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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2013-06-13 21:18:33 +0400 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2013-06-13 21:18:33 +0400 |
| commit | 509768f751986f171710319f44170e7dbab37394 (patch) | |
| tree | 30d116169464afb01cbfcab91d8f8d17ac3fe31e /include | |
| parent | 82ea4be61f45d1a7ba1f5151d90124d1ece0c07a (diff) | |
| parent | 902a75cdc491066514aac35b40a756aaa9740173 (diff) | |
| download | linux-509768f751986f171710319f44170e7dbab37394.tar.xz | |
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.h | 3 |
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 */ }; |