diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 221b810..4e4d0bc 100644
--- a/Documentation/sound/alsa/ALSA-Configuration.txt
+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -875,8 +875,7 @@
     		setup before initializing the codecs.  This option is
 		available only when CONFIG_SND_HDA_PATCH_LOADER=y is set.
 		See HD-Audio.txt for details.
-    beep_mode	- Selects the beep registration mode (0=off, 1=on, 2=
-		dynamic registration via mute switch on/off); the default
+    beep_mode	- Selects the beep registration mode (0=off, 1=on); default
 		value is set via CONFIG_SND_HDA_INPUT_BEEP_MODE kconfig.
     
     [Single (global) options]
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 03f7897..7456360 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -15,19 +15,24 @@
 
 ALC262
 ======
-  N/A
+  inv-dmic	Inverted internal mic workaround
 
 ALC267/268
 ==========
-  N/A
+  inv-dmic	Inverted internal mic workaround
 
-ALC269
+ALC269/270/275/276/280/282
 ======
   laptop-amic	Laptops with analog-mic input
   laptop-dmic	Laptops with digital-mic input
+  alc269-dmic	Enable ALC269(VA) digital mic workaround
+  alc271-dmic	Enable ALC271X digital mic workaround
+  inv-dmic	Inverted internal mic workaround
+  lenovo-dock   Enables docking station I/O for some Lenovos
 
 ALC662/663/272
 ==============
+  mario		Chromebook mario model fixup
   asus-mode1	ASUS
   asus-mode2	ASUS
   asus-mode3	ASUS
@@ -36,6 +41,7 @@
   asus-mode6	ASUS
   asus-mode7	ASUS
   asus-mode8	ASUS
+  inv-dmic	Inverted internal mic workaround
 
 ALC680
 ======
@@ -46,6 +52,7 @@
   acer-aspire-4930g	Acer Aspire 4930G/5930G/6530G/6930G/7730G
   acer-aspire-8930g	Acer Aspire 8330G/6935G
   acer-aspire		Acer Aspire others
+  inv-dmic	Inverted internal mic workaround
 
 ALC861/660
 ==========
diff --git a/MAINTAINERS b/MAINTAINERS
index 0ed7048..8ae601c 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -6765,9 +6765,11 @@
 
 TI LM49xxx FAMILY ASoC CODEC DRIVERS
 M:	M R Swami Reddy <mr.swami.reddy@ti.com>
+M:	Vishwas A Deshpande <vishwas.a.deshpande@ti.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	sound/soc/codecs/lm49453*
+F:	sound/soc/codecs/isabelle*
 
 TI TWL4030 SERIES SOC CODEC DRIVER
 M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c
index 84461fa..a310222 100644
--- a/arch/arm/mach-ux500/board-mop500.c
+++ b/arch/arm/mach-ux500/board-mop500.c
@@ -25,6 +25,7 @@
 #include <linux/mfd/tc3589x.h>
 #include <linux/mfd/tps6105x.h>
 #include <linux/mfd/abx500/ab8500-gpio.h>
+#include <linux/mfd/abx500/ab8500-codec.h>
 #include <linux/leds-lp5521.h>
 #include <linux/input.h>
 #include <linux/smsc911x.h>
@@ -97,6 +98,18 @@
 					0x7A, 0x00, 0x00},
 };
 
+/* ab8500-codec */
+static struct ab8500_codec_platform_data ab8500_codec_pdata = {
+	.amics =  {
+		.mic1_type = AMIC_TYPE_DIFFERENTIAL,
+		.mic2_type = AMIC_TYPE_DIFFERENTIAL,
+		.mic1a_micbias = AMIC_MICBIAS_VAMIC1,
+		.mic1b_micbias = AMIC_MICBIAS_VAMIC1,
+		.mic2_micbias = AMIC_MICBIAS_VAMIC2
+	},
+	.ear_cmv = EAR_CMV_0_95V
+};
+
 static struct gpio_keys_button snowball_key_array[] = {
 	{
 		.gpio           = 32,
@@ -195,6 +208,7 @@
 	.regulator	= ab8500_regulators,
 	.num_regulator	= ARRAY_SIZE(ab8500_regulators),
 	.gpio		= &ab8500_gpio_pdata,
+	.codec		= &ab8500_codec_pdata,
 };
 
