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

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or aplaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients (aseqdump) may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or aplaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc1... rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some applications , like browsers, can recover by readadjusting the timeline and restarting playback. Some can only recover after closing the application i.e mpv media player. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. some application can recover by readjusting the timeline and restarting playback. Some can only recover after closing the application. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in almost all other applications there is no error, such a web browsers as playing videos in Youtube, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Audio stops and soon after the error appears stopping the video. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Video stops, audio stops and soon after the error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Video stops, audio stops and rrror appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used for audio playback and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio" 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: (A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio") 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers , as playing videos in Youtube,, the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: (A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio") 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. in other applications such a web browsers the audio playback just stops. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP & sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: (A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio") 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop but with kernel 6.5rc7 and ALSA UMP sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: (A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio") 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop with kernel 6.5rc7 and ALSA UMP sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device Note: (A Focusrite 18i20 Gen 1 audio interface is used and set in Pavucontrol as "pro Audio") 4. play a video with the "video" application. 5. run aplaymidi -l or palaymidi --version 6. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop with kernel 6.5rc7 and ALSA UMP sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device 3. play a video with the "video" application. 4. run aplaymidi -l or palaymidi --version 5. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients may also cause this error. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - edited 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was edited from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop with kernel 6.5rc7 and ALSA UMP sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device 3. play a video with the "video" application. 4. run aplaymidi -l or palaymidi --version 5. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear) Repeat steps 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) This is not unique to aplaymidi, other ALSA clients also cause this error. Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplaymidi causes pa_stream_writeable_size() failed - connection terminated
by GitHub issues - opened 23 Aug '23

23 Aug '23

alsa-project/alsa-utils issue #232 was opened from symdeb: How to reproduce: 1. Plain Ubuntu 23.04 desktop with kernel 6.5rc7 and ALSA UMP sequencer enabled. 2. Insert a MIDI 2.0 UMP capable device 3. play a video with the "video" application. 4. run aplaymidi -l or palaymidi --version 5. Error appears. Unplug the device Wait for about 30 seconds (this is important, sooner the error will appear Repeat step 3 and 4 . The error won't occur. ![image](https://github.com/alsa-project/alsa-utils/assets/19564066/9bd6e993-be47-4715-8e6c-5f2a2242027c) Issue URL : https://github.com/alsa-project/alsa-utils/issues/232 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: fsl: merge DAI call back functions into ops
by Kuninori Morimoto 22 Aug '23

22 Aug '23

ALSA SoC merges DAI call backs into .ops. This patch merge these into one. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308152047.psX1QNDh-lkp@intel.com/ Cc: Randy Dunlap <rdunlap(a)infradead.org> Fixes: 446b31e89493 ("ASoC: soc-dai.h: remove unused call back functions") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 40a4a2667394..1671bcd4ee3d 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -222,6 +222,7 @@ static int psc_ac97_probe(struct snd_soc_dai *cpu_dai) * psc_ac97_dai_template: template CPU Digital Audio Interface */ static const struct snd_soc_dai_ops psc_ac97_analog_ops = { + .probe = psc_ac97_probe, .hw_params = psc_ac97_hw_analog_params, .trigger = psc_ac97_trigger, }; @@ -233,7 +234,6 @@ static const struct snd_soc_dai_ops psc_ac97_digital_ops = { static struct snd_soc_dai_driver psc_ac97_dai[] = { { .name = "mpc5200-psc-ac97.0", - .probe = psc_ac97_probe, .playback = { .stream_name = "AC97 Playback", .channels_min = 1, -- 2.25.1

3 2

[PATCH] ASoC: pxa: merge DAI call back functions into ops
by Kuninori Morimoto 22 Aug '23

22 Aug '23

ALSA SoC merges DAI call backs into .ops. This patch merge these into one. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202308151142.hoM5o9LV-lkp@intel.com/ Fixes: 446b31e89493 ("ASoC: soc-dai.h: remove unused call back functions") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/pxa/pxa2xx-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 10636506b622..437bfccd04f8 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -329,6 +329,8 @@ static int pxa2xx_i2s_remove(struct snd_soc_dai *dai) SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { + .probe = pxa2xx_i2s_probe, + .remove = pxa2xx_i2s_remove, .startup = pxa2xx_i2s_startup, .shutdown = pxa2xx_i2s_shutdown, .trigger = pxa2xx_i2s_trigger, @@ -338,8 +340,6 @@ static const struct snd_soc_dai_ops pxa_i2s_dai_ops = { }; static struct snd_soc_dai_driver pxa_i2s_dai = { - .probe = pxa2xx_i2s_probe, - .remove = pxa2xx_i2s_remove, .playback = { .channels_min = 2, .channels_max = 2, -- 2.25.1

2 1

[PATCH] ASoC: SOF: ipc4-topology: Add module parameter to ignore the CPC value
by Peter Ujfalusi 22 Aug '23

22 Aug '23

Add a new module parameter ipc4_ignore_cpc which can be used to force the kernel to ignore the queried CPC value for all firmware modules and use 0 instead. The CPC lookup is still done to report missing configurations and the debug print is going to be different to be explicit that the CPC is ignored and what was the value we would have used. The CPC value is sent to the firmware with the MOD_INIT_INSTANCE message and it is used by the firmware as a parameter for clock scaling. The flag is intended to be used only when there is a need to validate the firmware behavior regarding to clock scaling since the 0 CPC value will force the DSP to run in full speed, disabling the scaling and provides additional counter point to rule out clock management related issues. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 26 +++++++++++++++++++++++--- 1 file changed, 23 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 633f6040d712..733e5bff59a1 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -17,6 +17,19 @@ #include "ipc4-topology.h" #include "ops.h" +/* + * The ignore_cpc flag can be used to ignore the CPC value for all modules by + * using 0 instead. + * The CPC is sent to the firmware along with the SOF_IPC4_MOD_INIT_INSTANCE + * message and it is used for clock scaling. + * 0 as CPC value will instruct the firmware to use maximum frequency, thus + * deactivating the clock scaling. + */ +static bool ignore_cpc; +module_param_named(ipc4_ignore_cpc, ignore_cpc, bool, 0444); +MODULE_PARM_DESC(ipc4_ignore_cpc, + "Ignore CPC values. This option will disable clock scaling in firmware."); + #define SOF_IPC4_GAIN_PARAM_ID 0 #define SOF_IPC4_TPLG_ABI_SIZE 6 #define SOF_IPC4_CHAIN_DMA_BUF_SIZE_MS 2 @@ -970,9 +983,16 @@ sof_ipc4_update_resource_usage(struct snd_sof_dev *sdev, struct snd_sof_widget * /* Update base_config->cpc from the module manifest */ sof_ipc4_update_cpc_from_manifest(sdev, fw_module, base_config); - dev_dbg(sdev->dev, "%s: ibs / obs / cpc: %u / %u / %u\n", - swidget->widget->name, base_config->ibs, base_config->obs, - base_config->cpc); + if (ignore_cpc) { + dev_dbg(sdev->dev, "%s: ibs / obs: %u / %u, forcing cpc to 0 from %u\n", + swidget->widget->name, base_config->ibs, base_config->obs, + base_config->cpc); + base_config->cpc = 0; + } else { + dev_dbg(sdev->dev, "%s: ibs / obs / cpc: %u / %u / %u\n", + swidget->widget->name, base_config->ibs, base_config->obs, + base_config->cpc); + } } static int sof_ipc4_widget_assign_instance_id(struct snd_sof_dev *sdev, -- 2.41.0

2 1

[PATCH 1/3] ASoC: SOF: amd: Add sof support for vangogh platform
by Venkata Prasad Potturu 22 Aug '23

