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: (214 commits)
  ALSA: hda - Add pin-fix for HP dc5750
  ALSA: als4000: Fix potentially invalid DMA mode setup
  ALSA: als4000: enable burst mode
  ALSA: hda - Fix initial capsrc selection in patch_alc269()
  ASoC: TWL4030: Capture route runtime DAPM ordering fix
  ALSA: hda - Add PC-beep whitelist for an Intel board
  ALSA: hda - More relax for pending period handling
  ALSA: hda - Define AC_FMT_* constants
  ALSA: hda - Fix beep frequency on IDT 92HD73xx and 92HD71Bxx codecs
  ALSA: hda - Add support for HDMI HBR passthrough
  ALSA: hda - Set Stream Type in Stream Format according to AES0
  ALSA: hda - Fix Thinkpad X300 so SPDIF is not exposed
  ALSA: hda - FIX to not expose SPDIF on Thinkpad X301, since it does not have the ability to use SPDIF
  ASoC: wm9081: fix resource reclaim in wm9081_register error path
  ASoC: wm8978: fix a memory leak if a wm8978_register fail
  ASoC: wm8974: fix a memory leak if another WM8974 is registered
  ASoC: wm8961: fix resource reclaim in wm8961_register error path
  ASoC: wm8955: fix resource reclaim in wm8955_register error path
  ASoC: wm8940: fix a memory leak if wm8940_register return error
  ASoC: wm8904: fix resource reclaim in wm8904_register error path
  ...

7 files changed, 90 insertions, 8 deletions

diff --git a/include/sound/asound.h b/include/sound/asound.h
index 9f1eecf99e6b..a1803ecea34d 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -212,7 +212,11 @@ typedef int __bitwise snd_pcm_format_t;
 #define	SNDRV_PCM_FORMAT_S18_3BE	((__force snd_pcm_format_t) 41)	/* in three bytes */
 #define	SNDRV_PCM_FORMAT_U18_3LE	((__force snd_pcm_format_t) 42)	/* in three bytes */
 #define	SNDRV_PCM_FORMAT_U18_3BE	((__force snd_pcm_format_t) 43)	/* in three bytes */
-#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_U18_3BE
+#define	SNDRV_PCM_FORMAT_G723_24	((__force snd_pcm_format_t) 44) /* 8 samples in 3 bytes */
+#define	SNDRV_PCM_FORMAT_G723_24_1B	((__force snd_pcm_format_t) 45) /* 1 sample in 1 byte */
+#define	SNDRV_PCM_FORMAT_G723_40	((__force snd_pcm_format_t) 46) /* 8 Samples in 5 bytes */
+#define	SNDRV_PCM_FORMAT_G723_40_1B	((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */
+#define	SNDRV_PCM_FORMAT_LAST		SNDRV_PCM_FORMAT_G723_40_1B
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define	SNDRV_PCM_FORMAT_S16		SNDRV_PCM_FORMAT_S16_LE
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 6e3a29732dc4..85f1c6bf8566 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -174,6 +174,10 @@ struct snd_pcm_ops {
 #define SNDRV_PCM_FMTBIT_U18_3LE	(1ULL << SNDRV_PCM_FORMAT_U18_3LE)
 #define SNDRV_PCM_FMTBIT_S18_3BE	(1ULL << SNDRV_PCM_FORMAT_S18_3BE)
 #define SNDRV_PCM_FMTBIT_U18_3BE	(1ULL << SNDRV_PCM_FORMAT_U18_3BE)
+#define SNDRV_PCM_FMTBIT_G723_24	(1ULL << SNDRV_PCM_FORMAT_G723_24)
+#define SNDRV_PCM_FMTBIT_G723_24_1B	(1ULL << SNDRV_PCM_FORMAT_G723_24_1B)
+#define SNDRV_PCM_FMTBIT_G723_40	(1ULL << SNDRV_PCM_FORMAT_G723_40)
+#define SNDRV_PCM_FMTBIT_G723_40_1B	(1ULL << SNDRV_PCM_FORMAT_G723_40_1B)
 
 #ifdef SNDRV_LITTLE_ENDIAN
 #define SNDRV_PCM_FMTBIT_S16		SNDRV_PCM_FMTBIT_S16_LE
@@ -313,7 +317,7 @@ struct snd_pcm_runtime {
 	struct snd_pcm_mmap_control *control;
 
 	/* -- locking / scheduling -- */
-	unsigned int twake: 1;		/* do transfer (!poll) wakeup */
+	snd_pcm_uframes_t twake; 	/* do transfer (!poll) wakeup if non-zero */
 	wait_queue_head_t sleep;	/* poll sleep */
 	wait_queue_head_t tsleep;	/* transfer sleep */
 	struct fasync_struct *fasync;
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index c0227361a876..9d51d6f35893 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -12,6 +12,9 @@
  * published by the Free Software Foundation.
  */
 
+#define FSI_PORT_A	0
+#define FSI_PORT_B	1
+
 /* flags format
 
  * 0xABCDEEFF
@@ -55,12 +58,14 @@
 #define SH_FSI_GET_IFMT(x)	((x >> 8) & SH_FSI_FMT_MASK)
 #define SH_FSI_GET_OFMT(x)	((x >> 0) & SH_FSI_FMT_MASK)
 
-#define SH_FSI_FMT_MONO		(1 << 0)
-#define SH_FSI_FMT_MONO_DELAY	(1 << 1)
-#define SH_FSI_FMT_PCM		(1 << 2)
-#define SH_FSI_FMT_I2S		(1 << 3)
-#define SH_FSI_FMT_TDM		(1 << 4)
-#define SH_FSI_FMT_TDM_DELAY	(1 << 5)
+#define SH_FSI_FMT_MONO		0
+#define SH_FSI_FMT_MONO_DELAY	1
+#define SH_FSI_FMT_PCM		2
+#define SH_FSI_FMT_I2S		3
+#define SH_FSI_FMT_TDM		4
+#define SH_FSI_FMT_TDM_DELAY	5
+#define SH_FSI_FMT_SPDIF	6
+
 
 #define SH_FSI_IFMT_TDM_CH(x) \
 	(SH_FSI_IFMT(TDM)	| SH_FSI_SET_CH_I(x))
@@ -72,9 +77,41 @@
 #define SH_FSI_OFMT_TDM_DELAY_CH(x) \
 	(SH_FSI_OFMT(TDM_DELAY)	| SH_FSI_SET_CH_O(x))
 
+
+/*
+ * set_rate return value
+ *
+ * see ACKMD/BPFMD on
+ *     ACK_MD (FSI2)
+ *     CKG1   (FSI)
+ *
+ * err:  return value < 0
+ *
+ * 0x-00000AB
+ *
+ * A:  ACKMD value
+ * B:  BPFMD value
+ */
+
+#define SH_FSI_ACKMD_MASK	(0xF << 0)
+#define SH_FSI_ACKMD_512	(1 << 0)
+#define SH_FSI_ACKMD_256	(2 << 0)
+#define SH_FSI_ACKMD_128	(3 << 0)
+#define SH_FSI_ACKMD_64		(4 << 0)
+#define SH_FSI_ACKMD_32		(5 << 0)
+
+#define SH_FSI_BPFMD_MASK	(0xF << 4)
+#define SH_FSI_BPFMD_512	(1 << 4)
+#define SH_FSI_BPFMD_256	(2 << 4)
+#define SH_FSI_BPFMD_128	(3 << 4)
+#define SH_FSI_BPFMD_64		(4 << 4)
+#define SH_FSI_BPFMD_32		(5 << 4)
+#define SH_FSI_BPFMD_16		(6 << 4)
+
 struct sh_fsi_platform_info {
 	unsigned long porta_flags;
 	unsigned long portb_flags;
+	int (*set_rate)(int is_porta, int rate); /* for master mode */
 };
 
 extern struct snd_soc_dai fsi_soc_dai[2];
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 66ff4c124dbd..c5d9987bc897 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -273,6 +273,8 @@
 #define SND_SOC_DAPM_POST_PMD	0x8		/* after widget power down */
 #define SND_SOC_DAPM_PRE_REG	0x10	/* before audio path setup */
 #define SND_SOC_DAPM_POST_REG	0x20	/* after audio path setup */
+#define SND_SOC_DAPM_PRE_POST_PMD \
+				(SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)
 
 /* convenience event type detection */
 #define SND_SOC_DAPM_EVENT_ON(e)	\
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 697e7ffe39d7..65e9d03ed4f5 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -170,6 +170,21 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&xenum }
 
+#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\
+		xmin, xmax, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		  SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r_sx, \
+	.get = snd_soc_get_volsw_2r_sx, \
+	.put = snd_soc_put_volsw_2r_sx, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg_left, \
+		 .rreg = xreg_right, .shift = xshift, \
+		 .min = xmin, .max = xmax} }
+
+
 /*
  * Simplified versions of above macros, declaring a struct and calculating
  * ARRAY_SIZE internally
@@ -329,6 +344,12 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_limit_volume(struct snd_soc_codec *codec,
 	const char *name, int max);
+int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 
 /**
  * struct snd_soc_jack_pin - Describes a pin to update based on jack detection
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
index 3f428d53195b..6c6649656798 100644
--- a/include/sound/tlv320dac33-plat.h
+++ b/include/sound/tlv320dac33-plat.h
@@ -15,6 +15,8 @@
 
 struct tlv320dac33_platform_data {
 	int power_gpio;
+	int mode1_latency; /* latency caused by the i2c writes in us */
+	int auto_fifo_config; /* FIFO config based on the period size */
 	int keep_bclk;	/* Keep the BCLK running in FIFO modes */
 	u8 burst_bclkdiv;
 };
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
index 509efb050176..e475659bd3be 100644
--- a/include/sound/uda134x.h
+++ b/include/sound/uda134x.h
@@ -18,6 +18,18 @@ struct uda134x_platform_data {
 	struct l3_pins l3;
 	void (*power) (int);
 	int model;
+	/*
+	  ALSA SOC usually puts the device in standby mode when it's not used
+	  for sometime. If you unset is_powered_on_standby the driver will
+	  turn off the ADC/DAC when this callback is invoked and turn it back
+	  on when needed. Unfortunately this will result in a very light bump
+	  (it can be audible only with good earphones). If this bothers you
+	  set is_powered_on_standby, you will have slightly higher power
+	  consumption. Please note that sending the L3 command for ADC is
+	  enough to make the bump, so it doesn't make difference if you
+	  completely take off power from the codec.
+	*/
+	int is_powered_on_standby;
 #define UDA134X_UDA1340 1
 #define UDA134X_UDA1341 2
 #define UDA134X_UDA1344 3