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.
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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>"

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>

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 */

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))