ucm' CC utils.lo In file included from ../../../alsa-lib/src/ucm/ucm_local.h:41, from ../../../alsa-lib/src/ucm/utils.c:33: ../../../alsa-lib/include/use-case.h:45:10: fatal error: alsa/asoundlib.h: No such file or directory 45 | #include <alsa/asoundlib.h> | ^~~~~~~~~~~~~~~~~~ ``` Step to reproduce: ```sh git clone https://github.com/alsa-project/alsa-lib.git mkdir build cd alsa-lib libtoolize --force --copy --automake aclocal autoheader automake --foreign --copy --add-missing autoconf cd ../build ../alsa-lib/configure make ``` Thank you! Issue URL : https://github.com/alsa-project/alsa-lib/issues/136 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v3 1/2] ASoC: dt-bindings: imx-card: Add binding doc for imx sound card
by Shengjiu Wang 20 May '21

20 May '21

Imx-card is a new added machine driver for supporting ak4458/ak5558/ak5552/ak4497 codec on i.MX platforms. But these DAC/ADCs are not only supported codecs. This machine driver is designed to be a more common machine driver for i.MX platform, it can support widely cpu dai interface and codec dai interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3: - use a generic compatible string as Rob's comments. - change the file name changes in v2: - update doc accoring to Rob's comment, use the common porperties. .../bindings/sound/imx-audio-card.yaml | 122 ++++++++++++++++++ 1 file changed, 122 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-card.yaml diff --git a/Documentation/devicetree/bindings/sound/imx-audio-card.yaml b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml new file mode 100644 index 000000000000..d1816dd061cf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-card.yaml @@ -0,0 +1,122 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/imx-audio-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP i.MX audio sound card. + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +properties: + compatible: + enum: + - fsl,imx-audio-card + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User specified audio sound card name + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: + A list of the connections between audio components. Each entry is a + pair of strings, the first being the connection's sink, the second + being the connection's source. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + format: + description: audio format. + items: + enum: + - i2s + - dsp_b + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + properties: + sound-dai: true + + codec: + description: Holds subnode which indicates codec dai. + type: object + properties: + sound-dai: true + + fsl,mclk-equal-bclk: + description: Indicates mclk can be equal to bclk, especially for sai interface + $ref: /schemas/types.yaml#/definitions/flag + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - model + +additionalProperties: false + +examples: + - | + sound-ak4458 { + compatible = "fsl,imx-audio-card"; + model = "ak4458-audio"; + pri-dai-link { + link-name = "akcodec"; + format = "i2s"; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + fe-dai-link { + link-name = "HiFi-ASRC-FE"; + format = "i2s"; + cpu { + sound-dai = <&easrc>; + }; + }; + be-dai-link { + link-name = "HiFi-ASRC-BE"; + format = "dsp_b"; + dai-tdm-slot-num = <8>; + dai-tdm-slot-width = <32>; + fsl,mclk-equal-bclk; + cpu { + sound-dai = <&sai1>; + }; + codec { + sound-dai = <&ak4458_1>, <&ak4458_2>; + }; + }; + }; -- 2.27.0

3 3

[PATCH] ASoC: topology: Fix using uninitialized pointer
by Jaska Uimonen 20 May '21

20 May '21

The original patch changed kcontrol_type to a pointer. In some goto cases the pointer is assigned into a struct member as uninitialized and this will cause a runtime error with UBSan even if it isn't a real bug. So initialize the pointer to NULL. Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: d29d41e28eea ("ASoC: topology: Add support for multiple kcontrol types to a widget") Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e71d98d7b116..5e65b72910e9 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1422,7 +1422,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; - unsigned int *kcontrol_type; + unsigned int *kcontrol_type = NULL; struct snd_kcontrol_new *kc; int mixer_count = 0; int bytes_count = 0; -- 2.24.1

2 1

[PATCH] ASoC: cs35l33: fix an error code in probe()
by Dan Carpenter 20 May '21

20 May '21

This error path returns zero (success) but it should return -EINVAL. Fixes: 3333cb7187b9 ("ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/cs35l33.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c index 6f6b3c0c88b7..2a6f5e46d031 100644 --- a/sound/soc/codecs/cs35l33.c +++ b/sound/soc/codecs/cs35l33.c @@ -1202,6 +1202,7 @@ static int cs35l33_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS35L33 Device ID (%X). Expected ID %X\n", devid, CS35L33_CHIP_ID); + ret = -EINVAL; goto err_enable; } -- 2.30.2

3 2

[BUG] ALSA: usb-audio: Scarlett 2 mixer driver fails on ehci-pci
by Geoffrey D. Bennett 20 May '21

20 May '21

Hi there, I'm trying to track down an issue with my Scarlett Gen 2 mixer driver that has been reported by a few people, and I can now reliably reproduce the issue, but I need some help in figuring out where exactly the issue is and how to fix it please. The issue only occurs when attempting to use the interface on a USB port using ehci-pci. xhci_hcd USB ports work fine. The issue occurs when sending the first vendor-specific USB command, but only when sending from the kernel driver. Sending the same USB packets from user-space works fine(!). I did initially think that the fault could have been due to earlier USB messages putting the device into a state where it would reject the vendor-specific USB commands, but I have carefully ruled that out & have gotten identical usbmon traces from device power-on up until the device responds differently, the only difference beforehand being whether the USB packet was sent from the kernel driver or user-space. The messages look like this in "usbmon -s 10000 -fu" when sent from user-space (or when sent from the kernel driver when the interface is plugged in to an xhci_hcd port): ffff888125855200 1006026497 S Ci:2:040:0 s a1 00 0000 0005 0018 24 < ffff888125855200 1006026680 C Ci:2:040:0 0 24 = 66191018 73190604 01000000 01000000 00040000 00000000 And like this when sent from the kernel driver when the interface is plugged in to an ehci-pci port: ffff88810487a300 3686673995 S Ci:2:036:0 s a1 00 0000 0005 0018 24 < ffff88810487a300 3692178724 C Ci:2:036:0 -2 0 Identical messages sent according to usbmon, but they must be different somehow! The kernel code to send that message looks like this: return snd_usb_ctl_msg( dev, usb_sndctrlpipe(dev, 0), usb_req, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, 0, interface, buf, size); and helper.c snd_usb_ctl_msg() then calls usb_control_msg(). For sending arbitrary USB packets from user-space for testing, I'm using libusb and: int ret = usb_control_msg( devh, reqtype, request, value, index, buf, size, 1000 ); So, I presume usbmon isn't giving me the full story? How can I determine the difference between the kernel and the user-space usb_control_msg() functions? I see that I can #define EHCI_URB_TRACE in ehci-hcd.c. Can anyone with more experience than me let me know if I'm going in the right direction to track this down? Thanks, Geoffrey.

2 5

Add UCMs for Google Asus Nexus 7 and Acer Iconia A500
by GitHub pull_request - edited 20 May '21

20 May '21

alsa-project/alsa-ucm-conf pull request #92 was edited from digetx: @perexg As was discussed previouly at https://www.spinics.net/lists/alsa-devel/msg126382.html I'm opening PR with the UCMs for Nexus 7 and A500. Both these devices are supported by the mainline Linux kernel. You're right that Tegra ASoC driver doesn't set card's name properly. In particular I found that we need to set the card's `driver_name` to `tegra`, then UCM get a proper lookup path. I rebased UCMs on top of the recent alsa-ucm-conf master and switched Nexus 7 UCM to use generic RT5640 UCM. Everything works great once the card name of the kernel driver is corrected. I'll send the kernel patches ASAP for review. Nexus 7 alsa-info: https://alsa-project.org/db/?f=0a29f1dab39ec5b33e01215dee42dbfd1dbbf08e A500 alsa-info: https://alsa-project.org/db/?f=bde4f7938a39b1d26904b414969fc811e726242f The alsa-info is taken using the updated kernel ASoC driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add UCMs for Google Asus Nexus 7 and Acer Iconial A500
by GitHub pull_request - opened 20 May '21

20 May '21

alsa-project/alsa-ucm-conf pull request #92 was opened from digetx: @perexg As was discussed previouly at https://www.spinics.net/lists/alsa-devel/msg126382.html I'm opening PR with the UCMs for Nexus 7 and A500. Both these devices are supported by the mainline Linux kernel. You're right that Tegra ASoC driver doesn't set card's name properly. In particular I found that we need to set the card's `driver_name` to `tegra`, then UCM get a proper lookup path. I rebased UCMs on top of the recent alsa-ucm-conf master and switched Nexus 7 UCM to use generic RT5640 UCM. Everything works great once the card name of the kernel driver is corrected. I'll send the kernel patches ASAP for review. Nexus 7 alsa-info: https://alsa-project.org/db/?f=0a29f1dab39ec5b33e01215dee42dbfd1dbbf08e A500 alsa-info: https://alsa-project.org/db/?f=bde4f7938a39b1d26904b414969fc811e726242f The alsa-info is taken using the updated kernel ASoC driver. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/92.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

re: ALSA: usb-audio: Handle error for the current selector gracefully [ uninitialized variable issue ]
by Colin Ian King 20 May '21

20 May '21

Hi, Static analysis with Coverity on linux-next has detected an uninitialized variable issue with the following commit: commit 481f17c41803985446fd12887b2c042f9c43b0d5 Author: Takashi Iwai <tiwai(a)suse.de> Date: Tue May 18 17:21:12 2021 +0200 ALSA: usb-audio: Handle error for the current selector gracefully The branching is a bit convoluted and we end up with variable cur not being initialized. Analysis is as follows: 254static int __uac_clock_find_source(struct snd_usb_audio *chip, 255 const struct audioformat *fmt, int entity_id, 256 unsigned long *visited, bool validate) 257{ 258 union uac23_clock_source_desc *source; 259 union uac23_clock_selector_desc *selector; 260 union uac23_clock_multiplier_desc *multiplier; 1. var_decl: Declaring variable cur without initializer. 261 int ret, i, cur, err, pins, clock_id; 262 const u8 *sources; 263 int proto = fmt->protocol; 264 265 entity_id &= 0xff; 266 2. Condition test_and_set_bit(entity_id, visited), taking false branch. 267 if (test_and_set_bit(entity_id, visited)) { 268 usb_audio_warn(chip, 269 "%s(): recursive clock topology detected, id %d.\n", 270 __func__, entity_id); 271 return -EINVAL; 272 } 273 274 /* first, see if the ID we're looking for is a clock source already */ 275 source = snd_usb_find_clock_source(chip, entity_id, proto); 3. Condition source, taking false branch. 276 if (source) { 277 entity_id = GET_VAL(source, proto, bClockID); 278 if (validate && !uac_clock_source_is_valid(chip, fmt, 279 entity_id)) { 280 usb_audio_err(chip, 281 "clock source %d is not valid, cannot use\n", 282 entity_id); 283 return -ENXIO; 284 } 285 return entity_id; 286 } 287 288 selector = snd_usb_find_clock_selector(chip, entity_id, proto); 4. Condition selector, taking true branch. 289 if (selector) { 5. Condition proto == 48, taking true branch. 290 pins = GET_VAL(selector, proto, bNrInPins); 6. Condition proto == 48, taking true branch. 291 clock_id = GET_VAL(selector, proto, bClockID); 7. Condition proto == 48, taking true branch. 292 sources = GET_VAL(selector, proto, baCSourceID); 293 8. Condition pins == 1, taking false branch. 294 if (pins == 1) { 295 ret = 1; 296 goto find_source; 297 } 298 299 /* the entity ID we are looking for is a selector. 300 * find out what it currently selects */ 301 ret = uac_clock_selector_get_val(chip, clock_id); 9. Condition ret < 0, taking true branch. 302 if (ret < 0) { 10. Condition !chip->autoclock, taking false branch. 303 if (!chip->autoclock) 304 return ret; 11. Jumping to label find_others. 305 goto find_others; 306 } 307 308 /* Selector values are one-based */ 309 310 if (ret > pins || ret < 1) { 311 usb_audio_err(chip, 312 "%s(): selector reported illegal value, id %d, ret %d\n", 313 __func__, clock_id, ret); 314 315 if (!chip->autoclock) 316 return -EINVAL; 317 ret = 0; 318 goto find_others; 319 } 320 321 find_source: 322 cur = ret; 323 ret = __uac_clock_find_source(chip, fmt, 324 sources[ret - 1], 325 visited, validate); 326 if (ret > 0) { 327 err = uac_clock_selector_set_val(chip, entity_id, cur); 328 if (err < 0) 329 return err; 330 } 331 332 if (!validate || ret > 0 || !chip->autoclock) 333 return ret; 334 335 find_others: 336 /* The current clock source is invalid, try others. */ 12. Condition i <= pins, taking true branch. 337 for (i = 1; i <= pins; i++) { Uninitialized scalar variable (UNINIT) 13. uninit_use: Using uninitialized value cur. 338 if (i == cur) 339 continue; 340 Colin

2 1

[PATCH 0/8] ALSA: firewire-lib: start processing content of packet at the same cycle in several contexts for each direction
by Takashi Sakamoto 20 May '21

20 May '21

Hi, Within devices supported by drivers in ALSA firewire stack, only DICE ASICs support several pair of isochronous packet streams. As long as I experienced, the ASIC requires drivers to transfer packets at the same isochronous cycle. Unless, it generates no sound. Additionally, in future integration I have a plan to take drivers performing sequence replay for media clock recovery. It's convenient to manage the cycle to start processing content of packet in several IR contexts. This patchset is to start processing content of packet at the same cycle in several contexts for each direction, including code refactoring. Takashi Sakamoto (8): ALSA: firewire-lib: code refactoring for size of CIP header ALSA: firewire-lib: code refactoring for calculation of context payload ALSA: firewire-lib: code refactoring for selection of IT context header length ALSA: firewire-lib: start processing content of packet at the same cycle in several IR contexts ALSA: firewire-lib: skip initial packets instead of scheduling IR context ALSA: firewire-lib: code refactoring to start several IT/IR contexts ALSA: firewire-lib: start processing content of packet at the same cycle in several IT contexts ALSA: firewire-lib: change waking up timing to process packets sound/firewire/amdtp-stream.c | 492 ++++++++++++++------ sound/firewire/amdtp-stream.h | 53 ++- sound/firewire/bebob/bebob_stream.c | 29 +- sound/firewire/dice/dice-stream.c | 15 +- sound/firewire/digi00x/digi00x-stream.c | 7 +- sound/firewire/fireface/ff-stream.c | 7 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/motu/amdtp-motu.c | 9 +- sound/firewire/motu/motu-stream.c | 7 +- sound/firewire/oxfw/oxfw-stream.c | 14 +- sound/firewire/tascam/tascam-stream.c | 7 +- 11 files changed, 401 insertions(+), 247 deletions(-) -- 2.27.0

2 9

[PATCH 1/2] ASoC: topology: Fix some error codes in soc_tplg_dapm_widget_create()
by Dan Carpenter 20 May '21

20 May '21

Return -ENOMEM if these allocations fail. (The current code returns success). Fixes: d29d41e28eea ("ASoC: topology: Add support for multiple kcontrol types to a widget") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-topology.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e71d98d7b116..6b7a813bc264 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1481,13 +1481,17 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.num_kcontrols = le32_to_cpu(w->num_kcontrols); kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); - if (!kc) + if (!kc) { + ret = -ENOMEM; goto err; + } kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), GFP_KERNEL); - if (!kcontrol_type) + if (!kcontrol_type) { + ret = -ENOMEM; goto err; + } for (i = 0; i < w->num_kcontrols; i++) { control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; -- 2.30.2

2 3

[PATCH v6 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 20 May '21

20 May '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v5: - removed ref in qcom,mbhc-buttons-vthreshold-microvolt bindings to fix dt_bindings_check warning Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 148 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 298 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 705 ++++ 9 files changed, 5345 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 10

[PATCH v1 0/2] Unify NVIDIA Tegra ASoC machine drivers
by Dmitry Osipenko 19 May '21

19 May '21

During review of the RT5631 machine driver, Jon Hunter suggested that it will be better to squash all the current ASoC machine drivers into a single one. This suggestion is implemented by this patchset. The RT5631 support will come later with the ASUS Transformer changes. Dmitry Osipenko (2): of: base: Export of_device_compatible_match() ASoC: tegra: Unify ASoC machine drivers drivers/of/base.c | 1 + sound/soc/tegra/Kconfig | 12 + sound/soc/tegra/Makefile | 18 +- sound/soc/tegra/tegra_alc5632.c | 259 ---------- sound/soc/tegra/tegra_asoc_machine.c | 737 +++++++++++++++++++++++++++ sound/soc/tegra/tegra_asoc_machine.h | 44 ++ sound/soc/tegra/tegra_max98090.c | 276 ---------- sound/soc/tegra/tegra_rt5640.c | 222 -------- sound/soc/tegra/tegra_rt5677.c | 324 ------------ sound/soc/tegra/tegra_sgtl5000.c | 211 -------- sound/soc/tegra/tegra_wm8753.c | 185 ------- sound/soc/tegra/tegra_wm8903.c | 349 +++---------- sound/soc/tegra/tegra_wm9712.c | 166 ------ sound/soc/tegra/trimslice.c | 172 ------- 14 files changed, 860 insertions(+), 2116 deletions(-) delete mode 100644 sound/soc/tegra/tegra_alc5632.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.c create mode 100644 sound/soc/tegra/tegra_asoc_machine.h delete mode 100644 sound/soc/tegra/tegra_max98090.c delete mode 100644 sound/soc/tegra/tegra_rt5640.c delete mode 100644 sound/soc/tegra/tegra_rt5677.c delete mode 100644 sound/soc/tegra/tegra_sgtl5000.c delete mode 100644 sound/soc/tegra/tegra_wm8753.c delete mode 100644 sound/soc/tegra/tegra_wm9712.c delete mode 100644 sound/soc/tegra/trimslice.c -- 2.30.2

4 13

Intel HDA Driver Issue With PCM RW Interleaved
by re.mcclue 19 May '21

19 May '21

I'm trying to play PCM buffer single threaded without buffer underrun. Program source: https://gist.github.com/takaswie/d8a9553c230971d974a3dfc759b2ff67 (struct snd_pcm_sw_params).start_threshold is altered so as to allow specifying the start of audio frame transmission and prevent underrun issues. In my application, on ioctl(fd, SNDRV_PCM_IOCTL_STATUS, &status), I receive: File descriptor in bad state Output of $(dmesg) is: [ 69.013084] snd_hda_intel 0000:00:1f.3: Too big adjustment 128 What is causing this issue and how can I solve it? Thanks Sent with [ProtonMail](https://protonmail.com) Secure Email.

2 1

[PATCH v3 0/2] ASoC: meson: g12a-toacodec: add support for SM1
by Neil Armstrong 19 May '21

19 May '21

This patchset adds Amlogic SM1 support on the TOACODEC driver, first by switching to regmap fields for some bit fields to avoid code duplication, and then by adding the corresponding bits & struct for the SM1 changed bits. Changes since v2 at [2]: - use raw values instead of defines in REG_FIELD() for g12a_toacodec_match_data Changes since v1 at [1]: - switch to regmap field [1] https://lore.kernel.org/r/20210429170147.3615883-1-narmstrong@baylibre.com [2] https://lore.kernel.org/r/20210505072607.3815442-1-narmstrong@baylibre.com Neil Armstrong (2): ASoC: meson: g12a-toacodec: use regmap fields to prepare SM1 support ASoC: meson: g12a-toacodec: add support for SM1 TOACODEC sound/soc/meson/g12a-toacodec.c | 143 +++++++++++++++++++++++++++----- 1 file changed, 124 insertions(+), 19 deletions(-) -- 2.25.1

2 3

[PATCH] dt-bindings: sound: wm8750: convert to the json-schema
by Rafał Miłecki 19 May '21

19 May '21

From: Rafał Miłecki <rafal(a)milecki.pl> This helps validating DTS files. Signed-off-by: Rafał Miłecki <rafal(a)milecki.pl> --- .../devicetree/bindings/sound/wm8750.txt | 18 -------- .../devicetree/bindings/sound/wm8750.yaml | 42 +++++++++++++++++++ 2 files changed, 42 insertions(+), 18 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/wm8750.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8750.yaml diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt deleted file mode 100644 index 682f221f6f38..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8750.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8750 and WM8987 audio CODECs - -These devices support both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8750" or "wlf,wm8987" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8750: codec@1a { - compatible = "wlf,wm8750"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8750.yaml b/Documentation/devicetree/bindings/sound/wm8750.yaml new file mode 100644 index 000000000000..24246ac7bbdf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8750.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wm8750.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8750 and WM8987 audio CODECs + +description: | + These devices support both I2C and SPI (configured with pin strapping + on the board). + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + enum: + - wlf,wm8750 + - wlf,wm8987 + + reg: + description: + The I2C address of the device for I2C, the chip select number for SPI + maxItems: 1 + +additionalProperties: false + +required: + - reg + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8750"; + reg = <0x1a>; + }; + }; -- 2.26.2

3 5

[PATCH] ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware
by Ranjani Sridharan 19 May '21

19 May '21

BE hw_params op was recently added for SSP type DAIs. But sending the DAI_CONFIG IPC during hw_params is not supported with older firmware. So add an ABI check to avoid sending the IPC if the firmware ABI is older than 3.18. Fixes: e12be9fbfb91 ('ASoC: SOF: Intel: HDA: add hw params callback for SSP DAIs') Tested-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 8d7bab433fb3..c1f9f0f58464 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -421,11 +421,16 @@ static int ssp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct sof_ipc_dai_config *config; struct snd_sof_dai *sof_dai; struct sof_ipc_reply reply; int ret; + /* DAI_CONFIG IPC during hw_params is not supported in older firmware */ + if (v->abi_version < SOF_ABI_VER(3, 18, 0)) + return 0; + list_for_each_entry(sof_dai, &sdev->dai_list, list) { if (!sof_dai->cpu_dai_name || !sof_dai->dai_config) continue; -- 2.25.1

2 3

[PATCH] ASoC: meson: g12a-toacodec: add support for SM1 TOACODEC
by Neil Armstrong 19 May '21

19 May '21

This adds support for the TOACODEC found in Amlogic SM1 SoCs. The bits are shifted for more selection of clock sources, so this only maps the same support for G12A to the SM1 bits. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> --- sound/soc/meson/g12a-toacodec.c | 126 +++++++++++++++++++++++++++++++- 1 file changed, 124 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c index 9339fabccb79..9c1c6b65baad 100644 --- a/sound/soc/meson/g12a-toacodec.c +++ b/sound/soc/meson/g12a-toacodec.c @@ -23,15 +23,27 @@ #define CTRL0_ENABLE_SHIFT 31 #define CTRL0_DAT_SEL_SHIFT 14 #define CTRL0_DAT_SEL (0x3 << CTRL0_DAT_SEL_SHIFT) +#define CTRL0_DAT_SEL_SHIFT_SM1 18 +#define CTRL0_DAT_SEL_SM1 (0x3 << CTRL0_DAT_SEL_SHIFT_SM1) #define CTRL0_LANE_SEL 12 +#define CTRL0_LANE_SEL_SM1 16 #define CTRL0_LRCLK_SEL GENMASK(9, 8) +#define CTRL0_LRCLK_INV_SM1 BIT(10) +#define CTRL0_LRCLK_SEL_SM1 GENMASK(14, 12) #define CTRL0_BLK_CAP_INV BIT(7) +#define CTRL0_BLK_CAP_INV_SM1 BIT(9) #define CTRL0_BCLK_O_INV BIT(6) +#define CTRL0_BCLK_O_INV_SM1 BIT(8) #define CTRL0_BCLK_SEL GENMASK(5, 4) +#define CTRL0_BCLK_SEL_SM1 GENMASK(6, 4) #define CTRL0_MCLK_SEL GENMASK(2, 0) #define TOACODEC_OUT_CHMAX 2 +struct g12a_toacodec_match_data { + const struct snd_soc_component_driver *component_drv; +}; + static const char * const g12a_toacodec_mux_texts[] = { "I2S A", "I2S B", "I2S C", }; @@ -85,15 +97,73 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static int sm1_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_DAT_SEL_SM1, + FIELD_PREP(CTRL0_DAT_SEL_SM1, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL0_DAT_SEL_SM1 | + CTRL0_LRCLK_SEL_SM1 | + CTRL0_BCLK_SEL_SM1, + FIELD_PREP(CTRL0_DAT_SEL_SM1, mux) | + FIELD_PREP(CTRL0_LRCLK_SEL_SM1, mux) | + FIELD_PREP(CTRL0_BCLK_SEL_SM1, mux)); + + /* + * FIXME: + * On this soc, the glue gets the MCLK directly from the clock + * controller instead of going the through the TDM interface. + * + * Here we assume interface A uses clock A, etc ... While it is + * true for now, it could be different. Instead the glue should + * find out the clock used by the interface and select the same + * source. For that, we will need regmap backed clock mux which + * is a work in progress + */ + snd_soc_component_update_bits(component, e->reg, + CTRL0_MCLK_SEL, + FIELD_PREP(CTRL0_MCLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, CTRL0_DAT_SEL_SHIFT, g12a_toacodec_mux_texts); +static SOC_ENUM_SINGLE_DECL(sm1_toacodec_mux_enum, TOACODEC_CTRL0, + CTRL0_DAT_SEL_SHIFT_SM1, + g12a_toacodec_mux_texts); + static const struct snd_kcontrol_new g12a_toacodec_mux = SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum, snd_soc_dapm_get_enum_double, g12a_toacodec_mux_put_enum); +static const struct snd_kcontrol_new sm1_toacodec_mux = + SOC_DAPM_ENUM_EXT("Source", sm1_toacodec_mux_enum, + snd_soc_dapm_get_enum_double, + sm1_toacodec_mux_put_enum); + static const struct snd_kcontrol_new g12a_toacodec_out_enable = SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0, CTRL0_ENABLE_SHIFT, 1, 0); @@ -105,6 +175,13 @@ static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = { &g12a_toacodec_out_enable), }; +static const struct snd_soc_dapm_widget sm1_toacodec_widgets[] = { + SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0, + &sm1_toacodec_mux), + SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_out_enable), +}; + static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -175,6 +252,13 @@ static int g12a_toacodec_component_probe(struct snd_soc_component *c) CTRL0_BLK_CAP_INV); } +static int sm1_toacodec_component_probe(struct snd_soc_component *c) +{ + /* Initialize the static clock parameters */ + return snd_soc_component_write(c, TOACODEC_CTRL0, + CTRL0_BLK_CAP_INV_SM1); +} + static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { { "SRC", "I2S A", "IN A Playback" }, { "SRC", "I2S B", "IN B Playback" }, @@ -187,6 +271,10 @@ static const struct snd_kcontrol_new g12a_toacodec_controls[] = { SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), }; +static const struct snd_kcontrol_new sm1_toacodec_controls[] = { + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1, 3, 0), +}; + static const struct snd_soc_component_driver g12a_toacodec_component_drv = { .probe = g12a_toacodec_component_probe, .controls = g12a_toacodec_controls, @@ -199,25 +287,59 @@ static const struct snd_soc_component_driver g12a_toacodec_component_drv = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver sm1_toacodec_component_drv = { + .probe = sm1_toacodec_component_probe, + .controls = sm1_toacodec_controls, + .num_controls = ARRAY_SIZE(sm1_toacodec_controls), + .dapm_widgets = sm1_toacodec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sm1_toacodec_widgets), + .dapm_routes = g12a_toacodec_routes, + .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + static const struct regmap_config g12a_toacodec_regmap_cfg = { .reg_bits = 32, .val_bits = 32, .reg_stride = 4, }; +static const struct g12a_toacodec_match_data g12a_toacodec_match_data = { + .component_drv = &g12a_toacodec_component_drv, +}; + +static const struct g12a_toacodec_match_data sm1_toacodec_match_data = { + .component_drv = &sm1_toacodec_component_drv, +}; + static const struct of_device_id g12a_toacodec_of_match[] = { - { .compatible = "amlogic,g12a-toacodec", }, + { + .compatible = "amlogic,g12a-toacodec", + .data = &g12a_toacodec_match_data, + }, + { + .compatible = "amlogic,sm1-toacodec", + .data = &sm1_toacodec_match_data, + }, {} }; MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match); static int g12a_toacodec_probe(struct platform_device *pdev) { + const struct g12a_toacodec_match_data *data; struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; int ret; + data = device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + ret = device_reset(dev); if (ret) return ret; @@ -234,7 +356,7 @@ static int g12a_toacodec_probe(struct platform_device *pdev) } return devm_snd_soc_register_component(dev, - &g12a_toacodec_component_drv, g12a_toacodec_dai_drv, + data->component_drv, g12a_toacodec_dai_drv, ARRAY_SIZE(g12a_toacodec_dai_drv)); } -- 2.25.1

3 3

[PATCH] ASoC: soc-dai: fix up hw params only if it is needed
by Gyeongtaek Lee 19 May '21

19 May '21

If fixed hw params won't be used, fixing up isn't needed also. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dai.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 080fbe053fc5..4df1aae8abf3 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -327,14 +327,15 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = 0; - /* perform any topology hw_params fixups before DAI */ - ret = snd_soc_link_be_hw_params_fixup(rtd, params); - if (ret < 0) - goto end; - if (dai->driver->ops && - dai->driver->ops->hw_params) + dai->driver->ops->hw_params) { + /* perform any topology hw_params fixups before DAI */ + ret = snd_soc_link_be_hw_params_fixup(rtd, params); + if (ret < 0) + goto end; + ret = dai->driver->ops->hw_params(substream, params, dai); + } /* mark substream if succeeded */ if (ret == 0) base-commit: c06a2ba62fc401b7aaefd23f5d0bc06d2457ccc1 -- 2.21.0

2 1

[PATCH v11 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 19 May '21

19 May '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v11: - Stopped checking for presence of parent node in codec driver, as driver should never be called if parent device doesn't exist to call it. - Made codec subnode optional and stopped returning errors if it is missing. - Downgraded missing codec clock message from dev_err to dev_dbg. Users who choose to use the rk817 PMIC but do not choose to use the codec will not get an error message unless they have debugging enabled, whereas users who wish to use the codec but don't have the clock defined will not get an error message unless they have debug messages enabled. This seems like a fair compromise between mututally exclusive users. Note that errors with the clock or codec will still result in a dev_err. - Updated DT documentation to note that codec subnode is optional if no properties are to exist beneath it. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 188 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 83 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 539 ++++++++++++++++++ 7 files changed, 933 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

2 6

[PATCH 0/8] ALSA: firewire-lib: check cycle continuity
by Takashi Sakamoto 19 May '21

19 May '21

Hi, Current implementation of ALSA IEC 61883-1/6 packet streaming engine doesn't check whether received packets are exactly per isochronous cycle. This is required to process packets transferred from OXFW970-based devices and devices in RME Fireface series. However, the packet sequence with skipped cycle is inconvenient for media clock recovery. This patchset takes the engine to check cycle continuity at processing packets, including code refactoring. For RME Fireface series, the skipped cycle is handled as receiving an empty packet. For OXFW970-based devices, the skipped cycles are acceptable but media clock recovery is hard. Takashi Sakamoto (8): ALSA: firewire-lib: code refactoring to refer the same frame count per period in domain structure ALSA: firewire-lib: handle the case that empty isochronous packet payload for CIP ALSA: firewire-lib: code refactoring for sequence descriptor' ALSA: firewire-lib: code refactoring for helper function to compute OHCI 1394 cycle ALSA: firewire-lib: code refactoring for parser of IR context header ALSA: firewire-lib: code refactoring for check of CIP header about payload size ALSA: firewire-lib: check cycle continuity ALSA: firewire-lib: insert descriptor for skipped cycle sound/firewire/amdtp-stream.c | 172 ++++++++++++++++++++++------------ sound/firewire/amdtp-stream.h | 10 +- 2 files changed, 119 insertions(+), 63 deletions(-) -- 2.27.0

2 9

[PATCH][next] ALSA: rawmidi: fix incorrect array bounds check on clock_names
by Colin King 19 May '21

19 May '21

From: Colin Ian King <colin.king(a)canonical.com> The array bounds check on clock_names is currently checking the size of the entire array rather than the number of elements in the array leading to a potential array bounds read error. Fix this by using the ARRAY_SIZE macro instead of sizeof. Addresses-Coverity: ("Out-of-bounds read") Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/rawmidi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 4a6534db77d6..6c0a4a67ad2e 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1679,7 +1679,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, buffer_size, avail, xruns); if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; - if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + if (!snd_BUG_ON(clock_type >= ARRAY_SIZE(clock_names))) snd_iprintf(buffer, " Framing : tstamp\n" " Clock type : %s\n", -- 2.31.1

2 1

[PATCH 0/2] mfd: wcd934x: add mbhc interrupts
by Srinivas Kandagatla 19 May '21

19 May '21

This patchset adds interrupts required to enable MBHC (Mulit Button Headset Control) support for Jack and headset button detection in WCD938x Codec. Codec side patches are sent in a separate patch set. Tested it on Lenovo Yoga C630. --srini Srinivas Kandagatla (2): mfd: wcd934x: add macro for adding new interrupts mfd: wcd934x: add additional interrupts required for MBHC support drivers/mfd/wcd934x.c | 50 +++++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 26 deletions(-) -- 2.21.0

2 4

[bug report] ASoC: topology: Add support for multiple kcontrol types to a widget
by Dan Carpenter 19 May '21

19 May '21

Hello Jaska Uimonen, The patch d29d41e28eea: "ASoC: topology: Add support for multiple kcontrol types to a widget" from May 7, 2021, leads to the following static checker warning: sound/soc/soc-topology.c:1559 soc_tplg_dapm_widget_create() error: uninitialized symbol 'kcontrol_type'. sound/soc/soc-topology.c 1418 static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, 1419 struct snd_soc_tplg_dapm_widget *w) 1420 { 1421 struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; 1422 struct snd_soc_dapm_widget template, *widget; 1423 struct snd_soc_tplg_ctl_hdr *control_hdr; 1424 struct snd_soc_card *card = tplg->comp->card; 1425 unsigned int *kcontrol_type; 1426 struct snd_kcontrol_new *kc; 1427 int mixer_count = 0; 1428 int bytes_count = 0; 1429 int enum_count = 0; 1430 int ret = 0; 1431 int i; 1432 1433 if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1434 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1435 return -EINVAL; 1436 if (strnlen(w->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1437 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1438 return -EINVAL; 1439 1440 dev_dbg(tplg->dev, "ASoC: creating DAPM widget %s id %d\n", 1441 w->name, w->id); 1442 1443 memset(&template, 0, sizeof(template)); 1444 1445 /* map user to kernel widget ID */ 1446 template.id = get_widget_id(le32_to_cpu(w->id)); 1447 if ((int)template.id < 0) 1448 return template.id; 1449 1450 /* strings are allocated here, but used and freed by the widget */ 1451 template.name = kstrdup(w->name, GFP_KERNEL); 1452 if (!template.name) 1453 return -ENOMEM; 1454 template.sname = kstrdup(w->sname, GFP_KERNEL); 1455 if (!template.sname) { 1456 ret = -ENOMEM; 1457 goto err; 1458 } 1459 template.reg = le32_to_cpu(w->reg); 1460 template.shift = le32_to_cpu(w->shift); 1461 template.mask = le32_to_cpu(w->mask); 1462 template.subseq = le32_to_cpu(w->subseq); 1463 template.on_val = w->invert ? 0 : 1; 1464 template.off_val = w->invert ? 1 : 0; 1465 template.ignore_suspend = le32_to_cpu(w->ignore_suspend); 1466 template.event_flags = le16_to_cpu(w->event_flags); 1467 template.dobj.index = tplg->index; 1468 1469 tplg->pos += 1470 (sizeof(struct snd_soc_tplg_dapm_widget) + 1471 le32_to_cpu(w->priv.size)); 1472 1473 if (w->num_kcontrols == 0) { 1474 template.num_kcontrols = 0; 1475 goto widget; This patch changed kcontrol_type to a pointer. Originally if we hit this goto kcontrol_type was zero but now it's uninitialized. This will cause a runtime error with UBSan even if it isn't a real bug. 1476 } 1477 1478 control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; 1479 dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", 1480 w->name, w->num_kcontrols, control_hdr->type); 1481 1482 template.num_kcontrols = le32_to_cpu(w->num_kcontrols); 1483 kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); 1484 if (!kc) 1485 goto err; 1486 1487 kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), 1488 GFP_KERNEL); 1489 if (!kcontrol_type) 1490 goto err; 1491 1492 for (i = 0; i < w->num_kcontrols; i++) { 1493 control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; 1494 switch (le32_to_cpu(control_hdr->ops.info)) { 1495 case SND_SOC_TPLG_CTL_VOLSW: 1496 case SND_SOC_TPLG_CTL_STROBE: 1497 case SND_SOC_TPLG_CTL_VOLSW_SX: 1498 case SND_SOC_TPLG_CTL_VOLSW_XR_SX: 1499 case SND_SOC_TPLG_CTL_RANGE: 1500 case SND_SOC_TPLG_DAPM_CTL_VOLSW: 1501 /* volume mixer */ 1502 kc[i].index = mixer_count; 1503 kcontrol_type[i] = SND_SOC_TPLG_TYPE_MIXER; 1504 mixer_count++; 1505 ret = soc_tplg_dapm_widget_dmixer_create(tplg, &kc[i]); 1506 if (ret < 0) 1507 goto hdr_err; 1508 break; 1509 case SND_SOC_TPLG_CTL_ENUM: 1510 case SND_SOC_TPLG_CTL_ENUM_VALUE: 1511 case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: 1512 case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: 1513 case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: 1514 /* enumerated mixer */ 1515 kc[i].index = enum_count; 1516 kcontrol_type[i] = SND_SOC_TPLG_TYPE_ENUM; 1517 enum_count++; 1518 ret = soc_tplg_dapm_widget_denum_create(tplg, &kc[i]); 1519 if (ret < 0) 1520 goto hdr_err; 1521 break; 1522 case SND_SOC_TPLG_CTL_BYTES: 1523 /* bytes control */ 1524 kc[i].index = bytes_count; 1525 kcontrol_type[i] = SND_SOC_TPLG_TYPE_BYTES; 1526 bytes_count++; 1527 ret = soc_tplg_dapm_widget_dbytes_create(tplg, &kc[i]); 1528 if (ret < 0) 1529 goto hdr_err; 1530 break; 1531 default: 1532 dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", 1533 control_hdr->ops.get, control_hdr->ops.put, 1534 le32_to_cpu(control_hdr->ops.info)); 1535 ret = -EINVAL; 1536 goto hdr_err; 1537 } 1538 } 1539 1540 template.kcontrol_news = kc; 1541 1542 widget: 1543 ret = soc_tplg_widget_load(tplg, &template, w); 1544 if (ret < 0) 1545 goto hdr_err; 1546 1547 /* card dapm mutex is held by the core if we are loading topology 1548 * data during sound card init. */ 1549 if (card->instantiated) 1550 widget = snd_soc_dapm_new_control(dapm, &template); 1551 else 1552 widget = snd_soc_dapm_new_control_unlocked(dapm, &template); 1553 if (IS_ERR(widget)) { 1554 ret = PTR_ERR(widget); 1555 goto hdr_err; 1556 } 1557 1558 widget->dobj.type = SND_SOC_DOBJ_WIDGET; 1559 widget->dobj.widget.kcontrol_type = kcontrol_type; ^^^^^^^^^^^^^^ 1560 widget->dobj.ops = tplg->ops; 1561 widget->dobj.index = tplg->index; 1562 list_add(&widget->dobj.list, &tplg->comp->dobj_list); 1563 1564 ret = soc_tplg_widget_ready(tplg, widget, w); 1565 if (ret < 0) 1566 goto ready_err; regards, dan carpenter

1 0

[PATCH 0/7] ALSA: Prep work for PCI rescan support
by Takashi Iwai 18 May '21

18 May '21

Hi, this is a patch set as a prep work for the future-planned PCI rescan support(*). The essential part is the patch to track in-flight tasks, and the rest are cleanups and fixes in the core code. With those changes, the driver can perform a sort of software suspend on the device without the hardware intervention, which is required during the PCI BAR movement, for example. Takashi (*) https://lore.kernel.org/alsa-devel/e25017c6-e5e4-7a24-e793-14a2e70a434e@amd… === Takashi Iwai (7): ALSA: core: Use READ_ONCE() / WRITE_ONCE() for power state change ALSA: control: Add power state check commonly for ioctl handlers ALSA: control: Drop superfluous snd_power_wait() calls ALSA: control: Minor optimization for SNDRV_CTL_IOCTL_POWER_STATE ALSA: control: Track in-flight control read/write/tlv accesses ALSA: pcm: Block the release until the system resume finishes ALSA: Drop superfluous argument from snd_power_wait() include/sound/core.h | 39 +++++++++++++++++++++--- sound/core/control.c | 41 +++++++++++++------------- sound/core/control_compat.c | 17 +++++------ sound/core/init.c | 59 ++++++++++++++++++++++++++++++++----- sound/core/pcm_native.c | 6 +++- sound/soc/soc-core.c | 2 +- sound/usb/usx2y/us122l.c | 2 +- 7 files changed, 121 insertions(+), 45 deletions(-) -- 2.26.2

2 11

[PATCH -next] ASoC: hisilicon: fix missing clk_disable_unprepare() on error in hi6210_i2s_startup()
by Yang Yingliang 18 May '21

18 May '21

After calling clk_prepare_enable(), clk_disable_unprepare() need be called when calling clk_set_rate() failed. Fixes: 0bf750f4cbe1 ("ASoC: hisilicon: Add hi6210 i2s audio driver") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/hisilicon/hi6210-i2s.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index 907f5f1f7b44..ff05b9779e4b 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -102,18 +102,15 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, for (n = 0; n < i2s->clocks; n++) { ret = clk_prepare_enable(i2s->clk[n]); - if (ret) { - while (n--) - clk_disable_unprepare(i2s->clk[n]); - return ret; - } + if (ret) + goto err_unprepare_clk; } ret = clk_set_rate(i2s->clk[CLK_I2S_BASE], 49152000); if (ret) { dev_err(i2s->dev, "%s: setting 49.152MHz base rate failed %d\n", __func__, ret); - return ret; + goto err_unprepare_clk; } /* enable clock before frequency division */ @@ -165,6 +162,11 @@ static int hi6210_i2s_startup(struct snd_pcm_substream *substream, hi6210_write_reg(i2s, HII2S_SW_RST_N, val); return 0; + +err_unprepare_clk: + while (n--) + clk_disable_unprepare(i2s->clk[n]); + return ret; } static void hi6210_i2s_shutdown(struct snd_pcm_substream *substream, -- 2.25.1

2 1

[PATCH] ASoC: fsl: fix SND_SOC_IMX_RPMSG dependency
by Arnd Bergmann 18 May '21

18 May '21

From: Arnd Bergmann <arnd(a)arndb.de> Kconfig produces a warning with SND_SOC_FSL_RPMSG=y and SND_IMX_SOC=m: WARNING: unmet direct dependencies detected for SND_SOC_IMX_RPMSG Depends on [m]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_IMX_SOC [=m] && RPMSG [=y] Selected by [y]: - SND_SOC_FSL_RPMSG [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && COMMON_CLK [=y] && RPMSG [=y] && SND_IMX_SOC [=m]!=n Add a dependency to prevent this configuration. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0917d65d6921..556c284f49dd 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -119,6 +119,7 @@ config SND_SOC_FSL_RPMSG tristate "NXP Audio Base On RPMSG support" depends on COMMON_CLK depends on RPMSG + depends on SND_IMX_SOC || SND_IMX_SOC = n select SND_SOC_IMX_RPMSG if SND_IMX_SOC != n help Say Y if you want to add rpmsg audio support for the Freescale CPUs. -- 2.29.2

3 2

[PATCH -next] ASoC: soc-core: use DEVICE_ATTR_RO macro
by YueHaibing 18 May '21

18 May '21

Use DEVICE_ATTR_RO helper instead of plain DEVICE_ATTR, which makes the code a bit shorter and easier to read. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1c0904acb935..2d969e31f37d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -75,9 +75,9 @@ static ssize_t pmdown_time_show(struct device *dev, return sprintf(buf, "%ld\n", rtd->pmdown_time); } -static ssize_t pmdown_time_set(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) +static ssize_t pmdown_time_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) { struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; @@ -89,7 +89,7 @@ static ssize_t pmdown_time_set(struct device *dev, return count; } -static DEVICE_ATTR(pmdown_time, 0644, pmdown_time_show, pmdown_time_set); +static DEVICE_ATTR_RW(pmdown_time); static struct attribute *soc_dev_attrs[] = { &dev_attr_pmdown_time.attr, -- 2.17.1

2 3

[PATCH -next] ASoC: rk3328: fix missing clk_disable_unprepare() on error in rk3328_platform_probe()
by Yang Yingliang 18 May '21

18 May '21

Fix the missing clk_disable_unprepare() before return from rk3328_platform_probe() in the error handling case. Fixes: c32759035ad2 ("ASoC: rockchip: support ACODEC for rk3328") Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/rk3328_codec.c | 28 ++++++++++++++++++++++------ 1 file changed, 22 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rk3328_codec.c b/sound/soc/codecs/rk3328_codec.c index bfefefcc76d8..758d439e8c7a 100644 --- a/sound/soc/codecs/rk3328_codec.c +++ b/sound/soc/codecs/rk3328_codec.c @@ -474,7 +474,8 @@ static int rk3328_platform_probe(struct platform_device *pdev) rk3328->pclk = devm_clk_get(&pdev->dev, "pclk"); if (IS_ERR(rk3328->pclk)) { dev_err(&pdev->dev, "can't get acodec pclk\n"); - return PTR_ERR(rk3328->pclk); + ret = PTR_ERR(rk3328->pclk); + goto err_unprepare_mclk; } ret = clk_prepare_enable(rk3328->pclk); @@ -484,19 +485,34 @@ static int rk3328_platform_probe(struct platform_device *pdev) } base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err_unprepare_pclk; + } rk3328->regmap = devm_regmap_init_mmio(&pdev->dev, base, &rk3328_codec_regmap_config); - if (IS_ERR(rk3328->regmap)) - return PTR_ERR(rk3328->regmap); + if (IS_ERR(rk3328->regmap)) { + ret = PTR_ERR(rk3328->regmap); + goto err_unprepare_pclk; + } platform_set_drvdata(pdev, rk3328); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_rk3328, rk3328_dai, ARRAY_SIZE(rk3328_dai)); + if (ret) + goto err_unprepare_pclk; + + return 0; + +err_unprepare_pclk: + clk_disable_unprepare(rk3328->pclk); + +err_unprepare_mclk: + clk_disable_unprepare(rk3328->mclk); + return ret; } static const struct of_device_id rk3328_codec_of_match[] __maybe_unused = { -- 2.25.1

2 1

[PATCH alsa-ucm-conf 0/3] byt/cht: Add support for controlling mute LEDs using the new snd_ctl_led kmod
by Hans de Goede 18 May '21

18 May '21

Hi All, This series adds support for controlling spk/mic mute LEDs one some Bay- and Cherry-Trail devices using the new snd_ctl_led kmod functionality. This requires the latest alsa-lib and alsa-utils git code and when running on Fedora it also requires putting selinux in permissive mode for now. I've filed a bug against the selinux-policy to get this fixed: https://bugzilla.redhat.com/show_bug.cgi?id=1958210 Note the first patch is mostly unrelated, it is a resend of an earlier patch which seems to have fallen through the cracks. Regards, Hans Hans de Goede (3): cht-bsw-rt5672: Add support for the components string cht-bsw-rt5672: Add support for controlling speaker- and mic-mute LEDs bytcr-rt5640: Add support for controlling a speaker-mute LED ucm2/bytcr-rt5640/bytcr-rt5640.conf | 6 ++++ ucm2/cht-bsw-rt5672/HiFi.conf | 37 +++++++++++++++++++++++-- ucm2/cht-bsw-rt5672/cht-bsw-rt5672.conf | 6 ++++ 3 files changed, 47 insertions(+), 2 deletions(-) -- 2.31.1

2 4

[PATCH 0/2] ALSA: usb-audio: clock setup code refactoring
by Takashi Iwai 18 May '21

18 May '21

Hi, this is a couple of patches for refactoring and improvements on USB-audio clock setup code. By replacing with unions, we can handle both UAC2 and UAC3 equivalently. Takashi === Takashi Iwai (2): ALSA: usb-audio: Refactoring UAC2/3 clock setup code ALSA: usb-audio: Handle error for the current selector gracefully sound/usb/clock.c | 298 +++++++++++++++------------------------------- 1 file changed, 95 insertions(+), 203 deletions(-) -- 2.26.2

1 2

[PATCH v7 1/2] platform/x86: dell-privacy: Add support for Dell hardware privacy
by Perry Yuan 18 May '21