22 Aug '23

Add pci driver and platform driver to enable SOF support on ACP5x architecture based Vangogh platform. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/sof/amd/Kconfig | 12 ++- sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 8 +- sound/soc/sof/amd/acp.h | 9 +- sound/soc/sof/amd/pci-vangogh.c | 105 +++++++++++++++++++ sound/soc/sof/amd/vangogh.c | 156 +++++++++++++++++++++++++++++ 6 files changed, 290 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/amd/pci-vangogh.c create mode 100644 sound/soc/sof/amd/vangogh.c diff --git a/sound/soc/sof/amd/Kconfig b/sound/soc/sof/amd/Kconfig index 7dbc8df5cfe6..f2faa08f0c0e 100644 --- a/sound/soc/sof/amd/Kconfig +++ b/sound/soc/sof/amd/Kconfig @@ -2,7 +2,7 @@ # This file is provided under a dual BSD/GPLv2 license. When using or # redistributing this file, you may do so under either license. # -# Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. +# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. config SND_SOC_SOF_AMD_TOPLEVEL tristate "SOF support for AMD audio DSPs" @@ -34,6 +34,16 @@ config SND_SOC_SOF_AMD_RENOIR help Select this option for SOF support on AMD Renoir platform +config SND_SOC_SOF_AMD_VANGOGH + tristate "SOF support for VANGOGH" + depends on SND_SOC_SOF_PCI + select SND_SOC_SOF_AMD_COMMON + help + Select this option for SOF support + on AMD Vangogh platform. + Say Y if you want to enable SOF on Vangogh. + If unsure select "N". + config SND_SOC_SOF_AMD_REMBRANDT tristate "SOF support for REMBRANDT" depends on SND_SOC_SOF_PCI diff --git a/sound/soc/sof/amd/Makefile b/sound/soc/sof/amd/Makefile index ef9f7df4e379..f3b375e67a6f 100644 --- a/sound/soc/sof/amd/Makefile +++ b/sound/soc/sof/amd/Makefile @@ -2,13 +2,15 @@ # This file is provided under a dual BSD/GPLv2 license. When using or # redistributing this file, you may do so under either license. # -# Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. +# Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. snd-sof-amd-acp-objs := acp.o acp-loader.o acp-ipc.o acp-pcm.o acp-stream.o acp-trace.o acp-common.o snd-sof-amd-acp-$(CONFIG_SND_SOC_SOF_ACP_PROBES) = acp-probes.o snd-sof-amd-renoir-objs := pci-rn.o renoir.o snd-sof-amd-rembrandt-objs := pci-rmb.o rembrandt.o +snd-sof-amd-vangogh-objs := pci-vangogh.o vangogh.o obj-$(CONFIG_SND_SOC_SOF_AMD_COMMON) += snd-sof-amd-acp.o obj-$(CONFIG_SND_SOC_SOF_AMD_RENOIR) +=snd-sof-amd-renoir.o obj-$(CONFIG_SND_SOC_SOF_AMD_REMBRANDT) +=snd-sof-amd-rembrandt.o +obj-$(CONFIG_SND_SOC_SOF_AMD_VANGOGH) +=snd-sof-amd-vangogh.o diff --git a/sound/soc/sof/amd/acp-dsp-offset.h b/sound/soc/sof/amd/acp-dsp-offset.h index 920155dee819..19ef2b4c46d6 100644 --- a/sound/soc/sof/amd/acp-dsp-offset.h +++ b/sound/soc/sof/amd/acp-dsp-offset.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. + * Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. * * Author: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> */ @@ -49,27 +49,33 @@ #define ACP_CONTROL 0x1004 #define ACP3X_I2S_PIN_CONFIG 0x1400 +#define ACP5X_I2S_PIN_CONFIG 0x1400 #define ACP6X_I2S_PIN_CONFIG 0x1440 /* Registers offsets from ACP_PGFSM block */ #define ACP3X_PGFSM_BASE 0x141C +#define ACP5X_PGFSM_BASE 0x1424 #define ACP6X_PGFSM_BASE 0x1024 #define PGFSM_CONTROL_OFFSET 0x0 #define PGFSM_STATUS_OFFSET 0x4 #define ACP3X_CLKMUX_SEL 0x1424 +#define ACP5X_CLKMUX_SEL 0x142C #define ACP6X_CLKMUX_SEL 0x102C /* Registers from ACP_INTR block */ #define ACP3X_EXT_INTR_STAT 0x1808 +#define ACP5X_EXT_INTR_STAT 0x1808 #define ACP6X_EXT_INTR_STAT 0x1A0C #define ACP3X_DSP_SW_INTR_BASE 0x1814 +#define ACP5X_DSP_SW_INTR_BASE 0x1814 #define ACP6X_DSP_SW_INTR_BASE 0x1808 #define DSP_SW_INTR_CNTL_OFFSET 0x0 #define DSP_SW_INTR_STAT_OFFSET 0x4 #define DSP_SW_INTR_TRIG_OFFSET 0x8 #define ACP_ERROR_STATUS 0x18C4 #define ACP3X_AXI2DAGB_SEM_0 0x1880 +#define ACP5X_AXI2DAGB_SEM_0 0x1884 #define ACP6X_AXI2DAGB_SEM_0 0x1874 /* Registers from ACP_SHA block */ diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 72fa0af971f0..73c3e1373d42 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -3,7 +3,7 @@ * This file is provided under a dual BSD/GPLv2 license. When using or * redistributing this file, you may do so under either license. * - * Copyright(c) 2021 Advanced Micro Devices, Inc. All rights reserved. + * Copyright(c) 2021, 2023 Advanced Micro Devices, Inc. All rights reserved. * * Author: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> */ @@ -32,6 +32,7 @@ #define ACP_DSP_INTR_EN_MASK 0x00000001 #define ACP3X_SRAM_PTE_OFFSET 0x02050000 +#define ACP5X_SRAM_PTE_OFFSET 0x02050000 #define ACP6X_SRAM_PTE_OFFSET 0x03800000 #define PAGE_SIZE_4K_ENABLE 0x2 #define ACP_PAGE_SIZE 0x1000 @@ -56,9 +57,11 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 #define ACP_RN_PCI_ID 0x01 +#define ACP_VANGOGH_PCI_ID 0x50 #define ACP_RMB_PCI_ID 0x6F #define HOST_BRIDGE_CZN 0x1630 +#define HOST_BRIDGE_VGH 0x1645 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 #define ACP_PSP_TIMEOUT_US 1000000 @@ -163,6 +166,7 @@ struct acp_dsp_stream { struct sof_amd_acp_desc { unsigned int rev; + const char *name; unsigned int host_bridge_id; u32 pgfsm_base; u32 ext_intr_stat; @@ -253,6 +257,8 @@ extern struct snd_sof_dsp_ops sof_acp_common_ops; extern struct snd_sof_dsp_ops sof_renoir_ops; int sof_renoir_ops_init(struct snd_sof_dev *sdev); +extern struct snd_sof_dsp_ops sof_vangogh_ops; +int sof_vangogh_ops_init(struct snd_sof_dev *sdev); extern struct snd_sof_dsp_ops sof_rembrandt_ops; int sof_rembrandt_ops_init(struct snd_sof_dev *sdev); @@ -282,4 +288,5 @@ static inline const struct sof_amd_acp_desc *get_chip_info(struct snd_sof_pdata int acp_probes_register(struct snd_sof_dev *sdev); void acp_probes_unregister(struct snd_sof_dev *sdev); +extern struct snd_soc_acpi_mach snd_soc_acpi_amd_vangogh_sof_machines[]; #endif diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c new file mode 100644 index 000000000000..d8be42fbcb6d --- /dev/null +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Advanced Micro Devices, Inc. All rights reserved. +// +// Authors: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> + +/*. + * PCI interface for Vangogh ACP device + */ + +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/platform_device.h> +#include <sound/sof.h> +#include <sound/soc-acpi.h> + +#include "../ops.h" +#include "../sof-pci-dev.h" +#include "../../amd/mach-config.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define ACP5X_FUTURE_REG_ACLK_0 0x1864 + +static const struct sof_amd_acp_desc vangogh_chip_info = { + .rev = 5, + .name = "vangogh", + .host_bridge_id = HOST_BRIDGE_VGH, + .pgfsm_base = ACP5X_PGFSM_BASE, + .ext_intr_stat = ACP5X_EXT_INTR_STAT, + .dsp_intr_base = ACP5X_DSP_SW_INTR_BASE, + .sram_pte_offset = ACP5X_SRAM_PTE_OFFSET, + .hw_semaphore_offset = ACP5X_AXI2DAGB_SEM_0, + .acp_clkmux_sel = ACP5X_CLKMUX_SEL, + .probe_reg_offset = ACP5X_FUTURE_REG_ACLK_0, +}; + +static const struct sof_dev_desc vangogh_desc = { + .machines = snd_soc_acpi_amd_vangogh_sof_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .chip_info = &vangogh_chip_info, + .ipc_supported_mask = BIT(SOF_IPC), + .ipc_default = SOF_IPC, + .default_fw_path = { + [SOF_IPC] = "amd/sof", + }, + .default_tplg_path = { + [SOF_IPC] = "amd/sof-tplg", + }, + .default_fw_filename = { + [SOF_IPC] = "sof-vangogh.ri", + }, + .nocodec_tplg_filename = "sof-acp.tplg", + .ops = &sof_vangogh_ops, + .ops_init = sof_vangogh_ops_init, +}; + +static int acp_pci_vgh_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) +{ + unsigned int flag; + + if (pci->revision != ACP_VANGOGH_PCI_ID) + return -ENODEV; + + flag = snd_amd_acp_find_config(pci); + if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) + return -ENODEV; + + return sof_pci_probe(pci, pci_id); +}; + +static void acp_pci_vgh_remove(struct pci_dev *pci) +{ + sof_pci_remove(pci); +} + +/* PCI IDs */ +static const struct pci_device_id vgh_pci_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_AMD, ACP_PCI_DEV_ID), + .driver_data = (unsigned long)&vangogh_desc}, + { 0, } +}; +MODULE_DEVICE_TABLE(pci, vgh_pci_ids); + +/* pci_driver definition */ +static struct pci_driver snd_sof_pci_amd_vgh_driver = { + .name = KBUILD_MODNAME, + .id_table = vgh_pci_ids, + .probe = acp_pci_vgh_probe, + .remove = acp_pci_vgh_remove, + .driver = { + .pm = &sof_pci_pm, + }, +}; +module_pci_driver(snd_sof_pci_amd_vgh_driver); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); +MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c new file mode 100644 index 000000000000..f3f6bd790788 --- /dev/null +++ b/sound/soc/sof/amd/vangogh.c @@ -0,0 +1,156 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2023 Advanced Micro Devices, Inc. +// +// Authors: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> + +/* + * Hardware interface for Audio DSP on Vangogh platform + */ + +#include <linux/platform_device.h> +#include <linux/module.h> + +#include "../ops.h" +#include "../sof-audio.h" +#include "acp.h" +#include "acp-dsp-offset.h" + +#define I2S_HS_INSTANCE 0 +#define I2S_BT_INSTANCE 1 +#define I2S_SP_INSTANCE 2 +#define PDM_DMIC_INSTANCE 3 +#define I2S_HS_VIRTUAL_INSTANCE 4 + +static struct snd_soc_dai_driver vangogh_sof_dai[] = { + [I2S_HS_INSTANCE] = { + .id = I2S_HS_INSTANCE, + .name = "acp-sof-hs", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S HS controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_BT_INSTANCE] = { + .id = I2S_BT_INSTANCE, + .name = "acp-sof-bt", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S BT controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_SP_INSTANCE] = { + .id = I2S_SP_INSTANCE, + .name = "acp-sof-sp", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S SP controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [PDM_DMIC_INSTANCE] = { + .id = PDM_DMIC_INSTANCE, + .name = "acp-sof-dmic", + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 4, + .rate_min = 8000, + .rate_max = 48000, + }, + }, + + [I2S_HS_VIRTUAL_INSTANCE] = { + .id = I2S_HS_VIRTUAL_INSTANCE, + .name = "acp-sof-hs-virtual", + .playback = { + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 2, + .channels_max = 8, + .rate_min = 8000, + .rate_max = 96000, + }, + .capture = { + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S32_LE, + /* Supporting only stereo for I2S HS-Virtual controller capture */ + .channels_min = 2, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 48000, + }, + }, +}; + +/* Vangogh ops */ +struct snd_sof_dsp_ops sof_vangogh_ops; +EXPORT_SYMBOL_NS(sof_vangogh_ops, SND_SOC_SOF_AMD_COMMON); + +int sof_vangogh_ops_init(struct snd_sof_dev *sdev) +{ + /* common defaults */ + memcpy(&sof_vangogh_ops, &sof_acp_common_ops, sizeof(struct snd_sof_dsp_ops)); + + sof_vangogh_ops.drv = vangogh_sof_dai; + sof_vangogh_ops.num_drv = ARRAY_SIZE(vangogh_sof_dai); + + return 0; +} + +MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); +MODULE_DESCRIPTION("VANGOGH SOF Driver"); +MODULE_LICENSE("Dual BSD/GPL"); -- 2.25.1

2 3

[PATCH] sound/soc/codecs/rt5640.c: fix typos
by Senhong Liu 22 Aug '23

22 Aug '23

I noticed typos and i fixed them. Signed-off-by: Senhong Liu <liusenhong2022(a)email.szu.edu.cn> --- sound/soc/codecs/rt5640.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 8920726c38e8..15e1a62b9e57 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2570,7 +2570,7 @@ static void rt5640_enable_jack_detect(struct snd_soc_component *component, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { - dev_warn(component->dev, "Failed to reguest IRQ %d: %d\n", rt5640->irq, ret); + dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640_disable_jack_detect(component); return; } @@ -2625,7 +2625,7 @@ static void rt5640_enable_hda_jack_detect( NULL, rt5640_irq, IRQF_TRIGGER_RISING | IRQF_ONESHOT, "rt5640", rt5640); if (ret) { - dev_warn(component->dev, "Failed to reguest IRQ %d: %d\n", rt5640->irq, ret); + dev_warn(component->dev, "Failed to request IRQ %d: %d\n", rt5640->irq, ret); rt5640->irq = -ENXIO; return; } -- 2.25.1

2 2

[PATCH 0/4] ASoC: SOF: ipc4-topology: Fixes for pipelines with SRC
by Peter Ujfalusi 22 Aug '23

22 Aug '23

Hi, the SRC component in a pipeline provides flexibility on the sampling rate which was not handled previously. This series will improve the kernel side with the needed logic to be able to deal with the SRC type of components in pipelines. Regards, Peter --- Ranjani Sridharan (4): ASoC: SOF: ipc4-topology: Modify pipeline params based on SRC output format ASoC: SOF: ipc4-topology: Fix the output reference params for SRC ASoC: SOF: ipc4-topology: Fix pipeline params at the output of copier ASoC: SOF: ipc4-topology: Modify the reference output valid_bits for copier sound/soc/sof/ipc4-topology.c | 134 ++++++++++++++++++++++------------ 1 file changed, 89 insertions(+), 45 deletions(-) -- 2.41.0

2 5

[PATCH] soundwire: intel_ace2x: add DAI hw_params/prepare/hw_free callbacks
by Bard Liao 22 Aug '23