 static struct resource ab8500_resources[] = {
diff --git a/include/linux/ac97_codec.h b/include/linux/ac97_codec.h
deleted file mode 100644
index 0260c3e..0000000
--- a/include/linux/ac97_codec.h
+++ /dev/null
@@ -1,362 +0,0 @@
-#ifndef _AC97_CODEC_H_
-#define _AC97_CODEC_H_
-
-#include <linux/types.h>
-#include <linux/soundcard.h>
-
-/* AC97 1.0 */
-#define  AC97_RESET               0x0000      //
-#define  AC97_MASTER_VOL_STEREO   0x0002      // Line Out
-#define  AC97_HEADPHONE_VOL       0x0004      // 
-#define  AC97_MASTER_VOL_MONO     0x0006      // TAD Output
-#define  AC97_MASTER_TONE         0x0008      //
-#define  AC97_PCBEEP_VOL          0x000a      // none
-#define  AC97_PHONE_VOL           0x000c      // TAD Input (mono)
-#define  AC97_MIC_VOL             0x000e      // MIC Input (mono)
-#define  AC97_LINEIN_VOL          0x0010      // Line Input (stereo)
-#define  AC97_CD_VOL              0x0012      // CD Input (stereo)
-#define  AC97_VIDEO_VOL           0x0014      // none
-#define  AC97_AUX_VOL             0x0016      // Aux Input (stereo)
-#define  AC97_PCMOUT_VOL          0x0018      // Wave Output (stereo)
-#define  AC97_RECORD_SELECT       0x001a      //
-#define  AC97_RECORD_GAIN         0x001c
-#define  AC97_RECORD_GAIN_MIC     0x001e
-#define  AC97_GENERAL_PURPOSE     0x0020
-#define  AC97_3D_CONTROL          0x0022
-#define  AC97_MODEM_RATE          0x0024
-#define  AC97_POWER_CONTROL       0x0026
-
-/* AC'97 2.0 */
-#define AC97_EXTENDED_ID          0x0028       /* Extended Audio ID */
-#define AC97_EXTENDED_STATUS      0x002A       /* Extended Audio Status */
-#define AC97_PCM_FRONT_DAC_RATE   0x002C       /* PCM Front DAC Rate */
-#define AC97_PCM_SURR_DAC_RATE    0x002E       /* PCM Surround DAC Rate */
-#define AC97_PCM_LFE_DAC_RATE     0x0030       /* PCM LFE DAC Rate */
-#define AC97_PCM_LR_ADC_RATE      0x0032       /* PCM LR ADC Rate */
-#define AC97_PCM_MIC_ADC_RATE     0x0034       /* PCM MIC ADC Rate */
-#define AC97_CENTER_LFE_MASTER    0x0036       /* Center + LFE Master Volume */
-#define AC97_SURROUND_MASTER      0x0038       /* Surround (Rear) Master Volume */
-#define AC97_RESERVED_3A          0x003A       /* Reserved in AC '97 < 2.2 */
-
-/* AC'97 2.2 */
-#define AC97_SPDIF_CONTROL        0x003A       /* S/PDIF Control */
-
-/* range 0x3c-0x58 - MODEM */
-#define AC97_EXTENDED_MODEM_ID    0x003C
-#define AC97_EXTEND_MODEM_STAT    0x003E
-#define AC97_LINE1_RATE           0x0040
-#define AC97_LINE2_RATE           0x0042
-#define AC97_HANDSET_RATE         0x0044
-#define AC97_LINE1_LEVEL          0x0046
-#define AC97_LINE2_LEVEL          0x0048
-#define AC97_HANDSET_LEVEL        0x004A
-#define AC97_GPIO_CONFIG          0x004C
-#define AC97_GPIO_POLARITY        0x004E
-#define AC97_GPIO_STICKY          0x0050
-#define AC97_GPIO_WAKE_UP         0x0052
-#define AC97_GPIO_STATUS          0x0054
-#define AC97_MISC_MODEM_STAT      0x0056
-#define AC97_RESERVED_58          0x0058
-
-/* registers 0x005a - 0x007a are vendor reserved */
-
-#define AC97_VENDOR_ID1           0x007c
-#define AC97_VENDOR_ID2           0x007e
-
-/* volume control bit defines */
-#define AC97_MUTE                 0x8000
-#define AC97_MICBOOST             0x0040
-#define AC97_LEFTVOL              0x3f00
-#define AC97_RIGHTVOL             0x003f
-
-/* record mux defines */
-#define AC97_RECMUX_MIC           0x0000
-#define AC97_RECMUX_CD            0x0101
-#define AC97_RECMUX_VIDEO         0x0202
-#define AC97_RECMUX_AUX           0x0303
-#define AC97_RECMUX_LINE          0x0404
-#define AC97_RECMUX_STEREO_MIX    0x0505
-#define AC97_RECMUX_MONO_MIX      0x0606
-#define AC97_RECMUX_PHONE         0x0707
-
-/* general purpose register bit defines */
-#define AC97_GP_LPBK              0x0080       /* Loopback mode */
-#define AC97_GP_MS                0x0100       /* Mic Select 0=Mic1, 1=Mic2 */
-#define AC97_GP_MIX               0x0200       /* Mono output select 0=Mix, 1=Mic */
-#define AC97_GP_RLBK              0x0400       /* Remote Loopback - Modem line codec */
-#define AC97_GP_LLBK              0x0800       /* Local Loopback - Modem Line codec */
-#define AC97_GP_LD                0x1000       /* Loudness 1=on */
-#define AC97_GP_3D                0x2000       /* 3D Enhancement 1=on */
-#define AC97_GP_ST                0x4000       /* Stereo Enhancement 1=on */
-#define AC97_GP_POP               0x8000       /* Pcm Out Path, 0=pre 3D, 1=post 3D */
-
-/* extended audio status and control bit defines */
-#define AC97_EA_VRA               0x0001       /* Variable bit rate enable bit */
-#define AC97_EA_DRA               0x0002       /* Double-rate audio enable bit */
-#define AC97_EA_SPDIF             0x0004       /* S/PDIF Enable bit */
-#define AC97_EA_VRM               0x0008       /* Variable bit rate for MIC enable bit */
-#define AC97_EA_CDAC              0x0040       /* PCM Center DAC is ready (Read only) */
-#define AC97_EA_SDAC              0x0040       /* PCM Surround DACs are ready (Read only) */
-#define AC97_EA_LDAC              0x0080       /* PCM LFE DAC is ready (Read only) */
-#define AC97_EA_MDAC              0x0100       /* MIC ADC is ready (Read only) */
-#define AC97_EA_SPCV              0x0400       /* S/PDIF configuration valid (Read only) */
-#define AC97_EA_PRI               0x0800       /* Turns the PCM Center DAC off */
-#define AC97_EA_PRJ               0x1000       /* Turns the PCM Surround DACs off */
-#define AC97_EA_PRK               0x2000       /* Turns the PCM LFE DAC off */
-#define AC97_EA_PRL               0x4000       /* Turns the MIC ADC off */
-#define AC97_EA_SLOT_MASK         0xffcf       /* Mask for slot assignment bits */
-#define AC97_EA_SPSA_3_4          0x0000       /* Slot assigned to 3 & 4 */
-#define AC97_EA_SPSA_7_8          0x0010       /* Slot assigned to 7 & 8 */
-#define AC97_EA_SPSA_6_9          0x0020       /* Slot assigned to 6 & 9 */
-#define AC97_EA_SPSA_10_11        0x0030       /* Slot assigned to 10 & 11 */
-
-/* S/PDIF control bit defines */
-#define AC97_SC_PRO               0x0001       /* Professional status */
-#define AC97_SC_NAUDIO            0x0002       /* Non audio stream */
-#define AC97_SC_COPY              0x0004       /* Copyright status */
-#define AC97_SC_PRE               0x0008       /* Preemphasis status */
-#define AC97_SC_CC_MASK           0x07f0       /* Category Code mask */
-#define AC97_SC_L                 0x0800       /* Generation Level status */
-#define AC97_SC_SPSR_MASK         0xcfff       /* S/PDIF Sample Rate bits */
-#define AC97_SC_SPSR_44K          0x0000       /* Use 44.1kHz Sample rate */
-#define AC97_SC_SPSR_48K          0x2000       /* Use 48kHz Sample rate */
-#define AC97_SC_SPSR_32K          0x3000       /* Use 32kHz Sample rate */
-#define AC97_SC_DRS               0x4000       /* Double Rate S/PDIF */
-#define AC97_SC_V                 0x8000       /* Validity status */
-
-/* powerdown control and status bit defines */
-
-/* status */
-#define AC97_PWR_MDM              0x0010       /* Modem section ready */
-#define AC97_PWR_REF              0x0008       /* Vref nominal */
-#define AC97_PWR_ANL              0x0004       /* Analog section ready */
-#define AC97_PWR_DAC              0x0002       /* DAC section ready */
-#define AC97_PWR_ADC              0x0001       /* ADC section ready */
-
-/* control */
-#define AC97_PWR_PR0              0x0100       /* ADC and Mux powerdown */
-#define AC97_PWR_PR1              0x0200       /* DAC powerdown */
-#define AC97_PWR_PR2              0x0400       /* Output mixer powerdown (Vref on) */
-#define AC97_PWR_PR3              0x0800       /* Output mixer powerdown (Vref off) */
-#define AC97_PWR_PR4              0x1000       /* AC-link powerdown */
-#define AC97_PWR_PR5              0x2000       /* Internal Clk disable */
-#define AC97_PWR_PR6              0x4000       /* HP amp powerdown */
-#define AC97_PWR_PR7              0x8000       /* Modem off - if supported */
-
-/* extended audio ID register bit defines */
-#define AC97_EXTID_VRA            0x0001
-#define AC97_EXTID_DRA            0x0002
-#define AC97_EXTID_SPDIF          0x0004
-#define AC97_EXTID_VRM            0x0008
-#define AC97_EXTID_DSA0           0x0010
-#define AC97_EXTID_DSA1           0x0020
-#define AC97_EXTID_CDAC           0x0040
-#define AC97_EXTID_SDAC           0x0080
-#define AC97_EXTID_LDAC           0x0100
-#define AC97_EXTID_AMAP           0x0200
-#define AC97_EXTID_REV0           0x0400
-#define AC97_EXTID_REV1           0x0800
-#define AC97_EXTID_ID0            0x4000
-#define AC97_EXTID_ID1            0x8000
-
-/* extended status register bit defines */
-#define AC97_EXTSTAT_VRA          0x0001
-#define AC97_EXTSTAT_DRA          0x0002
-#define AC97_EXTSTAT_SPDIF        0x0004
-#define AC97_EXTSTAT_VRM          0x0008
-#define AC97_EXTSTAT_SPSA0        0x0010
-#define AC97_EXTSTAT_SPSA1        0x0020
-#define AC97_EXTSTAT_CDAC         0x0040
-#define AC97_EXTSTAT_SDAC         0x0080
-#define AC97_EXTSTAT_LDAC         0x0100
-#define AC97_EXTSTAT_MADC         0x0200
-#define AC97_EXTSTAT_SPCV         0x0400
-#define AC97_EXTSTAT_PRI          0x0800
-#define AC97_EXTSTAT_PRJ          0x1000
-#define AC97_EXTSTAT_PRK          0x2000
-#define AC97_EXTSTAT_PRL          0x4000
-
-/* extended audio ID register bit defines */
-#define AC97_EXTID_VRA            0x0001
-#define AC97_EXTID_DRA            0x0002
-#define AC97_EXTID_SPDIF          0x0004
-#define AC97_EXTID_VRM            0x0008
-#define AC97_EXTID_DSA0           0x0010
-#define AC97_EXTID_DSA1           0x0020
-#define AC97_EXTID_CDAC           0x0040
-#define AC97_EXTID_SDAC           0x0080
-#define AC97_EXTID_LDAC           0x0100
-#define AC97_EXTID_AMAP           0x0200
-#define AC97_EXTID_REV0           0x0400
-#define AC97_EXTID_REV1           0x0800
-#define AC97_EXTID_ID0            0x4000
-#define AC97_EXTID_ID1            0x8000
-
-/* extended status register bit defines */
-#define AC97_EXTSTAT_VRA          0x0001
-#define AC97_EXTSTAT_DRA          0x0002
-#define AC97_EXTSTAT_SPDIF        0x0004
-#define AC97_EXTSTAT_VRM          0x0008
-#define AC97_EXTSTAT_SPSA0        0x0010
-#define AC97_EXTSTAT_SPSA1        0x0020
-#define AC97_EXTSTAT_CDAC         0x0040
-#define AC97_EXTSTAT_SDAC         0x0080
-#define AC97_EXTSTAT_LDAC         0x0100
-#define AC97_EXTSTAT_MADC         0x0200
-#define AC97_EXTSTAT_SPCV         0x0400
-#define AC97_EXTSTAT_PRI          0x0800
-#define AC97_EXTSTAT_PRJ          0x1000
-#define AC97_EXTSTAT_PRK          0x2000
-#define AC97_EXTSTAT_PRL          0x4000
-
-/* useful power states */
-#define AC97_PWR_D0               0x0000      /* everything on */
-#define AC97_PWR_D1              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR4
-#define AC97_PWR_D2              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_D3              AC97_PWR_PR0|AC97_PWR_PR1|AC97_PWR_PR2|AC97_PWR_PR3|AC97_PWR_PR4
-#define AC97_PWR_ANLOFF          AC97_PWR_PR2|AC97_PWR_PR3  /* analog section off */
-
-/* Total number of defined registers.  */
-#define AC97_REG_CNT 64
-
-
-/* OSS interface to the ac97s.. */
-#define AC97_STEREO_MASK (SOUND_MASK_VOLUME|SOUND_MASK_PCM|\
-	SOUND_MASK_LINE|SOUND_MASK_CD|\
-	SOUND_MASK_ALTPCM|SOUND_MASK_IGAIN|\
-	SOUND_MASK_LINE1|SOUND_MASK_VIDEO)
-
-#define AC97_SUPPORTED_MASK (AC97_STEREO_MASK | \
-	SOUND_MASK_BASS|SOUND_MASK_TREBLE|\
-	SOUND_MASK_SPEAKER|SOUND_MASK_MIC|\
-	SOUND_MASK_PHONEIN|SOUND_MASK_PHONEOUT)
-
-#define AC97_RECORD_MASK (SOUND_MASK_MIC|\
-	SOUND_MASK_CD|SOUND_MASK_IGAIN|SOUND_MASK_VIDEO|\
-	SOUND_MASK_LINE1| SOUND_MASK_LINE|\
-	SOUND_MASK_PHONEIN)
-
-/* original check is not good enough in case FOO is greater than
- * SOUND_MIXER_NRDEVICES because the supported_mixers has exactly
- * SOUND_MIXER_NRDEVICES elements.
- * before matching the given mixer against the bitmask in supported_mixers we
- * check if mixer number exceeds maximum allowed size which is as mentioned
- * above SOUND_MIXER_NRDEVICES */
-#define supported_mixer(CODEC,FOO) ((FOO >= 0) && \
-                                    (FOO < SOUND_MIXER_NRDEVICES) && \
-                                    (CODEC)->supported_mixers & (1<<FOO) )
-
-struct ac97_codec {
-	/* Linked list of codecs */
-	struct list_head list;
-
-	/* AC97 controller connected with */
-	void *private_data;
-
-	char *name;
-	int id;
-	int dev_mixer; 
-	int type;
-	u32 model;
-
-	unsigned int modem:1;
-
-	struct ac97_ops *codec_ops;
-
-	/* controller specific lower leverl ac97 accessing routines.
-	   must be re-entrant safe */
-	u16  (*codec_read)  (struct ac97_codec *codec, u8 reg);
-	void (*codec_write) (struct ac97_codec *codec, u8 reg, u16 val);
-
-	/* Wait for codec-ready.  Ok to sleep here.  */
-	void  (*codec_wait)  (struct ac97_codec *codec);
-
-	/* callback used by helper drivers for interesting ac97 setups */
-	void  (*codec_unregister) (struct ac97_codec *codec);
-	
-	struct ac97_driver *driver;
-	void *driver_private;	/* Private data for the driver */
-	
-	spinlock_t lock;
-	
-	/* OSS mixer masks */
-	int modcnt;
-	int supported_mixers;
-	int stereo_mixers;
-	int record_sources;
-
-	/* Property flags */
-	int flags;
-
-	int bit_resolution;
-
-	/* OSS mixer interface */
-	int  (*read_mixer) (struct ac97_codec *codec, int oss_channel);
-	void (*write_mixer)(struct ac97_codec *codec, int oss_channel,
-			    unsigned int left, unsigned int right);
-	int  (*recmask_io) (struct ac97_codec *codec, int rw, int mask);
-	int  (*mixer_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
-
-	/* saved OSS mixer states */
-	unsigned int mixer_state[SOUND_MIXER_NRDEVICES];
-
-	/* Software Modem interface */
-	int  (*modem_ioctl)(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);
-};
-
-/*
- *	Operation structures for each known AC97 chip
- */
- 
-struct ac97_ops
-{
-	/* Initialise */
-	int (*init)(struct ac97_codec *c);
-	/* Amplifier control */
-	int (*amplifier)(struct ac97_codec *codec, int on);
-	/* Digital mode control */
-	int (*digital)(struct ac97_codec *codec, int slots, int rate, int mode);
-#define AUDIO_DIGITAL		0x8000
-#define AUDIO_PRO		0x4000
-#define AUDIO_DRS		0x2000
-#define AUDIO_CCMASK		0x003F
-	
-#define AC97_DELUDED_MODEM	1	/* Audio codec reports its a modem */
-#define AC97_NO_PCM_VOLUME	2	/* Volume control is missing 	   */
-#define AC97_DEFAULT_POWER_OFF 4 /* Needs warm reset to power up */
-};
-
-extern int ac97_probe_codec(struct ac97_codec *);
-
-extern struct ac97_codec *ac97_alloc_codec(void);
-extern void ac97_release_codec(struct ac97_codec *codec);
-
-struct ac97_driver {
-	struct list_head list;
-	char *name;
-	u32 codec_id;
-	u32 codec_mask;
-	int (*probe) (struct ac97_codec *codec, struct ac97_driver *driver);
-	void (*remove) (struct ac97_codec *codec, struct ac97_driver *driver);
-};
-
-/* quirk types */
-enum {
-	AC97_TUNE_DEFAULT = -1, /* use default from quirk list (not valid in list) */
-	AC97_TUNE_NONE = 0,     /* nothing extra to do */
-	AC97_TUNE_HP_ONLY,      /* headphone (true line-out) control as master only */
-	AC97_TUNE_SWAP_HP,      /* swap headphone and master controls */
-	AC97_TUNE_SWAP_SURROUND, /* swap master and surround controls */
-	AC97_TUNE_AD_SHARING,   /* for AD1985, turn on OMS bit and use headphone */
-	AC97_TUNE_ALC_JACK,     /* for Realtek, enable JACK detection */
-};
-
-struct ac97_quirk {
-	unsigned short vendor;  /* PCI vendor id */
-	unsigned short device;  /* PCI device id */
-	unsigned short mask;    /* device id bit mask, 0 = accept all */
-	const char *name;       /* name shown as info */
-	int type;               /* quirk type above */
-};
-
-#endif /* _AC97_CODEC_H_ */
diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h
index 56377df..cc0756a 100644
--- a/include/linux/dmaengine.h
+++ b/include/linux/dmaengine.h
@@ -670,6 +670,12 @@
 	return dmaengine_device_control(chan, DMA_RESUME, 0);
 }
 
+static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan,
+	dma_cookie_t cookie, struct dma_tx_state *state)
+{
+	return chan->device->device_tx_status(chan, cookie, state);
+}
+
 static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc)
 {
 	return desc->tx_submit(desc);
diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h
new file mode 100644
index 0000000..dc65292
--- /dev/null
+++ b/include/linux/mfd/abx500/ab8500-codec.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef AB8500_CORE_CODEC_H
+#define AB8500_CORE_CODEC_H
+
+/* Mic-types */
+enum amic_type {
+	AMIC_TYPE_SINGLE_ENDED,
+	AMIC_TYPE_DIFFERENTIAL
+};
+
+/* Mic-biases */
+enum amic_micbias {
+	AMIC_MICBIAS_VAMIC1,
+	AMIC_MICBIAS_VAMIC2
+};
+
+/* Bias-voltage */
+enum ear_cm_voltage {
+	EAR_CMV_0_95V,
+	EAR_CMV_1_10V,
+	EAR_CMV_1_27V,
+	EAR_CMV_1_58V
+};
+
+/* Analog microphone settings */
+struct amic_settings {
+	enum amic_type mic1_type;
+	enum amic_type mic2_type;
+	enum amic_micbias mic1a_micbias;
+	enum amic_micbias mic1b_micbias;
+	enum amic_micbias mic2_micbias;
+};
+
+/* Platform data structure for the audio-parts of the AB8500 */
+struct ab8500_codec_platform_data {
+	struct amic_settings amics;
+	enum ear_cm_voltage ear_cmv;
+};
+
+#endif
diff --git a/include/linux/mfd/abx500/ab8500.h b/include/linux/mfd/abx500/ab8500.h
index 91dd3ef..bc9b84b 100644
--- a/include/linux/mfd/abx500/ab8500.h
+++ b/include/linux/mfd/abx500/ab8500.h
@@ -266,6 +266,7 @@
 struct regulator_reg_init;
 struct regulator_init_data;
 struct ab8500_gpio_platform_data;
+struct ab8500_codec_platform_data;
 
