Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (549 commits)
  ALSA: hda - Fix ADC input-amp handling for Cx20549 codec
  ALSA: hda - Keep EAPD turned on for old Conexant chips
  ALSA: hda/realtek - Fix missing volume controls with ALC260
  ASoC: wm8940: Properly set codec->dapm.bias_level
  ALSA: hda - Fix pin-config for ASUS W90V
  ALSA: hda - Fix surround/CLFE headphone and speaker pins order
  ALSA: hda - Fix typo
  ALSA: Update the sound git tree URL
  ALSA: HDA: Add new revision for ALC662
  ASoC: max98095: Convert codec->hw_write to snd_soc_write
  ASoC: keep pointer to resource so it can be freed
  ASoC: sgtl5000: Fix wrong mask in some snd_soc_update_bits calls
  ASoC: wm8996: Fix wrong mask for setting WM8996_AIF_CLOCKING_2
  ASoC: da7210: Add support for line out and DAC
  ASoC: da7210: Add support for DAPM
  ALSA: hda/realtek - Fix DAC assignments of multiple speakers
  ASoC: Use SGTL5000_LINREG_VDDD_MASK instead of hardcoded mask value
  ASoC: Set sgtl5000->ldo in ldo_regulator_register
  ASoC: wm8996: Use SND_SOC_DAPM_AIF_OUT for AIF2 Capture
  ASoC: wm8994: Use SND_SOC_DAPM_AIF_OUT for AIF3 Capture
  ...

1 files changed, 12 insertions, 2 deletions

diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
index 19241576b6b..5ca122e5118 100644
--- a/sound/soc/codecs/88pm860x-codec.c
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -15,6 +15,7 @@
 #include <linux/platform_device.h>
 #include <linux/mfd/88pm860x.h>
 #include <linux/slab.h>
+#include <linux/delay.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -772,11 +773,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
 
 
 	SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
-			    PM860X_DAC_EN_2, 0, 0),
+			    SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
-			    PM860X_DAC_EN_2, 0, 0),
+			    SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
 			     PM860X_I2S_IFACE_3, 5, 1),
+	SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0),
 	SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
 	SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
 	SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
@@ -868,6 +870,11 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Left ADC", NULL, "Left ADC MOD"},
 	{"Right ADC", NULL, "Right ADC MOD"},
 
+	/* I2S Clock */
+	{"I2S DIN", NULL, "I2S CLK"},
+	{"I2S DIN1", NULL, "I2S CLK"},
+	{"I2S DOUT", NULL, "I2S CLK"},
+
 	/* PCM/AIF1 Inputs */
 	{"PCM SDO", NULL, "ADC Left Mux"},
 	{"PCM SDO", NULL, "ADCR EC Mux"},
@@ -1173,6 +1180,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Enable Audio PLL & Audio section */
+			data = AUDIO_PLL | AUDIO_SECTION_ON;
+			pm860x_reg_write(codec->control_data, REG_MISC2, data);
+			udelay(300);
 			data = AUDIO_PLL | AUDIO_SECTION_RESET
 				| AUDIO_SECTION_ON;
 			pm860x_reg_write(codec->control_data, REG_MISC2, data);