22 Aug '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The code is fork-lifted from intel.c and is mostly similar *except* for the SHIM configuration which cannot be done here with the introduction of HDAudio Extended links. The ACE2.x SOF side also requires the hw_free and trigger callbacks to be implemented for HDaudio DMA management Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel_ace2x.c | 283 ++++++++++++++++++++++++++++++++ 1 file changed, 283 insertions(+) diff --git a/drivers/soundwire/intel_ace2x.c b/drivers/soundwire/intel_ace2x.c index 1be0bea5f40f..a9d25ae0b73f 100644 --- a/drivers/soundwire/intel_ace2x.c +++ b/drivers/soundwire/intel_ace2x.c @@ -10,6 +10,7 @@ #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_intel.h> +#include <sound/pcm_params.h> #include <sound/hda-mlink.h> #include "cadence_master.h" #include "bus.h" @@ -191,10 +192,292 @@ static bool intel_check_cmdsync_unlocked(struct sdw_intel *sdw) return hdac_bus_eml_sdw_check_cmdsync_unlocked(sdw->link_res->hbus); } +/* DAI callbacks */ +static int intel_params_stream(struct sdw_intel *sdw, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + struct snd_pcm_hw_params *hw_params, + int link_id, int alh_stream_id) +{ + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_intel_stream_params_data params_data; + + params_data.substream = substream; + params_data.dai = dai; + params_data.hw_params = hw_params; + params_data.link_id = link_id; + params_data.alh_stream_id = alh_stream_id; + + if (res->ops && res->ops->params_stream && res->dev) + return res->ops->params_stream(res->dev, + &params_data); + return -EIO; +} + +static int intel_free_stream(struct sdw_intel *sdw, + struct snd_pcm_substream *substream, + struct snd_soc_dai *dai, + int link_id) + +{ + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_intel_stream_free_data free_data; + + free_data.substream = substream; + free_data.dai = dai; + free_data.link_id = link_id; + + if (res->ops && res->ops->free_stream && res->dev) + return res->ops->free_stream(res->dev, + &free_data); + + return 0; +} + /* * DAI operations */ +static int intel_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + struct sdw_cdns_pdi *pdi; + struct sdw_stream_config sconfig; + struct sdw_port_config *pconfig; + int ch, dir; + int ret; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + ch = params_channels(params); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = SDW_DATA_DIR_RX; + else + dir = SDW_DATA_DIR_TX; + + pdi = sdw_cdns_alloc_pdi(cdns, &cdns->pcm, ch, dir, dai->id); + + if (!pdi) { + ret = -EINVAL; + goto error; + } + + /* the SHIM will be configured in the callback functions */ + + sdw_cdns_config_stream(cdns, ch, dir, pdi); + + /* store pdi and state, may be needed in prepare step */ + dai_runtime->paused = false; + dai_runtime->suspended = false; + dai_runtime->pdi = pdi; + + /* Inform DSP about PDI stream number */ + ret = intel_params_stream(sdw, substream, dai, params, + sdw->instance, + pdi->intel_alh_id); + if (ret) + goto error; + + sconfig.direction = dir; + sconfig.ch_count = ch; + sconfig.frame_rate = params_rate(params); + sconfig.type = dai_runtime->stream_type; + + sconfig.bps = snd_pcm_format_width(params_format(params)); + + /* Port configuration */ + pconfig = kzalloc(sizeof(*pconfig), GFP_KERNEL); + if (!pconfig) { + ret = -ENOMEM; + goto error; + } + + pconfig->num = pdi->num; + pconfig->ch_mask = (1 << ch) - 1; + + ret = sdw_stream_add_master(&cdns->bus, &sconfig, + pconfig, 1, dai_runtime->stream); + if (ret) + dev_err(cdns->dev, "add master to stream failed:%d\n", ret); + + kfree(pconfig); +error: + return ret; +} + +static int intel_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + int ch, dir; + int ret = 0; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) { + dev_err(dai->dev, "failed to get dai runtime in %s\n", + __func__); + return -EIO; + } + + if (dai_runtime->suspended) { + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_pcm_hw_params *hw_params; + + hw_params = &rtd->dpcm[substream->stream].hw_params; + + dai_runtime->suspended = false; + + /* + * .prepare() is called after system resume, where we + * need to reinitialize the SHIM/ALH/Cadence IP. + * .prepare() is also called to deal with underflows, + * but in those cases we cannot touch ALH/SHIM + * registers + */ + + /* configure stream */ + ch = params_channels(hw_params); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dir = SDW_DATA_DIR_RX; + else + dir = SDW_DATA_DIR_TX; + + /* the SHIM will be configured in the callback functions */ + + sdw_cdns_config_stream(cdns, ch, dir, dai_runtime->pdi); + + /* Inform DSP about PDI stream number */ + ret = intel_params_stream(sdw, substream, dai, + hw_params, + sdw->instance, + dai_runtime->pdi->intel_alh_id); + } + + return ret; +} + +static int +intel_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dai_runtime *dai_runtime; + int ret; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return -EIO; + + /* + * The sdw stream state will transition to RELEASED when stream-> + * master_list is empty. So the stream state will transition to + * DEPREPARED for the first cpu-dai and to RELEASED for the last + * cpu-dai. + */ + ret = sdw_stream_remove_master(&cdns->bus, dai_runtime->stream); + if (ret < 0) { + dev_err(dai->dev, "remove master from stream %s failed: %d\n", + dai_runtime->stream->name, ret); + return ret; + } + + ret = intel_free_stream(sdw, substream, dai, sdw->instance); + if (ret < 0) { + dev_err(dai->dev, "intel_free_stream: failed %d\n", ret); + return ret; + } + + dai_runtime->pdi = NULL; + + return 0; +} + +static int intel_pcm_set_sdw_stream(struct snd_soc_dai *dai, + void *stream, int direction) +{ + return cdns_set_sdw_stream(dai, stream, direction); +} + +static void *intel_get_sdw_stream(struct snd_soc_dai *dai, + int direction) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_cdns_dai_runtime *dai_runtime; + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) + return ERR_PTR(-EINVAL); + + return dai_runtime->stream; +} + +static int intel_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_intel_link_res *res = sdw->link_res; + struct sdw_cdns_dai_runtime *dai_runtime; + int ret = 0; + + /* + * The .trigger callback is used to program HDaudio DMA and send required IPC to audio + * firmware. + */ + if (res->ops && res->ops->trigger) { + ret = res->ops->trigger(substream, cmd, dai); + if (ret < 0) + return ret; + } + + dai_runtime = cdns->dai_runtime_array[dai->id]; + if (!dai_runtime) { + dev_err(dai->dev, "failed to get dai runtime in %s\n", + __func__); + return -EIO; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_SUSPEND: + + /* + * The .prepare callback is used to deal with xruns and resume operations. + * In the case of xruns, the DMAs and SHIM registers cannot be touched, + * but for resume operations the DMAs and SHIM registers need to be initialized. + * the .trigger callback is used to track the suspend case only. + */ + + dai_runtime->suspended = true; + + break; + + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dai_runtime->paused = true; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dai_runtime->paused = false; + break; + default: + break; + } + + return ret; +} + static const struct snd_soc_dai_ops intel_pcm_dai_ops = { + .hw_params = intel_hw_params, + .prepare = intel_prepare, + .hw_free = intel_hw_free, + .trigger = intel_trigger, + .set_stream = intel_pcm_set_sdw_stream, + .get_stream = intel_get_sdw_stream, }; static const struct snd_soc_component_driver dai_component = { -- 2.25.1

2 1

[PATCH 1/2] ASoC: dt-bindings: nau8821: Add single-ended input feature
by Seven Lee 22 Aug '23

22 Aug '23

Add input with single-ended control. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml index fc2f4ce4db88..3e54abd4ca74 100644 --- a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -89,6 +89,12 @@ properties: $ref: /schemas/types.yaml#/definitions/uint32 default: 3072000 + nuvoton,left-input-single-end: + description: Enable left input with single-ended settings if set. + For the headset mic application, the single-ended control is + just limited to the left adc for design demand. + type: boolean + '#sound-dai-cells': const: 0 @@ -114,6 +120,7 @@ examples: nuvoton,jkdet-pull-enable; nuvoton,jkdet-pull-up; nuvoton,key-enable; + nuvoton,left-input-single-end; nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; nuvoton,micbias-voltage = <6>; nuvoton,vref-impedance = <2>; -- 2.25.1

5 6

Add a config for the Allen & Heath Zedi 10 mixer.
by GitHub pull_request - opened 22 Aug '23

22 Aug '23

alsa-project/alsa-ucm-conf pull request #341 was opened from pololevu: [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12402535… The mixer has an embedded 4x4 USB interface, this config surfaces the mixer channels as 4 mono inputs, 2 stereo inputs and 2 stereo outputs. Config based on the Behringer Flow8 config. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/341 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/341.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: linux-next: Tree for Aug 21 (sound/pci/hda/tas2781_hda_i2c.o)
by Randy Dunlap 21 Aug '23

21 Aug '23

On 8/20/23 23:46, Stephen Rothwell wrote: > Hi all, > > Changes since 20230818: on x86_64: ld: sound/pci/hda/tas2781_hda_i2c.o: in function `tas2781_save_calibration.isra.0': tas2781_hda_i2c.c:(.text+0x1075): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x108b): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x119b): undefined reference to `efi' ld: tas2781_hda_i2c.c:(.text+0x11b1): undefined reference to `efi' Full randconfig file is attached. -- ~Randy

1 0

[PATCH 0/3] ASoC: wm8782: Allow higher audio rates
by John Watts 21 Aug '23

21 Aug '23

The wm8782 supports higher audio rates than just 48kHz. This is configured by setting the FSAMPEN pin on the codec chip. This patch series introduces the 'max-rate' device tree property to allow indicating which rates are fully supported by the codec without accidentally overclocking. John Watts (3): ASoC: wm8782: Handle maximum audio rate at runtime ASoC: wm8782: Use max-rate device tree property ASoC: dt-bindings: wlf,wm8782: Add max-rate property .../devicetree/bindings/sound/wm8782.txt | 2 + sound/soc/codecs/wm8782.c | 49 ++++++++++++++----- 2 files changed, 39 insertions(+), 12 deletions(-) -- 2.41.0

3 8

[PATCH] [6.4.y] ASoC: SOF: intel: hda: Clean up link DMA for IPC3 during stop
by Jaroslav Kysela 21 Aug '23

21 Aug '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> commit 90219f1bd273055f1dc1d7bdc0965755b992c045 upstream. With IPC3, we reset hw_params during the stop trigger, so we should also clean up the link DMA during the stop trigger. Cc: <stable(a)vger.kernel.org> # 6.4.x Fixes: 1bf83fa6654c ("ASoC: SOF: Intel: hda-dai: Do not perform DMA cleanup during stop") Closes: https://github.com/thesofproject/linux/issues/4455 Closes: https://github.com/thesofproject/linux/issues/4482 Closes: https://bugzilla.kernel.org/show_bug.cgi?id=217673 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20230808110627.32375-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Note that many recent Intel based laptops are affected. Added missing code for 6.4 kernels to keep the fix simple not depending on the other changes. This commit is present in 6.5 tree already. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/sof/intel/hda-dai-ops.c | 13 ++++++++++++- sound/soc/sof/intel/hda-dai.c | 8 ++++---- sound/soc/sof/intel/hda.h | 2 ++ 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai-ops.c b/sound/soc/sof/intel/hda-dai-ops.c index 4b39cecacd68..5938046f46b2 100644 --- a/sound/soc/sof/intel/hda-dai-ops.c +++ b/sound/soc/sof/intel/hda-dai-ops.c @@ -289,16 +289,27 @@ static const struct hda_dai_widget_dma_ops hda_ipc4_chain_dma_ops = { static int hda_ipc3_post_trigger(struct snd_sof_dev *sdev, struct snd_soc_dai *cpu_dai, struct snd_pcm_substream *substream, int cmd) { + struct hdac_ext_stream *hext_stream = hda_get_hext_stream(sdev, cpu_dai, substream); struct snd_soc_dapm_widget *w = snd_soc_dai_get_widget(cpu_dai, substream->stream); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); switch (cmd) { case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: { struct snd_sof_dai_config_data data = { 0 }; + int ret; data.dai_data = DMA_CHAN_INVALID; - return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + ret = hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_HW_FREE, &data); + if (ret < 0) + return ret; + + if (cmd == SNDRV_PCM_TRIGGER_STOP) + return hda_link_dma_cleanup(substream, hext_stream, cpu_dai, codec_dai); + + break; } case SNDRV_PCM_TRIGGER_PAUSE_PUSH: return hda_dai_config(w, SOF_DAI_CONFIG_FLAGS_PAUSE, NULL); diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 44a5d94c5050..8a76320c3b99 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -91,10 +91,10 @@ hda_dai_get_ops(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai return sdai->platform_private; } -static int hda_link_dma_cleanup(struct snd_pcm_substream *substream, - struct hdac_ext_stream *hext_stream, - struct snd_soc_dai *cpu_dai, - struct snd_soc_dai *codec_dai) +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, + struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai, + struct snd_soc_dai *codec_dai) { struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(cpu_dai->component); const struct hda_dai_widget_dma_ops *ops = hda_dai_get_ops(substream, cpu_dai); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index c4befacde23e..94c738eae751 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -942,5 +942,7 @@ const struct hda_dai_widget_dma_ops * hda_select_dai_widget_ops(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); int hda_dai_config(struct snd_soc_dapm_widget *w, unsigned int flags, struct snd_sof_dai_config_data *data); +int hda_link_dma_cleanup(struct snd_pcm_substream *substream, struct hdac_ext_stream *hext_stream, + struct snd_soc_dai *cpu_dai, struct snd_soc_dai *codec_dai); #endif -- 2.41.0