 /**
  * struct ab8500_platform_data - AB8500 platform data
@@ -284,6 +285,7 @@
 	int num_regulator;
 	struct regulator_init_data *regulator;
 	struct ab8500_gpio_platform_data *gpio;
+	struct ab8500_codec_platform_data *codec;
 };
 
 extern int __devinit ab8500_init(struct ab8500 *ab8500,
diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h
new file mode 100644
index 0000000..0f25d16
--- /dev/null
+++ b/include/linux/platform_data/mmp_audio.h
@@ -0,0 +1,22 @@
+/*
+ *  MMP Platform AUDIO Management
+ *
+ *  Copyright (c) 2011 Marvell Semiconductors Inc.
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ */
+
+#ifndef MMP_AUDIO_H
+#define MMP_AUDIO_H
+
+struct mmp_audio_platdata {
+	u32 period_max_capture;
+	u32 buffer_max_capture;
+	u32 period_max_playback;
+	u32 buffer_max_playback;
+};
+
+#endif /* MMP_AUDIO_H */
diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h
new file mode 100644
index 0000000..26f406e
--- /dev/null
+++ b/include/sound/designware_i2s.h
@@ -0,0 +1,69 @@
+/*
+ * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#ifndef __SOUND_DESIGNWARE_I2S_H
+#define __SOUND_DESIGNWARE_I2S_H
+
+#include <linux/dmaengine.h>
+#include <linux/types.h>
+
+/*
+ * struct i2s_clk_config_data - represent i2s clk configuration data
+ * @chan_nr: number of channel
+ * @data_width: number of bits per sample (8/16/24/32 bit)
+ * @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz)
+ */
+struct i2s_clk_config_data {
+	int chan_nr;
+	u32 data_width;
+	u32 sample_rate;
+};
+
+struct i2s_platform_data {
+	#define DWC_I2S_PLAY	(1 << 0)
+	#define DWC_I2S_RECORD	(1 << 1)
+	unsigned int cap;
+	int channel;
+	u32 snd_fmts;
+	u32 snd_rates;
+
+	void *play_dma_data;
+	void *capture_dma_data;
+	bool (*filter)(struct dma_chan *chan, void *slave);
+	int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+};
+
+struct i2s_dma_data {
+	void *data;
+	dma_addr_t addr;
+	u32 max_burst;
+	enum dma_slave_buswidth addr_width;
+	bool (*filter)(struct dma_chan *chan, void *slave);
+};
+
+/* I2S DMA registers */
+#define I2S_RXDMA		0x01C0
+#define I2S_TXDMA		0x01C8
+
+#define TWO_CHANNEL_SUPPORT	2	/* up to 2.0 */
+#define FOUR_CHANNEL_SUPPORT	4	/* up to 3.1 */
+#define SIX_CHANNEL_SUPPORT	6	/* up to 5.1 */
+#define EIGHT_CHANNEL_SUPPORT	8	/* up to 7.1 */
+
+#endif /*  __SOUND_DESIGNWARE_I2S_H */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index a8fcaa6..b877334 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -39,6 +39,7 @@
 	const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config);
 int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd);
 snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream);
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream);
 
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
 	dma_filter_fn filter_fn, void *filter_data);
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 0d11128..c75c0d1 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -810,7 +810,7 @@
 int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, 
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var,
-			       struct snd_pcm_hw_constraint_list *l);
+			       const struct snd_pcm_hw_constraint_list *l);
 int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, 
 				  unsigned int cond,
 				  snd_pcm_hw_param_t var,
@@ -893,6 +893,7 @@
 
 int snd_pcm_limit_hw_rates(struct snd_pcm_runtime *runtime);
 unsigned int snd_pcm_rate_to_rate_bit(unsigned int rate);
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit);
 
 static inline void snd_pcm_set_runtime_buffer(struct snd_pcm_substream *substream,
 					      struct snd_dma_buffer *bufp)
@@ -1073,4 +1074,15 @@
 
 const char *snd_pcm_format_name(snd_pcm_format_t format);
 
+/**
+ * Get a string naming the direction of a stream
+ */
+static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
+{
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		return "Playback";
+	else
+		return "Capture";
+}
+
 #endif /* __SOUND_PCM_H */
diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h
index f494f1e..37ae12e 100644
--- a/include/sound/pcm_params.h
+++ b/include/sound/pcm_params.h
@@ -22,6 +22,8 @@
  *
  */
 
+#include <sound/pcm.h>
+
 int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, 
 			   struct snd_pcm_hw_params *params,
 			   snd_pcm_hw_param_t var, int *dir);
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index e3833d9..abe373d 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -229,6 +229,10 @@
 {	.id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \
 	.shift = wshift, .invert = winvert, \
 	.event = wevent, .event_flags = wflags}
+#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \
+{	.id = snd_soc_dapm_clock_supply, .name = wname, \
+	.reg = SND_SOC_NOPM, .event = dapm_clock_event, \
+	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
 
 /* generic widgets */
 #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \
@@ -245,6 +249,7 @@
 	.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
 	.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
 
+
 /* dapm kcontrol types */
 #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -327,6 +332,8 @@
 		   struct snd_kcontrol *kcontrol, int event);
 int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
+int dapm_clock_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol, int event);
 
 /* dapm controls */
 int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol,
@@ -367,6 +374,8 @@
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
+int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
+			    const struct snd_soc_dapm_route *route, int num);
 int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
 			     const struct snd_soc_dapm_route *route, int num);
 
@@ -432,6 +441,7 @@
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
 	snd_soc_dapm_supply,		/* power/clock supply */
 	snd_soc_dapm_regulator_supply,	/* external regulator */
+	snd_soc_dapm_clock_supply,	/* external clock */
 	snd_soc_dapm_aif_in,		/* audio interface input */
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
@@ -537,6 +547,8 @@
 	struct list_head dirty;
 	int inputs;
 	int outputs;
+
+	struct clk *clk;
 };
 
 struct snd_soc_dapm_update {
diff --git a/include/sound/soc.h b/include/sound/soc.h
index c703871..e063380 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -42,11 +42,22 @@
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
 	.max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \
+	((unsigned long)&(struct soc_mixer_control) \
+	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+	.min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert})
 #define SOC_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
 	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+#define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \
+	.put = snd_soc_put_volsw_range, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = xshift, .min = xmin,\
+		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -67,6 +78,16 @@
 		{.reg = xreg, .rreg = xreg, \
 		.shift = xshift, .rshift = xshift, \
 		.max = xmax, .min = xmin} }
+#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, xinvert, 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_range, \
+	.get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = xshift, .min = xmin,\
+		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
@@ -79,6 +100,13 @@
 	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
 					    xmax, xinvert) }
+#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \
+			   xmax, xinvert)		\
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
+	.info = snd_soc_info_volsw_range, \
+	.get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+	.private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \
+					    xshift, xmin, xmax, xinvert) }
 #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -97,6 +125,16 @@
 	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
 					    xmax, xinvert) }
+#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \
+			       xmax, xinvert, 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_range, \
+	.get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \
+	.private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \
+					    xshift, xmin, xmax, xinvert) }
 #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \
 {       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -460,6 +498,12 @@
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo);
+int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_get_volsw_range(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_bytes_info(struct snd_kcontrol *kcontrol,
@@ -785,13 +829,36 @@
 	/* config - must be set by machine driver */
 	const char *name;			/* Codec name */
 	const char *stream_name;		/* Stream name */
-	const char *codec_name;		/* for multi-codec */
-	const struct device_node *codec_of_node;
-	const char *platform_name;	/* for multi-platform */
-	const struct device_node *platform_of_node;
+	/*
+	 * You MAY specify the link's CPU-side device, either by device name,
+	 * or by DT/OF node, but not both. If this information is omitted,
+	 * the CPU-side DAI is matched using .cpu_dai_name only, which hence
+	 * must be globally unique. These fields are currently typically used
+	 * only for codec to codec links, or systems using device tree.
+	 */
+	const char *cpu_name;
+	const struct device_node *cpu_of_node;
+	/*
+	 * You MAY specify the DAI name of the CPU DAI. If this information is
+	 * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node
+	 * only, which only works well when that device exposes a single DAI.
+	 */
 	const char *cpu_dai_name;
-	const struct device_node *cpu_dai_of_node;
+	/*
+	 * You MUST specify the link's codec, either by device name, or by
+	 * DT/OF node, but not both.
+	 */
+	const char *codec_name;
+	const struct device_node *codec_of_node;
+	/* You MUST specify the DAI name within the codec */
 	const char *codec_dai_name;
+	/*
+	 * You MAY specify the link's platform/PCM/DMA driver, either by
+	 * device name, or by DT/OF node, but not both. Some forms of link
+	 * do not need a platform.
+	 */
+	const char *platform_name;
+	const struct device_node *platform_of_node;
 	int be_id;	/* optional ID for machine driver BE identification */
 
 	const struct snd_soc_pcm_stream *params;
diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h
new file mode 100644
index 0000000..1b365bf
--- /dev/null
+++ b/include/sound/spear_dma.h
@@ -0,0 +1,35 @@
+/*
+* linux/spear_dma.h
+*
+* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com)
+*
+* This program is free software; you can redistribute it and/or modify
+* it under the terms of the GNU General Public License as published by
+* the Free Software Foundation; either version 2 of the License, or
+* (at your option) any later version.
+*
+* This program is distributed in the hope that it will be useful,
+* but WITHOUT ANY WARRANTY; without even the implied warranty of
+* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+* GNU General Public License for more details.
+*
+* You should have received a copy of the GNU General Public License
+* along with this program; if not, write to the Free Software
+* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+*
+*/
+
+#ifndef SPEAR_DMA_H
+#define SPEAR_DMA_H
+
+#include <linux/dmaengine.h>
+
+struct spear_dma_data {
+	void *data;
+	dma_addr_t addr;
+	u32 max_burst;
+	enum dma_slave_buswidth addr_width;
+	bool (*filter)(struct dma_chan *chan, void *slave);
+};
+
+#endif /* SPEAR_DMA_H */
diff --git a/include/sound/spear_spdif.h b/include/sound/spear_spdif.h
new file mode 100644
index 0000000..a12f396
--- /dev/null
+++ b/include/sound/spear_spdif.h
@@ -0,0 +1,29 @@
+/*
+ * Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef __SOUND_SPDIF_H
+#define __SOUND_SPDIF_H
+
+struct spear_spdif_platform_data {
+	/* DMA params */
+	void *dma_params;
+	bool (*filter)(struct dma_chan *chan, void *slave);
+	void (*reset_perip)(void);
+};
+
+#endif /* SOUND_SPDIF_H */
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index 7067e2d..a64d8fe 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -38,21 +38,31 @@
 #define SNDRV_CTL_TLVT_DB_MINMAX 4	/* dB scale with min/max */
 #define SNDRV_CTL_TLVT_DB_MINMAX_MUTE 5	/* dB scale with min/max with mute */
 
+#define TLV_ITEM(type, ...) \
+	(type), TLV_LENGTH(__VA_ARGS__), __VA_ARGS__
+#define TLV_LENGTH(...) \
+	((unsigned int)sizeof((const unsigned int[]) { __VA_ARGS__ }))
+
+#define TLV_CONTAINER_ITEM(...) \
+	TLV_ITEM(SNDRV_CTL_TLVT_CONTAINER, __VA_ARGS__)
+#define DECLARE_TLV_CONTAINER(name, ...) \
+	unsigned int name[] = { TLV_CONTAINER_ITEM(__VA_ARGS__) }
+
 #define TLV_DB_SCALE_MASK	0xffff
 #define TLV_DB_SCALE_MUTE	0x10000
 #define TLV_DB_SCALE_ITEM(min, step, mute)			\
-	SNDRV_CTL_TLVT_DB_SCALE, 2 * sizeof(unsigned int),	\
-	(min), ((step) & TLV_DB_SCALE_MASK) | ((mute) ? TLV_DB_SCALE_MUTE : 0)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_SCALE,			\
+		 (min),					\
+		 ((step) & TLV_DB_SCALE_MASK) |		\
+			((mute) ? TLV_DB_SCALE_MUTE : 0))
 #define DECLARE_TLV_DB_SCALE(name, min, step, mute) \
 	unsigned int name[] = { TLV_DB_SCALE_ITEM(min, step, mute) }
 
 /* dB scale specified with min/max values instead of step */
 #define TLV_DB_MINMAX_ITEM(min_dB, max_dB)			\
-	SNDRV_CTL_TLVT_DB_MINMAX, 2 * sizeof(unsigned int),	\
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX, (min_dB), (max_dB))
 #define TLV_DB_MINMAX_MUTE_ITEM(min_dB, max_dB)			\
-	SNDRV_CTL_TLVT_DB_MINMAX_MUTE, 2 * sizeof(unsigned int),	\
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_MINMAX_MUTE, (min_dB), (max_dB))
 #define DECLARE_TLV_DB_MINMAX(name, min_dB, max_dB) \
 	unsigned int name[] = { TLV_DB_MINMAX_ITEM(min_dB, max_dB) }
 #define DECLARE_TLV_DB_MINMAX_MUTE(name, min_dB, max_dB) \
@@ -60,13 +70,16 @@
 
 /* linear volume between min_dB and max_dB (.01dB unit) */
 #define TLV_DB_LINEAR_ITEM(min_dB, max_dB)		    \
-	SNDRV_CTL_TLVT_DB_LINEAR, 2 * sizeof(unsigned int), \
-	(min_dB), (max_dB)
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_LINEAR, (min_dB), (max_dB))
 #define DECLARE_TLV_DB_LINEAR(name, min_dB, max_dB)	\
 	unsigned int name[] = { TLV_DB_LINEAR_ITEM(min_dB, max_dB) }
 
 /* dB range container */
 /* Each item is: <min> <max> <TLV> */
+#define TLV_DB_RANGE_ITEM(...) \
+	TLV_ITEM(SNDRV_CTL_TLVT_DB_RANGE, __VA_ARGS__)
+#define DECLARE_TLV_DB_RANGE(name, ...) \
+	unsigned int name[] = { TLV_DB_RANGE_ITEM(__VA_ARGS__) }
 /* The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR */
 #define TLV_DB_RANGE_HEAD(num)			\
 	SNDRV_CTL_TLVT_DB_RANGE, 6 * (num) * sizeof(unsigned int)
diff --git a/include/sound/vx_core.h b/include/sound/vx_core.h
index 5456343..4f67c76 100644
--- a/include/sound/vx_core.h
+++ b/include/sound/vx_core.h
@@ -341,7 +341,7 @@
 /*
  * PM
  */
-int snd_vx_suspend(struct vx_core *card, pm_message_t state);
+int snd_vx_suspend(struct vx_core *card);
 int snd_vx_resume(struct vx_core *card);
 
 /*
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index afef72c..0d7b25e 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -108,7 +108,7 @@
 
