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authorLinus Torvalds <torvalds@linux-foundation.org>2013-06-13 10:18:33 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2013-06-13 10:18:33 -0700
commit509768f751986f171710319f44170e7dbab37394 (patch)
tree30d116169464afb01cbfcab91d8f8d17ac3fe31e /include
parent82ea4be61f45d1a7ba1f5151d90124d1ece0c07a (diff)
parent902a75cdc491066514aac35b40a756aaa9740173 (diff)
downloadlinux-509768f751986f171710319f44170e7dbab37394.tar.gz
Merge tag 'asoc-v3.10-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound
Pull ASoC sound updates from Mark Brown:
 "Takashi is travelling at the minute and it'd be good to get the
  MAINTAINERS update in here merged so sending directly.

  As well as the usual driver specifics we've got a couple of core fixes
  here, one fixing capabilities for unidirectional streams and the other
  fixing suspend while audio streams are active.

  The suspend fix is a little involved but mostly as a result of
  removing some special casing that was doing the wrong thing."

* tag 'asoc-v3.10-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound:
  ASoC: tlv320aic3x: Remove deadlock from snd_soc_dapm_put_volsw_aic3x()
  ASoC: dapm: Treat DAI widgets like AIF widgets for power
  ASoC: arizona: Correct AEC loopback enable
  ASoC: pcm: Require both CODEC and CPU support when declaring stream caps
  MAINTAINERS: Remove myself from Wolfson maintainers
  ASoC: wm8994: Ensure microphone detection state is reset on removal
  ASoC: wm8994: Avoid leaking pm_runtime reference on removed jack race
  ASoC: cs42l52: fix hp_gain_enum shift value.
  ASoC: cs42l52: use correct PCM mixer TLV dB scale to match datasheet.
Diffstat (limited to 'include')
-rw-r--r--include/sound/soc-dapm.h3
1 files changed, 2 insertions, 1 deletions
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index d4609029f014..385c6329a967 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -450,7 +450,8 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_aif_in,		/* audio interface input */
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
-	snd_soc_dapm_dai,		/* link to DAI structure */
+	snd_soc_dapm_dai_in,		/* link to DAI structure */
+	snd_soc_dapm_dai_out,
 	snd_soc_dapm_dai_link,		/* link between two DAI structures */
 };