2 1

[PATCH v3 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 21 Aug '23

21 Aug '23

Create tas2781 side codec HDA driver for Lenovo Laptops. The quantity of the speakers has been define in ACPI. All of the tas2781s in the laptop will be aggregated as one audio speaker. The code supports realtek codec as the primary codec. Code offers several controls for digtial/analog gain setting during playback, and other for eq params setting in case of different audio profiles, such as music, voice, movie, etc. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v3: - oprimize the indentation - fix the ambiguous error message - The check of the given value to kcontrol's put func is in the sound/tas2781-comlib.c - In tasdevice_info_programs, corrected max value - Seperate snd_kcontrol_new tas2781_dsp_controls into two standalone kcontrol. - All the controls set as const static - Add static for page_array & rgno_array - Remove uncessary blank lines - Add descriptions for tas2781_save_calibration - remove global addr handling in the code - checking subid in switch statement in function tas2781_hda_bind - add force firmware load Kcontrol - rename the kcontrol name to be more undertandable - remove Superfluous cast in tasdevice_fw_ready - correct weird line break in function tas2781_acpi_get_i2c_resource - correct Referencing adev after acpi_dev_put() in tas2781_hda_read_acpi - As to checking the given value in tasdevice_set_profile_id, it seems done by the tasdevice_info_profile - replace strcpy with strscpy in tas2781_hda_read_acpi - rewrite the subid judgement - Add tiwai(a)suse.de into Cc list - remove the cast in tas2781_acpi_get_i2c_resource - remove else in tas2781_acpi_get_i2c_resource - fix the return value in tasdevice_set_profile_id - remove unneeded NL in tasdevice_config_get - Unifiy the comment style - remove ret = 0 in tasdevice_fw_ready - remove ret in tas2781_save_calibration - remove unused ret in tas2781_hda_playback - add force firmware load Kcontrol --- sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/tas2781_hda_i2c.c | 858 ++++++++++++++++++++++++++++++++ 3 files changed, 875 insertions(+) create mode 100644 sound/pci/hda/tas2781_hda_i2c.c diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 886255a03e8b..e66257277492 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -130,6 +130,21 @@ config SND_HDA_SCODEC_CS35L41_SPI comment "Set to Y if you want auto-loading the side codec driver" depends on SND_HDA=y && SND_HDA_SCODEC_CS35L41_SPI=m +config SND_HDA_SCODEC_TAS2781_I2C + tristate "Build TAS2781 HD-audio side codec support for I2C Bus" + depends on I2C + depends on ACPI + depends on SND_SOC + select SND_SOC_TAS2781_COMLIB + select SND_SOC_TAS2781_FMWLIB + select CRC32_SARWATE + help + Say Y or M here to include TAS2781 I2C HD-audio side codec support + in snd-hda-intel driver, such as ALC287. + +comment "Set to Y if you want auto-loading the side codec driver" + depends on SND_HDA=y && SND_HDA_SCODEC_TAS2781_I2C=m + config SND_HDA_CODEC_REALTEK tristate "Build Realtek HD-audio codec support" select SND_HDA_GENERIC diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index 00d306104484..1c76609690d6 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -32,6 +32,7 @@ snd-hda-scodec-cs35l41-objs := cs35l41_hda.o snd-hda-scodec-cs35l41-i2c-objs := cs35l41_hda_i2c.o snd-hda-scodec-cs35l41-spi-objs := cs35l41_hda_spi.o snd-hda-cs-dsp-ctls-objs := hda_cs_dsp_ctl.o +snd-hda-scodec-tas2781-i2c-objs := tas2781_hda_i2c.o # common driver obj-$(CONFIG_SND_HDA) := snd-hda-codec.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_HDA_SCODEC_CS35L41) += snd-hda-scodec-cs35l41.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_I2C) += snd-hda-scodec-cs35l41-i2c.o obj-$(CONFIG_SND_HDA_SCODEC_CS35L41_SPI) += snd-hda-scodec-cs35l41-spi.o obj-$(CONFIG_SND_HDA_CS_DSP_CONTROLS) += snd-hda-cs-dsp-ctls.o +obj-$(CONFIG_SND_HDA_SCODEC_TAS2781_I2C) += snd-hda-scodec-tas2781-i2c.o # this must be the last entry after codec drivers; # otherwise the codec patches won't be hooked before the PCI probe diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c new file mode 100644 index 000000000000..4ccbe12ad451 --- /dev/null +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -0,0 +1,858 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TAS2781 HDA I2C driver +// +// Copyright 2023 Texas Instruments, Inc. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> + +#include <linux/acpi.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <sound/hda_codec.h> +#include <sound/soc.h> +#include <sound/tas2781.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "hda_local.h" +#include "hda_auto_parser.h" +#include "hda_component.h" +#include "hda_jack.h" +#include "hda_generic.h" + +#define TASDEVICE_SPEAKER_CALIBRATION_SIZE 20 + +/* No standard control callbacks for SNDRV_CTL_ELEM_IFACE_CARD + * Define two controls, one is Volume control callbacks, the other is + * flag setting control callbacks. + */ + +/* Volume control callbacks for tas2781 */ +#define ACARD_SINGLE_RANGE_EXT_TLV(xname, xreg, xshift, xmin, xmax, xinvert, \ + xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xreg, .shift = xshift, \ + .rshift = xshift, .min = xmin, .max = xmax, \ + .invert = xinvert} } + +/* Flag control callbacks for tas2781 */ +#define ACARD_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_CARD, .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = xdata } + +enum calib_data { + R0_VAL = 0, + INV_R0, + R0LOW, + POWER, + TLIM, + CALIB_MAX +}; + +static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) +{ + struct tasdevice_priv *tas_priv = data; + struct acpi_resource_i2c_serialbus *sb; + + if (i2c_acpi_get_i2c_resource(ares, &sb)) { + if (tas_priv->ndev < TASDEVICE_MAX_CHANNELS && + sb->slave_address != TAS2781_GLOBAL_ADDR) { + tas_priv->tasdevice[tas_priv->ndev].dev_addr = + (unsigned int)sb->slave_address; + tas_priv->ndev++; + } + } + return 1; +} + +static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) +{ + struct acpi_device *adev; + struct device *physdev; + LIST_HEAD(resources); + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(p->dev, + "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + + ret = acpi_dev_get_resources(adev, &resources, tas2781_get_i2c_res, p); + if (ret < 0) + goto err; + + acpi_dev_free_resource_list(&resources); + strscpy(p->dev_name, hid, sizeof(p->dev_name)); + physdev = get_device(acpi_get_first_physical_node(adev)); + acpi_dev_put(adev); + + /* No side-effect to the playback even if subsystem_id is NULL*/ + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + + p->acpi_subsystem_id = sub; + + put_device(physdev); + + return 0; + +err: + dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); + + return ret; +} + +static void tas2781_hda_playback_hook(struct device *dev, int action) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + dev_dbg(tas_priv->dev, "%s: action = %d\n", __func__, action); + switch (action) { + case HDA_GEN_PCM_ACT_OPEN: + pm_runtime_get_sync(dev); + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 0); + mutex_unlock(&tas_priv->codec_lock); + break; + case HDA_GEN_PCM_ACT_CLOSE: + mutex_lock(&tas_priv->codec_lock); + tasdevice_tuning_switch(tas_priv, 1); + mutex_unlock(&tas_priv->codec_lock); + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + break; + default: + dev_dbg(tas_priv->dev, "Playback action not supported: %d\n", + action); + break; + } +} + +static int tasdevice_info_profile(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->rcabin.ncfgs - 1; + + return 0; +} + +static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->rcabin.profile_cfg_id; + + return 0; +} + +static int tasdevice_hda_clamp(int val, int max) +{ + if (val > max) + val = max; + + if (val < 0) + val = 0; + return val; +} + +static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + int nr_profile = ucontrol->value.integer.value[0]; + int max = tas_priv->rcabin.ncfgs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_profile, max); + + if (tas_priv->rcabin.profile_cfg_id != nr_profile) { + tas_priv->rcabin.profile_cfg_id = nr_profile; + ret = 1; + } + + return ret; +} + +static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_programs - 1; + + return 0; +} + +static int tasdevice_info_config(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_fw->nr_configurations - 1; + + return 0; +} + +static int tasdevice_program_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_prog; + + return 0; +} + +static int tasdevice_program_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_program = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_programs - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_program, max); + + if (tas_priv->cur_prog != nr_program) { + tas_priv->cur_prog = nr_program; + ret = 1; + } + + return ret; +} + +static int tasdevice_config_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = tas_priv->cur_conf; + + return 0; +} + +static int tasdevice_config_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct tasdevice_fw *tas_fw = tas_priv->fmw; + int nr_config = ucontrol->value.integer.value[0]; + int max = tas_fw->nr_configurations - 1; + int val, ret = 0; + + val = tasdevice_hda_clamp(nr_config, max); + + if (tas_priv->cur_conf != nr_config) { + tas_priv->cur_conf = nr_config; + ret = 1; + } + + return ret; +} + +/* + * tas2781_digital_getvol - get the volum control + * @kcontrol: control pointer + * @ucontrol: User data + * Customer Kcontrol for tas2781 is primarily for regmap booking, paging + * depends on internal regmap mechanism. + * tas2781 contains book and page two-level register map, especially + * book switching will set the register BXXP00R7F, after switching to the + * correct book, then leverage the mechanism for paging to access the + * register. + */ +static int tas2781_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_digital_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + return tasdevice_amp_getvol(tas_priv, ucontrol, mc); +} + +static int tas2781_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_digital_putvol. */ + return tasdevice_digital_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + /* The check of the given value is in tasdevice_amp_putvol. */ + return tasdevice_amp_putvol(tas_priv, ucontrol, mc); +} + +static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = (int)tas_priv->force_fwload_status; + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return 0; +} + +static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tasdevice_priv *tas_priv = snd_kcontrol_chip(kcontrol); + bool change, val = (bool)ucontrol->value.integer.value[0]; + + if (tas_priv->force_fwload_status == val) + change = false; + else { + change = true; + tas_priv->force_fwload_status = val; + } + dev_dbg(tas_priv->dev, "%s : Force FWload %s\n", __func__, + tas_priv->force_fwload_status ? "ON" : "OFF"); + + return change; +} + +static const struct snd_kcontrol_new tas2781_snd_controls[] = { + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, + 1, 0, 20, 