 #ifdef CONFIG_PM
 
-static int pxa2xx_ac97_do_suspend(struct snd_card *card, pm_message_t state)
+static int pxa2xx_ac97_do_suspend(struct snd_card *card)
 {
 	pxa2xx_audio_ops_t *platform_ops = card->dev->platform_data;
 
@@ -144,7 +144,7 @@
 	int ret = 0;
 
 	if (card)
-		ret = pxa2xx_ac97_do_suspend(card, PMSG_SUSPEND);
+		ret = pxa2xx_ac97_do_suspend(card);
 
 	return ret;
 }
@@ -160,10 +160,7 @@
 	return ret;
 }
 
-static const struct dev_pm_ops pxa2xx_ac97_pm_ops = {
-	.suspend	= pxa2xx_ac97_suspend,
-	.resume		= pxa2xx_ac97_resume,
-};
+static SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, pxa2xx_ac97_suspend, pxa2xx_ac97_resume);
 #endif
 
 static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
index f7c2bb0..eb4ceb7 100644
--- a/sound/atmel/abdac.c
+++ b/sound/atmel/abdac.c
@@ -535,9 +535,9 @@
 }
 
 #ifdef CONFIG_PM
-static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg)
+static int atmel_abdac_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_abdac *dac = card->private_data;
 
 	dw_dma_cyclic_stop(dac->dma.chan);
@@ -547,9 +547,9 @@
 	return 0;
 }
 
-static int atmel_abdac_resume(struct platform_device *pdev)
+static int atmel_abdac_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_abdac *dac = card->private_data;
 
 	clk_enable(dac->pclk);
@@ -559,9 +559,11 @@
 
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(atmel_abdac_pm, atmel_abdac_suspend, atmel_abdac_resume);
+#define ATMEL_ABDAC_PM_OPS	&atmel_abdac_pm
 #else
-#define atmel_abdac_suspend NULL
-#define atmel_abdac_resume NULL
+#define ATMEL_ABDAC_PM_OPS	NULL
 #endif
 
 static int __devexit atmel_abdac_remove(struct platform_device *pdev)
@@ -589,9 +591,9 @@
 	.remove		= __devexit_p(atmel_abdac_remove),
 	.driver		= {
 		.name	= "atmel_abdac",
+		.owner	= THIS_MODULE,
+		.pm	= ATMEL_ABDAC_PM_OPS,
 	},
-	.suspend	= atmel_abdac_suspend,
-	.resume		= atmel_abdac_resume,
 };
 
 static int __init atmel_abdac_init(void)
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index f5ded64..bf47025 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -1135,9 +1135,9 @@
 }
 
 #ifdef CONFIG_PM
-static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg)
+static int atmel_ac97c_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_ac97c *chip = card->private_data;
 
 	if (cpu_is_at32ap7000()) {
@@ -1151,9 +1151,9 @@
 	return 0;
 }
 
-static int atmel_ac97c_resume(struct platform_device *pdev)
+static int atmel_ac97c_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct atmel_ac97c *chip = card->private_data;
 
 	clk_enable(chip->pclk);
@@ -1165,9 +1165,11 @@
 	}
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(atmel_ac97c_pm, atmel_ac97c_suspend, atmel_ac97c_resume);
+#define ATMEL_AC97C_PM_OPS	&atmel_ac97c_pm
 #else
-#define atmel_ac97c_suspend NULL
-#define atmel_ac97c_resume NULL
+#define ATMEL_AC97C_PM_OPS	NULL
 #endif
 
 static int __devexit atmel_ac97c_remove(struct platform_device *pdev)
@@ -1210,9 +1212,9 @@
 	.remove		= __devexit_p(atmel_ac97c_remove),
 	.driver		= {
 		.name	= "atmel_ac97c",
+		.owner	= THIS_MODULE,
+		.pm	= ATMEL_AC97C_PM_OPS,
 	},
-	.suspend	= atmel_ac97c_suspend,
-	.resume		= atmel_ac97c_resume,
 };
 
 static int __init atmel_ac97c_init(void)
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 8f312fa..7ae6719 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -1250,10 +1250,10 @@
 int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime,
 			       unsigned int cond,
 			       snd_pcm_hw_param_t var,
-			       struct snd_pcm_hw_constraint_list *l)
+			       const struct snd_pcm_hw_constraint_list *l)
 {
 	return snd_pcm_hw_rule_add(runtime, cond, var,
-				   snd_pcm_hw_rule_list, l,
+				   snd_pcm_hw_rule_list, (void *)l,
 				   var, -1);
 }
 
diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c
index 9c9eff9..d4fc1bf 100644
--- a/sound/core/pcm_misc.c
+++ b/sound/core/pcm_misc.c
@@ -488,3 +488,21 @@
 	return SNDRV_PCM_RATE_KNOT;
 }
 EXPORT_SYMBOL(snd_pcm_rate_to_rate_bit);
+
+/**
+ * snd_pcm_rate_bit_to_rate - converts SNDRV_PCM_RATE_xxx bit to sample rate
+ * @rate_bit: the rate bit to convert
+ *
+ * Returns the sample rate that corresponds to the given SNDRV_PCM_RATE_xxx flag
+ * or 0 for an unknown rate bit
+ */
+unsigned int snd_pcm_rate_bit_to_rate(unsigned int rate_bit)
+{
+	unsigned int i;
+
+	for (i = 0; i < snd_pcm_known_rates.count; i++)
+		if ((1u << i) == rate_bit)
+			return snd_pcm_known_rates.list[i];
+	return 0;
+}
+EXPORT_SYMBOL(snd_pcm_rate_bit_to_rate);
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 8b5c36f..1128b35 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -1177,10 +1177,9 @@
 }
 
 #ifdef CONFIG_PM
-static int loopback_suspend(struct platform_device *pdev,
-				pm_message_t state)
+static int loopback_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct loopback *loopback = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1190,13 +1189,18 @@
 	return 0;
 }
 	
-static int loopback_resume(struct platform_device *pdev)
+static int loopback_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(loopback_pm, loopback_suspend, loopback_resume);
+#define LOOPBACK_PM_OPS	&loopback_pm
+#else
+#define LOOPBACK_PM_OPS	NULL
 #endif
 
 #define SND_LOOPBACK_DRIVER	"snd_aloop"
@@ -1204,12 +1208,10 @@
 static struct platform_driver loopback_driver = {
 	.probe		= loopback_probe,
 	.remove		= __devexit_p(loopback_remove),
-#ifdef CONFIG_PM
-	.suspend	= loopback_suspend,
-	.resume		= loopback_resume,
-#endif
 	.driver		= {
-		.name	= SND_LOOPBACK_DRIVER
+		.name	= SND_LOOPBACK_DRIVER,
+		.owner	= THIS_MODULE,
+		.pm	= LOOPBACK_PM_OPS,
 	},
 };
 
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index ad9434f..f7d3bfc 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1065,9 +1065,9 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_dummy_suspend(struct platform_device *pdev, pm_message_t state)
+static int snd_dummy_suspend(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 	struct snd_dummy *dummy = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1075,13 +1075,18 @@
 	return 0;
 }
 	
-static int snd_dummy_resume(struct platform_device *pdev)
+static int snd_dummy_resume(struct device *pdev)
 {
-	struct snd_card *card = platform_get_drvdata(pdev);
+	struct snd_card *card = dev_get_drvdata(pdev);
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_dummy_pm, snd_dummy_suspend, snd_dummy_resume);
+#define SND_DUMMY_PM_OPS	&snd_dummy_pm
+#else
+#define SND_DUMMY_PM_OPS	NULL
 #endif
 
 #define SND_DUMMY_DRIVER	"snd_dummy"
@@ -1089,12 +1094,10 @@
 static struct platform_driver snd_dummy_driver = {
 	.probe		= snd_dummy_probe,
 	.remove		= __devexit_p(snd_dummy_remove),
-#ifdef CONFIG_PM
-	.suspend	= snd_dummy_suspend,
-	.resume		= snd_dummy_resume,
-#endif
 	.driver		= {
-		.name	= SND_DUMMY_DRIVER
+		.name	= SND_DUMMY_DRIVER,
+		.owner	= THIS_MODULE,
+		.pm	= SND_DUMMY_PM_OPS,
 	},
 };
 
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 86f5fbc..bc03a20 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -139,7 +139,8 @@
 	.probe		= snd_mpu401_probe,
 	.remove		= __devexit_p(snd_mpu401_remove),
 	.driver		= {
-		.name	= SND_MPU401_DRIVER
+		.name	= SND_MPU401_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 7693079..cad73af 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -759,7 +759,8 @@
 	.probe		= snd_mtpav_probe,
 	.remove		= __devexit_p(snd_mtpav_remove),
 	.driver		= {
-		.name	= SND_MTPAV_DRIVER
+		.name	= SND_MTPAV_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index 621e60e..2d5514b 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -1040,7 +1040,8 @@
 	.probe  = snd_mts64_probe,
 	.remove = __devexit_p(snd_mts64_remove),
 	.driver = {
-		.name = PLATFORM_DRIVER
+		.name = PLATFORM_DRIVER,
+		.owner	= THIS_MODULE,
 	}
 };
 
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index 99704e6..6ca59fc 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -200,15 +200,18 @@
 }
 
 #ifdef CONFIG_PM
-static int pcsp_suspend(struct platform_device *dev, pm_message_t state)
+static int pcsp_suspend(struct device *dev)
 {
-	struct snd_pcsp *chip = platform_get_drvdata(dev);
+	struct snd_pcsp *chip = dev_get_drvdata(dev);
 	pcsp_stop_beep(chip);
 	snd_pcm_suspend_all(chip->pcm);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(pcsp_pm, pcsp_suspend, NULL);
+#define PCSP_PM_OPS	&pcsp_pm
 #else
-#define pcsp_suspend NULL
+#define PCSP_PM_OPS	NULL
 #endif	/* CONFIG_PM */
 
 static void pcsp_shutdown(struct platform_device *dev)
@@ -221,10 +224,10 @@
 	.driver		= {
 		.name	= "pcspkr",
 		.owner	= THIS_MODULE,
+		.pm	= PCSP_PM_OPS,
 	},
 	.probe		= pcsp_probe,
 	.remove		= __devexit_p(pcsp_remove),
-	.suspend	= pcsp_suspend,
 	.shutdown	= pcsp_shutdown,
 };
 
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 3e32bd3..8364855 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -829,7 +829,8 @@
 	.probe  = snd_portman_probe,
 	.remove = __devexit_p(snd_portman_remove),
 	.driver = {
-		.name = PLATFORM_DRIVER
+		.name = PLATFORM_DRIVER,
+		.owner	= THIS_MODULE,
 	}
 };
 
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index b2d0e8e..8670067 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -995,7 +995,8 @@
 	.probe		= snd_serial_probe,
 	.remove		= __devexit_p( snd_serial_remove),
 	.driver		= {
-		.name	= SND_SERIAL_DRIVER
+		.name	= SND_SERIAL_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
index 9d97478..d7d514d 100644
--- a/sound/drivers/virmidi.c
+++ b/sound/drivers/virmidi.c
@@ -142,7 +142,8 @@
 	.probe		= snd_virmidi_probe,
 	.remove		= __devexit_p(snd_virmidi_remove),
 	.driver		= {
-		.name	= SND_VIRMIDI_DRIVER
+		.name	= SND_VIRMIDI_DRIVER,
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index b8e5159..de5055a 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -725,7 +725,7 @@
 /*
  * suspend
  */
-int snd_vx_suspend(struct vx_core *chip, pm_message_t state)
+int snd_vx_suspend(struct vx_core *chip)
 {
 	unsigned int i;
 
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d7ccf28..f8fbe22 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -135,10 +135,9 @@
 	unsigned long mc_base_size;
 #ifdef OPTi93X
 	unsigned long mc_indir_index;
-	unsigned long mc_indir_size;
 	struct resource *res_mc_indir;
-	struct snd_wss *codec;
 #endif	/* OPTi93X */
+	struct snd_wss *codec;
 	unsigned long pwd_reg;
 
 	spinlock_t lock;
@@ -245,10 +244,8 @@
 	case OPTi9XX_HW_82C931:
 	case OPTi9XX_HW_82C933:
 		chip->mc_base = (hardware == OPTi9XX_HW_82C930) ? 0xf8f : 0xf8d;
-		if (!chip->mc_indir_index) {
+		if (!chip->mc_indir_index)
 			chip->mc_indir_index = 0xe0e;
-			chip->mc_indir_size = 2;
-		}
 		chip->password = 0xe4;
 		chip->pwd_reg = 0;
 		break;
@@ -351,7 +348,7 @@
 		(snd_opti9xx_read(chip, reg) & ~(mask)) | ((value) & (mask)))
 
 
-static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip,
+static int snd_opti9xx_configure(struct snd_opti9xx *chip,
 					   long port,
 					   int irq, int dma1, int dma2,
 					   long mpu_port, int mpu_irq)
@@ -403,7 +400,9 @@
 
