Merge "ASoC: pcm: update the start-up sequence for playback" into msm-4.8 (7618eca7) · Commits · e / devices / android_kernel_fairphone_FP3

sound/soc/soc-pcm.c

+21 −2

Original line number	Diff line number	Diff line
		@@ -781,6 +781,11 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)

		mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);

		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		snd_soc_dapm_stream_event(rtd,
		SNDRV_PCM_STREAM_PLAYBACK,
		SND_SOC_DAPM_STREAM_START);

		if (rtd->dai_link->ops && rtd->dai_link->ops->prepare) {
		ret = rtd->dai_link->ops->prepare(substream);
		if (ret < 0) {
		@@ -829,8 +834,15 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		cancel_delayed_work(&rtd->delayed_work);
		}

		snd_soc_dapm_stream_event(rtd, substream->stream,
		if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
		for (i = 0; i < rtd->num_codecs; i++) {
		codec_dai = rtd->codec_dais[i];
		if (codec_dai->capture_active == 1)
		snd_soc_dapm_stream_event(rtd,
		SNDRV_PCM_STREAM_CAPTURE,
		SND_SOC_DAPM_STREAM_START);
		}
		}

		for (i = 0; i < rtd->num_codecs; i++)
		snd_soc_dai_digital_mute(rtd->codec_dais[i], 0,
		@@ -838,6 +850,13 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
		snd_soc_dai_digital_mute(cpu_dai, 0, substream->stream);

		out:
		if (ret < 0 && substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
		pr_err("%s: Issue stop stream for codec_dai due to op failure %d = ret\n",
		__func__, ret);
		snd_soc_dapm_stream_event(rtd,
		SNDRV_PCM_STREAM_PLAYBACK,
		SND_SOC_DAPM_STREAM_STOP);
		}
		mutex_unlock(&rtd->pcm_mutex);
		return ret;
		}