0, tas2781_amp_getvol, + tas2781_amp_putvol, amp_vol_tlv), + ACARD_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, + 0, 0, 200, 1, tas2781_digital_getvol, + tas2781_digital_putvol, dvc_tlv), + ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tas2781_force_fwload_get, tas2781_force_fwload_put), +}; + +static const struct snd_kcontrol_new tas2781_prof_ctrl = { + .name = "Speaker Profile Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_profile, + .get = tasdevice_get_profile_id, + .put = tasdevice_set_profile_id, +}; + +static const struct snd_kcontrol_new tas2781_dsp_prog_ctrl = { + .name = "Speaker Program Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_programs, + .get = tasdevice_program_get, + .put = tasdevice_program_put, +}; + +static const struct snd_kcontrol_new tas2781_dsp_conf_ctrl = { + .name = "Speaker Config Id", + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .info = tasdevice_info_config, + .get = tasdevice_config_get, + .put = tasdevice_config_put, +}; + +static void tas2781_apply_calib(struct tasdevice_priv *tas_priv) +{ + static const unsigned char page_array[CALIB_MAX] = { + 0x17, 0x18, 0x18, 0x0d, 0x18 + }; + static const unsigned char rgno_array[CALIB_MAX] = { + 0x74, 0x0c, 0x14, 0x3c, 0x7c + }; + unsigned char *data; + int i, j, rc; + + for (i = 0; i < tas_priv->ndev; i++) { + data = tas_priv->cali_data.data + + i * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + for (j = 0; j < CALIB_MAX; j++) { + rc = tasdevice_dev_bulk_write(tas_priv, i, + TASDEVICE_REG(0, page_array[j], rgno_array[j]), + &(data[4 * j]), 4); + if (rc < 0) + dev_err(tas_priv->dev, + "chn %d calib %d bulk_wr err = %d\n", + i, j, rc); + } + } +} + +/* Update the calibrate data, including speaker impedance, f0, etc, into algo. + * Calibrate data is done by manufacturer in the factory. These data are used + * by Algo for calucating the speaker temperature, speaker membrance excursion + * and f0 in real time during playback. + */ +static int tas2781_save_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x02f9af02, 0x7734, 0x4233, 0xb4, 0x3d, + 0x93, 0xfe, 0x5a, 0xa3, 0x5d, 0xb3); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + /* Lenovo devices */ + if (tas_priv->catlog_id == LENOVO) + efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, 0x09, + 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + + tas_priv->cali_data.total_sz = 0; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + if (status == EFI_BUFFER_TOO_SMALL) { + /* Allocate data buffer of data_size bytes */ + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + tas_priv->cali_data.total_sz, GFP_KERNEL); + if (!tas_priv->cali_data.data) + return -ENOMEM; + /* Get variable contents into buffer */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + if (status != EFI_SUCCESS) + return -EINVAL; + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2781_apply_calib(tas_priv); + } else + tas_priv->cali_data.total_sz = 0; + + return 0; +} + +static void tasdev_fw_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_priv = context; + struct hda_codec *codec = tas_priv->codec; + int i, ret; + + pm_runtime_get_sync(tas_priv->dev); + mutex_lock(&tas_priv->codec_lock); + + ret = tasdevice_rca_parser(tas_priv, fmw); + if (ret) + goto out; + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_prof_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_prof_ctrl.name, ret); + goto out; + } + + for (i = 0; i < ARRAY_SIZE(tas2781_snd_controls); i++) { + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_snd_controls[i], tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_snd_controls[i].name, ret); + goto out; + } + } + + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; + scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", + codec->core.subsystem_id & 0xffff); + ret = tasdevice_dsp_parser(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "dspfw load %s error\n", + tas_priv->coef_binaryname); + tas_priv->fw_state = TASDEVICE_DSP_FW_FAIL; + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_prog_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_prog_ctrl.name, ret); + goto out; + } + + ret = snd_ctl_add(codec->card, + snd_ctl_new1(&tas2781_dsp_conf_ctrl, tas_priv)); + if (ret) { + dev_err(tas_priv->dev, + "Failed to add KControl %s = %d\n", + tas2781_dsp_conf_ctrl.name, ret); + goto out; + } + + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; + tasdevice_prmg_load(tas_priv, 0); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + tas2781_save_calibration(tas_priv); + +out: + if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) { + /*If DSP FW fail, kcontrol won't be created */ + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + } + mutex_unlock(&tas_priv->codec_lock); + if (fmw) + release_firmware(fmw); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_put_autosuspend(tas_priv->dev); +} + +static int tas2781_hda_bind(struct device *dev, struct device *master, + void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + struct hda_codec *codec; + unsigned int subid; + int ret; + + if (!comps || tas_priv->index < 0 || + tas_priv->index >= HDA_MAX_COMPONENTS) + return -EINVAL; + + comps = &comps[tas_priv->index]; + if (comps->dev) + return -EBUSY; + + codec = comps->codec; + subid = codec->core.subsystem_id >> 16; + + switch (subid) { + case 0x17aa: + tas_priv->catlog_id = LENOVO; + break; + default: + tas_priv->catlog_id = OTHERS; + break; + } + + pm_runtime_get_sync(dev); + + comps->dev = dev; + + strscpy(comps->name, dev_name(dev), sizeof(comps->name)); + + ret = tascodec_init(tas_priv, codec, tasdev_fw_ready); + if (ret) + return ret; + + comps->playback_hook = tas2781_hda_playback_hook; + + pm_runtime_mark_last_busy(dev); + pm_runtime_put_autosuspend(dev); + + return 0; +} + +static void tas2781_hda_unbind(struct device *dev, + struct device *master, void *master_data) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + struct hda_component *comps = master_data; + + if (comps[tas_priv->index].dev == dev) + memset(&comps[tas_priv->index], 0, sizeof(*comps)); + + tasdevice_config_info_remove(tas_priv); + tasdevice_dsp_remove(tas_priv); + + tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; +} + +static const struct component_ops tas2781_hda_comp_ops = { + .bind = tas2781_hda_bind, + .unbind = tas2781_hda_unbind, +}; + +static void tas2781_hda_remove(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + pm_runtime_get_sync(tas_priv->dev); + pm_runtime_disable(tas_priv->dev); + + component_del(tas_priv->dev, &tas2781_hda_comp_ops); + + pm_runtime_put_noidle(tas_priv->dev); + + tasdevice_remove(tas_priv); +} + +static int tas2781_hda_i2c_probe(struct i2c_client *clt) +{ + struct tasdevice_priv *tas_priv; + const char *device_name; + int ret; + + if (strstr(dev_name(&clt->dev), "TIAS2781")) + device_name = "TIAS2781"; + else + return -ENODEV; + + tas_priv = tasdevice_kzalloc(clt); + if (!tas_priv) + return -ENOMEM; + + tas_priv->irq_info.irq = clt->irq; + ret = tas2781_read_acpi(tas_priv, device_name); + if (ret) + return dev_err_probe(tas_priv->dev, ret, + "Platform not supported\n"); + + ret = tasdevice_init(tas_priv); + if (ret) + goto err; + + pm_runtime_set_autosuspend_delay(tas_priv->dev, 3000); + pm_runtime_use_autosuspend(tas_priv->dev); + pm_runtime_mark_last_busy(tas_priv->dev); + pm_runtime_set_active(tas_priv->dev); + pm_runtime_get_noresume(tas_priv->dev); + pm_runtime_enable(tas_priv->dev); + + pm_runtime_put_autosuspend(tas_priv->dev); + + ret = component_add(tas_priv->dev, &tas2781_hda_comp_ops); + if (ret) { + dev_err(tas_priv->dev, "Register component failed: %d\n", ret); + pm_runtime_disable(tas_priv->dev); + goto err; + } + + tas2781_reset(tas_priv); +err: + if (ret) + tas2781_hda_remove(&clt->dev); + return ret; +} + +static void tas2781_hda_i2c_remove(struct i2c_client *clt) +{ + tas2781_hda_remove(&clt->dev); +} + +static int tas2781_runtime_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int i; + + dev_dbg(tas_priv->dev, "Runtime Suspend\n"); + + mutex_lock(&tas_priv->codec_lock); + + if (tas_priv->playback_started) { + tasdevice_tuning_switch(tas_priv, 1); + tas_priv->playback_started = false; + } + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static int tas2781_runtime_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int ret; + + dev_dbg(tas_priv->dev, "Runtime Resume\n"); + + mutex_lock(&tas_priv->codec_lock); + + regcache_cache_only(tas_priv->regmap, false); + ret = regcache_sync(tas_priv->regmap); + if (ret) { + dev_err(tas_priv->dev, + "Failed to restore register cache: %d\n", ret); + goto out; + } + + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still works with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + +out: + mutex_unlock(&tas_priv->codec_lock); + + return ret; +} + +static int tas2781_system_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + dev_dbg(tas_priv->dev, "System Suspend\n"); + + ret = pm_runtime_force_suspend(dev); + if (ret) + return ret; + + /* Shutdown chip before system suspend */ + regcache_cache_only(tas_priv->regmap, false); + tasdevice_tuning_switch(tas_priv, 1); + regcache_cache_only(tas_priv->regmap, true); + regcache_mark_dirty(tas_priv->regmap); + + /* + * Reset GPIO may be shared, so cannot reset here. + * However beyond this point, amps may be powered down. + */ + return 0; +} + +static int tas2781_system_resume(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long calib_data_sz = + tas_priv->ndev * TASDEVICE_SPEAKER_CALIBRATION_SIZE; + int i, ret; + + dev_dbg(tas_priv->dev, "System Resume\n"); + + ret = pm_runtime_force_resume(dev); + if (ret) + return ret; + + mutex_lock(&tas_priv->codec_lock); + + for (i = 0; i < tas_priv->ndev; i++) { + tas_priv->tasdevice[i].cur_book = -1; + tas_priv->tasdevice[i].cur_prog = -1; + tas_priv->tasdevice[i].cur_conf = -1; + } + tas2781_reset(tas_priv); + tasdevice_prmg_load(tas_priv, tas_priv->cur_prog); + + /* If calibrated data occurs error, dsp will still work with default + * calibrated data inside algo. + */ + if (tas_priv->cali_data.total_sz > calib_data_sz) + tas2781_apply_calib(tas_priv); + mutex_unlock(&tas_priv->codec_lock); + + return 0; +} + +static const struct dev_pm_ops tas2781_hda_pm_ops = { + RUNTIME_PM_OPS(tas2781_runtime_suspend, tas2781_runtime_resume, NULL) + SYSTEM_SLEEP_PM_OPS(tas2781_system_suspend, tas2781_system_resume) +}; + +static const struct i2c_device_id tas2781_hda_i2c_id[] = { + { "tas2781-hda", 0 }, + {} +}; + +static const struct acpi_device_id tas2781_acpi_hda_match[] = { + {"TIAS2781", 0 }, + {} +}; +MODULE_DEVICE_TABLE(acpi, tas2781_acpi_hda_match); + +static struct i2c_driver tas2781_hda_i2c_driver = { + .driver = { + .name = "tas2781-hda", + .acpi_match_table = tas2781_acpi_hda_match, + .pm = &tas2781_hda_pm_ops, + }, + .id_table = tas2781_hda_i2c_id, + .probe_new = tas2781_hda_i2c_probe, + .remove = tas2781_hda_i2c_remove, +}; +module_i2c_driver(tas2781_hda_i2c_driver); + +MODULE_DESCRIPTION("TAS2781 HDA Driver"); +MODULE_AUTHOR("Shenghao Ding, TI, <shenghao-ding(a)ti.com>"); +MODULE_LICENSE("GPL"); +MODULE_IMPORT_NS(SND_SOC_TAS2781_FMWLIB); -- 2.34.1

4 12

[PATCH 1/3] ASoC: dt-bindings: fsl_easrc: Add support for imx8mp-easrc
by Adam Ford 21 Aug '23

21 Aug '23

The i.MX8MP appears to have the same easrc support as the Nano, so add imx8mp as an option with a fallback to imx8mn. Signed-off-by: Adam Ford <aford173(a)gmail.com> diff --git a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml index bdde68a1059c..2d53b3b10f2c 100644 --- a/Documentation/devicetree/bindings/sound/fsl,easrc.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,easrc.yaml @@ -14,7 +14,11 @@ properties: pattern: "^easrc@.*" compatible: - const: fsl,imx8mn-easrc + oneOf: + - items: + - enum: + - fsl,imx8mp-easrc + - const: fsl,imx8mn-easrc reg: maxItems: 1 -- 2.39.2

3 9

[PATCH] iommu: Remove the device_lock_assert() from __iommu_probe_device()
by Jason Gunthorpe 21 Aug '23

21 Aug '23

It turns out several drivers are calling of_dma_configure() outside the expected bus_type.dma_configure op. This ends up being mis-locked and triggers a lockdep assertion, or instance: iommu_probe_device_locked+0xd4/0xe4 of_iommu_configure+0x10c/0x200 of_dma_configure_id+0x104/0x3b8 a6xx_gmu_init+0x4c/0xccc [msm] a6xx_gpu_init+0x3ac/0x770 [msm] adreno_bind+0x174/0x2ac [msm] component_bind_all+0x118/0x24c msm_drm_bind+0x1e8/0x6c4 [msm] try_to_bring_up_aggregate_device+0x168/0x1d4 __component_add+0xa8/0x170 component_add+0x14/0x20 dsi_dev_attach+0x20/0x2c [msm] dsi_host_attach+0x9c/0x144 [msm] devm_mipi_dsi_attach+0x34/0xb4 lt9611uxc_attach_dsi.isra.0+0x84/0xfc [lontium_lt9611uxc] lt9611uxc_probe+0x5c8/0x68c [lontium_lt9611uxc] i2c_device_probe+0x14c/0x290 really_probe+0x148/0x2b4 __driver_probe_device+0x78/0x12c driver_probe_device+0x3c/0x160 __device_attach_driver+0xb8/0x138 bus_for_each_drv+0x84/0xe0 __device_attach+0xa8/0x1b0 device_initial_probe+0x14/0x20 bus_probe_device+0xb0/0xb4 deferred_probe_work_func+0x8c/0xc8 process_one_work+0x1ec/0x53c worker_thread+0x298/0x408 kthread+0x124/0x128 ret_from_fork+0x10/0x20 It is subtle and was never documented or enforced, but there has always been an assumption that of_dma_configure_id() is not concurrent. It makes several calls into the iommu layer that require this, including dev_iommu_get(). The majority of cases have been preventing concurrency using the device_lock(). Thus the new lock debugging added exposes an existing problem in drivers. On inspection this looks like a theoretical locking problem as generally the cases are already assuming they are the exclusive (single threaded) user of the target device. Sadly, there are deeper technical problems with all of the places doing this. There are several problemetic patterns: 1) Probe a driver on device A and then steal device B and use it as part of the driver operation. Since no driver was probed to device B it means we never called bus_type.dma_configure and thus the drivers hackily try to open code this. Unfortunately nothing prevents another driver from binding to device B and creating total chaos. eg vfio bind triggered by userspace 2) Probe a driver on device A and then create a new platform driver B for a fwnode that doesn't have one, then do #1 This has the same essential problem as #1, the new device is never probed so the hack call to of_dma_configure() is needed to setup DMA, and we are at risk of something else trying to use the device. 3) Probe a driver on device A but the of_node was incorrect for DMA so fix it by figuring out the right node and calling of_dma_configure() This will blow up in the iommu code if the driver is unprobed because the bus_type now assumes that dma_configure and dma_cleanup are strictly paired. Since dma_configure will have done the wrong thing due to the missing of_node, dma_cleanup will be unpaired and iommu_device_unuse_default_domain() will blow up. Further the driver operating on device A will not be protected against changes to the iommu domain since it never called iommu_device_use_default_domain() At least this case will not throw a lockdep warning as probe is done under the device_lock already. While it is tempting to just add a device_lock() around the few missing places, this doesn't solve the above underlying problems with those callers. For this reason remove the locking assertion in the iommu code so that theses driver problems continue to remain hidden. Add a comment to all the problematic call sites to encourage fixing and discourage future cut & paste expanding the problem. Add documentation to of_dma_configure_id() explaining the locking. Fixes: b19ca5922dff ("iommu: Do not attempt to re-lock the iommu device when probing") Reported-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Closes: https://lore.kernel.org/all/d090f196-a5c2-b188-31bf-e42553d8d251@samsung.co… Signed-off-by: Jason Gunthorpe <jgg(a)nvidia.com> --- drivers/bcma/main.c | 4 ++++ drivers/dma/qcom/hidma_mgmt.c | 4 ++++ drivers/gpu/drm/etnaviv/etnaviv_drv.c | 4 ++++ drivers/gpu/drm/msm/adreno/a6xx_gmu.c | 8 ++++++++ drivers/gpu/drm/sun4i/sun4i_backend.c | 4 ++++ drivers/gpu/drm/sun4i/sun8i_mixer.c | 4 ++++ drivers/gpu/drm/vc4/vc4_drv.c | 4 ++++ drivers/gpu/host1x/bus.c | 4 ++++ drivers/iommu/iommu.c | 12 ++++++++---- drivers/media/platform/qcom/venus/firmware.c | 4 ++++ drivers/media/platform/samsung/s5p-mfc/s5p_mfc.c | 4 ++++ drivers/net/wireless/ath/ath10k/snoc.c | 4 ++++ drivers/net/wireless/ath/ath11k/ahb.c | 4 ++++ drivers/of/device.c | 3 +++ sound/soc/bcm/bcm63xx-pcm-whistler.c | 4 ++++ 15 files changed, 67 insertions(+), 4 deletions(-) This is my suggestion to the issue, I don't like removing the locking assertion but if we can't have debug kernels reporting existing bugs then this would be the next best option. I've CC'd all the driver owners, if people have better explanations for what their drivers are doing please share :) diff --git a/drivers/bcma/main.c b/drivers/bcma/main.c index 7061d3ee836a15..4df97000d7e0a7 100644 --- a/drivers/bcma/main.c +++ b/drivers/bcma/main.c @@ -208,6 +208,10 @@ static void bcma_of_fill_device(struct device *parent, core->irq = bcma_of_get_irq(parent, core, 0); + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(&core->dev, node, false); } diff --git a/drivers/dma/qcom/hidma_mgmt.c b/drivers/dma/qcom/hidma_mgmt.c index 05e96b31d87169..2efd97747d37b3 100644 --- a/drivers/dma/qcom/hidma_mgmt.c +++ b/drivers/dma/qcom/hidma_mgmt.c @@ -390,6 +390,10 @@ static int __init hidma_mgmt_of_populate_channels(struct device_node *np) goto out; } new_pdev->dev.of_node = child; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(&new_pdev->dev, child, true); /* * It is assumed that calling of_msi_configure is safe on diff --git a/drivers/gpu/drm/etnaviv/etnaviv_drv.c b/drivers/gpu/drm/etnaviv/etnaviv_drv.c index 31a7f59ccb49ee..22729c66e39a65 100644 --- a/drivers/gpu/drm/etnaviv/etnaviv_drv.c +++ b/drivers/gpu/drm/etnaviv/etnaviv_drv.c @@ -635,6 +635,10 @@ static int etnaviv_pdev_probe(struct platform_device *pdev) * device as the GPU we found. This assumes that all Vivante * GPUs in the system share the same DMA constraints. */ + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ if (first_node) of_dma_configure(&pdev->dev, first_node, true); diff --git a/drivers/gpu/drm/msm/adreno/a6xx_gmu.c b/drivers/gpu/drm/msm/adreno/a6xx_gmu.c index 5deb79924897af..509bd8b7ec12ae 100644 --- a/drivers/gpu/drm/msm/adreno/a6xx_gmu.c +++ b/drivers/gpu/drm/msm/adreno/a6xx_gmu.c @@ -1492,6 +1492,10 @@ int a6xx_gmu_wrapper_init(struct a6xx_gpu *a6xx_gpu, struct device_node *node) gmu->dev = &pdev->dev; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(gmu->dev, node, true); pm_runtime_enable(gmu->dev); @@ -1556,6 +1560,10 @@ int a6xx_gmu_init(struct a6xx_gpu *a6xx_gpu, struct device_node *node) gmu->dev = &pdev->dev; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(gmu->dev, node, true); /* Fow now, don't do anything fancy until we get our feet under us */ diff --git a/drivers/gpu/drm/sun4i/sun4i_backend.c b/drivers/gpu/drm/sun4i/sun4i_backend.c index b11dbd50d73e98..30cbefb6799e51 100644 --- a/drivers/gpu/drm/sun4i/sun4i_backend.c +++ b/drivers/gpu/drm/sun4i/sun4i_backend.c @@ -800,6 +800,10 @@ static int sun4i_backend_bind(struct device *dev, struct device *master, * for us, and DRM doesn't do per-device allocation either, so * we would need to fix DRM first... */ + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(drm->dev, dev->of_node, true); if (ret) return ret; diff --git a/drivers/gpu/drm/sun4i/sun8i_mixer.c b/drivers/gpu/drm/sun4i/sun8i_mixer.c index 11d5244a5aa5fc..e26bea207599d3 100644 --- a/drivers/gpu/drm/sun4i/sun8i_mixer.c +++ b/drivers/gpu/drm/sun4i/sun8i_mixer.c @@ -399,6 +399,10 @@ static int sun8i_mixer_bind(struct device *dev, struct device *master, * DRM doesn't do per-device allocation either, so we * would need to fix DRM first... */ + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(drm->dev, dev->of_node, true); if (ret) return ret; diff --git a/drivers/gpu/drm/vc4/vc4_drv.c b/drivers/gpu/drm/vc4/vc4_drv.c index 823395c23cc308..82496d8fb50f84 100644 --- a/drivers/gpu/drm/vc4/vc4_drv.c +++ b/drivers/gpu/drm/vc4/vc4_drv.c @@ -305,6 +305,10 @@ static int vc4_drm_bind(struct device *dev) node = of_find_matching_node_and_match(NULL, vc4_dma_range_matches, NULL); if (node) { + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(dev, node, true); of_node_put(node); diff --git a/drivers/gpu/host1x/bus.c b/drivers/gpu/host1x/bus.c index 4d16a3396c4abf..5c7ef738a7d7d7 100644 --- a/drivers/gpu/host1x/bus.c +++ b/drivers/gpu/host1x/bus.c @@ -475,6 +475,10 @@ static int host1x_device_add(struct host1x *host1x, device->dev.bus = &host1x_bus_type; device->dev.parent = host1x->dev; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(&device->dev, host1x->dev->of_node, true); device->dev.dma_parms = &device->dma_parms; diff --git a/drivers/iommu/iommu.c b/drivers/iommu/iommu.c index 18162049bd2294..eb623a611f571d 100644 --- a/drivers/iommu/iommu.c +++ b/drivers/iommu/iommu.c @@ -458,8 +458,15 @@ static int __iommu_probe_device(struct device *dev, struct list_head *group_list * Allow __iommu_probe_device() to be safely called in parallel, * both dev->iommu_group and the initial setup of dev->iommu are * protected this way. + * + * We cannot have the assertion yet because of a number of drivers + * improperly calling of_dma_configure() and expecting to attach + * the iommu outside the bus_type.dma_configure callback. This messes + * up our ownership tracking, does not exclude other drivers from + * binding an operating the device, and is technically mislocked. + * + * device_lock_assert(dev); */ - device_lock_assert(dev); if (!ops) return -ENODEV; @@ -527,8 +534,6 @@ int iommu_probe_device_locked(struct device *dev) const struct iommu_ops *ops; int ret; - device_lock_assert(dev); - ret = __iommu_probe_device(dev, NULL); if (ret) return ret; @@ -1202,7 +1207,6 @@ void iommu_group_remove_device(struct device *dev) __iommu_group_remove_device(dev); } device_unlock(dev); - } EXPORT_SYMBOL_GPL(iommu_group_remove_device); diff --git a/drivers/media/platform/qcom/venus/firmware.c b/drivers/media/platform/qcom/venus/firmware.c index cfb11c551167ce..f70360c5534d64 100644 --- a/drivers/media/platform/qcom/venus/firmware.c +++ b/drivers/media/platform/qcom/venus/firmware.c @@ -296,6 +296,10 @@ int venus_firmware_init(struct venus_core *core) pdev->dev.of_node = np; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(&pdev->dev, np, true); if (ret) { dev_err(core->dev, "dma configure fail\n"); diff --git a/drivers/media/platform/samsung/s5p-mfc/s5p_mfc.c b/drivers/media/platform/samsung/s5p-mfc/s5p_mfc.c index e30e54935d79a9..40fa7b62545d7a 100644 --- a/drivers/media/platform/samsung/s5p-mfc/s5p_mfc.c +++ b/drivers/media/platform/samsung/s5p-mfc/s5p_mfc.c @@ -1114,6 +1114,10 @@ static struct device *s5p_mfc_alloc_memdev(struct device *dev, * to be treated as valid DMA masters we need a bit of a hack to force * them to inherit the MFC node's DMA configuration. */ + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(child, dev->of_node, true); if (device_add(child) == 0) { diff --git a/drivers/net/wireless/ath/ath10k/snoc.c b/drivers/net/wireless/ath/ath10k/snoc.c index 26214c00cd0d7b..977e5c6186b131 100644 --- a/drivers/net/wireless/ath/ath10k/snoc.c +++ b/drivers/net/wireless/ath/ath10k/snoc.c @@ -1614,6 +1614,10 @@ static int ath10k_fw_init(struct ath10k *ar) pdev->dev.of_node = node; + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(&pdev->dev, node, true); if (ret) { ath10k_err(ar, "dma configure fail: %d\n", ret); diff --git a/drivers/net/wireless/ath/ath11k/ahb.c b/drivers/net/wireless/ath/ath11k/ahb.c index 139da578831a44..a4d2a89ab96de0 100644 --- a/drivers/net/wireless/ath/ath11k/ahb.c +++ b/drivers/net/wireless/ath/ath11k/ahb.c @@ -1012,6 +1012,10 @@ static int ath11k_ahb_fw_resources_init(struct ath11k_base *ab) return PTR_ERR(pdev); } + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ ret = of_dma_configure(&pdev->dev, node, true); if (ret) { ath11k_err(ab, "dma configure fail: %d\n", ret); diff --git a/drivers/of/device.c b/drivers/of/device.c index 0f00f1b8070829..bc6f62ebcbd040 100644 --- a/drivers/of/device.c +++ b/drivers/of/device.c @@ -108,6 +108,9 @@ of_dma_set_restricted_buffer(struct device *dev, struct device_node *np) * If platform code needs to use its own special DMA configuration, it * can use a platform bus notifier and handle BUS_NOTIFY_ADD_DEVICE events * to fix up DMA configuration. + * + * This function should only be called from the bus_type.dma_configure callback + * and the driver_lock(dev) must be held. */ int of_dma_configure_id(struct device *dev, struct device_node *np, bool force_dma, const u32 *id) diff --git a/sound/soc/bcm/bcm63xx-pcm-whistler.c b/sound/soc/bcm/bcm63xx-pcm-whistler.c index 2c600b017524f3..d8e0520ca61155 100644 --- a/sound/soc/bcm/bcm63xx-pcm-whistler.c +++ b/sound/soc/bcm/bcm63xx-pcm-whistler.c @@ -354,6 +354,10 @@ static int bcm63xx_soc_pcm_new(struct snd_soc_component *component, i2s_priv = dev_get_drvdata(asoc_rtd_to_cpu(rtd, 0)->dev); + /* + * FIXME: of_dma_configure() should not be called outside + * bus_type.dma_configure + */ of_dma_configure(pcm->card->dev, pcm->card->dev->of_node, 1); ret = dma_coerce_mask_and_coherent(pcm->card->dev, DMA_BIT_MASK(32)); base-commit: 1bba4f1c0b4f2429e16da231a246ab4b99cc633c -- 2.41.0

4 7

[PATCH] ASoC: amd: yc: Add VivoBook Pro 15 to quirks list for acp6x
by BrenoRCBrito 21 Aug '23

21 Aug '23

VivoBook Pro 15 Ryzen Edition uses Ryzen 6800H processor, and adding to quirks list for acp6x will enable internal mic. Signed-off-by: BrenoRCBrito <brenorcbrito(a)gmail.com> --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index a2fe3bd4f9a1..a2661dd637a5 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "M3402RA"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."), + DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.40.1

2 1

[PATCH 0/4] ASoC: amd: acp: Add sound support for a line of HUAWEI laptops
by Marian Postevca 21 Aug '23

21 Aug '23

This series adds support for a line of HUAWEI laptops with AMD CPUs that connect using the ACP3x module to a ES8336 codec. The codec driver must be extended to support the S32 LE format and the MCLK div by 2 option. MCLK div by 2 is needed for one specific SKU which uses a 48Mhz MCLK which seems to be too high of a frequency for the codec and must be divided by 2. The acp legacy driver must also be extended by using callbacks so that the more complicated handling for this specific codec can be moved outside the more generic ACP code. The last patch tries to avoid anoying pop sounds when the speaker/headphones are enabled/disabled by delaying the handling of the GPIOs and using a mutex to avoid race conditions between the speaker power event callback and the trigger callback. Marian Postevca (4): ASoC: es8316: Enable support for S32 LE format and MCLK div by 2 ASoC: amd: acp: Add support for splitting the codec specific code from the ACP driver ASoC: amd: acp: Add machine driver that enables sound for systems with a ES8336 codec ASoC: amd: acp: Improve support for speaker power events sound/soc/amd/acp-config.c | 70 ++ sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-legacy-mach.c | 105 ++- sound/soc/amd/acp/acp-mach-common.c | 8 + sound/soc/amd/acp/acp-mach.h | 67 ++ sound/soc/amd/acp/acp-renoir.c | 4 + sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c | 615 ++++++++++++++++++ sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h | 12 + sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8316.h | 3 + 10 files changed, 886 insertions(+), 21 deletions(-) create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.c create mode 100644 sound/soc/amd/acp/acp3x-es83xx/acp3x-es83xx.h -- 2.39.1

4 25

[PATCH] ALSA: usb-audio: Fix init call orders for UAC1
by Takashi Iwai 21 Aug '23

21 Aug '23

There have been reports of USB-audio driver spewing errors at the probe time on a few devices like Jabra and Logitech. The suggested fix there couldn't be applied as is, unfortunately, because it'll likely break other devices. But, the patch suggested an interesting point: looking at the current init code in stream.c, one may notice that it does initialize differently from the device setup in endpoint.c. Namely, for UAC1, we should call snd_usb_init_pitch() and snd_usb_init_sample_rate() after setting the interface, while the init sequence at parsing calls them before setting the interface blindly. This patch changes the init sequence at parsing for UAC1 (and other devices that need a similar behavior) to be aligned with the rest of the code, setting the interface at first. And, this fixes the long-standing problems on a few UAC1 devices like Jabra / Logitech, as reported, too. Reported-and-tested-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> Closes: https://lore.kernel.org/r/202bbbc0f51522e8545783c4c5577d12a8e2d56d.camel@in… Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/stream.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index f10f4e6d3fb8..3d4add94e367 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -1093,6 +1093,7 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int i, altno, err, stream; struct audioformat *fp = NULL; struct snd_usb_power_domain *pd = NULL; + bool set_iface_first; int num, protocol; dev = chip->dev; @@ -1223,11 +1224,19 @@ static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, return err; } + set_iface_first = false; + if (protocol == UAC_VERSION_1 || + (chip->quirk_flags & QUIRK_FLAG_SET_IFACE_FIRST)) + set_iface_first = true; + /* try to set the interface... */ usb_set_interface(chip->dev, iface_no, 0); + if (set_iface_first) + usb_set_interface(chip->dev, iface_no, altno); snd_usb_init_pitch(chip, fp); snd_usb_init_sample_rate(chip, fp, fp->rate_max); - usb_set_interface(chip->dev, iface_no, altno); + if (!set_iface_first) + usb_set_interface(chip->dev, iface_no, altno); } return 0; } -- 2.35.3

1 0

[PATCH][next] ALSA: hda/realtek: Fix spelling mistake "powe" -> "power"
by Colin Ian King 21 Aug '23

21 Aug '23

There is a spelling mistake in a quirk entry. Fix it. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6707db55f9c5..54e17791c6a8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9967,7 +9967,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), - SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual power mode2 YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), -- 2.39.2

2 1

How to determine which files were read/parsed
by GitHub issues - opened 21 Aug '23

21 Aug '23

alsa-project/alsa-lib issue #345 was opened from bmariuszb: I would like to know which files were read by alsa doing `alsactl restore` or `alsactl init`. So far I know that there is a folder `/usr/share/alsa` and it has `init/00main`, and this is probably read first, because when I deleted whole alsa dir I got error about this file, then I don't know what is read. Also there is a file `/var/lib/alsa/asound.state` which from my understanding contains values for different controls that are visible in alsamixer. There is also "user" config file in `/etc/asound.conf` and optionally any user can create `~/.asound.conf`. Am I missing any other files? getting the list of files in order they are read for each card or all cards would be helpfull in debugging. For example by default I found out that my card HDA Intel PCH has two devices ALC1150 Analog and ALC1150 Digital. Digital is probably one of the six jacks in the back of my PC that are unplugged so I don't care about them right now. However Analog in my PC is connected to speakers and also from motherboard comes another cable to front jack panel that I plugged my headphones. I have a few problems with the default configuration: 1. Headphone slider in alsamixer, known as control doesn't work. 2. There are too many controls in Playback that are useless in my opinion. 3. I haven't found yet how headphone and speakers jacks where merged into one Analog device. I wonder if it's a software merge or a hardware thing. I would like to combine card 2: NVidia [HDA NVidia], device 3: HDMI 0 [X2483_2481] with the merged card 1 device 0 (headphones and speakers) and I was wondering if it would be better to merge all 3 devices at once. Correct me if I'm wrong, but from my understanding you have to connect different devices to be able to dynamically switch between them, because normally every process, e.g browser connects to a pcm device at startup and you can't change it later. So the actual audio switching is unmuting one thing and muting rest of them. All "things" are connected to one pcm, right? Please correct me if I'm wrong or use wrong terminology. ![image](https://github.com/alsa-project/alsa-lib/assets/57236506/2d73bb21-e9ee-4172-8e8e-04785cf0a887) Issue URL : https://github.com/alsa-project/alsa-lib/issues/345 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 1/2] ALSA: hda/tas2781: Fix acpi device refcount leak at tas2781_read_acpi()
by Takashi Iwai 20 Aug '23

20 Aug '23

The error path at tas2781_read_acpi() doesn't release the acpi_device adev but releases another device physdev instead. This results in a refcount leak. Fix it by replacing with the right object. Fixes: 5be27f1e3ec9 ("ALSA: hda/tas2781: Add tas2781 HDA driver") Reported-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Closes: https://lore.kernel.org/r/9f910785-e856-1539-e3e4-c9817af5fe67@linux.intel.… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 35dafc4aec4f..0968ae915fd0 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -118,7 +118,7 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) err: dev_err(p->dev, "read acpi error, ret: %d\n", ret); - put_device(physdev); + acpi_dev_put(adev); return ret; } -- 2.35.3

1 1

A potential Integer Overflow bug found in aplay/aplay.c
by GitHub issues - edited 20 Aug '23

20 Aug '23

alsa-project/alsa-utils issue #231 was edited from cascades-sjtu: Hi, I'm currently trying to use the static analysis tool [Infer](https://fbinfer.com) to find uncatched API-misuse bugs in OpenWrt packages, and I find a potential Integer Overflow in your project, version 1.2.9. The bug located in aplay/aplay.c. Firstly, the program tries to write the remains bytes in `audiobuf` with the length of `loaded` in line 2865, and `loaded` is later used as the parameter for `safe_read` in the loop, then `loaded` is used as the 2nd argument of `pct_write()` and finally after a multiply operation, it is used as the size of Malloc in `remap_data()`, as shown in the following code: ```cpp static void playback_go(int fd, size_t loaded, off_t count, int rtype, char *name) { int l, r; off_t written = 0; off_t c; header(rtype, name); set_params(); while (loaded > chunk_bytes && written < count && !in_aborting) { if (pcm_write(audiobuf + written, chunk_size) <= 0) return; written += chunk_bytes; loaded -= chunk_bytes; } if (written > 0 && loaded > 0) memmove(audiobuf, audiobuf + written, loaded); l = loaded; while (written < count && !in_aborting) { do { c = count - written; if (c > chunk_bytes) c = chunk_bytes; /* c < l, there is more data loaded * then we actually need to write */ if (c < l) l = c; c -= l; if (c == 0) break; r = safe_read(fd, audiobuf + l, c); if (r < 0) { perror(name); prg_exit(EXIT_FAILURE); } fdcount += r; if (r == 0) break; l += r; } while ((size_t)l < chunk_bytes); l = l * 8 / bits_per_frame; r = pcm_write(audiobuf, l); if (r != l) break; r = r * bits_per_frame / 8; written += r; l = 0; } if (!in_aborting) { snd_pcm_nonblock(handle, 0); snd_pcm_drain(handle); snd_pcm_nonblock(handle, nonblock); } } # in remap_data() chunk_bytes = count * bits_per_frame / 8; if (tmp_size < chunk_bytes) { free(tmp); tmp = malloc(chunk_bytes); if (!tmp) { error(_("not enough memory")); exit(1); } tmp_size = count; } ``` The parameter passed to Malloc may be overflowed so that the actual allocated memory is small. I also attached the analysis trace given by Infer FYI: ```json "trace": [ { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 2865, "col": 21, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Var" ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 0 ] ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2868, "col": 2, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2899, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2900, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "+", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2895, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "Cast", [ "Unsupported" ], [ "BinOp", "*", [ "Var" ], [ "Const", [ "Cint", 8 ] ] ] ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2896, "col": 7, "feature": [ "Call", "pcm_write" ] }, { "file": "aplay/aplay.c", "line": 2126, "col": 6, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2130, "col": 9, "feature": [ "Call", "remap_data" ] }, { "file": "aplay/aplay.c", "line": 2069, "col": 2, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Var" ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2070, "col": 6, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2072, "col": 9, "feature": [ "IntOverflow", "malloc", [ "Var" ] ] } ], ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/231 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

A potential Integer Overflow bug found in aplay/aplay.c
by GitHub issues - opened 20 Aug '23

20 Aug '23

alsa-project/alsa-utils issue #231 was opened from cascades-sjtu: Hi, I'm currently trying to use the static analysis tool [Infer](https://fbinfer.com) to find uncatched API-misuse bugs in OpenWrt packages, and I find a potential Integer Overflow in your project, version 1.2.9. The bug located in aplay/aplay.c. Firstly, the program tries to write the remains bytes in `audiobuf` with the length of `loaded` in line 2865, and `loaded` is later used as the parameter for `safe_read` in the loop, then `loaded` is used as the 2nd argument of `pct_write()` and finally after a multiply operation, it is used as the size of Malloc in `remap_data()`, as shown in the following code: ```cpp static void playback_go(int fd, size_t loaded, off_t count, int rtype, char *name) { int l, r; off_t written = 0; off_t c; header(rtype, name); set_params(); while (loaded > chunk_bytes && written < count && !in_aborting) { if (pcm_write(audiobuf + written, chunk_size) <= 0) return; written += chunk_bytes; loaded -= chunk_bytes; } if (written > 0 && loaded > 0) memmove(audiobuf, audiobuf + written, loaded); l = loaded; while (written < count && !in_aborting) { do { c = count - written; if (c > chunk_bytes) c = chunk_bytes; /* c < l, there is more data loaded * then we actually need to write */ if (c < l) l = c; c -= l; if (c == 0) break; r = safe_read(fd, audiobuf + l, c); if (r < 0) { perror(name); prg_exit(EXIT_FAILURE); } fdcount += r; if (r == 0) break; l += r; } while ((size_t)l < chunk_bytes); l = l * 8 / bits_per_frame; r = pcm_write(audiobuf, l); if (r != l) break; r = r * bits_per_frame / 8; written += r; l = 0; } if (!in_aborting) { snd_pcm_nonblock(handle, 0); snd_pcm_drain(handle); snd_pcm_nonblock(handle, nonblock); } } # in remap_data() chunk_bytes = count * bits_per_frame / 8; if (tmp_size < chunk_bytes) { free(tmp); tmp = malloc(chunk_bytes); if (!tmp) { error(_("not enough memory")); exit(1); } tmp_size = count; } ``` The parameter passed to Malloc may be overflowed so that the actual allocated memory is small. I also attached the analysis trace given by Infer FYI: ```json "trace": [ { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 940, "col": 14, "feature": [ "Input", "read" ] }, { "file": "aplay/aplay.c", "line": 2865, "col": 21, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Var" ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 0 ] ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2868, "col": 2, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2899, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2900, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "+", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2871, "col": 4, "feature": [ "Store", [ "Var" ], [ "BinOp", "-", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2872, "col": 8, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", ">", [ "Cast", [ "Unsupported" ], [ "Var" ] ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2878, "col": 8, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2879, "col": 5, "feature": [ "Store", [ "Var" ], [ "Var" ] ] }, { "file": "aplay/aplay.c", "line": 2895, "col": 3, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "Cast", [ "Unsupported" ], [ "BinOp", "*", [ "Var" ], [ "Const", [ "Cint", 8 ] ] ] ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2896, "col": 7, "feature": [ "Call", "pcm_write" ] }, { "file": "aplay/aplay.c", "line": 2126, "col": 6, "feature": [ "Prune", [ "UnOp", "!", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2130, "col": 9, "feature": [ "Call", "remap_data" ] }, { "file": "aplay/aplay.c", "line": 2069, "col": 2, "feature": [ "Store", [ "Var" ], [ "BinOp", "/", [ "BinOp", "*", [ "Var" ], [ "Var" ] ], [ "Cast", [ "Unsupported" ], [ "Const", [ "Cint", 8 ] ] ] ] ] }, { "file": "aplay/aplay.c", "line": 2070, "col": 6, "feature": [ "Prune", [ "BinOp", "<", [ "Var" ], [ "Var" ] ] ] }, { "file": "aplay/aplay.c", "line": 2072, "col": 9, "feature": [ "IntOverflow", "malloc", [ "Var" ] ] } ], ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/231 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

raw midi open fails results in verbose message to stdout
by GitHub issues - opened 20 Aug '23

20 Aug '23

alsa-project/alsa-lib issue #344 was opened from symdeb: ALSA lib rawmidi_hw.c:384:(snd_rawmidi_hw_open) open /dev/snd/umpC5D0 failed: No such file or directory Issue URL : https://github.com/alsa-project/alsa-lib/issues/344 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used interchangeably for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… The other Volt interfaces are similar so they surely could be accommodated in this profile as well if someone has them. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used interchangeably for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level signals or Hi-Z instruments. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - edited 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was edited from z411: This adds a profile for the (relatively recent) Universal Audio Volt 2 USB audio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2 profile for Universal Audio Volt 2
by GitHub pull_request - opened 19 Aug '23

19 Aug '23

alsa-project/alsa-ucm-conf pull request #340 was opened from z411: This adds a profile for the (relatively recent) Universal Audio Vol 2 USB aduio interface. It has a Stereo output and 2 mono inputs that are (like most budget interfaces) combined into a stereo input. This profile effectively splits these inputs into two Line inputs. These inputs can be used for XLR microphones, Line level inputs or Hi-Z inputs. The only weird thing is that in my machine the Mono Input 2 appears first in order but pretty sure I defined the prorities in the configuration file just fine. Everything works well on my machine. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12385352… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 1/2] ALSA: hda/tas2781: Add tas2781 HDA driver
by Shenghao Ding 18 Aug '23