 #else	/* OPTi93X */
 	case OPTi9XX_HW_82C931:
-	case OPTi9XX_HW_82C933:
+		/* disable 3D sound (set GPIO1 as output, low) */
+		snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(20), 0x04, 0x0c);
+	case OPTi9XX_HW_82C933: /* FALL THROUGH */
 		/*
 		 * The BTC 1817DW has QS1000 wavetable which is connected
 		 * to the serial digital input of the OPTI931.
@@ -696,8 +695,7 @@
 		if (value == snd_opti9xx_read(chip, OPTi9XX_MC_REG(1)))
 			return 0;
 #else	/* OPTi93X */
-	chip->res_mc_indir = request_region(chip->mc_indir_index,
-					    chip->mc_indir_size,
+	chip->res_mc_indir = request_region(chip->mc_indir_index, 2,
 					    "OPTi93x MC");
 	if (chip->res_mc_indir == NULL)
 		return -EBUSY;
@@ -770,8 +768,9 @@
 #ifdef OPTi93X
 	port = pnp_port_start(pdev, 0) - 4;
 	fm_port = pnp_port_start(pdev, 1) + 8;
-	chip->mc_indir_index = pnp_port_start(pdev, 3) + 2;
-	chip->mc_indir_size = pnp_port_len(pdev, 3) - 2;
+	/* adjust mc_indir_index - some cards report it at 0xe?d,
+	   other at 0xe?c but it really is always at 0xe?e */
+	chip->mc_indir_index = (pnp_port_start(pdev, 3) & ~0xf) | 0xe;
 #else
 	devmc = pnp_request_card_device(card, pid->devs[2].id, NULL);
 	if (devmc == NULL)
@@ -871,9 +870,7 @@
 			       &codec);
 	if (error < 0)
 		return error;
-#ifdef OPTi93X
 	chip->codec = codec;
-#endif
 	error = snd_wss_pcm(codec, 0, &pcm);
 	if (error < 0)
 		return error;
@@ -1054,11 +1051,55 @@
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_suspend(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	chip->codec->suspend(chip->codec);
+	return 0;
+}
+
+static int snd_opti9xx_resume(struct snd_card *card)
+{
+	struct snd_opti9xx *chip = card->private_data;
+	int error, xdma2;
+#if defined(CS4231) || defined(OPTi93X)
+	xdma2 = dma2;
+#else
+	xdma2 = -1;
+#endif
+
+	error = snd_opti9xx_configure(chip, port, irq, dma1, xdma2,
+				      mpu_port, mpu_irq);
+	if (error)
+		return error;
+	chip->codec->resume(chip->codec);
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+	return 0;
+}
+
+static int snd_opti9xx_isa_suspend(struct device *dev, unsigned int n,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(dev_get_drvdata(dev));
+}
+
+static int snd_opti9xx_isa_resume(struct device *dev, unsigned int n)
+{
+	return snd_opti9xx_resume(dev_get_drvdata(dev));
+}
+#endif
+
 static struct isa_driver snd_opti9xx_driver = {
 	.match		= snd_opti9xx_isa_match,
 	.probe		= snd_opti9xx_isa_probe,
 	.remove		= __devexit_p(snd_opti9xx_isa_remove),
-	/* FIXME: suspend/resume */
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_isa_suspend,
+	.resume		= snd_opti9xx_isa_resume,
+#endif
 	.driver		= {
 		.name	= DEV_NAME
 	},
@@ -1124,12 +1165,29 @@
 	snd_opti9xx_pnp_is_probed = 0;
 }
 
+#ifdef CONFIG_PM
+static int snd_opti9xx_pnp_suspend(struct pnp_card_link *pcard,
+				   pm_message_t state)
+{
+	return snd_opti9xx_suspend(pnp_get_card_drvdata(pcard));
+}
+
+static int snd_opti9xx_pnp_resume(struct pnp_card_link *pcard)
+{
+	return snd_opti9xx_resume(pnp_get_card_drvdata(pcard));
+}
+#endif
+
 static struct pnp_card_driver opti9xx_pnpc_driver = {
 	.flags		= PNP_DRIVER_RES_DISABLE,
 	.name		= "opti9xx",
 	.id_table	= snd_opti9xx_pnpids,
 	.probe		= snd_opti9xx_pnp_probe,
 	.remove		= __devexit_p(snd_opti9xx_pnp_remove),
+#ifdef CONFIG_PM
+	.suspend	= snd_opti9xx_pnp_suspend,
+	.resume		= snd_opti9xx_pnp_resume,
+#endif
 };
 #endif
 
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 49c8a0c..360b08b 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -1456,7 +1456,6 @@
 {
 	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
 				 SNDRV_PCM_INFO_MMAP_VALID |
-				 SNDRV_PCM_INFO_RESUME |
 				 SNDRV_PCM_INFO_SYNC_START),
 	.formats =		(SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM |
 				 SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE),
@@ -1657,6 +1656,10 @@
 			break;
 		}
 	}
+	/* Yamaha needs this to resume properly */
+	if (chip->hardware == WSS_HW_OPL3SA2)
+		snd_wss_out(chip, CS4231_PLAYBK_FORMAT,
+			    chip->image[CS4231_PLAYBK_FORMAT]);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 #if 1
 	snd_wss_mce_down(chip);
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 09d4648..7d8803a 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -69,7 +69,6 @@
 #include <linux/sound.h>
 #include <linux/slab.h>
 #include <linux/soundcard.h>
-#include <linux/ac97_codec.h>
 #include <linux/pci.h>
 #include <linux/bitops.h>
 #include <linux/interrupt.h>
@@ -199,6 +198,22 @@
         }                                          \
 })
 
+/* AC97 registers */
+#define AC97_MASTER_VOL_STEREO   0x0002      /* Line Out		*/
+#define AC97_PCBEEP_VOL          0x000a      /* none			*/
+#define AC97_PHONE_VOL           0x000c      /* TAD Input (mono)	*/
+#define AC97_MIC_VOL             0x000e      /* MIC Input (mono)	*/
+#define AC97_LINEIN_VOL          0x0010      /* Line Input (stereo)	*/
+#define AC97_CD_VOL              0x0012      /* CD Input (stereo)	*/
+#define AC97_AUX_VOL             0x0016      /* Aux Input (stereo)	*/
+#define AC97_PCMOUT_VOL          0x0018      /* Wave Output (stereo)	*/
+#define AC97_RECORD_SELECT       0x001a      /*			*/
+#define AC97_RECORD_GAIN         0x001c
+#define AC97_GENERAL_PURPOSE     0x0020
+#define AC97_3D_CONTROL          0x0022
+#define AC97_POWER_CONTROL       0x0026
+#define AC97_VENDOR_ID1           0x007c
+
 struct list_head cs4297a_devs = { &cs4297a_devs, &cs4297a_devs };
 
 typedef struct serdma_descr_s {
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 9dfc27b..ee895f3 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -1884,9 +1884,10 @@
 }
 
 #ifdef CONFIG_PM
-static int ali_suspend(struct pci_dev *pci, pm_message_t state)
+static int ali_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ali *chip = card->private_data;
 	struct snd_ali_image *im;
 	int i, j;
@@ -1929,13 +1930,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int ali_resume(struct pci_dev *pci)
+static int ali_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ali *chip = card->private_data;
 	struct snd_ali_image *im;
 	int i, j;
@@ -1982,6 +1984,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(ali_pm, ali_suspend, ali_resume);
+#define ALI_PM_OPS	&ali_pm
+#else
+#define ALI_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_ali_free(struct snd_ali * codec)
@@ -2299,10 +2306,9 @@
 	.id_table = snd_ali_ids,
 	.probe = snd_ali_probe,
 	.remove = __devexit_p(snd_ali_remove),
-#ifdef CONFIG_PM
-	.suspend = ali_suspend,
-	.resume = ali_resume,
-#endif
+	.driver = {
+		.pm = ALI_PM_OPS,
+	},
 };                                
 
 module_pci_driver(ali5451_driver);
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 59d6538..68c4469 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -766,9 +766,10 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_als300_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_als300_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_als300 *chip = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -777,13 +778,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_als300_resume(struct pci_dev *pci)
+static int snd_als300_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_als300 *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -802,6 +804,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_als300_pm, snd_als300_suspend, snd_als300_resume);
+#define SND_ALS300_PM_OPS	&snd_als300_pm
+#else
+#define SND_ALS300_PM_OPS	NULL
 #endif
 
 static int __devinit snd_als300_probe(struct pci_dev *pci,
@@ -857,10 +864,9 @@
 	.id_table = snd_als300_ids,
 	.probe = snd_als300_probe,
 	.remove = __devexit_p(snd_als300_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_als300_suspend,
-	.resume = snd_als300_resume,
-#endif
+	.driver = {
+		.pm = SND_ALS300_PM_OPS,
+	},
 };
 
 module_pci_driver(als300_driver);
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index 7d7f259..0eeca49 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -988,9 +988,10 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_als4000_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_als4000_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_card_als4000 *acard = card->private_data;
 	struct snd_sb *chip = acard->chip;
 
@@ -1001,13 +1002,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_als4000_resume(struct pci_dev *pci)
+static int snd_als4000_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_card_als4000 *acard = card->private_data;
 	struct snd_sb *chip = acard->chip;
 
@@ -1033,18 +1035,21 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_als4000_pm, snd_als4000_suspend, snd_als4000_resume);
+#define SND_ALS4000_PM_OPS	&snd_als4000_pm
+#else
+#define SND_ALS4000_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver als4000_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_als4000_ids,
 	.probe = snd_card_als4000_probe,
 	.remove = __devexit_p(snd_card_als4000_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_als4000_suspend,
-	.resume = snd_als4000_resume,
-#endif
+	.driver = {
+		.pm = SND_ALS4000_PM_OPS,
+	},
 };
 
 module_pci_driver(als4000_driver);
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 156a94f..31020d2 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1462,9 +1462,10 @@
 /*
  * power management
  */
-static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_atiixp_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp *chip = card->private_data;
 	int i;
 
@@ -1484,13 +1485,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_atiixp_resume(struct pci_dev *pci)
+static int snd_atiixp_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp *chip = card->private_data;
 	int i;
 
@@ -1526,6 +1528,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume);
+#define SND_ATIIXP_PM_OPS	&snd_atiixp_pm
+#else
+#define SND_ATIIXP_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
@@ -1705,10 +1712,9 @@
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_atiixp_suspend,
-	.resume = snd_atiixp_resume,
-#endif
+	.driver = {
+		.pm = SND_ATIIXP_PM_OPS,
+	},
 };
 
 module_pci_driver(atiixp_driver);
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 30a4fd9..79e204e 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1117,9 +1117,10 @@
 /*
  * power management
  */
-static int snd_atiixp_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_atiixp_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp_modem *chip = card->private_data;
 	int i;
 
@@ -1133,13 +1134,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_atiixp_resume(struct pci_dev *pci)
+static int snd_atiixp_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct atiixp_modem *chip = card->private_data;
 	int i;
 
@@ -1162,8 +1164,12 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_atiixp_pm, snd_atiixp_suspend, snd_atiixp_resume);
+#define SND_ATIIXP_PM_OPS	&snd_atiixp_pm
+#else
+#define SND_ATIIXP_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 #ifdef CONFIG_PROC_FS
 /*
@@ -1336,10 +1342,9 @@
 	.id_table = snd_atiixp_ids,
 	.probe = snd_atiixp_probe,
 	.remove = __devexit_p(snd_atiixp_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_atiixp_suspend,
-	.resume = snd_atiixp_resume,
-#endif
+	.driver = {
+		.pm = SND_ATIIXP_PM_OPS,
+	},
 };
 
 module_pci_driver(atiixp_modem_driver);
diff --git a/sound/pci/au88x0/au88x0_mixer.c b/sound/pci/au88x0/au88x0_mixer.c
index 557c782..fa13efb 100644
--- a/sound/pci/au88x0/au88x0_mixer.c
+++ b/sound/pci/au88x0/au88x0_mixer.c
@@ -10,6 +10,15 @@
 #include <sound/core.h>
 #include "au88x0.h"
 
+static int remove_ctl(struct snd_card *card, const char *name)
+{
+	struct snd_ctl_elem_id id;
+	memset(&id, 0, sizeof(id));
+	strcpy(id.name, name);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	return snd_ctl_remove_id(card, &id);
+}
+
 static int __devinit snd_vortex_mixer(vortex_t * vortex)
 {
 	struct snd_ac97_bus *pbus;
@@ -28,5 +37,7 @@
 	ac97.scaps = AC97_SCAP_NO_SPDIF;
 	err = snd_ac97_mixer(pbus, &ac97, &vortex->codec);
 	vortex->isquad = ((vortex->codec == NULL) ?  0 : (vortex->codec->ext_id&0x80));
+	remove_ctl(vortex->card, "Master Mono Playback Volume");
+	remove_ctl(vortex->card, "Master Mono Playback Switch");
 	return err;
 }
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index f0b4d74..4dddd87 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -2794,9 +2794,10 @@
 }
 
 static int
-snd_azf3328_suspend(struct pci_dev *pci, pm_message_t state)
+snd_azf3328_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_azf3328 *chip = card->private_data;
 	u16 *saved_regs_ctrl_u16;
 
@@ -2824,14 +2825,15 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
 static int
-snd_azf3328_resume(struct pci_dev *pci)
+snd_azf3328_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	const struct snd_azf3328 *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2859,18 +2861,21 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_azf3328_pm, snd_azf3328_suspend, snd_azf3328_resume);
+#define SND_AZF3328_PM_OPS	&snd_azf3328_pm
+#else
+#define SND_AZF3328_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver azf3328_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_azf3328_ids,
 	.probe = snd_azf3328_probe,
 	.remove = __devexit_p(snd_azf3328_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_azf3328_suspend,
-	.resume = snd_azf3328_resume,
-#endif
+	.driver = {
+		.pm = SND_AZF3328_PM_OPS,
+	},
 };
 
 module_pci_driver(azf3328_driver);
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index e76d68a..83277b7 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -1872,9 +1872,10 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_ca0106_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ca0106_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ca0106 *chip = card->private_data;
 	int i;
 
@@ -1889,13 +1890,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_ca0106_resume(struct pci_dev *pci)
+static int snd_ca0106_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ca0106 *chip = card->private_data;
 	int i;
 
