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| author | Linus Torvalds <torvalds@linux-foundation.org> | 2009-12-21 10:10:23 -0800 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2009-12-21 10:10:23 -0800 |
| commit | 9d5eb6787aeb1dd0a0c71c5c41ca1b4cac3e6be4 (patch) | |
| tree | 3953048d5c9c08d9e1f35056534289084b156776 /sound/soc/codecs/stac9766.c | |
| parent | 3d354cbc43db36e7e8b27ed78901064b87864ffc (diff) | |
| parent | 1f26cb92a28541da1c9b27571e40b08333595773 (diff) | |
| download | op-kernel-dev-9d5eb6787aeb1dd0a0c71c5c41ca1b4cac3e6be4.zip op-kernel-dev-9d5eb6787aeb1dd0a0c71c5c41ca1b4cac3e6be4.tar.gz | |
Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (24 commits)
ALSA: sbawe: fix memory detection
ALSA: fix incorrect rounding direction in snd_interval_ratnum()
ALSA: HDA: add powersaving hook for Realtek
ALSA: HDA: remove useless mixers on Aspire 8930G
ALSA: HDA: simplify Aspire 8930G verb array
ALSA: hda: Set Front Mic to input vref 50% for Lenovo 3000 Y410
ALSA: hda/realtek: Remove extra .capsrc_nids initialization for ALC889_INTEL
ALSA: Use kzalloc for allocating only one thing
ALSA: AACI: switch to per-pcm locking
ALSA: AACI: add double-rate support
ALSA: AACI: factor common hw_params logic into aaci_pcm_hw_params
ALSA: AACI: cleanup aaci_pcm_hw_params
ALSA: AACI: simplify codec rate information
ALSA: aaci - Fix a typo
ASoC: wm8974: fix a wrong bit definition
sound: sgio2audio/pdaudiocf/usb-audio: initialize PCM buffer
ALSA: hda - Fix quirk for Maxdata obook4-1
ALSA: hda - Fix missing capsrc_nids for ALC88x
ALSA: hda - Make use of beep device found in Dell Vostro 1015n
ALSA: hda - Fixed internal mic initialization for Dell Vostro 1015
...
Diffstat (limited to 'sound/soc/codecs/stac9766.c')
| -rw-r--r-- | sound/soc/codecs/stac9766.c | 18 |
1 files changed, 1 insertions, 17 deletions
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index bbc72c2..81b8c9d 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); vra |= 0x1; /* enable variable rate audio */ + vra &= ~0x4; /* disable SPDIF output */ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); @@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, return stac9766_ac97_write(codec, reg, runtime->rate); } -static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned short vra; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra &= !0x04; - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); - break; - } - return 0; -} - static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = { static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, - .trigger = ac97_digital_trigger, }; struct snd_soc_dai stac9766_dai[] = { |
