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author | Takashi Iwai <tiwai@suse.de> | 2014-05-14 14:23:48 +0200 |
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committer | Takashi Iwai <tiwai@suse.de> | 2014-05-14 14:23:48 +0200 |
commit | 927cdab3b6fe7e32990f5563d295eedfc355be6d (patch) | |
tree | 15923aeddbef73e103a96799af36fca56aff4e56 | |
parent | 7189eb9b8f7962474956196c301676470542f253 (diff) | |
parent | cf86197ec5655ed1f54f1f7ba34a01d576e3737d (diff) | |
download | op-kernel-dev-927cdab3b6fe7e32990f5563d295eedfc355be6d.zip op-kernel-dev-927cdab3b6fe7e32990f5563d295eedfc355be6d.tar.gz |
Merge tag 'asoc-v3.15-rc5-core' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-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.
-rw-r--r-- | MAINTAINERS | 6 | ||||
-rw-r--r-- | sound/soc/soc-dapm.c | 14 | ||||
-rw-r--r-- | sound/soc/soc-pcm.c | 2 |
3 files changed, 11 insertions, 11 deletions
diff --git a/MAINTAINERS b/MAINTAINERS index e67ea24..ab0baf7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2245,12 +2245,6 @@ L: linux-usb@vger.kernel.org S: Maintained F: drivers/usb/host/ohci-ep93xx.c -CIRRUS LOGIC CS4270 SOUND DRIVER -M: Timur Tabi <timur@tabi.org> -L: alsa-devel@alsa-project.org (moderated for non-subscribers) -S: Odd Fixes -F: sound/soc/codecs/cs4270* - CIRRUS LOGIC AUDIO CODEC DRIVERS M: Brian Austin <brian.austin@cirrus.com> M: Paul Handrigan <Paul.Handrigan@cirrus.com> diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7769b0a..6d6ceee 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 2cedf09..a391de0 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)) |