@@ -1922,6 +1924,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume);
+#define SND_CA0106_PM_OPS	&snd_ca0106_pm
+#else
+#define SND_CA0106_PM_OPS	NULL
 #endif
 
 // PCI IDs
@@ -1937,10 +1944,9 @@
 	.id_table = snd_ca0106_ids,
 	.probe = snd_ca0106_probe,
 	.remove = __devexit_p(snd_ca0106_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_ca0106_suspend,
-	.resume = snd_ca0106_resume,
-#endif
+	.driver = {
+		.pm = SND_CA0106_PM_OPS,
+	},
 };
 
 module_pci_driver(ca0106_driver);
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 3815bd4..b7d6f2b 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3338,9 +3338,10 @@
 	SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
 };
 
-static int snd_cmipci_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cmipci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cmipci *cm = card->private_data;
 	int i;
 
@@ -3361,13 +3362,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_cmipci_resume(struct pci_dev *pci)
+static int snd_cmipci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cmipci *cm = card->private_data;
 	int i;
 
@@ -3396,6 +3398,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_cmipci_pm, snd_cmipci_suspend, snd_cmipci_resume);
+#define SND_CMIPCI_PM_OPS	&snd_cmipci_pm
+#else
+#define SND_CMIPCI_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static struct pci_driver cmipci_driver = {
@@ -3403,10 +3410,9 @@
 	.id_table = snd_cmipci_ids,
 	.probe = snd_cmipci_probe,
 	.remove = __devexit_p(snd_cmipci_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_cmipci_suspend,
-	.resume = snd_cmipci_resume,
-#endif
+	.driver = {
+		.pm = SND_CMIPCI_PM_OPS,
+	},
 };
 	
 module_pci_driver(cmipci_driver);
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 33506ee..45a8317 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1997,9 +1997,10 @@
 
 #define CLKCR1_CKRA                             0x00010000L
 
-static int cs4281_suspend(struct pci_dev *pci, pm_message_t state)
+static int cs4281_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs4281 *chip = card->private_data;
 	u32 ulCLK;
 	unsigned int i;
@@ -2040,13 +2041,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int cs4281_resume(struct pci_dev *pci)
+static int cs4281_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs4281 *chip = card->private_data;
 	unsigned int i;
 	u32 ulCLK;
@@ -2082,6 +2084,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(cs4281_pm, cs4281_suspend, cs4281_resume);
+#define CS4281_PM_OPS	&cs4281_pm
+#else
+#define CS4281_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static struct pci_driver cs4281_driver = {
@@ -2089,10 +2096,9 @@
 	.id_table = snd_cs4281_ids,
 	.probe = snd_cs4281_probe,
 	.remove = __devexit_p(snd_cs4281_remove),
-#ifdef CONFIG_PM
-	.suspend = cs4281_suspend,
-	.resume = cs4281_resume,
-#endif
+	.driver = {
+		.pm = CS4281_PM_OPS,
+	},
 };
 	
 module_pci_driver(cs4281_driver);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index 6cc7404..1e007c7 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -30,7 +30,7 @@
 #include <linux/init.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 #include <sound/initval.h>
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
@@ -167,8 +167,9 @@
 	.probe = snd_card_cs46xx_probe,
 	.remove = __devexit_p(snd_card_cs46xx_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_cs46xx_suspend,
-	.resume = snd_cs46xx_resume,
+	.driver = {
+		.pm = &snd_cs46xx_pm,
+	},
 #endif
 };
 
diff --git a/include/sound/cs46xx.h b/sound/pci/cs46xx/cs46xx.h
similarity index 99%
rename from include/sound/cs46xx.h
rename to sound/pci/cs46xx/cs46xx.h
index e3005a6..29d8a8d 100644
--- a/include/sound/cs46xx.h
+++ b/sound/pci/cs46xx/cs46xx.h
@@ -23,10 +23,10 @@
  *
  */
 
-#include "pcm.h"
-#include "pcm-indirect.h"
-#include "rawmidi.h"
-#include "ac97_codec.h"
+#include <sound/pcm.h>
+#include <sound/pcm-indirect.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
 #include "cs46xx_dsp_spos.h"
 
 /*
@@ -1730,8 +1730,7 @@
 		      struct pci_dev *pci,
 		      int external_amp, int thinkpad,
 		      struct snd_cs46xx **rcodec);
-int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state);
-int snd_cs46xx_resume(struct pci_dev *pci);
+extern const struct dev_pm_ops snd_cs46xx_pm;
 
 int snd_cs46xx_pcm(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
 int snd_cs46xx_pcm_rear(struct snd_cs46xx *chip, int device, struct snd_pcm **rpcm);
diff --git a/include/sound/cs46xx_dsp_scb_types.h b/sound/pci/cs46xx/cs46xx_dsp_scb_types.h
similarity index 100%
rename from include/sound/cs46xx_dsp_scb_types.h
rename to sound/pci/cs46xx/cs46xx_dsp_scb_types.h
diff --git a/include/sound/cs46xx_dsp_spos.h b/sound/pci/cs46xx/cs46xx_dsp_spos.h
similarity index 100%
rename from include/sound/cs46xx_dsp_spos.h
rename to sound/pci/cs46xx/cs46xx_dsp_spos.h
diff --git a/include/sound/cs46xx_dsp_task_types.h b/sound/pci/cs46xx/cs46xx_dsp_task_types.h
similarity index 100%
rename from include/sound/cs46xx_dsp_task_types.h
rename to sound/pci/cs46xx/cs46xx_dsp_task_types.h
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 4fa5316..f75f5ff 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -61,7 +61,7 @@
 #include <sound/info.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include <asm/io.h>
 
@@ -3599,9 +3599,10 @@
 	BA1_CVOL,
 };
 
-int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cs46xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_cs46xx *chip = card->private_data;
 	int i, amp_saved;
 
@@ -3628,13 +3629,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_cs46xx_resume(struct pci_dev *pci)
+static int snd_cs46xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_cs46xx *chip = card->private_data;
 	int amp_saved;
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
@@ -3707,6 +3709,8 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_cs46xx_pm, snd_cs46xx_suspend, snd_cs46xx_resume);
 #endif /* CONFIG_PM */
 
 
diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c
index e377287..56fec0b 100644
--- a/sound/pci/cs46xx/dsp_spos.c
+++ b/sound/pci/cs46xx/dsp_spos.c
@@ -32,7 +32,7 @@
 #include <sound/control.h>
 #include <sound/info.h>
 #include <sound/asoundef.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include "cs46xx_lib.h"
 #include "dsp_spos.h"
diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c
index 00b148a..c2c695b 100644
--- a/sound/pci/cs46xx/dsp_spos_scb_lib.c
+++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c
@@ -31,7 +31,7 @@
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
-#include <sound/cs46xx.h>
+#include "cs46xx.h"
 
 #include "cs46xx_lib.h"
 #include "dsp_spos.h"
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 2c9697c..51f64ba 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -400,8 +400,9 @@
 	.probe = snd_cs5535audio_probe,
 	.remove = __devexit_p(snd_cs5535audio_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_cs5535audio_suspend,
-	.resume = snd_cs5535audio_resume,
+	.driver = {
+		.pm = &snd_cs5535audio_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/cs5535audio/cs5535audio.h b/sound/pci/cs5535audio/cs5535audio.h
index 51966d7..bb3cc64 100644
--- a/sound/pci/cs5535audio/cs5535audio.h
+++ b/sound/pci/cs5535audio/cs5535audio.h
@@ -94,10 +94,7 @@
 	struct cs5535audio_dma dmas[NUM_CS5535AUDIO_DMAS];
 };
 
-#ifdef CONFIG_PM
-int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state);
-int snd_cs5535audio_resume(struct pci_dev *pci);
-#endif
+extern const struct dev_pm_ops snd_cs5535audio_pm;
 
 #ifdef CONFIG_OLPC
 void __devinit olpc_prequirks(struct snd_card *card,
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index 185b000..6c34def 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -55,9 +55,10 @@
 
 }
 
-int snd_cs5535audio_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_cs5535audio_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs5535audio *cs5535au = card->private_data;
 	int i;
 
@@ -77,13 +78,14 @@
 		return -EIO;
 	}
 	pci_disable_device(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_cs5535audio_resume(struct pci_dev *pci)
+static int snd_cs5535audio_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct cs5535audio *cs5535au = card->private_data;
 	u32 tmp;
 	int timeout;
@@ -129,3 +131,4 @@
 	return 0;
 }
 
+SIMPLE_DEV_PM_OPS(snd_cs5535audio_pm, snd_cs5535audio_suspend, snd_cs5535audio_resume);
diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c
index d8a4423..8e40262 100644
--- a/sound/pci/ctxfi/ctatc.c
+++ b/sound/pci/ctxfi/ctatc.c
@@ -1537,7 +1537,7 @@
 }
 
 #ifdef CONFIG_PM
-static int atc_suspend(struct ct_atc *atc, pm_message_t state)
+static int atc_suspend(struct ct_atc *atc)
 {
 	int i;
 	struct hw *hw = atc->hw;
@@ -1553,7 +1553,7 @@
 
 	atc_release_resources(atc);
 
-	hw->suspend(hw, state);
+	hw->suspend(hw);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h
index 3a0def6..653e813 100644
--- a/sound/pci/ctxfi/ctatc.h
+++ b/sound/pci/ctxfi/ctatc.h
@@ -144,7 +144,7 @@
 	struct ct_timer *timer;
 
 #ifdef CONFIG_PM
-	int (*suspend)(struct ct_atc *atc, pm_message_t state);
+	int (*suspend)(struct ct_atc *atc);
 	int (*resume)(struct ct_atc *atc);
 #define NUM_PCMS (NUM_CTALSADEVS - 1)
 	struct snd_pcm *pcms[NUM_PCMS];
diff --git a/sound/pci/ctxfi/cthardware.h b/sound/pci/ctxfi/cthardware.h
index 908315b..c56fe53 100644
--- a/sound/pci/ctxfi/cthardware.h
+++ b/sound/pci/ctxfi/cthardware.h
@@ -73,7 +73,7 @@
 	int (*card_stop)(struct hw *hw);
 	int (*pll_init)(struct hw *hw, unsigned int rsr);
 #ifdef CONFIG_PM
-	int (*suspend)(struct hw *hw, pm_message_t state);
+	int (*suspend)(struct hw *hw);
 	int (*resume)(struct hw *hw, struct card_conf *info);
 #endif
 	int (*is_adc_source_selected)(struct hw *hw, enum ADCSRC source);
diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c
index a7df197..dc1969b 100644
--- a/sound/pci/ctxfi/cthw20k1.c
+++ b/sound/pci/ctxfi/cthw20k1.c
@@ -2086,7 +2086,7 @@
 }
 
 #ifdef CONFIG_PM
-static int hw_suspend(struct hw *hw, pm_message_t state)
+static int hw_suspend(struct hw *hw)
 {
 	struct pci_dev *pci = hw->pci;
 
@@ -2099,7 +2099,7 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c
index d6c54b5..9d1231d 100644
--- a/sound/pci/ctxfi/cthw20k2.c
+++ b/sound/pci/ctxfi/cthw20k2.c
@@ -2202,7 +2202,7 @@
 }
 
 #ifdef CONFIG_PM
-static int hw_suspend(struct hw *hw, pm_message_t state)
+static int hw_suspend(struct hw *hw)
 {
 	struct pci_dev *pci = hw->pci;
 
@@ -2210,7 +2210,7 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 
 	return 0;
 }
diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c
index 75aa2c3..e002183 100644
--- a/sound/pci/ctxfi/xfi.c
+++ b/sound/pci/ctxfi/xfi.c
@@ -126,21 +126,26 @@
 }
 
 #ifdef CONFIG_PM
-static int ct_card_suspend(struct pci_dev *pci, pm_message_t state)
+static int ct_card_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ct_atc *atc = card->private_data;
 
-	return atc->suspend(atc, state);
+	return atc->suspend(atc);
 }
 
-static int ct_card_resume(struct pci_dev *pci)
+static int ct_card_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ct_atc *atc = card->private_data;
 
 	return atc->resume(atc);
 }
+
+static SIMPLE_DEV_PM_OPS(ct_card_pm, ct_card_suspend, ct_card_resume);
+#define CT_CARD_PM_OPS	&ct_card_pm
+#else
+#define CT_CARD_PM_OPS	NULL
 #endif
 
 static struct pci_driver ct_driver = {
@@ -148,10 +153,9 @@
 	.id_table = ct_pci_dev_ids,
 	.probe = ct_card_probe,
 	.remove = __devexit_p(ct_card_remove),
-#ifdef CONFIG_PM
-	.suspend = ct_card_suspend,
-	.resume = ct_card_resume,
-#endif
+	.driver = {
+		.pm = CT_CARD_PM_OPS,
+	},
 };
 
 module_pci_driver(ct_driver);
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 0f8eda1..0ff754f 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -2205,9 +2205,10 @@
 
 #if defined(CONFIG_PM)
 
-static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_echo_suspend(struct device *dev)
 {
-	struct echoaudio *chip = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct echoaudio *chip = dev_get_drvdata(dev);
 
 	DE_INIT(("suspend start\n"));
 	snd_pcm_suspend_all(chip->analog_pcm);
@@ -2242,9 +2243,10 @@
 
 
 
-static int snd_echo_resume(struct pci_dev *pci)
+static int snd_echo_resume(struct device *dev)
 {
-	struct echoaudio *chip = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct echoaudio *chip = dev_get_drvdata(dev);
 	struct comm_page *commpage, *commpage_bak;
 	u32 pipe_alloc_mask;
 	int err;
@@ -2307,10 +2309,13 @@
 	return 0;
 }
 
