[PATCH 2/7] ASoC: soc-pcm: adjustment for DAI member 0 reset

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 



From: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>

commit 3635bf09a89cf ("ASoC: soc-pcm: add symmetry for channels and
sample bits") set 0 not only to dai->rate but also to dai->channels and
dai->sample_bits if DAI was not active at soc_pcm_close().

and

commit d3383420c969c ("ASoC: soc-pcm: move DAIs parameters cleaning into
hw_free()") moved it from soc_pcm_close() to soc_pcm_hw_free().

These happen at v3.14.
But, maybe because of branch merge conflict or something similar happen
then, soc_pcm_close() still has old settings
(care only dai->rate, doesn't care dai->channels/sample_bits).
This is 100% duplicated operation.

This patch removes soc_pcm_close() side operation which supposed to
already moved to soc_pcm_hw_free().

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@xxxxxxxxxxx>
---
 sound/soc/soc-pcm.c | 9 ---------
 1 file changed, 9 deletions(-)

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 8bc6983..690a912 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -687,15 +687,6 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 
 	snd_soc_runtime_deactivate(rtd, substream->stream);
 
-	/* clear the corresponding DAIs rate when inactive */
-	if (!cpu_dai->active)
-		cpu_dai->rate = 0;
-
-	for_each_rtd_codec_dai(rtd, i, codec_dai) {
-		if (!codec_dai->active)
-			codec_dai->rate = 0;
-	}
-
 	snd_soc_dai_digital_mute(cpu_dai, 1, substream->stream);
 
 	snd_soc_dai_shutdown(cpu_dai, substream);
-- 
2.7.4

_______________________________________________
Alsa-devel mailing list
Alsa-devel@xxxxxxxxxxxxxxxx
https://mailman.alsa-project.org/mailman/listinfo/alsa-devel



[Index of Archives]     [ALSA User]     [Linux Audio Users]     [Pulse Audio]     [Kernel Archive]     [Asterisk PBX]     [Photo Sharing]     [Linux Sound]     [Video 4 Linux]     [Gimp]     [Yosemite News]

  Powered by Linux