18 May '21

From: Perry Yuan <perry_yuan(a)dell.com> add support for Dell privacy driver for the Dell units equipped hardware privacy design, which protect users privacy of audio and camera from hardware level. Once the audio or camera privacy mode activated, any applications will not get any audio or video stream when user pressed ctrl+F4 hotkey, audio privacy mode will be enabled, micmute led will be also changed accordingly The micmute led is fully controlled by hardware & EC(embedded controller) and camera mute hotkey is Ctrl+F9. Currently design only emits SW_CAMERA_LENS_COVER event while the camera lens shutter will be changed by EC & HW(hardware) control *The flow is like this: 1) User presses key. HW does stuff with this key (timeout timer is started) 2) WMI event is emitted from BIOS to kernel 3) WMI event is received by dell-privacy 4) KEY_MICMUTE emitted from dell-privacy 5) Userland picks up key and modifies kcontrol for SW mute 6) Codec kernel driver catches and calls ledtrig_audio_set, like this: ledtrig_audio_set(LED_AUDIO_MICMUTE, rt715->micmute_led ? LED_ON :LED_OFF); 7) If "LED" is set to on dell-privacy notifies EC, and timeout is cancelled, HW mic mute activated. If EC not notified, HW mic mute will also be activated when timeout used up, it is just later than active ack Signed-off-by: Perry Yuan <perry_yuan(a)dell.com> --- v5 -> v6: * addressed feedback from Hans * addressed feedback from Pierre * optimize some debug format with dev_dbg() * remove platform driver,combined privacy acpi driver into single wmi driver file * optimize sysfs interface with string added to be more clearly reading * remove unused function and clear header file v4 -> v5: * addressed feedback from Randy Dunlap * addressed feedback from Pierre-Louis Bossart * rebase to latest 5.12 rc4 upstream kernel * fix some space alignment problem v3 -> v4: * fix format for Kconfig * add sysfs document * add flow comments to the privacy wmi/acpi driver * addressed feedback from Barnabás Pőcze[Thanks very much] * export privacy_valid to make the global state simpler to query * fix one issue which will block the dell-laptop driver to load when privacy driver invalid * addressed feedback from Pierre-Louis Bossart,remove the EC ID match v2 -> v3: * add sysfs attributes doc v1 -> v2: * query EC handle from EC driver directly. * fix some code style. * add KEY_END to keymap array. * clean platform device when cleanup called * use hexadecimal format for log print in dev_dbg * remove __set_bit for the report keys from probe. * fix keymap leak * add err_free_keymap in dell_privacy_wmi_probe * wmi driver will be unregistered if privacy_acpi_init() fails * add sysfs attribute files for user space query. * add leds micmute driver to privacy acpi * add more design info the commit info --- --- .../testing/sysfs-platform-dell-privacy-wmi | 55 +++ drivers/platform/x86/dell/Kconfig | 14 + drivers/platform/x86/dell/Makefile | 1 + drivers/platform/x86/dell/dell-laptop.c | 23 +- drivers/platform/x86/dell/dell-privacy-wmi.c | 394 ++++++++++++++++++ drivers/platform/x86/dell/dell-privacy-wmi.h | 23 + drivers/platform/x86/dell/dell-wmi.c | 8 +- 7 files changed, 511 insertions(+), 7 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.c create mode 100644 drivers/platform/x86/dell/dell-privacy-wmi.h diff --git a/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi new file mode 100644 index 000000000000..7f9e18705861 --- /dev/null +++ b/Documentation/ABI/testing/sysfs-platform-dell-privacy-wmi @@ -0,0 +1,55 @@ +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Display which dell hardware level privacy devices are supported + “Dell Privacy” is a set of HW, FW, and SW features to enhance + Dell’s commitment to platform privacy for MIC, Camera, and + ePrivacy screens. + The supported hardware privacy devices are: +Attributes: + Microphone Mute: + Identifies the local microphone can be muted by hardware, no applications + is available to capture system mic sound + + Camera Shutter: + Identifies camera shutter controlled by hardware, which is a micromechanical + shutter assembly that is built onto the camera module to block capturing images + from outside the laptop + + supported: + The privacy device is supported by this system + + unsupported: + The privacy device is not supported on this system + + For example to check which privacy devices are supported: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_supported_type + [Microphone Mute] [supported] + [Camera Shutter] [supported] + [ePrivacy Screen] [unsupported] + +What: /sys/bus/wmi/devices/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state +Date: Apr 2021 +KernelVersion: 5.13 +Contact: "perry.yuan(a)dell.com>" +Description: + Allow user space to check current dell privacy device state. + Describes the Device State class exposed by BIOS which can be + consumed by various applications interested in knowing the Privacy + feature capabilities +Attributes: + muted: + Identifies the privacy device is turned off and cannot send stream to OS applications + + unmuted: + Identifies the privacy device is turned on ,audio or camera driver can get + stream from mic and camera module to OS applications + + For example to check all supported current privacy device states: + + # cat /sys/bus/wmi/drivers/dell-privacy/6932965F-1671-4CEB-B988-D3AB0A901919/dell_privacy_current_state + [Microphone] [unmuted] + [Camera Shutter] [unmuted] diff --git a/drivers/platform/x86/dell/Kconfig b/drivers/platform/x86/dell/Kconfig index e0a55337f51a..05d124442b25 100644 --- a/drivers/platform/x86/dell/Kconfig +++ b/drivers/platform/x86/dell/Kconfig @@ -204,4 +204,18 @@ config DELL_WMI_SYSMAN To compile this driver as a module, choose M here: the module will be called dell-wmi-sysman. +config DELL_PRIVACY + tristate "Dell Hardware Privacy Support" + depends on ACPI + depends on ACPI_WMI + depends on INPUT + depends on DELL_LAPTOP + depends on LEDS_TRIGGER_AUDIO + select DELL_WMI + help + This driver provides integration with the "Dell Hardware Privacy" feature + of Dell laptops. + + To compile this driver as a module, choose M here: the module will + be called dell_privacy. endif # X86_PLATFORM_DRIVERS_DELL diff --git a/drivers/platform/x86/dell/Makefile b/drivers/platform/x86/dell/Makefile index d720a3e42ae3..7da0c33dfcca 100644 --- a/drivers/platform/x86/dell/Makefile +++ b/drivers/platform/x86/dell/Makefile @@ -19,3 +19,4 @@ obj-$(CONFIG_DELL_WMI_AIO) += dell-wmi-aio.o obj-$(CONFIG_DELL_WMI_DESCRIPTOR) += dell-wmi-descriptor.o obj-$(CONFIG_DELL_WMI_LED) += dell-wmi-led.o obj-$(CONFIG_DELL_WMI_SYSMAN) += dell-wmi-sysman/ +obj-$(CONFIG_DELL_PRIVACY) += dell-privacy-wmi.o diff --git a/drivers/platform/x86/dell/dell-laptop.c b/drivers/platform/x86/dell/dell-laptop.c index 70edc5bb3a14..e7ffc0b81208 100644 --- a/drivers/platform/x86/dell/dell-laptop.c +++ b/drivers/platform/x86/dell/dell-laptop.c @@ -31,6 +31,8 @@ #include "dell-rbtn.h" #include "dell-smbios.h" +#include "dell-privacy-wmi.h" + struct quirk_entry { bool touchpad_led; bool kbd_led_not_present; @@ -90,6 +92,7 @@ static struct rfkill *wifi_rfkill; static struct rfkill *bluetooth_rfkill; static struct rfkill *wwan_rfkill; static bool force_rfkill; +static bool has_privacy; module_param(force_rfkill, bool, 0444); MODULE_PARM_DESC(force_rfkill, "enable rfkill on non whitelisted models"); @@ -2206,10 +2209,16 @@ static int __init dell_init(void) if (dell_smbios_find_token(GLOBAL_MIC_MUTE_DISABLE) && dell_smbios_find_token(GLOBAL_MIC_MUTE_ENABLE)) { - micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); - ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); - if (ret < 0) - goto fail_led; + if (dell_privacy_present()) + has_privacy = true; + else + has_privacy = false; + if (!has_privacy) { + micmute_led_cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + ret = led_classdev_register(&platform_device->dev, &micmute_led_cdev); + if (ret < 0) + goto fail_led; + } } if (acpi_video_get_backlight_type() != acpi_backlight_vendor) @@ -2257,7 +2266,8 @@ static int __init dell_init(void) fail_get_brightness: backlight_device_unregister(dell_backlight_device); fail_backlight: - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); fail_led: dell_cleanup_rfkill(); fail_rfkill: @@ -2278,7 +2288,8 @@ static void __exit dell_exit(void) touchpad_led_exit(); kbd_led_exit(); backlight_device_unregister(dell_backlight_device); - led_classdev_unregister(&micmute_led_cdev); + if (!has_privacy) + led_classdev_unregister(&micmute_led_cdev); dell_cleanup_rfkill(); if (platform_device) { platform_device_unregister(platform_device); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.c b/drivers/platform/x86/dell/dell-privacy-wmi.c new file mode 100644 index 000000000000..4be7548d1a03 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.c @@ -0,0 +1,394 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ": " fmt + +#include <linux/acpi.h> +#include <linux/bitops.h> +#include <linux/input.h> +#include <linux/input/sparse-keymap.h> +#include <linux/list.h> +#include <linux/leds.h> +#include <linux/module.h> +#include <linux/wmi.h> + +#include "dell-privacy-wmi.h" + +#define DELL_PRIVACY_GUID "6932965F-1671-4CEB-B988-D3AB0A901919" +#define MICROPHONE_STATUS BIT(0) +#define CAMERA_STATUS BIT(1) +#define DELL_PRIVACY_AUDIO_EVENT 0x1 +#define DELL_PRIVACY_CAMERA_EVENT 0x2 +#define led_to_priv(c) container_of(c, struct privacy_wmi_data, cdev) + +/* + * The wmi_list is used to store the privacy_priv struct with mutex protecting + */ +static LIST_HEAD(wmi_list); +static DEFINE_MUTEX(list_mutex); + +struct privacy_wmi_data { + struct input_dev *input_dev; + struct wmi_device *wdev; + struct list_head list; + struct led_classdev cdev; + u32 features_present; + u32 last_status; +}; + +/* DELL Privacy Type */ +enum dell_hardware_privacy_type { + DELL_PRIVACY_TYPE_AUDIO = 0, + DELL_PRIVACY_TYPE_CAMERA, + DELL_PRIVACY_TYPE_SCREEN, + DELL_PRIVACY_TYPE_MAX, +}; + +static const char * const privacy_types[DELL_PRIVACY_TYPE_MAX] = { + [DELL_PRIVACY_TYPE_AUDIO] = "Microphone", + [DELL_PRIVACY_TYPE_CAMERA] = "Camera Shutter", + [DELL_PRIVACY_TYPE_SCREEN] = "ePrivacy Screen", +}; + +/* + * Keymap for WMI privacy events of type 0x0012 + */ +static const struct key_entry dell_wmi_keymap_type_0012[] = { + /* privacy mic mute */ + { KE_KEY, 0x0001, { KEY_MICMUTE } }, + /* privacy camera mute */ + { KE_SW, 0x0002, { SW_CAMERA_LENS_COVER } }, + { KE_END, 0}, +}; + +/* + * global privacy state for other modules to query if the privacy driver loaded successfully + * or no privacy WMI device was detected at initial probe phase + */ +bool dell_privacy_present(void) +{ + struct privacy_wmi_data *priv; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + mutex_unlock(&list_mutex); + + return priv; +} +EXPORT_SYMBOL_GPL(dell_privacy_present); + +/* + * The flow of privacy event: + * 1) User presses key. HW does stuff with this key (timeout is started) + * 2) WMI event is emitted from BIOS + * 3) WMI event is received by dell-privacy + * 4) KEY_MICMUTE emitted from dell-privacy + * 5) Userland picks up key and modifies kcontrol for SW mute + * 6) Codec kernel driver catches and calls ledtrig_audio_set defined by + * dell-privacy-acpi driver. Codec driver will call like this to switch micmute led state. + * ledtrig_audio_set(LED_AUDIO_MICMUTE, micmute_led ? LED_ON :LED_OFF); + * 7) If "LED" is set to on dell-privacy notifies EC,and timeout is cancelled, + * HW mic mute activated. + */ +bool dell_privacy_process_event(int type, int code, int status) +{ + struct privacy_wmi_data *priv; + const struct key_entry *key; + bool ret = false; + + mutex_lock(&list_mutex); + priv = list_first_entry_or_null(&wmi_list, + struct privacy_wmi_data, + list); + if (!priv) { + goto error; + } + + key = sparse_keymap_entry_from_scancode(priv->input_dev, (type << 16) | code); + if (!key) { + dev_warn(&priv->wdev->dev, "Unknown key with type 0x%04x and code 0x%04x pressed\n", + type, code); + goto error; + } + dev_dbg(&priv->wdev->dev, "Key with type 0x%04x and code 0x%04x pressed\n", type, code); + + switch (code) { + case DELL_PRIVACY_AUDIO_EVENT: /* Mic mute */ + case DELL_PRIVACY_CAMERA_EVENT: /* Camera mute */ + priv->last_status = status; + sparse_keymap_report_entry(priv->input_dev, key, 1, true); + ret = true; + break; + default: + dev_dbg(&priv->wdev->dev, "unknown event type 0x%04x 0x%04x\n", type, code); + } + +error: + mutex_unlock(&list_mutex); + return ret; +} +EXPORT_SYMBOL_GPL(dell_privacy_process_event); + +static ssize_t dell_privacy_supported_type_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + enum dell_hardware_privacy_type type; + u32 privacy_list; + char *s = buf; + + privacy_list = priv->features_present; + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_list & BIT(type)) + s += sprintf(s, "[%s] [supported]\n", privacy_types[type]); + else + s += sprintf(s, "[%s] [unsupported]\n", privacy_types[type]); + } + + return (s - buf); +} + +static ssize_t dell_privacy_current_state_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + u32 privacy_supported = priv->features_present; + enum dell_hardware_privacy_type type; + u32 privacy_state = priv->last_status; + char *s = buf; + + for (type = DELL_PRIVACY_TYPE_AUDIO; type < DELL_PRIVACY_TYPE_MAX; type++) { + if (privacy_supported & BIT(type)) { + if (privacy_state & BIT(type)) + s += sprintf(s, "[%s] [unmuted]\n", privacy_types[type]); + else + s += sprintf(s, "[%s] [muted]\n", privacy_types[type]); + } + } + + return (s - buf); +} + +static DEVICE_ATTR_RO(dell_privacy_supported_type); +static DEVICE_ATTR_RO(dell_privacy_current_state); + +static struct attribute *privacy_attributes[] = { + &dev_attr_dell_privacy_supported_type.attr, + &dev_attr_dell_privacy_current_state.attr, + NULL, +}; + +static const struct attribute_group privacy_attribute_group = { + .attrs = privacy_attributes +}; + +/* + * Describes the Device State class exposed by BIOS which can be consumed by + * various applications interested in knowing the Privacy feature capabilities. + * class DeviceState + * { + * [key, read] string InstanceName; + * [read] boolean ReadOnly; + * + * [WmiDataId(1), read] uint32 DevicesSupported; + * 0 - None; 0x1 - Microphone; 0x2 - Camera; 0x4 - ePrivacy Screen + * + * [WmiDataId(2), read] uint32 CurrentState; + * 0 - Off; 1 - On; Bit0 - Microphone; Bit1 - Camera; Bit2 - ePrivacyScreen + * }; + */ +static int get_current_status(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + union acpi_object *obj_present; + u32 *buffer; + int ret = 0; + + if (!priv) { + dev_err(&wdev->dev, "dell privacy priv is NULL\n"); + return -EINVAL; + } + /* check privacy support features and device states */ + obj_present = wmidev_block_query(wdev, 0); + if (!obj_present) { + dev_err(&wdev->dev, "failed to read Binary MOF\n"); + return -EIO; + } + + if (obj_present->type != ACPI_TYPE_BUFFER) { + dev_err(&wdev->dev, "Binary MOF is not a buffer!\n"); + ret = -EIO; + goto obj_free; + } + /* Although it's not technically a failure, this would lead to + * unexpected behavior + */ + if (obj_present->buffer.length != 8) { + dev_err(&wdev->dev, "Dell privacy buffer has unexpected length (%d)!\n", + obj_present->buffer.length); + ret = -EINVAL; + goto obj_free; + } + buffer = (u32 *)obj_present->buffer.pointer; + priv->features_present = buffer[0]; + priv->last_status = buffer[1]; + +obj_free: + kfree(obj_present); + return ret; +} + +static int dell_privacy_micmute_led_set(struct led_classdev *led_cdev, + enum led_brightness brightness) +{ + struct privacy_wmi_data *priv = led_to_priv(led_cdev); + static char *acpi_method = (char *)"ECAK"; + acpi_status status; + acpi_handle handle; + + handle = ec_get_handle(); + if (!handle) + return -EIO; + + if (!acpi_has_method(handle, acpi_method)) + return -EIO; + + status = acpi_evaluate_object(handle, acpi_method, NULL, NULL); + if (ACPI_FAILURE(status)) { + dev_err(&priv->wdev->dev, "Error setting privacy EC ack value: %s\n", + acpi_format_exception(status)); + return -EIO; + } + + return 0; +} + +/* + * Pressing the mute key activates a time delayed circuit to physically cut + * off the mute. The LED is in the same circuit, so it reflects the true + * state of the HW mute. The reason for the EC "ack" is so that software + * can first invoke a SW mute before the HW circuit is cut off. Without SW + * cutting this off first does not affect the time delayed muting or status + * of the LED but there is a possibility of a "popping" noise. + * + * If the EC receives the SW ack, the circuit will be activated before the + * delay completed. + * + * Exposing as an LED device allows the codec drivers notification path to + * EC ACK to work + */ +static int dell_privacy_leds_setup(struct device *dev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(dev); + + priv->cdev.name = "dell-privacy::micmute"; + priv->cdev.max_brightness = 1; + priv->cdev.brightness_set_blocking = dell_privacy_micmute_led_set; + priv->cdev.default_trigger = "audio-micmute"; + priv->cdev.brightness = ledtrig_audio_get(LED_AUDIO_MICMUTE); + return devm_led_classdev_register(dev, &priv->cdev); +} + +static int dell_privacy_wmi_probe(struct wmi_device *wdev, const void *context) +{ + struct privacy_wmi_data *priv; + struct key_entry *keymap; + int ret, i; + + ret = wmi_has_guid(DELL_PRIVACY_GUID); + if (!ret) + pr_debug("Unable to detect available Dell privacy devices!\n"); + + priv = devm_kzalloc(&wdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + dev_set_drvdata(&wdev->dev, priv); + priv->wdev = wdev; + /* create evdev passing interface */ + priv->input_dev = devm_input_allocate_device(&wdev->dev); + if (!priv->input_dev) + return -ENOMEM; + + /* remap the wmi keymap event to new keymap */ + keymap = kcalloc(ARRAY_SIZE(dell_wmi_keymap_type_0012), + sizeof(struct key_entry), GFP_KERNEL); + if (!keymap) + return -ENOMEM; + + /* remap the keymap code with Dell privacy key type 0x12 as prefix + * KEY_MICMUTE scancode will be reported as 0x120001 + */ + for (i = 0; i < ARRAY_SIZE(dell_wmi_keymap_type_0012); i++) { + keymap[i] = dell_wmi_keymap_type_0012[i]; + keymap[i].code |= (0x0012 << 16); + } + ret = sparse_keymap_setup(priv->input_dev, keymap, NULL); + kfree(keymap); + if (ret) + return ret; + + priv->input_dev->dev.parent = &wdev->dev; + priv->input_dev->name = "Dell Privacy Driver"; + priv->input_dev->id.bustype = BUS_HOST; + + ret = input_register_device(priv->input_dev); + if (ret) + return ret; + + ret = get_current_status(priv->wdev); + if (ret) + return ret; + + ret = devm_device_add_group(&wdev->dev, &privacy_attribute_group); + if (ret) + return ret; + + ret = dell_privacy_leds_setup(&priv->wdev->dev); + if (ret) + return ret; + + mutex_lock(&list_mutex); + list_add_tail(&priv->list, &wmi_list); + mutex_unlock(&list_mutex); + return 0; +} + +static int dell_privacy_wmi_remove(struct wmi_device *wdev) +{ + struct privacy_wmi_data *priv = dev_get_drvdata(&wdev->dev); + + mutex_lock(&list_mutex); + list_del(&priv->list); + mutex_unlock(&list_mutex); + return 0; +} + +static const struct wmi_device_id dell_wmi_privacy_wmi_id_table[] = { + { .guid_string = DELL_PRIVACY_GUID }, + { }, +}; + +static struct wmi_driver dell_privacy_wmi_driver = { + .driver = { + .name = "dell-privacy", + }, + .probe = dell_privacy_wmi_probe, + .remove = dell_privacy_wmi_remove, + .id_table = dell_wmi_privacy_wmi_id_table, +}; + +module_wmi_driver(dell_privacy_wmi_driver); + +MODULE_DEVICE_TABLE(wmi, dell_wmi_privacy_wmi_id_table); +MODULE_AUTHOR("Perry Yuan <perry_yuan(a)dell.com>"); +MODULE_DESCRIPTION("Dell Privacy WMI Driver"); +MODULE_LICENSE("GPL"); diff --git a/drivers/platform/x86/dell/dell-privacy-wmi.h b/drivers/platform/x86/dell/dell-privacy-wmi.h new file mode 100644 index 000000000000..04c49d2fa924 --- /dev/null +++ b/drivers/platform/x86/dell/dell-privacy-wmi.h @@ -0,0 +1,23 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Dell privacy notification driver + * + * Copyright (C) 2021 Dell Inc. All Rights Reserved. + */ + +#ifndef _DELL_PRIVACY_WMI_H_ +#define _DELL_PRIVACY_WMI_H_ + +#if IS_ENABLED(CONFIG_DELL_PRIVACY) +bool dell_privacy_present(void); +bool dell_privacy_process_event(int type, int code, int status); +#else /* CONFIG_DELL_PRIVACY */ +static inline bool dell_privacy_present(void) +{ + return -ENODEV; +} + +static inline bool dell_privacy_process_event(int type, int code, int status) +{} +#endif /* CONFIG_DELL_PRIVACY */ +#endif diff --git a/drivers/platform/x86/dell/dell-wmi.c b/drivers/platform/x86/dell/dell-wmi.c index bbdb3e860892..228e85a427f5 100644 --- a/drivers/platform/x86/dell/dell-wmi.c +++ b/drivers/platform/x86/dell/dell-wmi.c @@ -27,6 +27,7 @@ #include <acpi/video.h> #include "dell-smbios.h" #include "dell-wmi-descriptor.h" +#include "dell-privacy-wmi.h" MODULE_AUTHOR("Matthew Garrett <mjg(a)redhat.com>"); MODULE_AUTHOR("Pali Rohár <pali(a)kernel.org>"); @@ -427,7 +428,6 @@ static void dell_wmi_notify(struct wmi_device *wdev, switch (buffer_entry[1]) { case 0x0000: /* One key pressed or event occurred */ - case 0x0012: /* Event with extended data occurred */ if (len > 2) dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); @@ -439,6 +439,12 @@ static void dell_wmi_notify(struct wmi_device *wdev, dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[i]); break; + case 0x0012: + if ((len > 4) && dell_privacy_process_event(buffer_entry[1], buffer_entry[3], buffer_entry[4])) + /* privacy hotkey handle */; + else if (len > 2) + dell_wmi_process_key(wdev, buffer_entry[1], buffer_entry[2]); + break; default: /* Unknown event */ pr_info("Unknown WMI event type 0x%x\n", (int)buffer_entry[1]); -- 2.25.1

4 5

[PATCH v4 00/11] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 18 May '21

18 May '21

Hi, This patchset is revised version of my take 3: * https://lore.kernel.org/alsa-devel/20210518024326.67576-1-o-takashi@sakamoc… Changes from v3: * use unsigned int type instead of int type for storage of combination of enumeration-constants Takashi Sakamoto (11): Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro" ALSA: firewire-lib/motu: use int type for the value of bitwise OR with enumerator-constant ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: code refactoring to detect mackie models ALSA: oxfw: add explicit device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw: code refactoring for wrong_dbs quirk ALSA: oxfw: add quirk flag for blocking transmission method sound/firewire/Kconfig | 4 +- sound/firewire/amdtp-am824.c | 4 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 11 ++- sound/firewire/amdtp-stream.h | 5 +- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-stream.c | 25 +++--- sound/firewire/oxfw/oxfw.c | 137 ++++++++++++++--------------- sound/firewire/oxfw/oxfw.h | 13 ++- 11 files changed, 110 insertions(+), 98 deletions(-) -- 2.27.0

2 12

[PATCH] ALSA: line6: Fix racy initialization of LINE6 MIDI
by Takashi Iwai 18 May '21

18 May '21

The initialization of MIDI devices that are found on some LINE6 drivers are currently done in a racy way; namely, the MIDI buffer instance is allocated and initialized in each private_init callback while the communication with the interface is already started via line6_init_cap_control() call before that point. This may lead to Oops in line6_data_received() when a spurious event is received, as reported by syzkaller. This patch moves the MIDI initialization to line6_init_cap_control() as well instead of the too-lately-called private_init for avoiding the race. Also this reduces slightly more lines, so it's a win-win change. Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com Link: https://lore.kernel.org/r/000000000000a4be9405c28520de@google.com Link: https://lore.kernel.org/r/20210517132725.GA50495@hyeyoo Cc: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/driver.c | 4 ++++ sound/usb/line6/pod.c | 5 ----- sound/usb/line6/variax.c | 6 ------ 3 files changed, 4 insertions(+), 11 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index a030dd65eb28..9602929b7de9 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -699,6 +699,10 @@ static int line6_init_cap_control(struct usb_line6 *line6) line6->buffer_message = kmalloc(LINE6_MIDI_MESSAGE_MAXLEN, GFP_KERNEL); if (!line6->buffer_message) return -ENOMEM; + + ret = line6_init_midi(line6); + if (ret < 0) + return ret; } else { ret = line6_hwdep_init(line6); if (ret < 0) diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index cd44cb5f1310..16e644330c4d 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -376,11 +376,6 @@ static int pod_init(struct usb_line6 *line6, if (err < 0) return err; - /* initialize MIDI subsystem: */ - err = line6_init_midi(line6); - if (err < 0) - return err; - /* initialize PCM subsystem: */ err = line6_init_pcm(line6, &pod_pcm_properties); if (err < 0) diff --git a/sound/usb/line6/variax.c b/sound/usb/line6/variax.c index ed158f04de80..c2245aa93b08 100644 --- a/sound/usb/line6/variax.c +++ b/sound/usb/line6/variax.c @@ -159,7 +159,6 @@ static int variax_init(struct usb_line6 *line6, const struct usb_device_id *id) { struct usb_line6_variax *variax = line6_to_variax(line6); - int err; line6->process_message = line6_variax_process_message; line6->disconnect = line6_variax_disconnect; @@ -172,11 +171,6 @@ static int variax_init(struct usb_line6 *line6, if (variax->buffer_activate == NULL) return -ENOMEM; - /* initialize MIDI subsystem: */ - err = line6_init_midi(&variax->line6); - if (err < 0) - return err; - /* initiate startup procedure: */ schedule_delayed_work(&line6->startup_work, msecs_to_jiffies(VARIAX_STARTUP_DELAY1)); -- 2.26.2

1 0

[PATCH v3 00/11] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 18 May '21

18 May '21

Hi, This patchset is revised version of my take 2: * https://lore.kernel.org/alsa-devel/20210515071112.101535-1-o-takashi@sakamo… Changes from v2: * use int type instead of enumerated type for combination of enumeration-constants * fix semi-colon typo in subject line of 10th patch Regards Takashi Sakamoto (11): Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro" ALSA: firewire-lib/motu: use int type for the value of bitwise OR with enumerator-constant ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: code refactoring to detect mackie models ALSA: oxfw: add explicit device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw: code refactoring for wrong_dbs quirk ALSA: oxfw: add quirk flag for blocking transmission method sound/firewire/Kconfig | 4 +- sound/firewire/amdtp-am824.c | 4 +- sound/firewire/amdtp-am824.h | 2 +- sound/firewire/amdtp-stream.c | 11 ++- sound/firewire/amdtp-stream.h | 5 +- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-stream.c | 25 +++--- sound/firewire/oxfw/oxfw.c | 137 ++++++++++++++--------------- sound/firewire/oxfw/oxfw.h | 13 ++- 11 files changed, 110 insertions(+), 98 deletions(-) -- 2.27.0