18 Aug '23

Integrate tas2781 configs for Lenovo Laptops. All of the tas2781s in the laptop will be aggregated as one audio device. The code support realtek as the primary codec. Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of the certain one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Changes in v4: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://lore.kernel.org/oe-kbuild-all/202308172137.SthCPFbA-lkp@intel. | com/ - remove workaround code for 0x17aa38be, laptop vendor will fix it in acpi. - rename comp_match_tas2781_dev_name to avoid indentation - simplify the check of vendor id with Lenovo - ThinkPad is one of Lenovo's brands, I was suggested me to use ALC269_FIXUP_THINKPAD_ACPI. - Add comments on ACARD_SINGLE_RANGE_EXT_TLV - Add the range check for tas_priv->tasdevice[] in tas2781_acpi_get_i2c_res. - remove acpi_subsystem_id - Issue in Laptop 0x17aa38be ACPI talbe caused codec->bus->pci->subsystem_device is not equal to (codec->core.subsystem_id & 0xffff) in snd_hda_pick_fixup. The former is 0x3802 and the latter is 0x38be leads to getting the wrong fixup_id and enter into the wrong entry. Although, this issue has been raised to the laptop manufacturer, but the ACPI table is locked, cannot be changed any more. Correct the wrong entry in the code. - Rename "struct cs35l41_dev_name" to "struct scodec_dev_name" for all other side codecs instead of one certain codec. - Ignore the checkpatch complaints in alc269_fixup_tbl - Drop the hunk which is irrelevant with my code in alc_fixup_headset_mode_alc255_no_hp_mic - Add tiwai(a)suse.de into Cc list - remove useless index - combine ALC287_FIXUP_TAS2781_I2C_2 and ALC287_FIXUP_TAS2781_I2C_4 together as ALC287_FIXUP_TAS2781_I2C, The code view all the tas2781s in the laptop as one instance. - delete the white space at the end of the line in alc_fixup_headset_mode_alc255_no_hp_mic --- sound/pci/hda/patch_realtek.c | 88 +++++++++++++++++++++++++++++++++-- 1 file changed, 85 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 44fccfb93cff..ba1b02ed184a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6721,7 +6721,7 @@ static void comp_generic_playback_hook(struct hda_pcm_stream *hinfo, struct hda_ } } -struct cs35l41_dev_name { +struct scodec_dev_name { const char *bus; const char *hid; int index; @@ -6730,7 +6730,7 @@ struct cs35l41_dev_name { /* match the device name in a slightly relaxed manner */ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) { - struct cs35l41_dev_name *p = data; + struct scodec_dev_name *p = data; const char *d = dev_name(dev); int n = strlen(p->bus); char tmp[32]; @@ -6746,12 +6746,32 @@ static int comp_match_cs35l41_dev_name(struct device *dev, void *data) return !strcmp(d + n, tmp); } +static int comp_match_tas2781_dev_name(struct device *dev, + void *data) +{ + struct scodec_dev_name *p = data; + const char *d = dev_name(dev); + int n = strlen(p->bus); + char tmp[32]; + + /* check the bus name */ + if (strncmp(d, p->bus, n)) + return 0; + /* skip the bus number */ + if (isdigit(d[n])) + n++; + /* the rest must be exact matching */ + snprintf(tmp, sizeof(tmp), "-%s:00", p->hid); + + return !strcmp(d + n, tmp); +} + static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char *bus, const char *hid, int count) { struct device *dev = hda_codec_dev(cdc); struct alc_spec *spec = cdc->spec; - struct cs35l41_dev_name *rec; + struct scodec_dev_name *rec; int ret, i; switch (action) { @@ -6779,6 +6799,41 @@ static void cs35l41_generic_fixup(struct hda_codec *cdc, int action, const char } } +static void tas2781_generic_fixup(struct hda_codec *cdc, int action, + const char *bus, const char *hid) +{ + struct device *dev = hda_codec_dev(cdc); + struct alc_spec *spec = cdc->spec; + struct scodec_dev_name *rec; + int ret; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + rec = devm_kmalloc(dev, sizeof(*rec), GFP_KERNEL); + if (!rec) + return; + rec->bus = bus; + rec->hid = hid; + rec->index = 0; + spec->comps[0].codec = cdc; + component_match_add(dev, &spec->match, + comp_match_tas2781_dev_name, rec); + ret = component_master_add_with_match(dev, &comp_master_ops, + spec->match); + if (ret) + codec_err(cdc, + "Fail to register component aggregator %d\n", + ret); + else + spec->gen.pcm_playback_hook = + comp_generic_playback_hook; + break; + case HDA_FIXUP_ACT_FREE: + component_master_del(dev, &comp_master_ops); + break; + } +} + static void cs35l41_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { cs35l41_generic_fixup(cdc, action, "i2c", "CSC3551", 2); @@ -6806,6 +6861,12 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st cs35l41_generic_fixup(cdc, action, "i2c", "CLSA0101", 2); } +static void tas2781_fixup_i2c(struct hda_codec *cdc, + const struct hda_fixup *fix, int action) +{ + tas2781_generic_fixup(cdc, action, "i2c", "TIAS2781"); +} + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7231,6 +7292,7 @@ enum { ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, + ALC287_FIXUP_TAS2781_I2C, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9309,6 +9371,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC287_FIXUP_TAS2781_I2C] = { + .type = HDA_FIXUP_FUNC, + .v.func = tas2781_fixup_i2c, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9884,6 +9952,20 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x387d, "Yoga S780-16 pro Quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x387e, "Yoga S780-16 pro Quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3881, "YB9 dual powe mode2 YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3884, "Y780 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x3886, "Y780 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a7, "Y780P AMD YG dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38a8, "Y780P AMD VECO dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38bf, "Yoga S980-14.5 proX LX Dual", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38c3, "Y980 DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cb, "Y790 YG DUAL", ALC287_FIXUP_TAS2781_I2C), + SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), 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), -- 2.34.1

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[PATCH] ASoC: amd: acp: Add kcontrols and widgets per-codec in common code
by Alper Nebi Yasak 18 Aug '23

18 Aug '23

Commit 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") adds card kcontrols and DAPM widgets corresponding to jacks for Headphone and Headset Mic. But these were already being added in acp-legacy-mach.c and acp-sof-mach.c, so this causes a probe error: sof_mach rt5682s-hs-rt1019: control 2:0:0:Headphone Jack Switch:0 is already present sof_mach rt5682s-hs-rt1019: ASoC: Failed to add Headphone Jack Switch: -16 sof_mach rt5682s-hs-rt1019: devm_snd_soc_register_card(sof-rt5682s-hs-rt1019) failed: -16 sof_mach: probe of rt5682s-hs-rt1019 failed with error -16 Removing the new duplicates from the common code might be enough to fix the issue. But some of the codecs use different kcontrols and widgets. NAU8821 alone has an "Int Mic". MAX98360A has a single "Spk", while RT1019 has "Left Spk" and "Right Spk" (and two codec instances with name prefixes). Since there are per-codec differences, add the kcontrols and widgets in the common code as each dai-link is being initialized, instead of in the two machine drivers' probe(). Also, MAX98388 has Left and Right instances, but uses a single "SPK" widget. That seems weird, so normalize it to use two widgets (one per each instance like RT1019) while we're adding controls. Fixes: 7ac3404c2e21 ("ASoC: amd: acp: Map missing jack kcontrols") Signed-off-by: Alper Nebi Yasak <alpernebiyasak(a)gmail.com> --- Applies onto linux-next. sound/soc/amd/acp/acp-legacy-mach.c | 22 +--------- sound/soc/amd/acp/acp-mach-common.c | 68 +++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-sof-mach.c | 21 +-------- 3 files changed, 67 insertions(+), 44 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 676ad50638d0..6d57d17ddfd7 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -75,23 +75,6 @@ static struct acp_card_drvdata rt5682s_rt1019_rmb_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), - -}; - -static const struct snd_soc_dapm_widget acp_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - static int acp_asoc_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -110,11 +93,8 @@ static int acp_asoc_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index f3abaa182fbb..a06af82b8056 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -513,6 +513,16 @@ SND_SOC_DAILINK_DEF(rt1019, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC1019:00", "rt1019-aif"), COMP_CODEC("i2c-10EC1019:01", "rt1019-aif"))); +static const struct snd_kcontrol_new rt1019_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + +static const struct snd_soc_dapm_widget rt1019_widgets[] = { + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), +}; + static const struct snd_soc_dapm_route rt1019_map_lr[] = { { "Left Spk", NULL, "Left SPO" }, { "Right Spk", NULL, "Right SPO" }, @@ -533,10 +543,25 @@ static int acp_card_rt1019_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != RT1019) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, rt1019_widgets, + ARRAY_SIZE(rt1019_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, rt1019_controls, + ARRAY_SIZE(rt1019_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, rt1019_map_lr, ARRAY_SIZE(rt1019_map_lr)); } @@ -667,6 +692,14 @@ static const struct snd_soc_ops acp_card_rt1019_ops = { SND_SOC_DAILINK_DEF(max98360a, DAILINK_COMP_ARRAY(COMP_CODEC("MX98360A:00", "HiFi"))); +static const struct snd_kcontrol_new max98360a_controls[] = { + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget max98360a_widgets[] = { + SND_SOC_DAPM_SPK("Spk", NULL), +}; + static const struct snd_soc_dapm_route max98360a_map[] = { {"Spk", NULL, "Speaker"}, }; @@ -675,10 +708,25 @@ static int acp_card_maxim_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; struct acp_card_drvdata *drvdata = card->drvdata; + int ret; if (drvdata->amp_codec_id != MAX98360A) return -EINVAL; + ret = snd_soc_dapm_new_controls(&card->dapm, max98360a_widgets, + ARRAY_SIZE(max98360a_widgets)); + if (ret) { + dev_err(rtd->dev, "unable to add widget dapm controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_add_card_controls(card, max98360a_controls, + ARRAY_SIZE(max98360a_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98360a_map, ARRAY_SIZE(max98360a_map)); } @@ -744,13 +792,19 @@ SND_SOC_DAILINK_DEF(max98388, DAILINK_COMP_ARRAY(COMP_CODEC("i2c-ADS8388:00", "max98388-aif1"), COMP_CODEC("i2c-ADS8388:01", "max98388-aif1"))); +static const struct snd_kcontrol_new max98388_controls[] = { + SOC_DAPM_PIN_SWITCH("Left Spk"), + SOC_DAPM_PIN_SWITCH("Right Spk"), +}; + static const struct snd_soc_dapm_widget max98388_widgets[] = { - SND_SOC_DAPM_SPK("SPK", NULL), + SND_SOC_DAPM_SPK("Left Spk", NULL), + SND_SOC_DAPM_SPK("Right Spk", NULL), }; static const struct snd_soc_dapm_route max98388_map[] = { - { "SPK", NULL, "Left BE_OUT" }, - { "SPK", NULL, "Right BE_OUT" }, + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, }; static struct snd_soc_codec_conf max98388_conf[] = { @@ -803,6 +857,14 @@ static int acp_card_max98388_init(struct snd_soc_pcm_runtime *rtd) /* Don't need to add routes if widget addition failed */ return ret; } + + ret = snd_soc_add_card_controls(card, max98388_controls, + ARRAY_SIZE(max98388_controls)); + if (ret) { + dev_err(rtd->dev, "unable to add card controls, ret %d\n", ret); + return ret; + } + return snd_soc_dapm_add_routes(&rtd->card->dapm, max98388_map, ARRAY_SIZE(max98388_map)); } diff --git a/sound/soc/amd/acp/acp-sof-mach.c b/sound/soc/amd/acp/acp-sof-mach.c index a1c893f33f74..5223033a122f 100644 --- a/sound/soc/amd/acp/acp-sof-mach.c +++ b/sound/soc/amd/acp/acp-sof-mach.c @@ -94,22 +94,6 @@ static struct acp_card_drvdata sof_nau8821_max98388_data = { .tdm_mode = false, }; -static const struct snd_kcontrol_new acp_controls[] = { - SOC_DAPM_PIN_SWITCH("Headphone Jack"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Spk"), - SOC_DAPM_PIN_SWITCH("Left Spk"), - SOC_DAPM_PIN_SWITCH("Right Spk"), -}; - -static const struct snd_soc_dapm_widget acp_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_SPK("Spk", NULL), - SND_SOC_DAPM_SPK("Left Spk", NULL), - SND_SOC_DAPM_SPK("Right Spk", NULL), -}; - static int acp_sof_probe(struct platform_device *pdev) { struct snd_soc_card *card = NULL; @@ -128,11 +112,8 @@ static int acp_sof_probe(struct platform_device *pdev) card->dev = dev; card->owner = THIS_MODULE; card->name = pdev->id_entry->name; - card->dapm_widgets = acp_widgets; - card->num_dapm_widgets = ARRAY_SIZE(acp_widgets); - card->controls = acp_controls; - card->num_controls = ARRAY_SIZE(acp_controls); card->drvdata = (struct acp_card_drvdata *)pdev->id_entry->driver_data; + /* Widgets and controls added per-codec in acp-mach-common.c */ acp_card_drvdata = card->drvdata; dmi_id = dmi_first_match(acp_quirk_table); base-commit: 47762f08697484cf0c2f2904b8c52375ed26c8cb -- 2.40.1

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