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: (290 commits)
  ALSA: pcm - Update document about xrun_debug proc file
  ALSA: lx6464es - support standard alsa module parameters
  ALSA: snd_usb_caiaq: set mixername
  ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205)
  ALSA: use card device as parent for jack input-devices
  ALSA: sound/ps3: Correct existing and add missing annotations
  ALSA: sound/ps3: Restructure driver source
  ALSA: sound/ps3: Fix checkpatch issues
  ASoC: Fix lm4857 control
  ALSA: ctxfi - Clear PCM resources at hw_params and hw_free
  ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callbacks
  ALSA: ctxfi - Add missing start check in atc_pcm_playback_start()
  ALSA: ctxfi - Add use_system_timer module option
  ALSA: usb - Add boot quirk for C-Media 6206 USB Audio
  ALSA: ctxfi - Fix wrong model id for UAA
  ALSA: ctxfi - Clean up probe routines
  ALSA: hda - Fix the previous tagra-8ch patch
  ALSA: hda - Add 7.1 support for MSI GX620
  ALSA: pcm - A helper function to compose PCM stream name for debug prints
  ALSA: emu10k1 - Fix minimum periods for efx playback
  ...

1 files changed, 27 insertions, 21 deletions

diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 523bad077fa0..abed37acf787 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
+static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
+				 struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	u16 status, rate;
+
+	BUG_ON(event != SND_SOC_DAPM_PRE_PMD);
+
+	/* Gracefully shut down the voice interface. */
+	status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000;
+	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
+	schedule_timeout_interruptible(msecs_to_jiffies(1));
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
+	ac97_write(codec, AC97_EXTENDED_MID, status);
+
+	return 0;
+}
+
+
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
 SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
-SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1),
+SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
+		   wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
 SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
 SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
@@ -689,7 +710,7 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int source)
 	Ndiv = target / source;
 	if ((Ndiv < 5) || (Ndiv > 12))
 		printk(KERN_WARNING
-			"WM9713 PLL N value %d out of recommended range!\n",
+			"WM9713 PLL N value %u out of recommended range!\n",
 			Ndiv);
 
 	pll_div->n = Ndiv;
@@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
-				 struct snd_soc_dai *dai)
-{
-	struct snd_soc_codec *codec = dai->codec;
-	u16 status, rate;
-
-	/* Gracefully shut down the voice interface. */
-	status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
-	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
-	schedule_timeout_interruptible(msecs_to_jiffies(1));
-	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
-	ac97_write(codec, AC97_EXTENDED_MID, status);
-}
-
 static int ac97_hifi_prepare(struct snd_pcm_substream *substream,
 			     struct snd_soc_dai *dai)
 {
@@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
 
 static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
 	.hw_params	= wm9713_pcm_hw_params,
-	.shutdown	= wm9713_voiceshutdown,
 	.set_clkdiv	= wm9713_set_dai_clkdiv,
 	.set_pll	= wm9713_set_dai_pll,
 	.set_fmt	= wm9713_set_dai_fmt,
@@ -1035,13 +1040,13 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "HiFi Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
@@ -1051,7 +1056,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = WM9713_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &wm9713_dai_ops_aux,
 	},
 	{
@@ -1069,6 +1074,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
 	.ops = &wm9713_dai_ops_voice,
+	.symmetric_rates = 1,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);