2 15

[PATCH] sound: rawmidi: Add framing mode
by David Henningsson 18 May '21

18 May '21

This commit adds a new framing mode that frames all MIDI data into 32-byte frames with a timestamp. The main benefit is that we can get accurate timestamps even if userspace wakeup and processing is not immediate. Testing on a Celeron N3150 with this mode has a max jitter of 2.8 ms, compared to the in-kernel seq implementation which has a max jitter of 5 ms during idle and much worse when running scheduler stress tests in parallel. Signed-off-by: David Henningsson <coding(a)diwic.se> --- Hi, it's been a while. Both because I've been busy with things that needed to take priority, and because I wanted to do a little more than just compile testing it before submitting; hopefully this one is mergable. That testing showed that I needed to change from "u8" to "__u8" in the headers. Other than that I think it is just the changes that Takashi Iwai pointed out in v6. include/sound/rawmidi.h | 2 + include/uapi/sound/asound.h | 30 +++++++++++- sound/core/rawmidi.c | 93 ++++++++++++++++++++++++++++++++++++- sound/core/rawmidi_compat.c | 4 +- 4 files changed, 124 insertions(+), 5 deletions(-) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 334842daa904..989e1517332d 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -81,6 +81,8 @@ struct snd_rawmidi_substream { bool opened; /* open flag */ bool append; /* append flag (merge more streams) */ bool active_sensing; /* send active sensing when close */ + unsigned int framing; /* whether to frame input data */ + unsigned int clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ size_t bytes; struct snd_rawmidi *rmidi; diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..d17c061950df 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -710,7 +710,7 @@ enum { * Raw MIDI section - /dev/snd/midi?? */ -#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 1) +#define SNDRV_RAWMIDI_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 2) enum { SNDRV_RAWMIDI_STREAM_OUTPUT = 0, @@ -736,12 +736,38 @@ struct snd_rawmidi_info { unsigned char reserved[64]; /* reserved for future use */ }; +#define SNDRV_RAWMIDI_MODE_FRAMING_MASK (7<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_SHIFT 0 +#define SNDRV_RAWMIDI_MODE_FRAMING_NONE (0<<0) +#define SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP (1<<0) +#define SNDRV_RAWMIDI_MODE_CLOCK_MASK (7<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_SHIFT 3 +#define SNDRV_RAWMIDI_MODE_CLOCK_NONE (0<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_REALTIME (1<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC (2<<3) +#define SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW (3<<3) + +#define SNDRV_RAWMIDI_FRAMING_DATA_LENGTH 16 + +struct snd_rawmidi_framing_tstamp { + /* For now, frame_type is always 0. Midi 2.0 is expected to add new + * types here. Applications are expected to skip unknown frame types. + */ + __u8 frame_type; + __u8 length; /* number of valid bytes in data field */ + __u8 reserved[2]; + __u32 tv_nsec; /* nanoseconds */ + __u64 tv_sec; /* seconds */ + __u8 data[SNDRV_RAWMIDI_FRAMING_DATA_LENGTH]; +} __packed; + struct snd_rawmidi_params { int stream; size_t buffer_size; /* queue size in bytes */ size_t avail_min; /* minimum avail bytes for wakeup */ unsigned int no_active_sensing: 1; /* do not send active sensing byte in close() */ - unsigned char reserved[16]; /* reserved for future use */ + unsigned int mode; /* For input data only, frame incoming data */ + unsigned char reserved[12]; /* reserved for future use */ }; #ifndef __KERNEL__ diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index aca00af93afe..4a6534db77d6 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -680,9 +680,12 @@ static int resize_runtime_buffer(struct snd_rawmidi_runtime *runtime, bool is_input) { char *newbuf, *oldbuf; + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; if (params->buffer_size < 32 || params->buffer_size > 1024L * 1024L) return -EINVAL; + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP && (params->buffer_size & 0x1f) != 0) + return -EINVAL; if (params->avail_min < 1 || params->avail_min > params->buffer_size) return -EINVAL; if (params->buffer_size != runtime->buffer_size) { @@ -720,8 +723,24 @@ EXPORT_SYMBOL(snd_rawmidi_output_params); int snd_rawmidi_input_params(struct snd_rawmidi_substream *substream, struct snd_rawmidi_params *params) { + unsigned int framing = params->mode & SNDRV_RAWMIDI_MODE_FRAMING_MASK; + unsigned int clock_type = params->mode & SNDRV_RAWMIDI_MODE_CLOCK_MASK; + int err; + + if (framing == SNDRV_RAWMIDI_MODE_FRAMING_NONE && clock_type != SNDRV_RAWMIDI_MODE_CLOCK_NONE) + return -EINVAL; + else if (clock_type > SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW) + return -EINVAL; + if (framing > SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) + return -EINVAL; snd_rawmidi_drain_input(substream); - return resize_runtime_buffer(substream->runtime, params, true); + err = resize_runtime_buffer(substream->runtime, params, true); + if (err < 0) + return err; + + substream->framing = framing; + substream->clock_type = clock_type; + return 0; } EXPORT_SYMBOL(snd_rawmidi_input_params); @@ -963,6 +982,62 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, return -ENOIOCTLCMD; } +static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream, + const unsigned char *buffer, int src_count, const struct timespec64 *tstamp) +{ + struct snd_rawmidi_runtime *runtime = substream->runtime; + struct snd_rawmidi_framing_tstamp *dest_ptr; + struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec }; + int dest_frames = 0; + int orig_count = src_count; + int frame_size = sizeof(struct snd_rawmidi_framing_tstamp); + + BUILD_BUG_ON(frame_size != 0x20); + if (snd_BUG_ON((runtime->hw_ptr & 0x1f) != 0)) + return -EINVAL; + + while (src_count > 0) { + if ((int)(runtime->buffer_size - runtime->avail) < frame_size) { + runtime->xruns += src_count; + break; + } + if (src_count >= SNDRV_RAWMIDI_FRAMING_DATA_LENGTH) + frame.length = SNDRV_RAWMIDI_FRAMING_DATA_LENGTH; + else { + frame.length = src_count; + memset(frame.data, 0, SNDRV_RAWMIDI_FRAMING_DATA_LENGTH); + } + memcpy(frame.data, buffer, frame.length); + buffer += frame.length; + src_count -= frame.length; + dest_ptr = (struct snd_rawmidi_framing_tstamp *) (runtime->buffer + runtime->hw_ptr); + *dest_ptr = frame; + runtime->avail += frame_size; + runtime->hw_ptr += frame_size; + runtime->hw_ptr %= runtime->buffer_size; + dest_frames++; + } + return orig_count - src_count; +} + +static struct timespec64 get_framing_tstamp(struct snd_rawmidi_substream *substream) +{ + struct timespec64 ts64 = {0, 0}; + + switch (substream->clock_type) { + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC_RAW: + ktime_get_raw_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_MONOTONIC: + ktime_get_ts64(&ts64); + break; + case SNDRV_RAWMIDI_MODE_CLOCK_REALTIME: + ktime_get_real_ts64(&ts64); + break; + } + return ts64; +} + /** * snd_rawmidi_receive - receive the input data from the device * @substream: the rawmidi substream @@ -977,6 +1052,7 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, const unsigned char *buffer, int count) { unsigned long flags; + struct timespec64 ts64 = get_framing_tstamp(substream); int result = 0, count1; struct snd_rawmidi_runtime *runtime = substream->runtime; @@ -987,8 +1063,11 @@ int snd_rawmidi_receive(struct snd_rawmidi_substream *substream, "snd_rawmidi_receive: input is not active!!!\n"); return -EINVAL; } + spin_lock_irqsave(&runtime->lock, flags); - if (count == 1) { /* special case, faster code */ + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + result = receive_with_tstamp_framing(substream, buffer, count, &ts64); + } else if (count == 1) { /* special case, faster code */ substream->bytes++; if (runtime->avail < runtime->buffer_size) { runtime->buffer[runtime->hw_ptr++] = buffer[0]; @@ -1541,6 +1620,8 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, struct snd_rawmidi_substream *substream; struct snd_rawmidi_runtime *runtime; unsigned long buffer_size, avail, xruns; + unsigned int clock_type; + static const char *clock_names[4] = { "none", "realtime", "monotonic", "monotonic raw" }; rmidi = entry->private_data; snd_iprintf(buffer, "%s\n\n", rmidi->name); @@ -1596,6 +1677,14 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, " Avail : %lu\n" " Overruns : %lu\n", buffer_size, avail, xruns); + if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { + clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; + if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + snd_iprintf(buffer, + " Framing : tstamp\n" + " Clock type : %s\n", + clock_names[clock_type]); + } } } } diff --git a/sound/core/rawmidi_compat.c b/sound/core/rawmidi_compat.c index 7397130976d0..68a93443583c 100644 --- a/sound/core/rawmidi_compat.c +++ b/sound/core/rawmidi_compat.c @@ -13,7 +13,8 @@ struct snd_rawmidi_params32 { u32 buffer_size; u32 avail_min; unsigned int no_active_sensing; /* avoid bit-field */ - unsigned char reserved[16]; + unsigned int mode; + unsigned char reserved[12]; } __attribute__((packed)); static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, @@ -25,6 +26,7 @@ static int snd_rawmidi_ioctl_params_compat(struct snd_rawmidi_file *rfile, if (get_user(params.stream, &src->stream) || get_user(params.buffer_size, &src->buffer_size) || get_user(params.avail_min, &src->avail_min) || + get_user(params.mode, &src->mode) || get_user(val, &src->no_active_sensing)) return -EFAULT; params.no_active_sensing = val; -- 2.25.1

3 3

[PATCH] ALSA: dice: fix stream format for TC Electronic Konnekt Live at high sampling transfer frequency
by Takashi Sakamoto 18 May '21

18 May '21

At high sampling transfer frequency, TC Electronic Konnekt Live transfers/receives 6 audio data frames in multi bit linear audio data channel of data block in CIP payload. Current hard-coded stream format is wrong. Cc: <stable(a)vger.kernel.org> Fixes: f1f0f330b1d0 ("ALSA: dice: add parameters of stream formats for models produced by TC Electronic") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-tcelectronic.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/firewire/dice/dice-tcelectronic.c b/sound/firewire/dice/dice-tcelectronic.c index a8875d24ba2a..43a3bcb15b3d 100644 --- a/sound/firewire/dice/dice-tcelectronic.c +++ b/sound/firewire/dice/dice-tcelectronic.c @@ -38,8 +38,8 @@ static const struct dice_tc_spec konnekt_24d = { }; static const struct dice_tc_spec konnekt_live = { - .tx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, - .rx_pcm_chs = {{16, 16, 16}, {0, 0, 0} }, + .tx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, + .rx_pcm_chs = {{16, 16, 6}, {0, 0, 0} }, .has_midi = true, }; -- 2.27.0

2 1

[PATCH] ALSA: dice: disable double_pcm_frames mode for M-Audio Profire 610, 2626 and Avid M-Box 3 Pro
by Takashi Sakamoto 18 May '21

18 May '21

ALSA dice driver detects jumbo payload at high sampling transfer frequency for below models: * Avid M-Box 3 Pro * M-Audio Profire 610 * M-Audio Profire 2626 Although many DICE-based devices have a quirk at high sampling transfer frequency to multiplex double number of PCM frames into data block than the number in IEC 61883-1/6, the above devices are just compliant to IEC 61883-1/6. This commit disables the mode of double_pcm_frames for the models. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-pcm.c | 4 ++-- sound/firewire/dice/dice-stream.c | 2 +- sound/firewire/dice/dice.c | 24 ++++++++++++++++++++++++ sound/firewire/dice/dice.h | 3 ++- 4 files changed, 29 insertions(+), 4 deletions(-) diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index af8a90ee40f3..a69ca1111b03 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -218,7 +218,7 @@ static int pcm_open(struct snd_pcm_substream *substream) if (frames_per_period > 0) { // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { frames_per_period *= 2; frames_per_buffer *= 2; } @@ -273,7 +273,7 @@ static int pcm_hw_params(struct snd_pcm_substream *substream, mutex_lock(&dice->mutex); // For double_pcm_frame quirk. - if (rate > 96000) { + if (rate > 96000 && !dice->disable_double_pcm_frames) { events_per_period /= 2; events_per_buffer /= 2; } diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c index 1a14c083e8ce..c4dfe76500c2 100644 --- a/sound/firewire/dice/dice-stream.c +++ b/sound/firewire/dice/dice-stream.c @@ -181,7 +181,7 @@ static int keep_resources(struct snd_dice *dice, struct amdtp_stream *stream, // as 'Dual Wire'. // For this quirk, blocking mode is required and PCM buffer size should // be aligned to SYT_INTERVAL. - double_pcm_frames = rate > 96000; + double_pcm_frames = (rate > 96000 && !dice->disable_double_pcm_frames); if (double_pcm_frames) { rate /= 2; pcm_chs *= 2; diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 107a81691f0e..239d164b0eea 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -21,6 +21,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. #define OUI_PRESONUS 0x000a92 #define OUI_HARMAN 0x000fd7 +#define OUI_AVID 0x00a07e #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 @@ -222,6 +223,14 @@ static int dice_probe(struct fw_unit *unit, (snd_dice_detect_formats_t)entry->driver_data; } + // Below models are compliant to IEC 61883-1/6 and have no quirk at high sampling transfer + // frequency. + // * Avid M-Box 3 Pro + // * M-Audio Profire 610 + // * M-Audio Profire 2626 + if (entry->vendor_id == OUI_MAUDIO || entry->vendor_id == OUI_AVID) + dice->disable_double_pcm_frames = true; + spin_lock_init(&dice->lock); mutex_init(&dice->mutex); init_completion(&dice->clock_accepted); @@ -278,7 +287,22 @@ static void dice_bus_reset(struct fw_unit *unit) #define DICE_INTERFACE 0x000001 +#define DICE_DEV_ENTRY_TYPICAL(vendor, model, data) \ + { \ + .match_flags = IEEE1394_MATCH_VENDOR_ID | \ + IEEE1394_MATCH_MODEL_ID | \ + IEEE1394_MATCH_SPECIFIER_ID | \ + IEEE1394_MATCH_VERSION, \ + .vendor_id = (vendor), \ + .model_id = (model), \ + .specifier_id = (vendor), \ + .version = DICE_INTERFACE, \ + .driver_data = (kernel_ulong_t)(data), \ + } + static const struct ieee1394_device_id dice_id_table[] = { + // Avid M-Box 3 Pro. To match in probe function. + DICE_DEV_ENTRY_TYPICAL(OUI_AVID, 0x000004, snd_dice_detect_extension_formats), /* M-Audio Profire 2626 has a different value in version field. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index adc6f7c84460..3c967d1b3605 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -109,7 +109,8 @@ struct snd_dice { struct fw_iso_resources rx_resources[MAX_STREAMS]; struct amdtp_stream tx_stream[MAX_STREAMS]; struct amdtp_stream rx_stream[MAX_STREAMS]; - bool global_enabled; + bool global_enabled:1; + bool disable_double_pcm_frames:1; struct completion clock_accepted; unsigned int substreams_counter; -- 2.27.0

2 1

[PATCH v2 00/10] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 18 May '21

18 May '21

Hi, This patchset is reveised version of my initial post: * https://lore.kernel.org/alsa-devel/20210514085435.92807-1-o-takashi@sakamoc… Changes from v1: * revert Kconfig fix * fix detection of mackie models * add quirk flag for blocking transmission Regards Takashi Sakamoto (10): Revert "ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro" ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: code refactoring to detect mackie models ALSA: oxfw: add explicit device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw; code refactoring for wrong_dbs quirk ALSA: oxfw: add quirk flag for blocking transmission method sound/firewire/Kconfig | 4 +- sound/firewire/amdtp-stream.c | 7 +- sound/firewire/bebob/bebob.c | 2 +- sound/firewire/oxfw/oxfw-stream.c | 25 +++--- sound/firewire/oxfw/oxfw.c | 137 ++++++++++++++---------------- sound/firewire/oxfw/oxfw.h | 12 ++- 6 files changed, 98 insertions(+), 89 deletions(-) -- 2.27.0

2 13

[PATCH v5 0/9] ASoC: codecs: add wcd938x support
by Srinivas Kandagatla 18 May '21

18 May '21

This patchset adds support for Qualcomm WCD938X codec. Qualcomm WCD9380/WCD9385 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 4 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 7 x TX diff inputs, 8 DMICs and MBHC. Eventhough this device has two SoundWire devices, only tx device has access to main codec Control/Status Registers! For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on SM8250 MTP device. Thanks, srini Changes since v4: - updated dt-bindings and example as suggested by RobH - moved SoundWire specific functions to sdw driver. - fixed wrong indentation reported by kernel test robot Srinivas Kandagatla (9): ASoC: dt-bindings: wcd938x: add bindings for wcd938x ASoC: codecs: wcd-clsh: add new version support ASoC: codecs: wcd938x: add basic driver ASoC: dt-bindings: wcd938x-sdw: add bindings for wcd938x-sdw ASoC: codecs: wcd938x-sdw: add SoundWire driver ASoC: codecs: wcd938x: add basic controls ASoC: codecs: wcd938x: add playback dapm widgets ASoC: codecs: wcd938x: add capture dapm widgets ASoC: codecs: wcd938x: add audio routing .../bindings/sound/qcom,wcd938x-sdw.yaml | 70 + .../bindings/sound/qcom,wcd938x.yaml | 149 + sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/wcd-clsh-v2.c | 348 +- sound/soc/codecs/wcd-clsh-v2.h | 16 + sound/soc/codecs/wcd938x-sdw.c | 298 ++ sound/soc/codecs/wcd938x.c | 3753 +++++++++++++++++ sound/soc/codecs/wcd938x.h | 705 ++++ 9 files changed, 5346 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd938x.yaml create mode 100644 sound/soc/codecs/wcd938x-sdw.c create mode 100644 sound/soc/codecs/wcd938x.c create mode 100644 sound/soc/codecs/wcd938x.h -- 2.21.0

2 11

[PATCH] dt-bindings: More removals of type references on common properties
by Rob Herring 17 May '21

17 May '21

Users of common properties shouldn't have a type definition as the common schemas already have one. A few new ones slipped in and *-names was missed in the last clean-up pass. Drop all the unnecessary type references in the tree. A meta-schema update to catch these is pending. Cc: Luca Ceresoli <luca(a)lucaceresoli.net> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Olivier Moysan <olivier.moysan(a)foss.st.com> Cc: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Georgi Djakov <djakov(a)kernel.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Orson Zhai <orsonzhai(a)gmail.com> Cc: Baolin Wang <baolin.wang7(a)gmail.com> Cc: Chunyan Zhang <zhang.lyra(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Fabrice Gasnier <fabrice.gasnier(a)st.com> Cc: Odelu Kukatla <okukatla(a)codeaurora.org> Cc: Alex Elder <elder(a)kernel.org> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: linux-clk(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-iio(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-input(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/clock/idt,versaclock5.yaml | 2 -- .../devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 1 - Documentation/devicetree/bindings/input/input.yaml | 1 - Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml | 1 - Documentation/devicetree/bindings/net/qcom,ipa.yaml | 1 - .../devicetree/bindings/power/supply/sc2731-charger.yaml | 2 +- Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml | 2 +- 7 files changed, 2 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml index c268debe5b8d..28675b0b80f1 100644 --- a/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml +++ b/Documentation/devicetree/bindings/clock/idt,versaclock5.yaml @@ -60,7 +60,6 @@ properties: maxItems: 2 idt,xtal-load-femtofarads: - $ref: /schemas/types.yaml#/definitions/uint32 minimum: 9000 maximum: 22760 description: Optional load capacitor for XTAL1 and XTAL2 @@ -84,7 +83,6 @@ patternProperties: enum: [ 1800000, 2500000, 3300000 ] idt,slew-percent: description: The Slew rate control for CMOS single-ended. - $ref: /schemas/types.yaml#/definitions/uint32 enum: [ 80, 85, 90, 100 ] required: diff --git a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml index 6f2398cdc82d..1e7894e524f9 100644 --- a/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml +++ b/Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml @@ -102,7 +102,6 @@ patternProperties: st,adc-channel-names: description: List of single-ended channel names. - $ref: /schemas/types.yaml#/definitions/string-array st,filter-order: description: | diff --git a/Documentation/devicetree/bindings/input/input.yaml b/Documentation/devicetree/bindings/input/input.yaml index 74244d21d2b3..d41d8743aad4 100644 --- a/Documentation/devicetree/bindings/input/input.yaml +++ b/Documentation/devicetree/bindings/input/input.yaml @@ -38,6 +38,5 @@ properties: Duration in seconds which the key should be kept pressed for device to reset automatically. Device with key pressed reset feature can specify this property. - $ref: /schemas/types.yaml#/definitions/uint32 additionalProperties: true diff --git a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml index cb6498108b78..36c955965d90 100644 --- a/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml +++ b/Documentation/devicetree/bindings/interconnect/qcom,rpmh.yaml @@ -92,7 +92,6 @@ properties: this interconnect to send RPMh commands. qcom,bcm-voter-names: - $ref: /schemas/types.yaml#/definitions/string-array description: | Names for each of the qcom,bcm-voters specified. diff --git a/Documentation/devicetree/bindings/net/qcom,ipa.yaml b/Documentation/devicetree/bindings/net/qcom,ipa.yaml index 7443490d4cc6..5fe6d3dceb08 100644 --- a/Documentation/devicetree/bindings/net/qcom,ipa.yaml +++ b/Documentation/devicetree/bindings/net/qcom,ipa.yaml @@ -105,7 +105,6 @@ properties: - description: Whether the IPA clock is enabled (if valid) qcom,smem-state-names: - $ref: /schemas/types.yaml#/definitions/string-array description: The names of the state bits used for SMP2P output items: - const: ipa-clock-enabled-valid diff --git a/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml b/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml index db1aa238cda5..b62c2431f94e 100644 --- a/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml +++ b/Documentation/devicetree/bindings/power/supply/sc2731-charger.yaml @@ -20,7 +20,7 @@ properties: maxItems: 1 phys: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle to the USB phy monitored-battery: diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index b4c190bddd84..61802a11baf4 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -49,7 +49,7 @@ properties: maxItems: 1 memory-region: - $ref: /schemas/types.yaml#/definitions/phandle + maxItems: 1 description: phandle to a node describing reserved memory (System RAM memory) The M core can't access all the DDR memory space on some platform, -- 2.27.0

6 9

[RFC PATCH] sound: line6: Fix race condition in line6_probe
by Hyeonggon Yoo 17 May '21

17 May '21

syzbot reported general protection fault in midibuf_is_full. the cause is linemidi pointer in struct usb_line6 isn't properly initialized. the pointer isn't initialized because there is race condition in line6_probe. it calls line6_init_cap_control first, which submits urb. and then it initializes it's data using private_init function. so it's possible line6_data_received is called before it's data isn't initialized. Link: https://lkml.org/lkml/2021/5/17/543 Reported-by: syzbot+0d2b3feb0a2887862e06@syzkallerlkml..appspotmail.com Signed-off-by: Hyeonggon Yoo <42.hyeyoo(a)gmail.com> --- sound/usb/line6/driver.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index a030dd65eb28..2c183a2a30f0 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -788,17 +788,17 @@ int line6_probe(struct usb_interface *interface, line6_get_usb_properties(line6); + /* initialize device data based on device: */ + ret = private_init(line6, id); + if (ret < 0) + goto error; + if (properties->capabilities & LINE6_CAP_CONTROL) { ret = line6_init_cap_control(line6); if (ret < 0) goto error; } - /* initialize device data based on device: */ - ret = private_init(line6, id); - if (ret < 0) - goto error; - /* creation of additional special files should go here */ dev_info(&interface->dev, "Line 6 %s now attached\n", -- 2.25.1

2 5

[PATCH 00/11] ALSA: usx2y: Fixes and cleanups
by Takashi Iwai 17 May '21

17 May '21

Hi, this is a patch series for usx2y drivers. Most part of changes are about cleanup of the code. The current usx2y driver is fairly unmaintained code, and this was the cause of some overseen bugs (or rather confusion) as happened in UMN fiasco. So the initial goal is to make a bit more readable for further reviews. The remaining part of changes are actual fixes. At this time, I concentrated only on the obvious issues that could be spotted out after the cleanup. The driver is likely broken in other ways, but the identification and the fix would need the actual device, so I leave them for now. Takashi === Takashi Iwai (11): ALSA: usx2y: Avoid camelCase ALSA: usx2y: Fix spaces ALSA: usx2y: Coding style fixes ALSA: usx2y: Fix potential leaks of uninitialized memory ALSA: usx2y: Avoid self-killing ALSA: usx2y: Fix potential memory leaks ALSA: usxy2: Fix potential doubly allocations ALSA: usx2y: Fix shmem initialization ALSA: usx2y: Don't call free_pages_exact() with NULL address ALSA: usx2y: Cleanup probe and disconnect callbacks ALSA: usx2y: Nuke pcm_list sound/usb/usx2y/us122l.c | 61 +-- sound/usb/usx2y/us122l.h | 2 +- sound/usb/usx2y/usX2Yhwdep.c | 135 +++--- sound/usb/usx2y/usX2Yhwdep.h | 2 +- sound/usb/usx2y/usb_stream.c | 82 ++-- sound/usb/usx2y/usb_stream.h | 23 +- sound/usb/usx2y/usbus428ctldefs.h | 104 ++--- sound/usb/usx2y/usbusx2y.c | 389 +++++++++-------- sound/usb/usx2y/usbusx2y.h | 65 ++- sound/usb/usx2y/usbusx2yaudio.c | 684 ++++++++++++++++-------------- sound/usb/usx2y/usx2yhwdeppcm.c | 627 ++++++++++++++------------- sound/usb/usx2y/usx2yhwdeppcm.h | 4 +- 12 files changed, 1146 insertions(+), 1032 deletions(-) -- 2.26.2

1 11

[syzbot] general protection fault in line6_midibuf_write (2)
by syzbot 17 May '21

