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authorMark Brown <broonie@linaro.org>2014-05-22 03:23:30 +0400
committerMark Brown <broonie@linaro.org>2014-05-22 03:23:30 +0400
commit266bd275b929ca9e36a1e8570748d873337ed9a1 (patch)
tree3c5ae7fa204f95411fcc4b7c97db16995cace3a7 /sound/soc
parentd6d211db37e75de2ddc3a4f979038c40df7cc79c (diff)
parentcf86197ec5655ed1f54f1f7ba34a01d576e3737d (diff)
downloadlinux-266bd275b929ca9e36a1e8570748d873337ed9a1.tar.xz
Merge tag 'asoc-v3.15-rc5-core' into asoc-linus
ASoC: Core fixes for v3.15 A few things here: - Fix the creation of spurious CODEC<->CODEC links which caused DAPM to have audio paths which shouldn't be present causing spurious powerups and potential audible issues for users. - Ensure the suspend->off transition doesn't have spurious transitions to prepare added to the sequence. - Fix incorrect skipping of PCM suspension for active audio streams. - Remove Timur Tabi from the CS4270 maintainers, Cirrus are now doing this and Timur no longer has the boards that he was using. # gpg: Signature made Wed 14 May 2014 12:59:19 BST using RSA key ID 7EA229BD # gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" # gpg: aka "Mark Brown <broonie@debian.org>" # gpg: aka "Mark Brown <broonie@kernel.org>" # gpg: aka "Mark Brown <broonie@tardis.ed.ac.uk>" # gpg: aka "Mark Brown <broonie@linaro.org>" # gpg: aka "Mark Brown <Mark.Brown@linaro.org>"
Diffstat (limited to 'sound/soc')
-rw-r--r--sound/soc/soc-dapm.c14
-rw-r--r--sound/soc/soc-pcm.c2
2 files changed, 11 insertions, 5 deletions
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 7769b0a2bc5a..6d6ceee447d5 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -1612,8 +1612,11 @@ static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
"ASoC: Failed to turn on bias: %d\n", ret);
}
- /* Prepare for a STADDBY->ON or ON->STANDBY transition */
- if (d->bias_level != d->target_bias_level) {
+ /* Prepare for a transition to ON or away from ON */
+ if ((d->target_bias_level == SND_SOC_BIAS_ON &&
+ d->bias_level != SND_SOC_BIAS_ON) ||
+ (d->target_bias_level != SND_SOC_BIAS_ON &&
+ d->bias_level == SND_SOC_BIAS_ON)) {
ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
if (ret != 0)
dev_err(d->dev,
@@ -3475,8 +3478,11 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
cpu_dai = rtd->cpu_dai;
codec_dai = rtd->codec_dai;
- /* dynamic FE links have no fixed DAI mapping */
- if (rtd->dai_link->dynamic)
+ /*
+ * dynamic FE links have no fixed DAI mapping.
+ * CODEC<->CODEC links have no direct connection.
+ */
+ if (rtd->dai_link->dynamic || rtd->dai_link->params)
continue;
/* there is no point in connecting BE DAI links with dummies */
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 2cedf09f6d96..a391de058037 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1675,7 +1675,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream,
be->dpcm[stream].state = SND_SOC_DPCM_STATE_STOP;
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
- if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)
+ if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START)
continue;
if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream))