Commit 266bd275 authored by Mark Brown's avatar Mark Brown
Browse files

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>"
parents d6d211db cf86197e

MAINTAINERS

+0 −6
Original line number Diff line number Diff line
@@ -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>

sound/soc/soc-dapm.c

+10 −4
Original line number Diff line number Diff line
@@ -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 */

sound/soc/soc-pcm.c

+1 −1
Original line number Diff line number Diff line
@@ -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))

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