17 May '21

Hello, syzbot found the following issue on: HEAD commit: e5242861 usb: gadget: s3c: Fix the error handling path in .. git tree: https://git.kernel.org/pub/scm/linux/kernel/git/gregkh/usb.git usb-testing console output: https://syzkaller.appspot.com/x/log.txt?x=15dd2111d00000 kernel config: https://syzkaller.appspot.com/x/.config?x=9f35f6ae509ee286 dashboard link: https://syzkaller.appspot.com/bug?extid=0d2b3feb0a2887862e06 Unfortunately, I don't have any reproducer for this issue yet. IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+0d2b3feb0a2887862e06(a)syzkaller.appspotmail.com general protection fault, probably for non-canonical address 0xdffffc0000000019: 0000 [#1] SMP KASAN KASAN: null-ptr-deref in range [0x00000000000000c8-0x00000000000000cf] CPU: 0 PID: 27882 Comm: syz-executor.1 Not tainted 5.12.0-rc4-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 RIP: 0010:midibuf_is_full sound/usb/line6/midibuf.c:44 [inline] RIP: 0010:line6_midibuf_write+0x30/0x3f0 sound/usb/line6/midibuf.c:91 Code: 49 89 f5 41 54 55 89 d5 53 48 89 fb 4c 8d 7b 18 48 83 ec 28 e8 a1 12 95 fc 4c 89 fa 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <0f> b6 04 02 84 c0 74 08 3c 03 0f 8e 09 03 00 00 44 8b 63 18 31 ff RSP: 0018:ffffc900000079f0 EFLAGS: 00010002 RAX: dffffc0000000000 RBX: 00000000000000b0 RCX: ffffc900055a9000 RDX: 0000000000000019 RSI: ffffffff84ab519f RDI: 00000000000000b0 RBP: 0000000000000000 R08: 0000000000000000 R09: ffffffff87ac9e83 R10: ffffffff84ab1930 R11: 0000000000000001 R12: 00000000000000b0 R13: ffff88811544da80 R14: ffff8881161ca584 R15: 00000000000000c8 FS: 00007fe616b12700(0000) GS:ffff8881f6a00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fef908ea130 CR3: 000000010da07000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 Call Trace: <IRQ> line6_data_received+0x281/0x520 sound/usb/line6/driver.c:296 __usb_hcd_giveback_urb+0x2b0/0x5c0 drivers/usb/core/hcd.c:1656 usb_hcd_giveback_urb+0x367/0x410 drivers/usb/core/hcd.c:1726 dummy_timer+0x11f4/0x32a0 drivers/usb/gadget/udc/dummy_hcd.c:1971 call_timer_fn+0x1a5/0x630 kernel/time/timer.c:1431 expire_timers kernel/time/timer.c:1476 [inline] __run_timers.part.0+0x67c/0xa10 kernel/time/timer.c:1745 __run_timers kernel/time/timer.c:1726 [inline] run_timer_softirq+0x80/0x120 kernel/time/timer.c:1758 __do_softirq+0x1b0/0x944 kernel/softirq.c:345 invoke_softirq kernel/softirq.c:221 [inline] __irq_exit_rcu kernel/softirq.c:422 [inline] irq_exit_rcu+0x110/0x1a0 kernel/softirq.c:434 sysvec_apic_timer_interrupt+0x6a/0x90 arch/x86/kernel/apic/apic.c:1100 </IRQ> asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:632 RIP: 0010:_raw_spin_unlock_irqrestore+0x3a/0x50 kernel/locking/spinlock.c:192 Code: 10 e8 6a 0c 76 fb 48 89 ef e8 32 c0 76 fb 81 e3 00 02 00 00 75 17 9c 58 f6 c4 02 75 17 48 85 db 74 01 fb 65 ff 0d 96 a5 50 7a <5b> 5d c3 e8 5e f2 94 fb eb e2 e8 b7 85 fe ff eb e2 0f 1f 44 00 00 RSP: 0018:ffffc900148ffe18 EFLAGS: 00000246 RAX: 0000000000000006 RBX: 0000000000000200 RCX: 1ffffffff11996b1 RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffffffff85b14972 RBP: ffff8881068a4008 R08: 0000000000000001 R09: 0000000000000001 R10: ffffffff81463c08 R11: 0000000000000000 R12: 0000000000000216 R13: ffff8881068a4008 R14: fffffffffffffff0 R15: 1ffff9200291ffcf spin_unlock_irqrestore include/linux/spinlock.h:409 [inline] raw_ioctl_ep0_stall drivers/usb/gadget/legacy/raw_gadget.c:751 [inline] raw_ioctl+0xed8/0x2720 drivers/usb/gadget/legacy/raw_gadget.c:1247 vfs_ioctl fs/ioctl.c:48 [inline] __do_sys_ioctl fs/ioctl.c:753 [inline] __se_sys_ioctl fs/ioctl.c:739 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:739 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x466217 Code: 3c 1c 48 f7 d8 49 39 c4 72 b8 e8 a4 48 02 00 85 c0 78 bd 48 83 c4 08 4c 89 e0 5b 41 5c c3 0f 1f 44 00 00 b8 10 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 bc ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007fe616b100b8 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 00007fe616b11130 RCX: 0000000000466217 RDX: 0000000000000000 RSI: 0000000000005501 RDI: 0000000000000005 RBP: 0000000000000005 R08: 0000000000000010 R09: 00312e6364755f79 R10: 00007fe616b0fe57 R11: 0000000000000246 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000020000000 R15: 00000000000010d0 Modules linked in: ---[ end trace 71f0ad4466d7d24b ]--- RIP: 0010:midibuf_is_full sound/usb/line6/midibuf.c:44 [inline] RIP: 0010:line6_midibuf_write+0x30/0x3f0 sound/usb/line6/midibuf.c:91 Code: 49 89 f5 41 54 55 89 d5 53 48 89 fb 4c 8d 7b 18 48 83 ec 28 e8 a1 12 95 fc 4c 89 fa 48 b8 00 00 00 00 00 fc ff df 48 c1 ea 03 <0f> b6 04 02 84 c0 74 08 3c 03 0f 8e 09 03 00 00 44 8b 63 18 31 ff RSP: 0018:ffffc900000079f0 EFLAGS: 00010002 RAX: dffffc0000000000 RBX: 00000000000000b0 RCX: ffffc900055a9000 RDX: 0000000000000019 RSI: ffffffff84ab519f RDI: 00000000000000b0 RBP: 0000000000000000 R08: 0000000000000000 R09: ffffffff87ac9e83 R10: ffffffff84ab1930 R11: 0000000000000001 R12: 00000000000000b0 R13: ffff88811544da80 R14: ffff8881161ca584 R15: 00000000000000c8 FS: 00007fe616b12700(0000) GS:ffff8881f6a00000(0000) knlGS:0000000000000000 CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 CR2: 00007fef908ea130 CR3: 000000010da07000 CR4: 00000000001506f0 DR0: 0000000000000000 DR1: 0000000000000000 DR2: 0000000000000000 DR3: 0000000000000000 DR6: 00000000fffe0ff0 DR7: 0000000000000400 --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

[PATCH v3 00/16] Replace some bad characters on documents
by Mauro Carvalho Chehab 17 May '21

17 May '21

The conversion tools used during DocBook/LaTeX/html/Markdown->ReST conversion and some cut-and-pasted text contain some characters that aren't easily reachable on standard keyboards and/or could cause troubles when parsed by the documentation build system. Replace the occurences of the following characters: - U+00a0 (' '): NO-BREAK SPACE - U+00ad (''): SOFT HYPHEN - U+2010 ('‐'): HYPHEN - U+2217 ('∗'): ASTERISK OPERATOR - U+feff (''): BOM I'll submit in separate another series to address other character occurrences. v3: - removed curly commas and changed the patch descriptions. v2: - removed EM/EN dashes and changed the patch descriptions. Mauro Carvalho Chehab (16): docs: hwmon: ir36021.rst: replace some characters docs: admin-guide: reporting-issues.rst: replace some characters docs: trace: coresight: coresight-etm4x-reference.rst: replace some characters docs: driver-api: ioctl.rst: replace some characters docs: driver-api: media: drivers: zoran.rst: replace some characters docs: usb: replace some characters docs: userspace-api: media: v4l: dev-decoder.rst: replace some characters docs: userspace-api: media: dvb: intro.rst: replace some characters docs: vm: zswap.rst: replace some characters docs: filesystems: ext4: blockgroup.rst: replace some characters docs: networking: device_drivers: replace some characters docs: PCI: acpi-info.rst: replace some characters docs: sound: kernel-api: writing-an-alsa-driver.rst: replace some characters docs: firmware-guide: acpi: dsd: graph.rst: replace some characters docs: virt: kvm: api.rst: replace some characters docs: RCU: replace some characters Documentation/PCI/acpi-info.rst | 18 ++--- .../Data-Structures/Data-Structures.rst | 46 ++++++------ .../Expedited-Grace-Periods.rst | 36 +++++----- .../Tree-RCU-Memory-Ordering.rst | 2 +- .../RCU/Design/Requirements/Requirements.rst | 70 +++++++++---------- .../admin-guide/reporting-issues.rst | 2 +- Documentation/driver-api/ioctl.rst | 8 +-- .../driver-api/media/drivers/zoran.rst | 2 +- Documentation/filesystems/ext4/blockgroup.rst | 2 +- .../firmware-guide/acpi/dsd/graph.rst | 2 +- Documentation/hwmon/ir36021.rst | 2 +- .../device_drivers/ethernet/intel/i40e.rst | 6 +- .../device_drivers/ethernet/intel/iavf.rst | 2 +- .../kernel-api/writing-an-alsa-driver.rst | 2 +- .../coresight/coresight-etm4x-reference.rst | 2 +- Documentation/usb/ehci.rst | 2 +- Documentation/usb/gadget_printer.rst | 2 +- .../userspace-api/media/dvb/intro.rst | 4 +- .../userspace-api/media/v4l/dev-decoder.rst | 2 +- Documentation/virt/kvm/api.rst | 28 ++++---- Documentation/vm/zswap.rst | 4 +- 21 files changed, 122 insertions(+), 122 deletions(-) -- 2.31.1

3 4

[PATCH v2 1/2] ASoC: imx-akcodec: Add imx-akcodec machine driver
by Shengjiu Wang 17 May '21

17 May '21

Add machine driver for i.MX boards that have AK4458/AK5558/AK4497/AK5552 DAC/ADC attached to SAI interface. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Cosmin-Gabriel Samoila <cosmin.samoila(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Signed-off-by: Adrian Alonso <adrian.alonso(a)nxp.com> --- changes in v2: - update the code according to the change of binding doc sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-akcodec.c | 799 ++++++++++++++++++++++++++++++++++++ 3 files changed, 813 insertions(+) create mode 100644 sound/soc/fsl/imx-akcodec.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 0917d65d6921..5720ad36494f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -350,6 +350,18 @@ config SND_SOC_IMX_RPMSG Say Y if you want to add support for SoC audio on an i.MX board with a rpmsg devices. +config SND_SOC_IMX_AKCODEC + tristate "SoC Audio support for i.MX boards with AKCODEC" + depends on OF && I2C + select SND_SOC_AK4458 + select SND_SOC_AK5558 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + help + SoC Audio support for i.MX boards with AK4458/AK4497/AK5558/AK5552 + Say Y if you want to add support for SoC audio on an i.MX board with + an AK4458/AK4497/AK5558/AK5552 DAC/ADC. + endif # SND_IMX_SOC endmenu diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index f146ce464acd..ee7caed0c758 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -71,6 +71,7 @@ snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-audmix-objs := imx-audmix.o snd-soc-imx-hdmi-objs := imx-hdmi.o snd-soc-imx-rpmsg-objs := imx-rpmsg.o +snd-soc-imx-akcodec-objs := imx-akcodec.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o @@ -79,3 +80,4 @@ obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_AUDMIX) += snd-soc-imx-audmix.o obj-$(CONFIG_SND_SOC_IMX_HDMI) += snd-soc-imx-hdmi.o obj-$(CONFIG_SND_SOC_IMX_RPMSG) += snd-soc-imx-rpmsg.o +obj-$(CONFIG_SND_SOC_IMX_AKCODEC) += snd-soc-imx-akcodec.o diff --git a/sound/soc/fsl/imx-akcodec.c b/sound/soc/fsl/imx-akcodec.c new file mode 100644 index 000000000000..586ecac8c0bf --- /dev/null +++ b/sound/soc/fsl/imx-akcodec.c @@ -0,0 +1,799 @@ +// SPDX-License-Identifier: GPL-2.0+ +// Copyright 2017-2021 NXP + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/of_device.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/pcm.h> +#include <sound/soc-dapm.h> +#include <sound/simple_card_utils.h> + +#include "fsl_sai.h" + +enum codec_type { + CODEC_AK5558 = 1, + CODEC_AK4458, + CODEC_AK4497, + CODEC_AK5552, +}; + +/* + * Mapping LRCK fs and frame width, table 3 & 4 in datasheet + * @rmin: min rate + * @rmax: max rate + * @wmin: min frame ratio + * @wmax: max frame ratio + */ +struct imx_akcodec_fs_mul { + unsigned int rmin; + unsigned int rmax; + unsigned int wmin; + unsigned int wmax; +}; + +/* + * Mapping TDM mode and frame width + */ +struct imx_akcodec_tdm_fs_mul { + unsigned int min; + unsigned int max; + unsigned int mul; +}; + +/* + * struct imx_akcodec_plat_data - specific info for codecs + * + * @fs_mul: ratio of mclk/fs for normal mode + * @tdm_fs_mul: ratio of mclk/fs for tdm mode + * @support_rates: supported sample rate + * @support_tdm_rates: supported sample rate for tdm mode + * @support_channels: supported channels + * @support_tdm_channels: supported channels for tdm mode + * @num_fs_mul: ARRAY_SIZE of fs_mul + * @num_tdm_fs_mul: ARRAY_SIZE of tdm_fs_mul + * @num_rates: ARRAY_SIZE of support_rates + * @num_tdm_rates: ARRAY_SIZE of support_tdm_rates + * @num_channels: ARRAY_SIZE of support_channels + * @num_tdm_channels: ARRAY_SIZE of support_tdm_channels + * @one2one_ratio: if mclk/bclk = 1 or not + * @type: codec type + */ +struct imx_akcodec_plat_data { + struct imx_akcodec_fs_mul *fs_mul; + struct imx_akcodec_tdm_fs_mul *tdm_fs_mul; + const u32 *support_rates; + const u32 *support_tdm_rates; + const u32 *support_channels; + const u32 *support_tdm_channels; + unsigned int num_fs_mul; + unsigned int num_tdm_fs_mul; + unsigned int num_rates; + unsigned int num_tdm_rates; + unsigned int num_channels; + unsigned int num_tdm_channels; + unsigned int num_codecs; + bool one2one_ratio; + enum codec_type type; +}; + +/* + * struct imx_akcodec_data - platform device data + * + * @card: card instance + * @plat_data: pointer of imx_akcodec_plat_data + * @dapm_routes: pointer of dapm_routes + * @num_dapm_routes: number of dapm_routes + * @slots: slots value + * @slot_width: slot width value + * @tdm_mode: tdm mode enabled or not + * @asrc_rate: asrc rates + * @asrc_format: asrc format + */ +struct imx_akcodec_data { + struct snd_soc_card card; + const struct imx_akcodec_plat_data *plat_data; + struct snd_soc_dapm_route *dapm_routes; + int num_dapm_routes; + unsigned int slots; + unsigned int slot_width; + bool tdm_mode; + u32 asrc_rate; + u32 asrc_format; +}; + +struct imx_akcodec_fs_mul ak4458_fs_mul[] = { + /* Normal, < 32kHz */ + { .rmin = 8000, .rmax = 24000, .wmin = 1024, .wmax = 1024, }, + /* Normal, 32kHz */ + { .rmin = 32000, .rmax = 32000, .wmin = 256, .wmax = 1024, }, + /* Normal */ + { .rmin = 44100, .rmax = 48000, .wmin = 256, .wmax = 768, }, + /* Double */ + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 512, }, + /* Quad */ + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 256, }, + /* Oct */ + { .rmin = 352800, .rmax = 384000, .wmin = 32, .wmax = 128, }, + /* Hex */ + { .rmin = 705600, .rmax = 768000, .wmin = 16, .wmax = 64, }, +}; + +struct imx_akcodec_tdm_fs_mul ak4458_tdm_fs_mul[] = { + /* + * Table 13 - Audio Interface Format + * For TDM mode, MCLK should is set to + * obtained from 2 * slots * slot_width + */ + { .min = 128, .max = 128, .mul = 256 }, /* TDM128 */ + { .min = 256, .max = 256, .mul = 512 }, /* TDM256 */ + { .min = 512, .max = 512, .mul = 1024 }, /* TDM512 */ +}; + +struct imx_akcodec_fs_mul ak4497_fs_mul[] = { + /** + * Table 7 - mapping multiplier and speed mode + * Tables 8 & 9 - mapping speed mode and LRCK fs + */ + { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, /* Normal, <= 32kHz */ + { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, /* Normal */ + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, /* Double */ + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 128, }, /* Quad */ + { .rmin = 352800, .rmax = 384000, .wmin = 128, .wmax = 128, }, /* Oct */ + { .rmin = 705600, .rmax = 768000, .wmin = 64, .wmax = 64, }, /* Hex */ +}; + +/* + * Auto MCLK selection based on LRCK for Normal Mode + * (Table 4 from datasheet) + */ +struct imx_akcodec_fs_mul ak5558_fs_mul[] = { + { .rmin = 8000, .rmax = 32000, .wmin = 1024, .wmax = 1024, }, + { .rmin = 44100, .rmax = 48000, .wmin = 512, .wmax = 512, }, + { .rmin = 88200, .rmax = 96000, .wmin = 256, .wmax = 256, }, + { .rmin = 176400, .rmax = 192000, .wmin = 128, .wmax = 128, }, + { .rmin = 352800, .rmax = 384000, .wmin = 64, .wmax = 64, }, + { .rmin = 705600, .rmax = 768000, .wmin = 32, .wmax = 32, }, +}; + +/* + * MCLK and BCLK selection based on TDM mode + * because of SAI we also add the restriction: MCLK >= 2 * BCLK + * (Table 9 from datasheet) + */ +struct imx_akcodec_tdm_fs_mul ak5558_tdm_fs_mul[] = { + { .min = 128, .max = 128, .mul = 256 }, + { .min = 256, .max = 256, .mul = 512 }, + { .min = 512, .max = 512, .mul = 1024 }, +}; + +static const u32 akcodec_rates[] = { + 8000, 11025, 16000, 22050, 32000, 44100, 48000, 88200, + 96000, 176400, 192000, 352800, 384000, 705600, 768000, +}; + +static const u32 akcodec_tdm_rates[] = { + 8000, 16000, 32000, 48000, 96000, +}; + +static const u32 ak4458_channels[] = { + 1, 2, 4, 6, 8, 10, 12, 14, 16, +}; + +static const u32 ak4458_tdm_channels[] = { + 1, 2, 3, 4, 5, 6, 7, 8, 16, +}; + +static const u32 ak5558_channels[] = { + 1, 2, 4, 6, 8, +}; + +static const u32 ak5558_tdm_channels[] = { + 1, 2, 3, 4, 5, 6, 7, 8, +}; + +static const struct imx_akcodec_plat_data imx8mq_akcodec = { + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .one2one_ratio = false, +}; + +static const struct imx_akcodec_plat_data imx_akcodec = { + .support_rates = akcodec_rates, + .num_rates = ARRAY_SIZE(akcodec_rates), + .support_tdm_rates = akcodec_tdm_rates, + .num_tdm_rates = ARRAY_SIZE(akcodec_tdm_rates), + .one2one_ratio = true, +}; + +static bool format_is_dsd(struct snd_pcm_hw_params *params) +{ + snd_pcm_format_t format = params_format(params); + + switch (format) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + return true; + default: + return false; + } +} + +static bool format_is_tdm(unsigned int dai_fmt) +{ + if ((dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_DSP_B) + return true; + else + return false; +} + +static unsigned long akcodec_get_mclk_rate(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + bool tdm_mode) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(rtd->card); + const struct imx_akcodec_plat_data *plat_data = data->plat_data; + unsigned int width = data->slots * data->slot_width; + unsigned int rate = params_rate(params); + int i; + + if (data->tdm_mode) { + for (i = 0; i < plat_data->num_tdm_fs_mul; i++) { + /* min = max = slots * slots_width */ + if (width != plat_data->tdm_fs_mul[i].min) + continue; + return rate * plat_data->tdm_fs_mul[i].mul; + } + } else { + for (i = 0; i < plat_data->num_fs_mul; i++) { + if (rate >= plat_data->fs_mul[i].rmin && + rate <= plat_data->fs_mul[i].rmax) { + width = max(width, plat_data->fs_mul[i].wmin); + width = min(width, plat_data->fs_mul[i].wmax); + + /* Adjust SAI bclk:mclk ratio */ + width *= plat_data->one2one_ratio ? 1 : 2; + + return rate * width; + } + } + } + + /* Let DAI manage clk frequency by default */ + return 0; +} + +static int imx_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + bool is_dsd = format_is_dsd(params); + struct device *dev = card->dev; + struct snd_soc_dai *codec_dai; + unsigned long mclk_freq; + unsigned int fmt; + bool tdm_mode; + int ret, i; + + tdm_mode = format_is_tdm(rtd->dai_link->dai_fmt); + + if (tdm_mode) { + data->slots = 8; + data->slot_width = 32; + fmt = rtd->dai_link->dai_fmt; + } else { + if (is_dsd) { + data->slots = 1; + data->slot_width = params_width(params); + fmt = (rtd->dai_link->dai_fmt & ~SND_SOC_DAIFMT_FORMAT_MASK) | + SND_SOC_DAIFMT_PDM; + } else { + data->slots = 2; + data->slot_width = params_physical_width(params); + fmt = (rtd->dai_link->dai_fmt & ~SND_SOC_DAIFMT_FORMAT_MASK) | + SND_SOC_DAIFMT_I2S; + } + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set cpu dai fmt: %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_tdm_slot(cpu_dai, + BIT(data->slots) - 1, + BIT(data->slots) - 1, + data->slots, data->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set cpu dai tdm slot: %d\n", ret); + return ret; + } + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set codec dai[%d] fmt: %d\n", i, ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(codec_dai, + BIT(data->slots) - 1, + BIT(data->slots) - 1, + data->slots, data->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set codec dai[%d] tdm slot: %d\n", i, ret); + return ret; + } + } + + /* Set MCLK freq */ + mclk_freq = akcodec_get_mclk_rate(substream, params, tdm_mode); + /* Use the maximum freq from DSD512 (512*44100 = 22579200) */ + if (is_dsd) + mclk_freq = 22579200; + + ret = snd_soc_dai_set_sysclk(cpu_dai, FSL_SAI_CLK_MAST1, mclk_freq, + SND_SOC_CLOCK_OUT); + if (ret && ret != -ENOTSUPP) + dev_err(dev, "failed to set cpui dai mclk1 rate (%lu): %d\n", + mclk_freq, ret); + return ret; +} + +static int ak5558_hw_rule_rate(struct snd_pcm_hw_params *p, struct snd_pcm_hw_rule *r) +{ + struct imx_akcodec_data *data = r->private; + struct snd_interval t = { .min = 8000, .max = 8000, }; + unsigned long mclk_freq; + unsigned int fs; + int i; + + fs = hw_param_interval(p, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + fs *= data->tdm_mode ? 8 : 2; + + /* Identify maximum supported rate */ + for (i = 0; i < ARRAY_SIZE(akcodec_rates); i++) { + mclk_freq = fs * akcodec_rates[i]; + /* Adjust SAI bclk:mclk ratio */ + mclk_freq *= data->plat_data->one2one_ratio ? 1 : 2; + + /* Skip rates for which MCLK is beyond supported value */ + if (mclk_freq > 36864000) + continue; + + if (t.max < akcodec_rates[i]) + t.max = akcodec_rates[i]; + } + + return snd_interval_refine(hw_param_interval(p, r->var), &t); +} + +static int imx_aif_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + static struct snd_pcm_hw_constraint_list constraint_rates; + static struct snd_pcm_hw_constraint_list constraint_channels; + int ret = 0; + + data->tdm_mode = format_is_tdm(rtd->dai_link->dai_fmt); + if (data->tdm_mode) { + constraint_channels.list = data->plat_data->support_tdm_channels; + constraint_channels.count = data->plat_data->num_tdm_channels; + constraint_rates.list = data->plat_data->support_tdm_rates; + constraint_rates.count = data->plat_data->num_tdm_rates; + } else { + constraint_channels.list = data->plat_data->support_channels; + constraint_channels.count = data->plat_data->num_channels; + constraint_rates.list = data->plat_data->support_rates; + constraint_rates.count = data->plat_data->num_rates; + } + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraint_channels); + if (ret) + return ret; + + ret = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraint_rates); + if (ret) + return ret; + + if (data->plat_data->type == CODEC_AK5558) + ret = snd_pcm_hw_rule_add(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + ak5558_hw_rule_rate, data, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); + + return ret; +} + +static struct snd_soc_ops imx_aif_ops = { + .hw_params = imx_aif_hw_params, + .startup = imx_aif_startup, +}; + +static struct snd_soc_ops imx_aif_ops_be = { + .hw_params = imx_aif_hw_params, +}; + +static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_card *card = rtd->card; + struct imx_akcodec_data *data = snd_soc_card_get_drvdata(card); + struct snd_interval *rate; + struct snd_mask *mask; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + rate->max = data->asrc_rate; + rate->min = data->asrc_rate; + + mask = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + snd_mask_none(mask); + snd_mask_set(mask, data->asrc_format); + + return 0; +} + +static int imx_akcodec_parse_of(struct snd_soc_card *card, + u32 *asrc_format, + u32 *asrc_rate, + enum codec_type *codec_type, + unsigned int *num_codecs) +{ + struct device_node *np; + struct device_node *codec = NULL; + struct device_node *platform = NULL; + struct device_node *cpu = NULL; + struct device *dev = card->dev; + struct snd_soc_dai_link *link; + struct of_phandle_args args; + struct snd_soc_dai_link_component *dlc; + int ret, num_links; + u32 width; + + ret = snd_soc_of_parse_card_name(card, "model"); + if (ret) { + dev_err(dev, "Error parsing card name: %d\n", ret); + return ret; + } + + /* DAPM routes */ + if (of_property_read_bool(dev->of_node, "audio-routing")) { + ret = snd_soc_of_parse_audio_routing(card, "audio-routing"); + if (ret) + return ret; + } + + /* Populate links */ + num_links = of_get_child_count(dev->of_node); + + /* Allocate the DAI link array */ + card->dai_link = devm_kcalloc(dev, num_links, sizeof(*link), GFP_KERNEL); + if (!card->dai_link) + return -ENOMEM; + + card->num_links = num_links; + link = card->dai_link; + + for_each_child_of_node(dev->of_node, np) { + dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + ret = -ENOMEM; + goto err_put_np; + } + + link->cpus = &dlc[0]; + link->platforms = &dlc[1]; + + link->num_cpus = 1; + link->num_platforms = 1; + + ret = of_property_read_string(np, "link-name", &link->name); + if (ret) { + dev_err(card->dev, "error getting codec dai_link name\n"); + goto err_put_np; + } + + cpu = of_get_child_by_name(np, "cpu"); + if (!cpu) { + dev_err(dev, "%s: Can't find cpu DT node\n", link->name); + ret = -EINVAL; + goto err; + } + + ret = of_parse_phandle_with_args(cpu, "sound-dai", + "#sound-dai-cells", 0, &args); + if (ret) { + dev_err(card->dev, "%s: error getting cpu phandle\n", link->name); + goto err; + } + link->cpus->of_node = args.np; + link->platforms->of_node = link->cpus->of_node; + link->id = args.args[0]; + + ret = snd_soc_of_get_dai_name(cpu, &link->cpus->dai_name); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(card->dev, "%s: error getting cpu dai name: %d\n", + link->name, ret); + goto err; + } + + codec = of_get_child_by_name(np, "codec"); + if (codec) { + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + if (ret < 0) { + if (ret != -EPROBE_DEFER) + dev_err(dev, "%s: codec dai not found: %d\n", + link->name, ret); + goto err; + } + + *num_codecs = link->num_codecs; + + if (!strcmp(link->codecs->dai_name, "ak4458-aif")) + *codec_type = CODEC_AK4458; + else if (!strcmp(link->codecs->dai_name, "ak4497-aif")) + *codec_type = CODEC_AK4497; + else if (!strcmp(link->codecs->dai_name, "ak5558-aif")) + *codec_type = CODEC_AK5558; + else if (!strcmp(link->codecs->dai_name, "ak5552-aif")) + *codec_type = CODEC_AK5552; + + } else { + dlc = devm_kzalloc(dev, sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + ret = -ENOMEM; + goto err; + } + + link->codecs = dlc; + link->num_codecs = 1; + + link->codecs->dai_name = "snd-soc-dummy-dai"; + link->codecs->name = "snd-soc-dummy"; + } + + if (!strncmp(link->name, "HiFi-ASRC-FE", 12)) { + /* DPCM frontend */ + link->dynamic = 1; + link->dpcm_merged_chan = 1; + + ret = of_property_read_u32(args.np, "fsl,asrc-rate", asrc_rate); + if (ret) { + dev_err(dev, "failed to get output rate\n"); + ret = -EINVAL; + goto err; + } + + ret = of_property_read_u32(args.np, "fsl,asrc-format", asrc_format); + if (ret) { + /* Fallback to old binding; translate to asrc_format */ + ret = of_property_read_u32(args.np, "fsl,asrc-width", &width); + if (ret) { + dev_err(dev, + "failed to decide output format\n"); + goto err; + } + + if (width == 24) + *asrc_format = SNDRV_PCM_FORMAT_S24_LE; + else + *asrc_format = SNDRV_PCM_FORMAT_S16_LE; + } + } else if (!strncmp(link->name, "HiFi-ASRC-BE", 12)) { + /* DPCM backend */ + link->no_pcm = 1; + link->platforms->of_node = NULL; + link->platforms->name = "snd-soc-dummy"; + + link->be_hw_params_fixup = be_hw_params_fixup; + link->ops = &imx_aif_ops_be; + } else { + link->ops = &imx_aif_ops; + } + + if (link->no_pcm || link->dynamic) + snd_soc_dai_link_set_capabilities(link); + + ret = asoc_simple_parse_daifmt(dev, np, codec, + NULL, &link->dai_fmt); + if (ret) + link->dai_fmt = SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS | + SND_SOC_DAIFMT_I2S; + + link->ignore_pmdown_time = 1; + link->stream_name = link->name; + link++; + + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); + } + + return 0; +err: + of_node_put(cpu); + of_node_put(codec); + of_node_put(platform); +err_put_np: + of_node_put(np); + return ret; +} + +static int imx_akcodec_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *link_be = NULL, *link; + struct imx_akcodec_plat_data *plat_data; + struct imx_akcodec_data *data; + int ret, i; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + plat_data = devm_kzalloc(&pdev->dev, sizeof(*plat_data), GFP_KERNEL); + if (!plat_data) + return -ENOMEM; + + data->plat_data = plat_data; + data->card.dev = &pdev->dev; + + memcpy(plat_data, of_device_get_match_data(&pdev->dev), + sizeof(struct imx_akcodec_plat_data)); + + dev_set_drvdata(&pdev->dev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = imx_akcodec_parse_of(&data->card, + &data->asrc_format, + &data->asrc_rate, + &plat_data->type, + &plat_data->num_codecs); + if (ret) + return ret; + + data->num_dapm_routes = plat_data->num_codecs + 1; + data->dapm_routes = devm_kcalloc(&pdev->dev, data->num_dapm_routes, + sizeof(struct snd_soc_dapm_route), + GFP_KERNEL); + if (!data->dapm_routes) + return -ENOMEM; + + /* configure the dapm routes */ + switch (plat_data->type) { + case CODEC_AK4458: + case CODEC_AK4497: + if (plat_data->num_codecs == 1) { + data->dapm_routes[0].sink = "Playback"; + data->dapm_routes[0].source = "CPU-Playback"; + i = 1; + } else { + for (i = 0; i < plat_data->num_codecs; i++) { + data->dapm_routes[i].sink = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", + i + 1, "Playback"); + data->dapm_routes[i].source = "CPU-Playback"; + } + } + data->dapm_routes[i].sink = "CPU-Playback"; + data->dapm_routes[i].source = "ASRC-Playback"; + break; + case CODEC_AK5558: + case CODEC_AK5552: + if (plat_data->num_codecs == 1) { + data->dapm_routes[0].sink = "CPU-Capture"; + data->dapm_routes[0].source = "Capture"; + i = 1; + } else { + for (i = 0; i < plat_data->num_codecs; i++) { + data->dapm_routes[i].source = + devm_kasprintf(&pdev->dev, GFP_KERNEL, "%d %s", + i + 1, "Capture"); + data->dapm_routes[i].sink = "CPU-Capture"; + } + } + data->dapm_routes[i].sink = "ASRC-Capture"; + data->dapm_routes[i].source = "CPU-Capture"; + break; + default: + break; + } + + switch (plat_data->type) { + case CODEC_AK4458: + plat_data->fs_mul = ak4458_fs_mul; + plat_data->num_fs_mul = ARRAY_SIZE(ak4458_fs_mul); + plat_data->tdm_fs_mul = ak4458_tdm_fs_mul; + plat_data->num_tdm_fs_mul = ARRAY_SIZE(ak4458_tdm_fs_mul); + plat_data->support_channels = ak4458_channels; + plat_data->num_channels = ARRAY_SIZE(ak4458_channels); + plat_data->support_tdm_channels = ak4458_tdm_channels; + plat_data->num_tdm_channels = ARRAY_SIZE(ak4458_tdm_channels); + break; + case CODEC_AK4497: + plat_data->fs_mul = ak4497_fs_mul; + plat_data->num_fs_mul = ARRAY_SIZE(ak4497_fs_mul); + plat_data->support_channels = ak4458_channels; + plat_data->num_channels = ARRAY_SIZE(ak4458_channels); + break; + case CODEC_AK5558: + case CODEC_AK5552: + plat_data->fs_mul = ak5558_fs_mul; + plat_data->num_fs_mul = ARRAY_SIZE(ak5558_fs_mul); + plat_data->tdm_fs_mul = ak5558_tdm_fs_mul; + plat_data->num_tdm_fs_mul = ARRAY_SIZE(ak5558_tdm_fs_mul); + plat_data->support_channels = ak5558_channels; + plat_data->num_channels = ARRAY_SIZE(ak5558_channels); + plat_data->support_tdm_channels = ak5558_tdm_channels; + plat_data->num_tdm_channels = ARRAY_SIZE(ak5558_tdm_channels); + break; + default: + break; + } + + /* with asrc as front end */ + if (data->card.num_links == 3) { + data->card.dapm_routes = data->dapm_routes; + data->card.num_dapm_routes = data->num_dapm_routes; + for_each_card_prelinks(&data->card, i, link) { + if (link->no_pcm == 1) + link_be = link; + } + for_each_card_prelinks(&data->card, i, link) { + if (link->dynamic == 1 && link_be) { + link->dpcm_playback = link_be->dpcm_playback; + link->dpcm_capture = link_be->dpcm_capture; + } + } + } + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + return 0; +} + +static const struct of_device_id imx_akcodec_dt_ids[] = { + { .compatible = "fsl,imx8mq-audio-akcodec", .data = &imx8mq_akcodec, }, + { .compatible = "fsl,imx-audio-akcodec", .data = &imx_akcodec, }, + { }, +}; +MODULE_DEVICE_TABLE(of, imx_akcodec_dt_ids); + +static struct platform_driver imx_akcodec_driver = { + .driver = { + .name = "imx-akcodec", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_akcodec_dt_ids, + }, + .probe = imx_akcodec_probe, +}; +module_platform_driver(imx_akcodec_driver); + +MODULE_DESCRIPTION("Freescale i.MX AK CODEC ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-akcodec"); -- 2.27.0

3 3

Not building on Raspbian Buster
by GitHub issues - opened 17 May '21

17 May '21

alsa-project/alsa-utils issue #89 was opened from riban-bw: Make fails during linking because the object files are not created. If the compiler is run then the object file will build but the prefix added by Makefile seems to stop the copmilation from occuring. This is building v1.2.4. (Master refuses to build due to dependancy on later version of asoundlib.) ``` Making all in include make[1]: Entering directory '/root/alsa-utils/include' (CDPATH="${ZSH_VERSION+.}:" && cd .. && /bin/bash /root/alsa-utils/missing autoheader) rm -f stamp-h1 touch aconfig.h.in cd .. && /bin/bash ./config.status include/aconfig.h config.status: creating include/aconfig.h config.status: include/aconfig.h is unchanged make all-am make[2]: Entering directory '/root/alsa-utils/include' Updating version.h make[2]: Leaving directory '/root/alsa-utils/include' make[1]: Leaving directory '/root/alsa-utils/include' Making all in alsactl make[1]: Entering directory '/root/alsa-utils/alsactl' Making all in init make[2]: Entering directory '/root/alsa-utils/alsactl/init' make[2]: Nothing to be done for 'all'. make[2]: Leaving directory '/root/alsa-utils/alsactl/init' make[2]: Entering directory '/root/alsa-utils/alsactl' source='alsactl.c' object='alsactl-alsactl.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-alsactl.o `test -f 'alsactl.c' || echo './'`alsactl.c source='state.c' object='alsactl-state.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-state.o `test -f 'state.c' || echo './'`state.c source='lock.c' object='alsactl-lock.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-lock.o `test -f 'lock.c' || echo './'`lock.c source='utils.c' object='alsactl-utils.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-utils.o `test -f 'utils.c' || echo './'`utils.c source='init_parse.c' object='alsactl-init_parse.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-init_parse.o `test -f 'init_parse.c' || echo './'`init_parse.c source='init_ucm.c' object='alsactl-init_ucm.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-init_ucm.o `test -f 'init_ucm.c' || echo './'`init_ucm.c source='daemon.c' object='alsactl-daemon.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-daemon.o `test -f 'daemon.c' || echo './'`daemon.c source='monitor.c' object='alsactl-monitor.o' libtool=no \ DEPDIR=.deps depmode=none /bin/bash ../depcomp \ gcc -DHAVE_CONFIG_H -I. -I../include -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -c -o alsactl-monitor.o `test -f 'monitor.c' || echo './'`monitor.c gcc -D_GNU_SOURCE -D__USE_GNU -DSYS_ASOUNDRC=\"/var/lib/alsa/asound.state\" -DSYS_LOCKFILE=\"/var/lock/asound.state.lock\" -DSYS_PIDFILE=\"/var/run/alsactl.pid\" -O2 -Wall -pipe -g -o alsactl alsactl-alsactl.o alsactl-state.o alsactl-lock.o alsactl-utils.o alsactl-init_parse.o alsactl-init_ucm.o alsactl-daemon.o alsactl-monitor.o -lasound -lm -ldl -lpthread gcc: error: alsactl-alsactl.o: No such file or directory gcc: error: alsactl-state.o: No such file or directory gcc: error: alsactl-lock.o: No such file or directory gcc: error: alsactl-utils.o: No such file or directory gcc: error: alsactl-init_parse.o: No such file or directory gcc: error: alsactl-init_ucm.o: No such file or directory gcc: error: alsactl-daemon.o: No such file or directory gcc: error: alsactl-monitor.o: No such file or directory make[2]: *** [Makefile:527: alsactl] Error 1 make[2]: Leaving directory '/root/alsa-utils/alsactl' make[1]: *** [Makefile:811: all-recursive] Error 1 make[1]: Leaving directory '/root/alsa-utils/alsactl' make: *** [Makefile:429: all-recursive] Error 1 ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/89 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/2] ASoC: dwc: add a quirk DW_I2S_QUIRK_STOP_ON_SHUTDOWN to dwc driver
by Vijendar Mukunda 17 May '21

17 May '21

For CZ/StoneyRidge platforms, ACP DMA between ACP SRAM and I2S FIFO should be stopped before stopping I2S Controller DMA. When DMA is progressing and stop request received, while DMA transfer ongoing between ACP SRAM and I2S FIFO, Stopping I2S DMA prior to ACP DMA stop resulting DMA Channel stop failure. This issue can't be fixed in ACP DMA driver due to design constraint. Add a quirk DW_I2S_QUIRK_STOP_ON_SHUTDOWN in dwc driver to fix DMA stop failure by invoking i2s_stop() sequence in shutdown() callback. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- include/sound/designware_i2s.h | 1 + sound/soc/dwc/dwc-i2s.c | 7 ++++++- 2 files changed, 7 insertions(+), 1 deletion(-) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 80d275b..a700d20 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -34,6 +34,7 @@ struct i2s_platform_data { #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) #define DW_I2S_QUIRK_16BIT_IDX_OVERRIDE (1 << 2) + #define DW_I2S_QUIRK_STOP_ON_SHUTDOWN (1 << 3) unsigned int quirks; unsigned int i2s_reg_comp1; unsigned int i2s_reg_comp2; diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index fd41602..f3a681f 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -308,6 +308,10 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, static void dw_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + if (dev->quirks & DW_I2S_QUIRK_STOP_ON_SHUTDOWN) + i2s_stop(dev, substream); snd_soc_dai_set_dma_data(dai, substream, NULL); } @@ -342,7 +346,8 @@ static int dw_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev->active--; - i2s_stop(dev, substream); + if (!(dev->quirks & DW_I2S_QUIRK_STOP_ON_SHUTDOWN)) + i2s_stop(dev, substream); break; default: ret = -EINVAL; -- 2.7.4

4 11

[Question] Different routing + sample formats for input and output
by GitHub issues - edited 17 May '21

17 May '21

alsa-project/alsa-lib issue #135 was edited from flatmax: One soundcard I use has different output and input sample formats. The input format is S32_LE. The output format is S24_3LE. There is also a routing change I like to do. My current .asoundrc (below) uses the asym plugin to manage input and output difference. The asym plugin creates problems for some unknown reason where I get overruns : `ALSA lib pcm.c:8545:(snd_pcm_recover) overrun occurred` If I switch my setup to using the soundcard for output and a second USB microphone for input, then I don't get any overruns. I still use the asym plugin to make it work and it works nicely. I have also tried simply the plug pluging, however I still get overruns. Is there any way to setup the one soundcard with routing and format changes without using the asym plugin ? Or is there a better way to setup the one soundcard with the required routing and format changes ? ``` ########## BEGIN ~/.asoundrc ################### @hooks [ { func load files [ "~/.asoundrc" ] errors false } ] pcm.!default { type asym playback.pcm "UMC1820_12ch_play" capture.pcm "UMC1820_10ch_capt" } ctl.!default { type hw card "UMC1820" } pcm.UMC1820_12ch_play { type route; slave.pcm "hw:UMC1820"; slave.format S24_3LE; slave.channels 12; ttable.0.2 1 ttable.1.3 1 ttable.2.4 1 ttable.3.5 1 ttable.4.6 1 ttable.5.7 1 ttable.6.8 1 ttable.7.9 1 ttable.8.10 1 ttable.9.11 1 ttable.10.0 1 ttable.11.1 1 } pcm.UMC1820_10ch_capt { type route; slave.pcm "hw:UMC1820"; slave.format S32_LE; slave.channels 10; ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 ttable.4.4 1 ttable.5.5 1 ttable.6.6 1 ttable.7.7 1 ttable.8.8 1 ttable.9.9 1 } ########## END ~/.asoundrc ################### ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/135 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[Question] Different routing + sample formats for input and output
by GitHub issues - edited 17 May '21

17 May '21

alsa-project/alsa-lib issue #135 was edited from flatmax: One soundcard I use has different output and input sample formats. The input format is S32_LE. The output format is S24_3LE. There is also a routing change I like to do. My current .asoundrc (below) uses the asym plugin to manage input and output difference. The asym plugin creates problems for some unknown reason where I get overruns : `ALSA lib pcm.c:8545:(snd_pcm_recover) overrun occurred` If I switch my setup to using the soundcard for output and a second USB microphone for input, then I don't get any overruns. I still use the asym plugin to make it work and it works nicely. I have also tried simply the plug pluging, however I still get overruns. Is there any way to setup the one soundcard with routing and format changes without using the asym plugin ? Or is there a better way to setup the one soundcard with the required routing and format changes ? ``` ########## BEGIN ~/.asoundrc ################### @hooks [ { func load files [ "~/.asoundrc" ] errors false } ] pcm.!default { type asym playback.pcm "UMC1820_12ch_play" capture.pcm "UMC1820_10ch_capt" } ctl.!default { type hw card "UMC1820_8ch" } pcm.UMC1820_12ch_play { type route; slave.pcm "hw:UMC1820"; slave.format S24_3LE; slave.channels 12; ttable.0.2 1 ttable.1.3 1 ttable.2.4 1 ttable.3.5 1 ttable.4.6 1 ttable.5.7 1 ttable.6.8 1 ttable.7.9 1 ttable.8.10 1 ttable.9.11 1 ttable.10.0 1 ttable.11.1 1 } pcm.UMC1820_10ch_capt { type route; slave.pcm "hw:UMC1820"; slave.format S32_LE; slave.channels 10; ttable.0.0 1 ttable.1.1 1 ttable.2.2 1 ttable.3.3 1 ttable.4.4 1 ttable.5.5 1 ttable.6.6 1 ttable.7.7 1 ttable.8.8 1 ttable.9.9 1 } ########## END ~/.asoundrc ################### ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/135 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[Question] Different routing + sample formats for input and output
by GitHub issues - opened 17 May '21

17 May '21

alsa-project/alsa-lib issue #135 was opened from flatmax: One soundcard I use has different output and input sample formats. The input format is S32_LE. The output format is S24_3LE. There is also a routing change I like to do. My current .asoundrc (below) uses the asym plugin to manage input and output difference. The asym plugin creates problems for some unknown reason where I get overruns : `ALSA lib pcm.c:8545:(snd_pcm_recover) overrun occurred` If I switch my setup to using the soundcard for output and a second USB microphone for input, then I don't get any overruns. I still use the asym plugin to make it work and it works nicely. I have also tried simply the plug pluging, however I still get overruns. Is there any way to setup the one soundcard with routing and format changes without using the asym plugin ? Or is there a better way to setup the one soundcard with the required routing and format changes ? ``` ########## BEGIN ~/.asoundrc ################### @hooks [ { func load files [ "~/.asoundrc" ] errors false } ] pcm.!default { type asym playback.pcm "UMC1820_12ch_play" capture.pcm "UMM6_capt" } ctl.!default { type hw card "UMC1820_8ch" } pcm.UMC1820_12ch_play { type route; slave.pcm "hw:UMC1820"; slave.format S24_3LE; slave.channels 12; ttable.0.2 1 ttable.1.3 1 ttable.2.4 1 ttable.3.5 1 ttable.4.6 1 ttable.5.7 1 ttable.6.8 1 ttable.7.9 1 ttable.8.10 1 ttable.9.11 1 ttable.10.0 1 ttable.11.1 1 } pcm.UMM6_capt { type linear slave.pcm "hw:UMM6" slave.format S16_LE; } ########## END ~/.asoundrc ################### ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/135 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 00/40] Use ASCII subset instead of UTF-8 alternate symbols
by Mauro Carvalho Chehab 15 May '21

15 May '21

This series contain basically a cleanup from all those years of converting files to ReST. During the conversion period, several tools like LaTeX, pandoc, DocBook and some specially-written scripts were used in order to convert existing documents. Such conversion tools - plus some text editor like LibreOffice or similar - have a set of rules that turns some typed ASCII characters into UTF-8 alternatives, for instance converting commas into curly commas and adding non-breakable spaces. All of those are meant to produce better results when the text is displayed in HTML or PDF formats. While it is perfectly fine to use UTF-8 characters in Linux, and specially at the documentation, it is better to stick to the ASCII subset on such particular case, due to a couple of reasons: 1. it makes life easier for tools like grep; 2. they easier to edit with the some commonly used text/source code editors. Also, Sphinx already do such conversion automatically outside literal blocks, as described at: https://docutils.sourceforge.io/docs/user/smartquotes.html In this series, the following UTF-8 symbols are replaced: - U+00a0 (' '): NO-BREAK SPACE - U+00ad (''): SOFT HYPHEN - U+00b4 ('´'): ACUTE ACCENT - U+00d7 ('×'): MULTIPLICATION SIGN - U+2010 ('‐'): HYPHEN - U+2018 ('‘'): LEFT SINGLE QUOTATION MARK - U+2019 ('’'): RIGHT SINGLE QUOTATION MARK - U+201c ('“'): LEFT DOUBLE QUOTATION MARK - U+201d ('”'): RIGHT DOUBLE QUOTATION MARK - U+2212 ('−'): MINUS SIGN - U+2217 ('∗'): ASTERISK OPERATOR - U+feff (''): ZERO WIDTH NO-BREAK SPACE (BOM) --- v2: - removed EM/EN DASH conversion from this patchset; - removed a few fixes, as those were addressed on a separate series. PS.: The first version of this series was posted with a different name: https://lore.kernel.org/lkml/cover.1620641727.git.mchehab+huawei@kernel.org/ I also changed the patch texts, in order to better describe the patches goals. Mauro Carvalho Chehab (40): docs: hwmon: Use ASCII subset instead of UTF-8 alternate symbols docs: admin-guide: Use ASCII subset instead of UTF-8 alternate symbols docs: admin-guide: media: ipu3.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: admin-guide: perf: imx-ddr.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: admin-guide: pm: Use ASCII subset instead of UTF-8 alternate symbols docs: trace: coresight: coresight-etm4x-reference.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: driver-api: ioctl.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: driver-api: thermal: Use ASCII subset instead of UTF-8 alternate symbols docs: driver-api: media: drivers: Use ASCII subset instead of UTF-8 alternate symbols docs: driver-api: firmware: other_interfaces.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: fault-injection: nvme-fault-injection.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: usb: Use ASCII subset instead of UTF-8 alternate symbols docs: process: code-of-conduct.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: userspace-api: media: fdl-appendix.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: userspace-api: media: v4l: Use ASCII subset instead of UTF-8 alternate symbols docs: userspace-api: media: dvb: Use ASCII subset instead of UTF-8 alternate symbols docs: vm: zswap.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: filesystems: f2fs.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: filesystems: ext4: Use ASCII subset instead of UTF-8 alternate symbols docs: kernel-hacking: Use ASCII subset instead of UTF-8 alternate symbols docs: hid: Use ASCII subset instead of UTF-8 alternate symbols docs: security: tpm: tpm_event_log.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: security: keys: trusted-encrypted.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: networking: scaling.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: networking: devlink: devlink-dpipe.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: networking: device_drivers: Use ASCII subset instead of UTF-8 alternate symbols docs: x86: Use ASCII subset instead of UTF-8 alternate symbols docs: scheduler: sched-deadline.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: power: powercap: powercap.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: ABI: Use ASCII subset instead of UTF-8 alternate symbols docs: PCI: acpi-info.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: gpu: Use ASCII subset instead of UTF-8 alternate symbols docs: sound: kernel-api: writing-an-alsa-driver.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: arm64: arm-acpi.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: infiniband: tag_matching.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: misc-devices: ibmvmc.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: firmware-guide: acpi: lpit.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: firmware-guide: acpi: dsd: graph.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: virt: kvm: api.rst: Use ASCII subset instead of UTF-8 alternate symbols docs: RCU: Use ASCII subset instead of UTF-8 alternate symbols ...sfs-class-chromeos-driver-cros-ec-lightbar | 2 +- .../ABI/testing/sysfs-devices-platform-ipmi | 2 +- .../testing/sysfs-devices-platform-trackpoint | 2 +- Documentation/ABI/testing/sysfs-devices-soc | 4 +- Documentation/PCI/acpi-info.rst | 22 +- .../Data-Structures/Data-Structures.rst | 52 ++-- .../Expedited-Grace-Periods.rst | 40 +-- .../Tree-RCU-Memory-Ordering.rst | 10 +- .../RCU/Design/Requirements/Requirements.rst | 122 ++++----- Documentation/admin-guide/media/ipu3.rst | 2 +- Documentation/admin-guide/perf/imx-ddr.rst | 2 +- Documentation/admin-guide/pm/intel_idle.rst | 4 +- Documentation/admin-guide/pm/intel_pstate.rst | 4 +- Documentation/admin-guide/ras.rst | 86 +++--- .../admin-guide/reporting-issues.rst | 2 +- Documentation/arm64/arm-acpi.rst | 8 +- .../driver-api/firmware/other_interfaces.rst | 2 +- Documentation/driver-api/ioctl.rst | 8 +- .../media/drivers/sh_mobile_ceu_camera.rst | 8 +- .../driver-api/media/drivers/zoran.rst | 2 +- .../driver-api/thermal/cpu-idle-cooling.rst | 14 +- .../driver-api/thermal/intel_powerclamp.rst | 6 +- .../thermal/x86_pkg_temperature_thermal.rst | 2 +- .../fault-injection/nvme-fault-injection.rst | 2 +- Documentation/filesystems/ext4/attributes.rst | 20 +- Documentation/filesystems/ext4/bigalloc.rst | 6 +- Documentation/filesystems/ext4/blockgroup.rst | 8 +- Documentation/filesystems/ext4/blocks.rst | 2 +- Documentation/filesystems/ext4/directory.rst | 16 +- Documentation/filesystems/ext4/eainode.rst | 2 +- Documentation/filesystems/ext4/inlinedata.rst | 6 +- Documentation/filesystems/ext4/inodes.rst | 6 +- Documentation/filesystems/ext4/journal.rst | 8 +- Documentation/filesystems/ext4/mmp.rst | 2 +- .../filesystems/ext4/special_inodes.rst | 4 +- Documentation/filesystems/ext4/super.rst | 10 +- Documentation/filesystems/f2fs.rst | 4 +- .../firmware-guide/acpi/dsd/graph.rst | 2 +- Documentation/firmware-guide/acpi/lpit.rst | 2 +- Documentation/gpu/i915.rst | 2 +- Documentation/gpu/komeda-kms.rst | 2 +- Documentation/hid/hid-sensor.rst | 70 ++--- Documentation/hid/intel-ish-hid.rst | 246 +++++++++--------- Documentation/hwmon/ir36021.rst | 2 +- Documentation/hwmon/ltc2992.rst | 2 +- Documentation/hwmon/pm6764tr.rst | 2 +- Documentation/infiniband/tag_matching.rst | 4 +- Documentation/kernel-hacking/hacking.rst | 2 +- Documentation/kernel-hacking/locking.rst | 2 +- Documentation/misc-devices/ibmvmc.rst | 8 +- .../device_drivers/ethernet/intel/i40e.rst | 8 +- .../device_drivers/ethernet/intel/iavf.rst | 4 +- .../device_drivers/ethernet/netronome/nfp.rst | 12 +- .../networking/devlink/devlink-dpipe.rst | 2 +- Documentation/networking/scaling.rst | 18 +- Documentation/power/powercap/powercap.rst | 210 +++++++-------- Documentation/process/code-of-conduct.rst | 2 +- Documentation/scheduler/sched-deadline.rst | 2 +- .../security/keys/trusted-encrypted.rst | 4 +- Documentation/security/tpm/tpm_event_log.rst | 2 +- .../kernel-api/writing-an-alsa-driver.rst | 68 ++--- .../coresight/coresight-etm4x-reference.rst | 16 +- Documentation/usb/ehci.rst | 2 +- Documentation/usb/gadget_printer.rst | 2 +- Documentation/usb/mass-storage.rst | 36 +-- .../media/dvb/audio-set-bypass-mode.rst | 2 +- .../userspace-api/media/dvb/audio.rst | 2 +- .../userspace-api/media/dvb/dmx-fopen.rst | 2 +- .../userspace-api/media/dvb/dmx-fread.rst | 2 +- .../media/dvb/dmx-set-filter.rst | 2 +- .../userspace-api/media/dvb/intro.rst | 6 +- .../userspace-api/media/dvb/video.rst | 2 +- .../userspace-api/media/fdl-appendix.rst | 64 ++--- .../userspace-api/media/v4l/crop.rst | 16 +- .../userspace-api/media/v4l/dev-decoder.rst | 6 +- .../userspace-api/media/v4l/diff-v4l.rst | 2 +- .../userspace-api/media/v4l/open.rst | 2 +- .../media/v4l/vidioc-cropcap.rst | 4 +- Documentation/virt/kvm/api.rst | 28 +- Documentation/vm/zswap.rst | 4 +- Documentation/x86/resctrl.rst | 2 +- Documentation/x86/sgx.rst | 4 +- 82 files changed, 693 insertions(+), 693 deletions(-) -- 2.30.2