+static SIMPLE_DEV_PM_OPS(snd_echo_pm, snd_echo_suspend, snd_echo_resume);
+#define SND_ECHO_PM_OPS	&snd_echo_pm
+#else
+#define SND_ECHO_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
-
 static void __devexit snd_echo_remove(struct pci_dev *pci)
 {
 	struct echoaudio *chip;
@@ -2333,10 +2338,9 @@
 	.id_table = snd_echo_ids,
 	.probe = snd_echo_probe,
 	.remove = __devexit_p(snd_echo_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_echo_suspend,
-	.resume = snd_echo_resume,
-#endif /* CONFIG_PM */
+	.driver = {
+		.pm = SND_ECHO_PM_OPS,
+	},
 };
 
 module_pci_driver(echo_driver);
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 7fdbbe4..ddac4e6 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -207,9 +207,10 @@
 
 
 #ifdef CONFIG_PM
-static int snd_emu10k1_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_emu10k1_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_emu10k1 *emu = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -231,13 +232,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_emu10k1_resume(struct pci_dev *pci)
+static int snd_emu10k1_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_emu10k1 *emu = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -261,17 +263,21 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_emu10k1_pm, snd_emu10k1_suspend, snd_emu10k1_resume);
+#define SND_EMU10K1_PM_OPS	&snd_emu10k1_pm
+#else
+#define SND_EMU10K1_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver emu10k1_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_emu10k1_ids,
 	.probe = snd_card_emu10k1_probe,
 	.remove = __devexit_p(snd_card_emu10k1_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_emu10k1_suspend,
-	.resume = snd_emu10k1_resume,
-#endif
+	.driver = {
+		.pm = SND_EMU10K1_PM_OPS,
+	},
 };
 
 module_pci_driver(emu10k1_driver);
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 3821c81..f7e6f73 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2033,9 +2033,10 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_ensoniq_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ensoniq_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ensoniq *ensoniq = card->private_data;
 	
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -2058,13 +2059,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_ensoniq_resume(struct pci_dev *pci)
+static int snd_ensoniq_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct ensoniq *ensoniq = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2087,8 +2089,12 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif /* CONFIG_PM */
 
+static SIMPLE_DEV_PM_OPS(snd_ensoniq_pm, snd_ensoniq_suspend, snd_ensoniq_resume);
+#define SND_ENSONIQ_PM_OPS	&snd_ensoniq_pm
+#else
+#define SND_ENSONIQ_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static int __devinit snd_ensoniq_create(struct snd_card *card,
 				     struct pci_dev *pci,
@@ -2493,10 +2499,9 @@
 	.id_table = snd_audiopci_ids,
 	.probe = snd_audiopci_probe,
 	.remove = __devexit_p(snd_audiopci_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_ensoniq_suspend,
-	.resume = snd_ensoniq_resume,
-#endif
+	.driver = {
+		.pm = SND_ENSONIQ_PM_OPS,
+	},
 };
 	
 module_pci_driver(ens137x_driver);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 82c8d8c..dbb8180 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1321,35 +1321,30 @@
 	return change;
 }
 
-static unsigned int db_scale_master[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_master,
 	0, 54, TLV_DB_SCALE_ITEM(-3600, 50, 1),
 	54, 63, TLV_DB_SCALE_ITEM(-900, 100, 0),
-};
+);
 
-static unsigned int db_scale_audio1[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio1,
 	0, 8, TLV_DB_SCALE_ITEM(-3300, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-900, 150, 0),
-};
+);
 
-static unsigned int db_scale_audio2[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_audio2,
 	0, 8, TLV_DB_SCALE_ITEM(-3450, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-1050, 150, 0),
-};
+);
 
-static unsigned int db_scale_mic[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_mic,
 	0, 8, TLV_DB_SCALE_ITEM(-2400, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(0, 150, 0),
-};
+);
 
-static unsigned int db_scale_line[] = {
-	TLV_DB_RANGE_HEAD(2),
+static const DECLARE_TLV_DB_RANGE(db_scale_line,
 	0, 8, TLV_DB_SCALE_ITEM(-3150, 300, 1),
 	8, 15, TLV_DB_SCALE_ITEM(-750, 150, 0),
-};
+);
 
 static const DECLARE_TLV_DB_SCALE(db_scale_capture, 0, 150, 0);
 
@@ -1474,9 +1469,10 @@
 };
 
 
-static int es1938_suspend(struct pci_dev *pci, pm_message_t state)
+static int es1938_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1938 *chip = card->private_data;
 	unsigned char *s, *d;
 
@@ -1494,13 +1490,14 @@
 	}
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int es1938_resume(struct pci_dev *pci)
+static int es1938_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1938 *chip = card->private_data;
 	unsigned char *s, *d;
 
@@ -1534,6 +1531,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(es1938_pm, es1938_suspend, es1938_resume);
+#define ES1938_PM_OPS	&es1938_pm
+#else
+#define ES1938_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef SUPPORT_JOYSTICK
@@ -1887,10 +1889,9 @@
 	.id_table = snd_es1938_ids,
 	.probe = snd_es1938_probe,
 	.remove = __devexit_p(snd_es1938_remove),
-#ifdef CONFIG_PM
-	.suspend = es1938_suspend,
-	.resume = es1938_resume,
-#endif
+	.driver = {
+		.pm = ES1938_PM_OPS,
+	},
 };
 
 module_pci_driver(es1938_driver);
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index 52b5c0b..fb4c90b9 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2381,9 +2381,10 @@
 /*
  * PM support
  */
-static int es1968_suspend(struct pci_dev *pci, pm_message_t state)
+static int es1968_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1968 *chip = card->private_data;
 
 	if (! chip->do_pm)
@@ -2398,13 +2399,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int es1968_resume(struct pci_dev *pci)
+static int es1968_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct es1968 *chip = card->private_data;
 	struct esschan *es;
 
@@ -2454,6 +2456,11 @@
 	chip->in_suspend = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(es1968_pm, es1968_suspend, es1968_resume);
+#define ES1968_PM_OPS	&es1968_pm
+#else
+#define ES1968_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef SUPPORT_JOYSTICK
@@ -2903,10 +2910,9 @@
 	.id_table = snd_es1968_ids,
 	.probe = snd_es1968_probe,
 	.remove = __devexit_p(snd_es1968_remove),
-#ifdef CONFIG_PM
-	.suspend = es1968_suspend,
-	.resume = es1968_resume,
-#endif
+	.driver = {
+		.pm = ES1968_PM_OPS,
+	},
 };
 
 module_pci_driver(es1968_driver);
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index b32e802..522c870 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1369,9 +1369,10 @@
 	FM801_CODEC_CTRL, FM801_I2S_MODE, FM801_VOLUME, FM801_GEN_CTRL,
 };
 
-static int snd_fm801_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_fm801_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct fm801 *chip = card->private_data;
 	int i;
 
@@ -1385,13 +1386,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_fm801_resume(struct pci_dev *pci)
+static int snd_fm801_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct fm801 *chip = card->private_data;
 	int i;
 
@@ -1414,17 +1416,21 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_fm801_pm, snd_fm801_suspend, snd_fm801_resume);
+#define SND_FM801_PM_OPS	&snd_fm801_pm
+#else
+#define SND_FM801_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver fm801_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_fm801_ids,
 	.probe = snd_card_fm801_probe,
 	.remove = __devexit_p(snd_card_fm801_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_fm801_suspend,
-	.resume = snd_fm801_resume,
-#endif
+	.driver = {
+		.pm = SND_FM801_PM_OPS,
+	},
 };
 
 module_pci_driver(fm801_driver);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index d030797..194d625 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -53,15 +53,14 @@
 	  driver. This interface is used to generate digital beeps.
 
 config SND_HDA_INPUT_BEEP_MODE
-	int "Digital beep registration mode (0=off, 1=on, 2=mute sw on/off)"
+	int "Digital beep registration mode (0=off, 1=on)"
 	depends on SND_HDA_INPUT_BEEP=y
 	default "1"
-	range 0 2
+	range 0 1
 	help
 	  Set 0 to disable the digital beep interface for HD-audio by default.
 	  Set 1 to always enable the digital beep interface for HD-audio by
-	  default. Set 2 to control the beep device registration to input
-	  layer using a "Beep Switch" in mixer applications.
+	  default.
 
 config SND_HDA_INPUT_JACK
 	bool "Support jack plugging notification via input layer"
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index f7520b9..647218d 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -727,7 +727,7 @@
 			models++;
 		}
 	}
-	if (id < 0) {
+	if (id < 0 && quirk) {
 		q = snd_pci_quirk_lookup(codec->bus->pci, quirk);
 		if (q) {
 			id = q->value;
@@ -736,7 +736,7 @@
 #endif
 		}
 	}
-	if (id < 0) {
+	if (id < 0 && quirk) {
 		for (q = quirk; q->subvendor; q++) {
 			unsigned int vendorid =
 				q->subdevice | (q->subvendor << 16);
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 60738e5..0bc2315 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -162,50 +162,20 @@
 	return 0;
 }
 
-static void snd_hda_do_register(struct work_struct *work)
-{
-	struct hda_beep *beep =
-		container_of(work, struct hda_beep, register_work);
-
-	mutex_lock(&beep->mutex);
-	if (beep->enabled && !beep->dev)
-		snd_hda_do_attach(beep);
-	mutex_unlock(&beep->mutex);
-}
-
-static void snd_hda_do_unregister(struct work_struct *work)
-{
-	struct hda_beep *beep =
-		container_of(work, struct hda_beep, unregister_work.work);
-
-	mutex_lock(&beep->mutex);
-	if (!beep->enabled && beep->dev)
-		snd_hda_do_detach(beep);
-	mutex_unlock(&beep->mutex);
-}
-
 int snd_hda_enable_beep_device(struct hda_codec *codec, int enable)
 {
 	struct hda_beep *beep = codec->beep;
-	enable = !!enable;
-	if (beep == NULL)
+	if (!beep)
 		return 0;
+	enable = !!enable;
 	if (beep->enabled != enable) {
 		beep->enabled = enable;
 		if (!enable) {
+			cancel_work_sync(&beep->beep_work);
 			/* turn off beep */
 			snd_hda_codec_write(beep->codec, beep->nid, 0,
 						  AC_VERB_SET_BEEP_CONTROL, 0);
 		}
-		if (beep->mode == HDA_BEEP_MODE_SWREG) {
-			if (enable) {
-				cancel_delayed_work(&beep->unregister_work);
-				schedule_work(&beep->register_work);
-			} else {
-				schedule_delayed_work(&beep->unregister_work,
-									   HZ);
-			}
-		}
 		return 1;
 	}
 	return 0;
@@ -215,6 +185,7 @@
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 {
 	struct hda_beep *beep;
+	int err;
 
 	if (!snd_hda_get_bool_hint(codec, "beep"))
 		return 0; /* disabled explicitly by hints */
@@ -232,21 +203,16 @@
 
 	beep->nid = nid;
 	beep->codec = codec;
-	beep->mode = codec->beep_mode;
 	codec->beep = beep;
 
-	INIT_WORK(&beep->register_work, &snd_hda_do_register);
-	INIT_DELAYED_WORK(&beep->unregister_work, &snd_hda_do_unregister);
 	INIT_WORK(&beep->beep_work, &snd_hda_generate_beep);
 	mutex_init(&beep->mutex);
 
-	if (beep->mode == HDA_BEEP_MODE_ON) {
-		int err = snd_hda_do_attach(beep);
-		if (err < 0) {
-			kfree(beep);
-			codec->beep = NULL;
-			return err;
-		}
+	err = snd_hda_do_attach(beep);
+	if (err < 0) {
+		kfree(beep);
+		codec->beep = NULL;
+		return err;
 	}
 
 	return 0;
@@ -257,8 +223,6 @@
 {
 	struct hda_beep *beep = codec->beep;
 	if (beep) {
-		cancel_work_sync(&beep->register_work);
-		cancel_delayed_work(&beep->unregister_work);
 		if (beep->dev)
 			snd_hda_do_detach(beep);
 		codec->beep = NULL;
@@ -266,3 +230,31 @@
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
+
+/* get/put callbacks for beep mute mixer switches */
+int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_beep *beep = codec->beep;
+	if (beep) {
+		ucontrol->value.integer.value[0] =
+			ucontrol->value.integer.value[1] =
+			beep->enabled;
+		return 0;
+	}
+	return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+}
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_get_beep);
+
+int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_beep *beep = codec->beep;
+	if (beep)
+		snd_hda_enable_beep_device(codec,
+					   *ucontrol->value.integer.value);
+	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+}
+EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index 55f0647..4dc6933 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -26,21 +26,16 @@
 
 #define HDA_BEEP_MODE_OFF	0
 #define HDA_BEEP_MODE_ON	1
-#define HDA_BEEP_MODE_SWREG	2
 
 /* beep information */
 struct hda_beep {
 	struct input_dev *dev;
 	struct hda_codec *codec;
-	unsigned int mode;
 	char phys[32];
 	int tone;
 	hda_nid_t nid;
 	unsigned int enabled:1;
-	unsigned int request_enable:1;
 	unsigned int linear_tone:1;	/* linear tone for IDT/STAC codec */
-	struct work_struct register_work; /* registration work */
-	struct delayed_work unregister_work; /* unregistration work */
 	struct work_struct beep_work; /* scheduled task for beep event */
 	struct mutex mutex;
 };
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 51cb2a2..88a9c20 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2676,25 +2676,6 @@
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put);
 
