diff mbox series

ASoC: soc-pcm: fix regression in soc_new_pcm()

Message ID 20200218103824.26708-1-stephan@gerhold.net
State Accepted
Commit a4877a6fb2bd2e356a5eaacd86d6b6d69ff84e69
Headers show
Series ASoC: soc-pcm: fix regression in soc_new_pcm() | expand

Commit Message

Stephan Gerhold Feb. 18, 2020, 10:38 a.m. UTC 
Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai")
swapped the SNDRV_PCM_STREAM_* parameter in the
snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only
for codec2codec links. For normal links it breaks registration of
playback/capture-only PCM devices.

E.g. on qcom/apq8016_sbc there is usually one playback-only and one
capture-only PCM device, but they disappeared after the commit.

The codec2codec case was added in commit a342031cdd08
("ASoC: create pcm for codec2codec links as well") as an extra check
(e.g. `playback = playback && cpu_playback->channels_min`).

We should be able to simplify the code by checking directly for
the correct stream type in the loop.
This also fixes the regression because we check for PLAYBACK for
both codec and cpu dai again when codec2codec is not used.

Cc: Sameer Pujar <spujar@nvidia.com>
Cc: Jerome Brunet <jbrunet@baylibre.com>
Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
---
Original report of the regression: https://lore.kernel.org/alsa-devel/20200217144120.GA243254@gerhold.net/

I'm still quite confused about the crash that was fixed by the commit
that introduced the regression... If anything, the original code
added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well")
should have been too "restrictive" since it checked for both
PLAYBACK and CAPTURE for the cpu dai in the codec2codec case...

Audio works fine gain on apq8016_sbc with this patch,
but I don't have any device with codec2codec to test this.
---
 sound/soc/soc-pcm.c | 15 ++++++---------
 1 file changed, 6 insertions(+), 9 deletions(-)
diff mbox series

Patch

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index ff1b7c7078e5..cfa24d214a9c 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2888,22 +2888,19 @@  int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 		capture = rtd->dai_link->dpcm_capture;
 	} else {
 		/* Adapt stream for codec2codec links */
-		struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ?
-			&cpu_dai->driver->playback : &cpu_dai->driver->capture;
-		struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ?
-			&cpu_dai->driver->capture : &cpu_dai->driver->playback;
+		int cpu_capture = rtd->dai_link->params ?
+			SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+		int cpu_playback = rtd->dai_link->params ?
+			SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
 
 		for_each_rtd_codec_dai(rtd, i, codec_dai) {
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) &&
-			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_CAPTURE))
+			    snd_soc_dai_stream_valid(cpu_dai,   cpu_playback))
 				playback = 1;
 			if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) &&
-			    snd_soc_dai_stream_valid(cpu_dai,   SNDRV_PCM_STREAM_PLAYBACK))
+			    snd_soc_dai_stream_valid(cpu_dai,   cpu_capture))
 				capture = 1;
 		}
-
-		capture = capture && cpu_capture->channels_min;
-		playback = playback && cpu_playback->channels_min;
 	}
 
 	if (rtd->dai_link->playback_only) {