4 13

[PATCH 0/7] ALSA: oxfw: code refactoring for quirks specific to ASICs
by Takashi Sakamoto 15 May '21

15 May '21

Hi, This patchset is for code refactoring to ALSA OXFW driver, mainly regarding to ASIC types and quirks. Takashi Sakamoto (7): ALSA: oxfw: code refactoring for existent device entry with specifier_id and version ALSA: oxfw: add device entry for Loud Technologies Tapco Link.FireWire 4x6 ALSA: oxfw: add explicit device entry for Loud Technologies Mackie Onyx Sattelite ALSA: oxfw: add comment for the type of ASICs ALSA: oxfw: code refactoring for jumbo-payload quirk in OXFW970 ALSA: firewire-lib: code refactoring for jumbo payload quirk ALSA: oxfw; code refactoring for wrong_dbs quirk sound/firewire/amdtp-stream.c | 7 +- sound/firewire/oxfw/oxfw-stream.c | 20 ++--- sound/firewire/oxfw/oxfw.c | 123 +++++++++++++----------------- sound/firewire/oxfw/oxfw.h | 10 ++- 4 files changed, 76 insertions(+), 84 deletions(-) -- 2.27.0

1 8

[PATCH v10 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 14 May '21

14 May '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 186 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 83 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 558 ++++++++++++++++++ 7 files changed, 950 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

2 8

[PATCH v9 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 14 May '21

14 May '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 186 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 85 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 558 ++++++++++++++++++ 7 files changed, 952 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

3 10

[PATCH v6] ASoC: topology: Add support for multiple kcontrol types to a widget
by Jaska Uimonen 14 May '21

14 May '21

Current dapm widget has a single variable to describe its kcontrol's type. As there can be many kcontrols in one widget it is inherently presumed that the types are the same. Lately there has been use cases where different types of kcontrols would be needed for a single widget. Thus add pointer to dapm widget to hold an array for different kcontrol types and modify the kcontrol creation to operate in a loop based on individual kcontrol type. Change control creation and deletion to use individual kcontrol types in SOF driver. This is done in the same patch for not breaking bisect. SOF driver is also currently the only one using the dapm widget kcontrol_type. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Hi, This is a patch for enabling multiple different types of kcontrols for a dapm widget. It refactors the dapm kcontrol parsing by lifting the creation loop and memory allocation up one level and making the type variable to hold multiple type values. Sof driver is modified to use this type information in the same patch as it is quite cumbersome to split the changes nicely. changes since v5: - remove extra free reported by coccinelle (alloc was changed to devm_kcalloc) changes since v4: - change title from dapm -> topology - remove cover letter - cc maintainers, me learning list behavior, apologies for the noise changes since v3: - fix sparse warnings for not using le32_to_cpu changes since v2: - only 1 patch for not breaking bisect - sof internal reviews -> minor changes + reviewed by tags - no upstream comments after v2 -> assuming I'm on the right track here Changes since v1: - don't add kcontrol type to core i.e. to kcontrol and kcontrol_news - extend existing dapm widget kcontrol_type into holding an array of types include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 466 +++++++++++++++++------------------ sound/soc/sof/topology.c | 15 +- 3 files changed, 233 insertions(+), 250 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index 328cf763d9b4..4afd667e124c 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -54,7 +54,7 @@ struct snd_soc_dobj_control { /* dynamic widget object */ struct snd_soc_dobj_widget { - unsigned int kcontrol_type; /* kcontrol type: mixer, enum, bytes */ + unsigned int *kcontrol_type; /* kcontrol type: mixer, enum, bytes */ }; /* generic dynamic object - all dynamic objects belong to this struct */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 73076d425efb..e71d98d7b116 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1203,249 +1203,216 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg, return ret; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_dmixer_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { - struct snd_kcontrol_new *kc; struct soc_mixer_control *sm; struct snd_soc_tplg_mixer_control *mc; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (kc == NULL) - return NULL; - - for (i = 0; i < num_kcontrols; i++) { - mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; - - /* validate kcontrol */ - if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_sm; + int err; - sm = devm_kzalloc(tplg->dev, sizeof(*sm), GFP_KERNEL); - if (sm == NULL) - goto err_sm; + mc = (struct snd_soc_tplg_mixer_control *)tplg->pos; - tplg->pos += (sizeof(struct snd_soc_tplg_mixer_control) + - le32_to_cpu(mc->priv.size)); + /* validate kcontrol */ + if (strnlen(mc->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", - mc->hdr.name, i); + sm = devm_kzalloc(tplg->dev, sizeof(*sm), GFP_KERNEL); + if (!sm) + return -ENOMEM; - kc[i].private_value = (long)sm; - kc[i].name = devm_kstrdup(tplg->dev, mc->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_sm; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(mc->hdr.access); + tplg->pos += sizeof(struct snd_soc_tplg_mixer_control) + + le32_to_cpu(mc->priv.size); - /* we only support FL/FR channel mapping atm */ - sm->reg = tplc_chan_get_reg(tplg, mc->channel, - SNDRV_CHMAP_FL); - sm->rreg = tplc_chan_get_reg(tplg, mc->channel, - SNDRV_CHMAP_FR); - sm->shift = tplc_chan_get_shift(tplg, mc->channel, - SNDRV_CHMAP_FL); - sm->rshift = tplc_chan_get_shift(tplg, mc->channel, - SNDRV_CHMAP_FR); + dev_dbg(tplg->dev, " adding DAPM widget mixer control %s\n", + mc->hdr.name); - sm->max = le32_to_cpu(mc->max); - sm->min = le32_to_cpu(mc->min); - sm->invert = le32_to_cpu(mc->invert); - sm->platform_max = le32_to_cpu(mc->platform_max); - sm->dobj.index = tplg->index; - INIT_LIST_HEAD(&sm->dobj.list); - - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&mc->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &mc->hdr, mc->hdr.name); - goto err_sm; - } + kc->private_value = (long)sm; + kc->name = devm_kstrdup(tplg->dev, mc->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(mc->hdr.access); + + /* we only support FL/FR channel mapping atm */ + sm->reg = tplc_chan_get_reg(tplg, mc->channel, + SNDRV_CHMAP_FL); + sm->rreg = tplc_chan_get_reg(tplg, mc->channel, + SNDRV_CHMAP_FR); + sm->shift = tplc_chan_get_shift(tplg, mc->channel, + SNDRV_CHMAP_FL); + sm->rshift = tplc_chan_get_shift(tplg, mc->channel, + SNDRV_CHMAP_FR); + + sm->max = le32_to_cpu(mc->max); + sm->min = le32_to_cpu(mc->min); + sm->invert = le32_to_cpu(mc->invert); + sm->platform_max = le32_to_cpu(mc->platform_max); + sm->dobj.index = tplg->index; + INIT_LIST_HEAD(&sm->dobj.list); + + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&mc->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &mc->hdr, mc->hdr.name); + return err; + } - /* create any TLV data */ - err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", - mc->hdr.name); - goto err_sm; - } + /* create any TLV data */ + err = soc_tplg_create_tlv(tplg, kc, &mc->hdr); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to create TLV %s\n", + mc->hdr.name); + return err; + } - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)mc); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - mc->hdr.name); - goto err_sm; - } + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)mc); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + mc->hdr.name); + return err; } - return kc; -err_sm: - return NULL; + return 0; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_denum_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { - struct snd_kcontrol_new *kc; struct snd_soc_tplg_enum_control *ec; struct soc_enum *se; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (kc == NULL) - return NULL; - - for (i = 0; i < num_kcontrols; i++) { - ec = (struct snd_soc_tplg_enum_control *)tplg->pos; - /* validate kcontrol */ - if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_se; + int err; - se = devm_kzalloc(tplg->dev, sizeof(*se), GFP_KERNEL); - if (se == NULL) - goto err_se; + ec = (struct snd_soc_tplg_enum_control *)tplg->pos; + /* validate kcontrol */ + if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + - le32_to_cpu(ec->priv.size)); + se = devm_kzalloc(tplg->dev, sizeof(*se), GFP_KERNEL); + if (!se) + return -ENOMEM; - dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", - ec->hdr.name); + tplg->pos += (sizeof(struct snd_soc_tplg_enum_control) + + le32_to_cpu(ec->priv.size)); - kc[i].private_value = (long)se; - kc[i].name = devm_kstrdup(tplg->dev, ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_se; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(ec->hdr.access); + dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", + ec->hdr.name); - /* we only support FL/FR channel mapping atm */ - se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); - se->shift_l = tplc_chan_get_shift(tplg, ec->channel, - SNDRV_CHMAP_FL); - se->shift_r = tplc_chan_get_shift(tplg, ec->channel, - SNDRV_CHMAP_FR); + kc->private_value = (long)se; + kc->name = devm_kstrdup(tplg->dev, ec->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(ec->hdr.access); - se->items = le32_to_cpu(ec->items); - se->mask = le32_to_cpu(ec->mask); - se->dobj.index = tplg->index; + /* we only support FL/FR channel mapping atm */ + se->reg = tplc_chan_get_reg(tplg, ec->channel, SNDRV_CHMAP_FL); + se->shift_l = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FL); + se->shift_r = tplc_chan_get_shift(tplg, ec->channel, + SNDRV_CHMAP_FR); - switch (le32_to_cpu(ec->hdr.ops.info)) { - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - err = soc_tplg_denum_create_values(tplg, se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create values for %s\n", - ec->hdr.name); - goto err_se; - } - fallthrough; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - err = soc_tplg_denum_create_texts(tplg, se, ec); - if (err < 0) { - dev_err(tplg->dev, "ASoC: could not create texts for %s\n", - ec->hdr.name); - goto err_se; - } - break; - default: - dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", - ec->hdr.ops.info, ec->hdr.name); - goto err_se; - } + se->items = le32_to_cpu(ec->items); + se->mask = le32_to_cpu(ec->mask); + se->dobj.index = tplg->index; - /* map io handlers */ - err = soc_tplg_kcontrol_bind_io(&ec->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &ec->hdr, ec->hdr.name); - goto err_se; + switch (le32_to_cpu(ec->hdr.ops.info)) { + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + err = soc_tplg_denum_create_values(tplg, se, ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: could not create values for %s\n", + ec->hdr.name); + return err; } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)ec); + fallthrough; + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + err = soc_tplg_denum_create_texts(tplg, se, ec); if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", + dev_err(tplg->dev, "ASoC: could not create texts for %s\n", ec->hdr.name); - goto err_se; + return err; } + break; + default: + dev_err(tplg->dev, "ASoC: invalid enum control type %d for %s\n", + ec->hdr.ops.info, ec->hdr.name); + return -EINVAL; } - return kc; + /* map io handlers */ + err = soc_tplg_kcontrol_bind_io(&ec->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &ec->hdr, ec->hdr.name); + return err; + } -err_se: - return NULL; + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)ec); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + ec->hdr.name); + return err; + } + + return 0; } -static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( - struct soc_tplg *tplg, int num_kcontrols) +static int soc_tplg_dapm_widget_dbytes_create(struct soc_tplg *tplg, struct snd_kcontrol_new *kc) { struct snd_soc_tplg_bytes_control *be; struct soc_bytes_ext *sbe; - struct snd_kcontrol_new *kc; - int i, err; - - kc = devm_kcalloc(tplg->dev, num_kcontrols, sizeof(*kc), GFP_KERNEL); - if (!kc) - return NULL; + int err; - for (i = 0; i < num_kcontrols; i++) { - be = (struct snd_soc_tplg_bytes_control *)tplg->pos; + be = (struct snd_soc_tplg_bytes_control *)tplg->pos; - /* validate kcontrol */ - if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == - SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - goto err_sbe; + /* validate kcontrol */ + if (strnlen(be->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == + SNDRV_CTL_ELEM_ID_NAME_MAXLEN) + return -EINVAL; - sbe = devm_kzalloc(tplg->dev, sizeof(*sbe), GFP_KERNEL); - if (sbe == NULL) - goto err_sbe; + sbe = devm_kzalloc(tplg->dev, sizeof(*sbe), GFP_KERNEL); + if (!sbe) + return -ENOMEM; - tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + - le32_to_cpu(be->priv.size)); + tplg->pos += (sizeof(struct snd_soc_tplg_bytes_control) + + le32_to_cpu(be->priv.size)); - dev_dbg(tplg->dev, - "ASoC: adding bytes kcontrol %s with access 0x%x\n", - be->hdr.name, be->hdr.access); + dev_dbg(tplg->dev, + "ASoC: adding bytes kcontrol %s with access 0x%x\n", + be->hdr.name, be->hdr.access); - kc[i].private_value = (long)sbe; - kc[i].name = devm_kstrdup(tplg->dev, be->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) - goto err_sbe; - kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; - kc[i].access = le32_to_cpu(be->hdr.access); + kc->private_value = (long)sbe; + kc->name = devm_kstrdup(tplg->dev, be->hdr.name, GFP_KERNEL); + if (!kc->name) + return -ENOMEM; + kc->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kc->access = le32_to_cpu(be->hdr.access); - sbe->max = le32_to_cpu(be->max); - INIT_LIST_HEAD(&sbe->dobj.list); + sbe->max = le32_to_cpu(be->max); + INIT_LIST_HEAD(&sbe->dobj.list); - /* map standard io handlers and check for external handlers */ - err = soc_tplg_kcontrol_bind_io(&be->hdr, &kc[i], tplg); - if (err) { - soc_control_err(tplg, &be->hdr, be->hdr.name); - goto err_sbe; - } - - /* pass control to driver for optional further init */ - err = soc_tplg_init_kcontrol(tplg, &kc[i], - (struct snd_soc_tplg_ctl_hdr *)be); - if (err < 0) { - dev_err(tplg->dev, "ASoC: failed to init %s\n", - be->hdr.name); - goto err_sbe; - } + /* map standard io handlers and check for external handlers */ + err = soc_tplg_kcontrol_bind_io(&be->hdr, kc, tplg); + if (err) { + soc_control_err(tplg, &be->hdr, be->hdr.name); + return err; } - return kc; - -err_sbe: + /* pass control to driver for optional further init */ + err = soc_tplg_init_kcontrol(tplg, kc, + (struct snd_soc_tplg_ctl_hdr *)be); + if (err < 0) { + dev_err(tplg->dev, "ASoC: failed to init %s\n", + be->hdr.name); + return err; + } - return NULL; + return 0; } static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, @@ -1455,8 +1422,13 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, struct snd_soc_dapm_widget template, *widget; struct snd_soc_tplg_ctl_hdr *control_hdr; struct snd_soc_card *card = tplg->comp->card; - unsigned int kcontrol_type; + unsigned int *kcontrol_type; + struct snd_kcontrol_new *kc; + int mixer_count = 0; + int bytes_count = 0; + int enum_count = 0; int ret = 0; + int i; if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) @@ -1499,7 +1471,6 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, le32_to_cpu(w->priv.size)); if (w->num_kcontrols == 0) { - kcontrol_type = 0; template.num_kcontrols = 0; goto widget; } @@ -1508,57 +1479,66 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", w->name, w->num_kcontrols, control_hdr->type); - switch (le32_to_cpu(control_hdr->ops.info)) { - case SND_SOC_TPLG_CTL_VOLSW: - case SND_SOC_TPLG_CTL_STROBE: - case SND_SOC_TPLG_CTL_VOLSW_SX: - case SND_SOC_TPLG_CTL_VOLSW_XR_SX: - case SND_SOC_TPLG_CTL_RANGE: - case SND_SOC_TPLG_DAPM_CTL_VOLSW: - kcontrol_type = SND_SOC_TPLG_TYPE_MIXER; /* volume mixer */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_dmixer_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; - goto hdr_err; - } - break; - case SND_SOC_TPLG_CTL_ENUM: - case SND_SOC_TPLG_CTL_ENUM_VALUE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: - case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: - kcontrol_type = SND_SOC_TPLG_TYPE_ENUM; /* enumerated mixer */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_denum_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; - goto hdr_err; - } - break; - case SND_SOC_TPLG_CTL_BYTES: - kcontrol_type = SND_SOC_TPLG_TYPE_BYTES; /* bytes control */ - template.num_kcontrols = le32_to_cpu(w->num_kcontrols); - template.kcontrol_news = - soc_tplg_dapm_widget_dbytes_create(tplg, - template.num_kcontrols); - if (!template.kcontrol_news) { - ret = -ENOMEM; + template.num_kcontrols = le32_to_cpu(w->num_kcontrols); + kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); + if (!kc) + goto err; + + kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), + GFP_KERNEL); + if (!kcontrol_type) + goto err; + + for (i = 0; i < w->num_kcontrols; i++) { + control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + switch (le32_to_cpu(control_hdr->ops.info)) { + case SND_SOC_TPLG_CTL_VOLSW: + case SND_SOC_TPLG_CTL_STROBE: + case SND_SOC_TPLG_CTL_VOLSW_SX: + case SND_SOC_TPLG_CTL_VOLSW_XR_SX: + case SND_SOC_TPLG_CTL_RANGE: + case SND_SOC_TPLG_DAPM_CTL_VOLSW: + /* volume mixer */ + kc[i].index = mixer_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_MIXER; + mixer_count++; + ret = soc_tplg_dapm_widget_dmixer_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + case SND_SOC_TPLG_CTL_ENUM: + case SND_SOC_TPLG_CTL_ENUM_VALUE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: + case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: + /* enumerated mixer */ + kc[i].index = enum_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_ENUM; + enum_count++; + ret = soc_tplg_dapm_widget_denum_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + case SND_SOC_TPLG_CTL_BYTES: + /* bytes control */ + kc[i].index = bytes_count; + kcontrol_type[i] = SND_SOC_TPLG_TYPE_BYTES; + bytes_count++; + ret = soc_tplg_dapm_widget_dbytes_create(tplg, &kc[i]); + if (ret < 0) + goto hdr_err; + break; + default: + dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", + control_hdr->ops.get, control_hdr->ops.put, + le32_to_cpu(control_hdr->ops.info)); + ret = -EINVAL; goto hdr_err; } - break; - default: - dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", - control_hdr->ops.get, control_hdr->ops.put, - le32_to_cpu(control_hdr->ops.info)); - ret = -EINVAL; - goto hdr_err; } + template.kcontrol_news = kc; + widget: ret = soc_tplg_widget_load(tplg, &template, w); if (ret < 0) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 59abcfc9bd55..92d346bbd357 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1063,6 +1063,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, scontrol->min_volume_step = le32_to_cpu(mc->min); scontrol->max_volume_step = le32_to_cpu(mc->max); scontrol->num_channels = le32_to_cpu(mc->num_channels); + scontrol->control_data->index = kc->index; /* set cmd for mixer control */ if (le32_to_cpu(mc->max) == 1) { @@ -1140,7 +1141,7 @@ static int sof_control_load_enum(struct snd_soc_component *scomp, scontrol->comp_id = sdev->next_comp_id; scontrol->num_channels = le32_to_cpu(ec->num_channels); - + scontrol->control_data->index = kc->index; scontrol->cmd = SOF_CTRL_CMD_ENUM; dev_dbg(scomp->dev, "tplg: load kcontrol index %d chans %d comp_id %d\n", @@ -1188,6 +1189,7 @@ static int sof_control_load_bytes(struct snd_soc_component *scomp, scontrol->comp_id = sdev->next_comp_id; scontrol->cmd = SOF_CTRL_CMD_BINARY; + scontrol->control_data->index = kc->index; dev_dbg(scomp->dev, "tplg: load kcontrol index %d chans %d\n", scontrol->comp_id, scontrol->num_channels); @@ -2133,7 +2135,7 @@ static int sof_get_control_data(struct snd_soc_component *scomp, for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; - switch (widget->dobj.widget.kcontrol_type) { + switch (widget->dobj.widget.kcontrol_type[i]) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; wdata[i].control = sm->dobj.private; @@ -2147,8 +2149,8 @@ static int sof_get_control_data(struct snd_soc_component *scomp, wdata[i].control = se->dobj.private; break; default: - dev_err(scomp->dev, "error: unknown kcontrol type %d in widget %s\n", - widget->dobj.widget.kcontrol_type, + dev_err(scomp->dev, "error: unknown kcontrol type %u in widget %s\n", + widget->dobj.widget.kcontrol_type[i], widget->name); return -EINVAL; } @@ -2164,7 +2166,8 @@ static int sof_get_control_data(struct snd_soc_component *scomp, return -EINVAL; /* make sure data is valid - data can be updated at runtime */ - if (wdata[i].pdata->magic != SOF_ABI_MAGIC) + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && + wdata[i].pdata->magic != SOF_ABI_MAGIC) return -EINVAL; *size += wdata[i].pdata->size; @@ -2605,7 +2608,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, } for (i = 0; i < widget->num_kcontrols; i++) { kc = &widget->kcontrol_news[i]; - switch (dobj->widget.kcontrol_type) { + switch (widget->dobj.widget.kcontrol_type[i]) { case SND_SOC_TPLG_TYPE_MIXER: sm = (struct soc_mixer_control *)kc->private_value; scontrol = sm->dobj.private; -- 2.24.1

2 1

[PATCH] ASoC: qcom: lpass-cpu: Fix pop noise during audio capture begin
by Srinivasa Rao Mandadapu 14 May '21

14 May '21

This patch fixes PoP noise of around 15ms observed during audio capture begin. Enables BCLK and LRCLK in snd_soc_dai_ops startup call for introducing some delay before capture start and clock enable. Signed-off-by: Judy Hsiao <judyhsiao(a)chromium.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 37 +++++++++++++++++++++++++++++++++---- 1 file changed, 33 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index c62d2612e8f5..3dcfc2f115a7 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -73,14 +73,28 @@ static int lpass_cpu_daiops_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); - int ret; + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret = -EINVAL; ret = clk_prepare_enable(drvdata->mi2s_osr_clk[dai->driver->id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s osr clk: %d\n", ret); return ret; } - ret = clk_prepare(drvdata->mi2s_bit_clk[dai->driver->id]); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_fields_write(i2sctl->spken, id, + LPAIF_I2SCTL_SPKEN_ENABLE); + } else { + ret = regmap_fields_write(i2sctl->micen, id, + LPAIF_I2SCTL_MICEN_ENABLE); + } + if (ret) + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", + ret); + + ret = clk_prepare_enable(drvdata->mi2s_bit_clk[dai->driver->id]); if (ret) { dev_err(dai->dev, "error in enabling mi2s bit clk: %d\n", ret); clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); @@ -93,9 +107,23 @@ static void lpass_cpu_daiops_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct lpass_data *drvdata = snd_soc_dai_get_drvdata(dai); + struct lpaif_i2sctl *i2sctl = drvdata->i2sctl; + unsigned int id = dai->driver->id; + int ret = -EINVAL; clk_disable_unprepare(drvdata->mi2s_osr_clk[dai->driver->id]); - clk_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + clk_disable_unprepare(drvdata->mi2s_bit_clk[dai->driver->id]); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_fields_write(i2sctl->spken, id, + LPAIF_I2SCTL_SPKEN_DISABLE); + } else { + ret = regmap_fields_write(i2sctl->micen, id, + LPAIF_I2SCTL_MICEN_DISABLE); + } + if (ret) + dev_err(dai->dev, "error writing to i2sctl reg: %d\n", + ret); } static int lpass_cpu_daiops_hw_params(struct snd_pcm_substream *substream, @@ -308,7 +336,8 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, ret); clk_disable(drvdata->mi2s_bit_clk[dai->driver->id]); - + break; + default: break; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

2 3

[PATCH -next] ASoC: intel/boards: add missing MODULE_DEVICE_TABLE
by Zou Wei 14 May '21

14 May '21

This patch adds missing MODULE_DEVICE_TABLE definition which generates correct modalias for automatic loading of this driver when it is built as an external module. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Zou Wei <zou_wei(a)huawei.com> --- sound/soc/intel/boards/sof_da7219_max98373.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index f3cb077..8d1ad89 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -440,6 +440,7 @@ static const struct platform_device_id board_ids[] = { }, { } }; +MODULE_DEVICE_TABLE(platform, board_ids); static struct platform_driver audio = { .probe = audio_probe, diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 58548ea..cf1d053 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -968,6 +968,7 @@ static const struct platform_device_id board_ids[] = { }, { } }; +MODULE_DEVICE_TABLE(platform, board_ids); static struct platform_driver sof_audio = { .probe = sof_audio_probe, -- 2.6.2

