diff options
| author | Mark Brown <broonie@linaro.org> | 2014-05-22 00:23:30 +0100 |
|---|---|---|
| committer | Mark Brown <broonie@linaro.org> | 2014-05-22 00:23:30 +0100 |
| commit | 266bd275b929ca9e36a1e8570748d873337ed9a1 (patch) | |
| tree | 3c5ae7fa204f95411fcc4b7c97db16995cace3a7 /sound/soc | |
| parent | d6d211db37e75de2ddc3a4f979038c40df7cc79c (diff) | |
| parent | cf86197ec5655ed1f54f1f7ba34a01d576e3737d (diff) | |
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.c | 14 | ||||
| -rw-r--r-- | sound/soc/soc-pcm.c | 2 |
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)) |