3 4

[PATCH AUTOSEL 5.12 090/116] ASoC: rt286: Generalize support for ALC3263 codec
by Sasha Levin 14 May '21

14 May '21

From: David Ward <david.ward(a)gatech.edu> [ Upstream commit aa2f9c12821e6a4ba1df4fb34a3dbc6a2a1ee7fe ] The ALC3263 codec on the XPS 13 9343 is also found on the Latitude 13 7350 and Venue 11 Pro 7140. They require the same handling for the combo jack to work with a headset: GPIO pin 6 must be set. The HDA driver always sets this pin on the ALC3263, which it distinguishes by the codec vendor/device ID 0x10ec0288 and PCI subsystem vendor ID 0x1028 (Dell). The ASoC driver does not use PCI, so adapt this check to use DMI to determine if Dell is the system vendor. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=150601 BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205961 Signed-off-by: David Ward <david.ward(a)gatech.edu> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210418134658.4333-6-david.ward@gatech.edu Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt286.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8abe232ca4a4..2f18f5114f7e 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1117,12 +1117,11 @@ static const struct dmi_system_id force_combo_jack_table[] = { { } }; -static const struct dmi_system_id dmi_dell_dino[] = { +static const struct dmi_system_id dmi_dell[] = { { - .ident = "Dell Dino", + .ident = "Dell", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc."), - DMI_MATCH(DMI_PRODUCT_NAME, "XPS 13 9343") } }, { } @@ -1133,7 +1132,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, { struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); struct rt286_priv *rt286; - int i, ret, val; + int i, ret, vendor_id; rt286 = devm_kzalloc(&i2c->dev, sizeof(*rt286), GFP_KERNEL); @@ -1149,14 +1148,15 @@ static int rt286_i2c_probe(struct i2c_client *i2c, } ret = regmap_read(rt286->regmap, - RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &val); + RT286_GET_PARAM(AC_NODE_ROOT, AC_PAR_VENDOR_ID), &vendor_id); if (ret != 0) { dev_err(&i2c->dev, "I2C error %d\n", ret); return ret; } - if (val != RT286_VENDOR_ID && val != RT288_VENDOR_ID) { + if (vendor_id != RT286_VENDOR_ID && vendor_id != RT288_VENDOR_ID) { dev_err(&i2c->dev, - "Device with ID register %#x is not rt286\n", val); + "Device with ID register %#x is not rt286\n", + vendor_id); return -ENODEV; } @@ -1180,8 +1180,8 @@ static int rt286_i2c_probe(struct i2c_client *i2c, if (pdata) rt286->pdata = *pdata; - if (dmi_check_system(force_combo_jack_table) || - dmi_check_system(dmi_dell_dino)) + if ((vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) || + dmi_check_system(force_combo_jack_table)) rt286->pdata.cbj_en = true; regmap_write(rt286->regmap, RT286_SET_AUDIO_POWER, AC_PWRST_D3); @@ -1220,7 +1220,7 @@ static int rt286_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL3, 0xf777, 0x4737); regmap_update_bits(rt286->regmap, RT286_DEPOP_CTRL4, 0x00ff, 0x003f); - if (dmi_check_system(dmi_dell_dino)) { + if (vendor_id == RT288_VENDOR_ID && dmi_check_system(dmi_dell)) { regmap_update_bits(rt286->regmap, RT286_SET_GPIO_MASK, 0x40, 0x40); regmap_update_bits(rt286->regmap, -- 2.30.2

2 2

Add some CLOVE SSIDs of ALC293
by Pshou 14 May '21

14 May '21

Hi Alsa-devel ML: Can you help me add some source code to patch_realtek.c? Add some CLEVO SSID of alc293. Best regards Pshou

2 3

Add some CLOVE SSIDs of ALC293
by Pshou 14 May '21

14 May '21

From: Pshou Sent: Friday, May 14, 2021 1:23 PM To: 'alsa-devel(a)alsa-project.org' <alsa-devel(a)alsa-project.org> Cc: Kailang <kailang(a)realtek.com>; 'Takashi Iwai' <tiwai(a)suse.de>; Steve Wang <chirowang(a)realtek.com> Subject: Add some CLOVE SSIDs of ALC293 Hi Alsa-devel ML: Can you help me add some source code to patch_realtek.c? Add some CLEVO SSID of alc293. Best regards Pshou

1 0

[PATCH -next] ALSA: gus: Replace unsafe strcpy() with strscpy()
by YueHaibing 14 May '21

14 May '21

Fix smatch warning: sound/isa/gus/gus_main.c:396 snd_gus_check_version() error: strcpy() 'card->longname' too large for 'card->shortname' (80 vs 32) Even if this is not a real bug since the longest length of card->longname now is 31, replace strcpy() with strscpy() in order to avoid possible future mistake. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/isa/gus/gus_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index b7518122a10d..7524ce5785de 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -384,7 +384,7 @@ static int snd_gus_check_version(struct snd_gus_card * gus) } } } - strcpy(card->shortname, card->longname); + strscpy(card->shortname, card->longname, sizeof(card->shortname)); gus->uart_enable = 1; /* standard GUSes doesn't have midi uart trouble */ snd_gus_init_control(gus); return 0; -- 2.17.1

2 1

[PATCH 0/5] ALSA: firewire: misc fixes
by Takashi Sakamoto 14 May '21

14 May '21

Hi, This patchset includes misc fixes for bebob, oxfw, dice, and firewire-lib modules. The most significant bug to be fixed was introduced by my work for AMDTP domain during 2019-2020. Takashi Sakamoto (5): ALSA: dice: fix stream format at middle sampling rate for Alesis iO 26 ALSA: bebob/oxfw: fix Kconfig entry for Mackie d.2 Pro ALSA: firewire-lib: fix check of jumbo payload ALSA: firewire-lib: fix calculation for size of IR context payload ALSA: firewire-lib: fix amdtp_packet tracepoints event in terms of packet_index sound/firewire/Kconfig | 4 +-- sound/firewire/amdtp-stream-trace.h | 6 ++--- sound/firewire/amdtp-stream.c | 42 +++++++++++++++++------------ sound/firewire/bebob/bebob.c | 2 +- sound/firewire/dice/dice-alesis.c | 2 +- sound/firewire/oxfw/oxfw.c | 1 - 6 files changed, 32 insertions(+), 25 deletions(-) -- 2.27.0

2 6

[PATCH v2] ALSA: hda: generic: Remove redundant assignment to dac
by Jiapeng Chong 14 May '21

14 May '21

Variable dac is set to zero, but this value is never read as it is overwritten or not used later on, hence it is a redundant assignment and can be removed. Clean up the following clang-analyzer warning: sound/pci/hda/hda_generic.c:1436:4: warning: Value stored to 'dac' is never read [clang-analyzer-deadcode.DeadStores]. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- Changes in v2: -For the follow advice: https://lore.kernel.org/patchwork/patch/1423536/ sound/pci/hda/hda_generic.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b638fc2..fc02957 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1433,7 +1433,7 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, path = snd_hda_add_new_path(codec, dac, pin, 0); } if (!path) { - dac = dacs[i] = 0; + dacs[i] = 0; badness += bad->no_dac; } else { /* print_nid_path(codec, "output", path); */ -- 1.8.3.1

2 1

[PATCH] ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware
by Ranjani Sridharan 13 May '21

13 May '21

BE hw_params op was recently added for SSP type DAIs. But sending the DAI_CONFIG IPC during hw_params is not supported with older firmware. So add an ABI check to avoid sending the IPC if the firmware ABI is older than 3.18. Fixes: e12be9fbfb91 ('ASoC: SOF: Intel: HDA: add hw params callback for SSP DAIs') Tested-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 8d7bab433fb3..c1f9f0f58464 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -421,11 +421,16 @@ static int ssp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct sof_ipc_dai_config *config; struct snd_sof_dai *sof_dai; struct sof_ipc_reply reply; int ret; + /* DAI_CONFIG IPC during hw_params is not supported in older firmware */ + if (v->abi_version < SOF_ABI_VER(3, 18, 0)) + return 0; + list_for_each_entry(sof_dai, &sdev->dai_list, list) { if (!sof_dai->cpu_dai_name || !sof_dai->dai_config) continue; -- 2.25.1

1 0

X-Fi USB HD - switching from USB1.1 to USB2?
by Pavel Hofman 13 May '21

13 May '21

Hi, The X-Fi USB HD device (and likely the similar 5.1 model too) can run at USB1.1 (with limited samplerates) and USB2 modes (with full specs). In linux the device stays at USB1.1. If connected to Windows with stock windows USB driver the same happens. However, when the Creative proprietary driver is installed, the device switches to USB2 and supports all features. When rebooting to linux without re-plugging, the USB2 mode stays and the device offers its USB2 configurations to snd-usb-audio. We captured USB packets in linux and Win7 with wireshark - both dumps are included. The windows dump shows that in windows as well as in linux the device starts as device 3.5.0 at USB1.1: DEVICE DESCRIPTOR bLength: 18 bDescriptorType: 0x01 (DEVICE) bcdUSB: 0x0110 bDeviceClass: Device (0x00) bDeviceSubClass: 0 bDeviceProtocol: 0 (Use class code info from Interface Descriptors) bMaxPacketSize0: 64 idVendor: Creative Technology, Ltd (0x041e) idProduct: Unknown (0x30d7) bcdDevice: 0x0100 iManufacturer: 1 iProduct: 2 iSerialNumber: 3 bNumConfigurations: 1 After USB1.1 enumeration, the windows host controlled by the vendor driver sends a request USR_FUNCTION_ABORT_PIPE to EP IN 6: The EP IN 6 descriptor (from USB1.1 configuration): ENDPOINT DESCRIPTOR bLength: 7 bDescriptorType: 0x05 (ENDPOINT) bEndpointAddress: 0x86 IN Endpoint:6 1... .... = Direction: IN Endpoint .... 0110 = Endpoint Number: 0x6 bmAttributes: 0x03 .... ..11 = Transfertype: Interrupt-Transfer (0x3) wMaxPacketSize: 64 ...0 0... .... .... = Transactions per microframe: 1 (0) .... ..00 0100 0000 = Maximum Packet Size: 64 bInterval: 1 The request sent by the host is: USB URB [Source: host] [Destination: 3.5.6] USBPcap pseudoheader length: 27 IRP ID: 0xfffffa80073ee120 IRP USBD_STATUS: USBD_STATUS_SUCCESS (0x00000000) URB Function: URB_FUNCTION_ABORT_PIPE (0x0002) IRP information: 0x00, Direction: FDO -> PDO URB bus id: 3 Device address: 5 Endpoint: 0x86, Direction: IN URB transfer type: USB IRP Info (0xfe) Packet Data Length: 0 The device responds with a proprietary response, with a different source 3.5.127: USB URB [Source: 3.5.127] [Destination: host] USBPcap pseudoheader length: 27 IRP ID: 0xfffffa80073ee120 IRP USBD_STATUS: USBD_STATUS_SUCCESS (0x00000000) URB Function: URB_FUNCTION_ABORT_PIPE (0x0002) IRP information: 0x01, Direction: PDO -> FDO 0000 000. = Reserved: 0x00 .... ...1 = Direction: PDO -> FDO (0x1) URB bus id: 3 Device address: 5 Endpoint: 0xff, Direction: IN 1... .... = Direction: IN (1) .... 1111 = Endpoint number: 15 URB transfer type: Unknown (0xff) [Expert Info (Error/Malformed): USBPcap did not recognize URB Function code (report to desowin.org/USBPcap)] [USBPcap did not recognize URB Function code (report to desowin.org/USBPcap)] [Severity level: Error] [Group: Malformed] Packet Data Length: 0 Immediately after that the device re-connects as 3.6.0 and reports its USB2.0 description: DEVICE DESCRIPTOR bLength: 18 bDescriptorType: 0x01 (DEVICE) bcdUSB: 0x0200 bDeviceClass: Device (0x00) bDeviceSubClass: 0 bDeviceProtocol: 0 (Use class code info from Interface Descriptors) bMaxPacketSize0: 64 idVendor: Creative Technology, Ltd (0x041e) idProduct: Unknown (0x30d7) bcdDevice: 0x0100 iManufacturer: 1 iProduct: 2 iSerialNumber: 3 bNumConfigurations: 1 This is a regular USB-audio device which works OK in snd-usb-audio, as tested after rebooting from windows to linux. Please is there any similar case handled by the existing alsa usb driver which we could try to modify and test? My USB skills are not up to writing the required code myself from scratch, unfortunately. Thanks a lot, Pavel.

2 3

no_period_wakeup, axfer and --sched-model=timer
by Amadeusz Sławiński 13 May '21

13 May '21

Hi, I was checking some stuff relater to NO_PERIOD_WAKEUP and noticed that axfer has support for using either --sched-model=irq or --sched-model=timer. However from few quick tests it seems like it doesn't work? $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC283 Analog [ALC283 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 When using --sched-model=irq it transfers data until I press Ctrl+C $ axfer transfer playback --sched-model=irq -D hw:0,0 -r 48000 -c2 -f S16_LE /dev/urandom PLAYBACK: Format 'Signed 16 bit Little Endian', Rate 48000 Hz, Channels 'Stereo' ^CPLAYBACK: Expected 4611686018427387903frames, Actual 163960frames Aborted by signal: Interrupt However with --sched-model=timer it time outs by itself: $ axfer transfer playback --sched-model=timer -D hw:0,0 -r 48000 -c2 -f S16_LE /dev/urandom PLAYBACK: Format 'Signed 16 bit Little Endian', Rate 48000 Hz, Channels 'Stereo' Fail to process frames: Connection timed out PLAYBACK: Expected 4611686018427387903frames, Actual 16304frames How well is NO_PERIOD_WAKEUP tested/supported? Is it a bug in axfer or perhaps some issue in kernel code? From some debugging I did, I have my suspicions that it gets stuck on poll in: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… waiting for runtime->sleep to wake it, but seems like it never happens. What do you think? Amadeusz

3 5

alsa-lib dmix segfault; latest Git
by Mark Hills 13 May '21

13 May '21

The current Git master looks very broken, segfaulting immediately. It seems to be caused by commit 27f4a85a95 ("pcm: direct - move the direct struct init to _snd_pcm_direct_new()") Here's the trace: $ aplay -D plug:master testing.wav Program received signal SIGSEGV, Segmentation fault. 0x00007ffff7f14ea6 in snd_pcm_dmix_open (pcmp=0x7fffffffe170, name=0x0, opts=0x7fffffffdef0, params=0x7fffffffdf30, root=0x43b1e0, sconf=0x455ae0, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm_dmix.c:1017 1017 pcm->ops = &snd_pcm_dmix_ops; (gdb) bt #0 0x00007ffff7f14ea6 in snd_pcm_dmix_open (pcmp=0x7fffffffe170, name=0x0, opts=0x7fffffffdef0, params=0x7fffffffdf30, root=0x43b1e0, sconf=0x455ae0, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm_dmix.c:1017 #1 0x00007ffff7f15763 in _snd_pcm_dmix_open (pcmp=0x7fffffffe170, name=0x0, root=0x43b1e0, conf=0x43b620, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm_dmix.c:1374 #2 0x00007ffff7ec72d8 in snd_pcm_open_conf (pcmp=0x7fffffffe170, name=0x0, pcm_root=0x43b1e0, pcm_conf=0x43b620, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm.c:2609 #3 0x00007ffff7ec7a1b in snd1_pcm_open_named_slave (pcmp=0x7fffffffe170, name=0x0, root=0x43b1e0, conf=0x43b620, stream=SND_PCM_STREAM_PLAYBACK, mode=0, parent_conf=0x43c390) at pcm.c:2823 #4 0x00007ffff7ee38a3 in snd_pcm_open_slave (pcmp=0x7fffffffe170, root=0x43b1e0, conf=0x43b620, stream=SND_PCM_STREAM_PLAYBACK, mode=0, parent_conf=0x43c390) at pcm_local.h:1008 #5 0x00007ffff7ee87a3 in _snd_pcm_route_open (pcmp=0x7fffffffe420, name=0x43cc50 "master", root=0x43b1e0, conf=0x43c390, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm_route.c:1367 #6 0x00007ffff7ec72d8 in snd_pcm_open_conf (pcmp=0x7fffffffe420, name=0x43cc50 "master", pcm_root=0x43b1e0, pcm_conf=0x43c390, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm.c:2609 #7 0x00007ffff7ec7547 in snd_pcm_open_noupdate (pcmp=0x7fffffffe420, root=0x43b1e0, name=0x43cc50 "master", stream=SND_PCM_STREAM_PLAYBACK, mode=0, hop=1) at pcm.c:2668 #8 0x00007ffff7ec79f4 in snd1_pcm_open_named_slave (pcmp=0x7fffffffe420, name=0x0, root=0x43b1e0, conf=0x43c320, stream=SND_PCM_STREAM_PLAYBACK, mode=0, parent_conf=0x43d120) at pcm.c:2821 #9 0x00007ffff7ef57ba in snd_pcm_open_slave (pcmp=0x7fffffffe420, root=0x43b1e0, conf=0x43c320, stream=SND_PCM_STREAM_PLAYBACK, mode=0, parent_conf=0x43d120) at pcm_local.h:1008 #10 0x00007ffff7ef824e in _snd_pcm_plug_open (pcmp=0x413768 <handle>, name=0x7fffffffebbb "plug:master", root=0x43b1e0, conf=0x43d120, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm_plug.c:1324 #11 0x00007ffff7ec72d8 in snd_pcm_open_conf (pcmp=0x413768 <handle>, name=0x7fffffffebbb "plug:master", pcm_root=0x43b1e0, pcm_conf=0x43d120, stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm.c:2609 #12 0x00007ffff7ec7547 in snd_pcm_open_noupdate (pcmp=0x413768 <handle>, root=0x43b1e0, name=0x7fffffffebbb "plug:master", stream=SND_PCM_STREAM_PLAYBACK, mode=0, hop=0) at pcm.c:2668 #13 0x00007ffff7ec7610 in snd_pcm_open (pcmp=0x413768 <handle>, name=0x7fffffffebbb "plug:master", stream=SND_PCM_STREAM_PLAYBACK, mode=0) at pcm.c:2698 #14 0x00000000004054ab in main (argc=4, argv=0x7fffffffe898) at aplay.c:850 -- Mark

2 1

pulse.conf overrides ~/.asoundrc for default
by GitHub issues - edited 13 May '21

13 May '21

alsa-project/alsa-lib issue #134 was edited from flatmax: On many systems pulse audio sets the default device to pulse. This overrides the settings in ~/.asoundrc. This is further complicated by using a tool like pasuspender and finding that pulse's default device is still set and overriding ~/.asoundrc It would seem more empowering to let the user override the default device in ~/.asoundrc and have that take effect over pulse.conf which typically resides in /usr/share/alsa/alsa.conf.d For that matter, it would be more empowering to the user to override everything (well almost any thing) using ~/.asoundrc, whether it is in alsa,conf.d or elsewhere. Issue URL : https://github.com/alsa-project/alsa-lib/issues/134 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pulse.conf overrides ~/.asoundrc for default
by GitHub issues - edited 13 May '21

13 May '21

alsa-project/alsa-lib issue #134 was edited from flatmax: On many systems pulse audio sets the default device to pulse. This overrides the settings in ~/.asoundrc. This is further complicated by using a tool like pasuspender and finding that pulse's default device is still set and overriding ~/.asoundrc It would seem more empowering to let the user override the default device in ~/.asoundrc and have that take effect over pulse.conf which typically resides in /usr/share/alsa/alsa.conf.d For that matter, it would be more empowering to the user to override anything using ~/.asoundrc, whether it is in alsa,conf.d or elsewhere. Issue URL : https://github.com/alsa-project/alsa-lib/issues/134 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pulse.conf overrides ~/.asoundrc for default
by GitHub issues - edited 13 May '21

13 May '21

alsa-project/alsa-lib issue #134 was edited from flatmax: On many systems pulse audio sets the default device to pulse. This overrides the settings in ~/.asoundrc. This is further complicated by using a tool like pasuspender and finding that pulse's default device is still set and overriding ~/.asoundrc It would seem more empowering to let the user override the default device in ~/.asoundrc and have that take effect over pulse.conf which typically resides in /usr/share/alsa/alsa.conf.d Issue URL : https://github.com/alsa-project/alsa-lib/issues/134 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pulse.conf overrides ~/.asoundrc for default
by GitHub issues - opened 13 May '21

13 May '21

alsa-project/alsa-lib issue #134 was opened from flatmax: On many systems pulse audio sets the default device to pulse. This overrides the settings in ~/.asoundrc. This is further complicated by using a tool like pasuspender and finding that pulse's default device is still set and overriding ~/.asoundrc It would seem more empowering to let the user override the default device in ~/.asoundrc and have that take effect over pulse.conf which resides in /usr/share/alsa/alsa.conf.d Issue URL : https://github.com/alsa-project/alsa-lib/issues/134 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

re: ASoC: q6dsp: q6afe: remove unneeded dead-store initialization
by Colin Ian King 12 May '21

12 May '21

Hi, Static analysis with Coverity on linux-next has detected an issue with the following commit: commit 5f1b95d08de712327e452d082a50fded435ec884 Author: Yang Li <yang.lee(a)linux.alibaba.com> Date: Sun Apr 25 18:12:33 2021 +0800 ASoC: q6dsp: q6afe: remove unneeded dead-store initialization the analysis is as follows: 1181 int q6afe_port_stop(struct q6afe_port *port) 1182 { 1183 struct afe_port_cmd_device_stop *stop; 1184 struct q6afe *afe = port->afe; 1185 struct apr_pkt *pkt; 1. var_decl: Declaring variable port_id without initializer. 1186 int port_id; 1187 int ret = 0; 1188 int index, pkt_size; 1189 void *p; 1190 1191 index = port->token; 2. Condition index < 0, taking false branch. 3. Condition index >= 127, taking false branch. 1192 if (index < 0 || index >= AFE_PORT_MAX) { 1193 dev_err(afe->dev, "AFE port index[%d] invalid!\n", index); 1194 return -EINVAL; 1195 } 1196 1197 pkt_size = APR_HDR_SIZE + sizeof(*stop); 1198 p = kzalloc(pkt_size, GFP_KERNEL); 4. Condition !p, taking false branch. 1199 if (!p) 1200 return -ENOMEM; 1201 1202 pkt = p; 1203 stop = p + APR_HDR_SIZE; 1204 1205 pkt->hdr.hdr_field = APR_HDR_FIELD(APR_MSG_TYPE_SEQ_CMD, 1206 APR_HDR_LEN(APR_HDR_SIZE), 1207 APR_PKT_VER); 1208 pkt->hdr.pkt_size = pkt_size; 1209 pkt->hdr.src_port = 0; 1210 pkt->hdr.dest_port = 0; 1211 pkt->hdr.token = index; 1212 pkt->hdr.opcode = AFE_PORT_CMD_DEVICE_STOP; Uninitialized scalar variable (UNINIT) 5. uninit_use: Using uninitialized value port_id. 1213 stop->port_id = port_id; 1214 stop->reserved = 0; the commit removed the initialization of port_id = port->id, and now we have a regression where stop->port_id is being assigned with a garbage uninitialized value in port_id. Perhaps the fix needs reverting. As it stands, I don't know why clang was reporting this as an error. Colin

2 1

[PATCH AUTOSEL 4.4 4/4] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index dcefb12557f1..7fed8d1bb79c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1182,11 +1182,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

1 0

[PATCH AUTOSEL 4.9 6/7] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 6089ed6efc8d..8d99ac931ff6 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1165,11 +1165,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

1 0

[PATCH AUTOSEL 4.14 10/12] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 14881fe80a21..1833deefe1af 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1212,11 +1212,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

1 0

[PATCH AUTOSEL 4.19 15/18] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index b9f7b23ae378..6099a9f1cb3d 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1214,11 +1214,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

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[PATCH AUTOSEL 5.4 20/23] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index efceeae09045..7ac3f04ca8c0 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1202,11 +1202,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

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[PATCH AUTOSEL 5.10 31/34] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 96903295a967..7c49a7e92dd2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1202,11 +1202,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

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[PATCH AUTOSEL 5.11 32/35] ALSA: hda: generic: change the DAC ctl name for LO+SPK or LO+HP
by Sasha Levin 12 May '21

12 May '21

From: Hui Wang <hui.wang(a)canonical.com> [ Upstream commit f48652bbe3ae62ba2835a396b7e01f063e51c4cd ] Without this change, the DAC ctl's name could be changed only when the machine has both Speaker and Headphone, but we met some machines which only has Lineout and Headhpone, and the Lineout and Headphone share the Audio Mixer0 and DAC0, the ctl's name is set to "Front". On most of machines, the "Front" is used for Speaker only or Lineout only, but on this machine it is shared by Lineout and Headphone, This introduces an issue in the pipewire and pulseaudio, suppose users want the Headphone to be on and the Speaker/Lineout to be off, they could turn off the "Front", this works on most of the machines, but on this machine, the "Front" couldn't be turned off otherwise the headphone will be off too. Here we do some change to let the ctl's name change to "Headphone+LO" on this machine, and pipewire and pulseaudio already could handle "Headphone+LO" and "Speaker+LO". (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/747) BugLink: http://bugs.launchpad.net/bugs/804178 Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Link: https://lore.kernel.org/r/20210504073917.22406-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_generic.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 96903295a967..7c49a7e92dd2 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -1202,11 +1202,17 @@ static const char *get_line_out_pfx(struct hda_codec *codec, int ch, *index = ch; return "Headphone"; case AUTO_PIN_LINE_OUT: - /* This deals with the case where we have two DACs and - * one LO, one HP and one Speaker */ - if (!ch && cfg->speaker_outs && cfg->hp_outs) { - bool hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); - bool spk_lo_shared = !path_has_mixer(codec, spec->speaker_paths[0], ctl_type); + /* This deals with the case where one HP or one Speaker or + * one HP + one Speaker need to share the DAC with LO + */ + if (!ch) { + bool hp_lo_shared = false, spk_lo_shared = false; + + if (cfg->speaker_outs) + spk_lo_shared = !path_has_mixer(codec, + spec->speaker_paths[0], ctl_type); + if (cfg->hp_outs) + hp_lo_shared = !path_has_mixer(codec, spec->hp_paths[0], ctl_type); if (hp_lo_shared && spk_lo_shared) return spec->vmaster_mute.hook ? "PCM" : "Master"; if (hp_lo_shared) -- 2.30.2

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