-#ifdef CONFIG_SND_HDA_INPUT_BEEP
-/**
- * snd_hda_mixer_amp_switch_put_beep - Put callback for a beep AMP switch
- *
- * This function calls snd_hda_enable_beep_device(), which behaves differently
- * depending on beep_mode option.
- */
-int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
-				      struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	long *valp = ucontrol->value.integer.value;
-
-	snd_hda_enable_beep_device(codec, *valp);
-	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-}
-EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep);
-#endif /* CONFIG_SND_HDA_INPUT_BEEP */
-
 /*
  * bound volume controls
  *
@@ -3509,22 +3490,52 @@
 EXPORT_SYMBOL_HDA(snd_hda_codec_set_power_to_all);
 
 /*
+ *  supported power states check
+ */
+static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state)
+{
+	int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE);
+
+	if (sup < 0)
+		return false;
+	if (sup & power_state)
+		return true;
+	else
+		return false;
+}
+
+/*
  * set power state of the codec
  */
 static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state)
 {
+	int count;
+	unsigned int state;
+
 	if (codec->patch_ops.set_power_state) {
 		codec->patch_ops.set_power_state(codec, fg, power_state);
 		return;
 	}
 
 	/* this delay seems necessary to avoid click noise at power-down */
-	if (power_state == AC_PWRST_D3)
-		msleep(100);
-	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
-			    power_state);
-	snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+	if (power_state == AC_PWRST_D3) {
+		/* transition time less than 10ms for power down */
+		bool epss = snd_hda_codec_get_supported_ps(codec, fg, AC_PWRST_EPSS);
+		msleep(epss ? 10 : 100);
+	}
+
+	/* repeat power states setting at most 10 times*/
+	for (count = 0; count < 10; count++) {
+		snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+				    power_state);
+		snd_hda_codec_set_power_to_all(codec, fg, power_state, true);
+		state = snd_hda_codec_read(codec, fg, 0,
+					   AC_VERB_GET_POWER_STATE, 0);
+		if (!(state & AC_PWRST_ERROR))
+			break;
+	}
 }
 
 #ifdef CONFIG_SND_HDA_HWDEP
@@ -3545,7 +3556,7 @@
 static void hda_call_codec_suspend(struct hda_codec *codec)
 {
 	if (codec->patch_ops.suspend)
-		codec->patch_ops.suspend(codec, PMSG_SUSPEND);
+		codec->patch_ops.suspend(codec);
 	hda_cleanup_all_streams(codec);
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
@@ -4418,6 +4429,13 @@
 	cancel_delayed_work_sync(&codec->power_work);
 
 	spin_lock(&codec->power_lock);
+	/* If the power down delayed work was cancelled above before starting,
+	 * then there is no need to go through power up here.
+	 */
+	if (codec->power_on) {
+		spin_unlock(&codec->power_lock);
+		return;
+	}
 	trace_hda_power_up(codec);
 	snd_hda_update_power_acct(codec);
 	codec->power_on = 1;
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2fdaadb..c422d33 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -323,6 +323,9 @@
 #define AC_PWRST_D1			0x01
 #define AC_PWRST_D2			0x02
 #define AC_PWRST_D3			0x03
+#define AC_PWRST_ERROR                  (1<<8)
+#define AC_PWRST_CLK_STOP_OK            (1<<9)
+#define AC_PWRST_SETTING_RESET          (1<<10)
 
 /* Processing capabilies */
 #define AC_PCAP_BENIGN			(1<<0)
@@ -703,7 +706,7 @@
 	void (*set_power_state)(struct hda_codec *codec, hda_nid_t fg,
 				unsigned int power_state);
 #ifdef CONFIG_PM
-	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*suspend)(struct hda_codec *codec);
 	int (*resume)(struct hda_codec *codec);
 #endif
 #ifdef CONFIG_SND_HDA_POWER_SAVE
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 7757536..c8aced1 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -72,7 +72,7 @@
 static char *patch[SNDRV_CARDS];
 #endif
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-static int beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
+static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] =
 					CONFIG_SND_HDA_INPUT_BEEP_MODE};
 #endif
 
@@ -103,9 +103,9 @@
 MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface.");
 #endif
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
-module_param_array(beep_mode, int, NULL, 0444);
+module_param_array(beep_mode, bool, NULL, 0444);
 MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode "
-			    "(0=off, 1=on, 2=mute switch on/off) (default=1).");
+			    "(0=off, 1=on) (default=1).");
 #endif
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -151,6 +151,7 @@
 			 "{Intel, CPT},"
 			 "{Intel, PPT},"
 			 "{Intel, LPT},"
+			 "{Intel, HPT},"
 			 "{Intel, PBG},"
 			 "{Intel, SCH},"
 			 "{ATI, SB450},"
@@ -535,6 +536,7 @@
 #define AZX_DCAPS_BUFSIZE	(1 << 21)	/* no buffer size alignment */
 #define AZX_DCAPS_ALIGN_BUFSIZE	(1 << 22)	/* buffer size alignment */
 #define AZX_DCAPS_4K_BDLE_BOUNDARY (1 << 23)	/* BDLE in 4k boundary */
+#define AZX_DCAPS_POSFIX_COMBO  (1 << 24)	/* Use COMBO as default */
 
 /* quirks for ATI SB / AMD Hudson */
 #define AZX_DCAPS_PRESET_ATI_SB \
@@ -2403,9 +2405,10 @@
  * power management
  */
 
-static int azx_suspend(struct pci_dev *pci, pm_message_t state)
+static int azx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 	struct azx_pcm *p;
 
@@ -2424,13 +2427,14 @@
 		pci_disable_msi(chip->pci);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int azx_resume(struct pci_dev *pci)
+static int azx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -2455,6 +2459,12 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume);
+#define AZX_PM_OPS	&azx_pm
+#else
+#define azx_suspend(dev)
+#define azx_resume(dev)
+#define AZX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 
@@ -2521,13 +2531,13 @@
 			   disabled ? "Disabling" : "Enabling",
 			   pci_name(chip->pci));
 		if (disabled) {
-			azx_suspend(pci, PMSG_FREEZE);
+			azx_suspend(&pci->dev);
 			chip->disabled = true;
 			snd_hda_lock_devices(chip->bus);
 		} else {
 			snd_hda_unlock_devices(chip->bus);
 			chip->disabled = false;
-			azx_resume(pci);
+			azx_resume(&pci->dev);
 		}
 	}
 }
@@ -2731,6 +2741,10 @@
 		snd_printd(SFX "Using LPIB position fix\n");
 		return POS_FIX_LPIB;
 	}
+	if (chip->driver_caps & AZX_DCAPS_POSFIX_COMBO) {
+		snd_printd(SFX "Using COMBO position fix\n");
+		return POS_FIX_COMBO;
+	}
 	return POS_FIX_AUTO;
 }
 
@@ -3243,7 +3257,7 @@
 	/* CPT */
 	{ PCI_DEVICE(0x8086, 0x1c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE },
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* PBG */
 	{ PCI_DEVICE(0x8086, 0x1d20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
@@ -3251,11 +3265,15 @@
 	/* Panther Point */
 	{ PCI_DEVICE(0x8086, 0x1e20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE},
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* Lynx Point */
 	{ PCI_DEVICE(0x8086, 0x8c20),
 	  .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP |
-	  AZX_DCAPS_BUFSIZE},
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
+	/* Haswell */
+	{ PCI_DEVICE(0x8086, 0x0c0c),
+	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
+	  AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO },
 	/* SCH */
 	{ PCI_DEVICE(0x8086, 0x811b),
 	  .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP |
@@ -3341,6 +3359,10 @@
 	/* VIA VT8251/VT8237A */
 	{ PCI_DEVICE(0x1106, 0x3288),
 	  .driver_data = AZX_DRIVER_VIA | AZX_DCAPS_POSFIX_VIA },
+	/* VIA GFX VT7122/VX900 */
+	{ PCI_DEVICE(0x1106, 0x9170), .driver_data = AZX_DRIVER_GENERIC },
+	/* VIA GFX VT6122/VX11 */
+	{ PCI_DEVICE(0x1106, 0x9140), .driver_data = AZX_DRIVER_GENERIC },
 	/* SIS966 */
 	{ PCI_DEVICE(0x1039, 0x7502), .driver_data = AZX_DRIVER_SIS },
 	/* ULI M5461 */
@@ -3398,10 +3420,9 @@
 	.id_table = azx_ids,
 	.probe = azx_probe,
 	.remove = __devexit_p(azx_remove),
-#ifdef CONFIG_PM
-	.suspend = azx_suspend,
-	.resume = azx_resume,
-#endif
+	.driver = {
+		.pm = AZX_PM_OPS,
+	},
 };
 
 module_pci_driver(azx_driver);
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 2dd1c11..aaccc02 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -127,10 +127,15 @@
 static void jack_detect_update(struct hda_codec *codec,
 			       struct hda_jack_tbl *jack)
 {
-	if (jack->jack_dirty || !jack->jack_detect) {
+	if (!jack->jack_dirty)
+		return;
+
+	if (jack->phantom_jack)
+		jack->pin_sense = AC_PINSENSE_PRESENCE;
+	else
 		jack->pin_sense = read_pin_sense(codec, jack->nid);
-		jack->jack_dirty = 0;
-	}
+
+	jack->jack_dirty = 0;
 }
 
 /**
@@ -264,8 +269,8 @@
  * This assigns a jack-detection kctl to the given pin.  The kcontrol
  * will have the given name and index.
  */
-int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
-			  const char *name, int idx)
+static int __snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
+			  const char *name, int idx, bool phantom_jack)
 {
 	struct hda_jack_tbl *jack;
 	struct snd_kcontrol *kctl;
@@ -283,47 +288,81 @@
 	if (err < 0)
 		return err;
 	jack->kctl = kctl;
+	jack->phantom_jack = !!phantom_jack;
+
 	state = snd_hda_jack_detect(codec, nid);
 	snd_kctl_jack_report(codec->bus->card, kctl, state);
 #ifdef CONFIG_SND_HDA_INPUT_JACK
-	jack->type = get_input_jack_type(codec, nid);
-	err = snd_jack_new(codec->bus->card, name, jack->type, &jack->jack);
-	if (err < 0)
-		return err;
-	jack->jack->private_data = jack;
-	jack->jack->private_free = hda_free_jack_priv;
-	snd_jack_report(jack->jack, state ? jack->type : 0);
+	if (!phantom_jack) {
+		jack->type = get_input_jack_type(codec, nid);
+		err = snd_jack_new(codec->bus->card, name, jack->type,
+				   &jack->jack);
+		if (err < 0)
+			return err;
+		jack->jack->private_data = jack;
+		jack->jack->private_free = hda_free_jack_priv;
+		snd_jack_report(jack->jack, state ? jack->type : 0);
+	}
 #endif
 	return 0;
 }
+
+int snd_hda_jack_add_kctl(struct hda_codec *codec, hda_nid_t nid,
+			  const char *name, int idx)
+{
+	return __snd_hda_jack_add_kctl(codec, nid, name, idx, false);
+}
 EXPORT_SYMBOL_HDA(snd_hda_jack_add_kctl);
 
+/* get the unique index number for the given kctl name */
+static int get_unique_index(struct hda_codec *codec, const char *name, int idx)
+{
+	struct hda_jack_tbl *jack;
+	int i, len = strlen(name);
+ again:
+	jack = codec->jacktbl.list;
+	for (i = 0; i < codec->jacktbl.used; i++, jack++) {
+		/* jack->kctl.id contains "XXX Jack" name string with index */
+		if (jack->kctl &&
+		    !strncmp(name, jack->kctl->id.name, len) &&
+		    !strcmp(" Jack", jack->kctl->id.name + len) &&
+		    jack->kctl->id.index == idx) {
+			idx++;
+			goto again;
+		}
+	}
+	return idx;
+}
+
 static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid,
-			 const struct auto_pin_cfg *cfg,
-			 char *lastname, int *lastidx)
+			 const struct auto_pin_cfg *cfg)
 {
 	unsigned int def_conf, conn;
 	char name[44];
 	int idx, err;
+	bool phantom_jack;
 
 	if (!nid)
 		return 0;
-	if (!is_jack_detectable(codec, nid))
-		return 0;
 	def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	conn = get_defcfg_connect(def_conf);
-	if (conn != AC_JACK_PORT_COMPLEX)
+	if (conn == AC_JACK_PORT_NONE)
 		return 0;
+	phantom_jack = (conn != AC_JACK_PORT_COMPLEX) ||
+		       !is_jack_detectable(codec, nid);
 
 	snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), &idx);
-	if (!strcmp(name, lastname) && idx == *lastidx)
-		idx++;
-	strncpy(lastname, name, 44);
-	*lastidx = idx;
-	err = snd_hda_jack_add_kctl(codec, nid, name, idx);
+	if (phantom_jack)
+		/* Example final name: "Internal Mic Phantom Jack" */
+		strncat(name, " Phantom", sizeof(name) - strlen(name) - 1);
+	idx = get_unique_index(codec, name, idx);
+	err = __snd_hda_jack_add_kctl(codec, nid, name, idx, phantom_jack);
 	if (err < 0)
 		return err;
-	return snd_hda_jack_detect_enable(codec, nid, 0);
+
+	if (!phantom_jack)
+		return snd_hda_jack_detect_enable(codec, nid, 0);
+	return 0;
 }
 
 /**
@@ -333,42 +372,41 @@
 			   const struct auto_pin_cfg *cfg)
 {
 	const hda_nid_t *p;
-	int i, err, lastidx = 0;
-	char lastname[44] = "";
+	int i, err;
 
 	for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) {
 		if (*p == *cfg->line_out_pins) /* might be duplicated */
 			break;
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0; i < cfg->num_inputs; i++) {
-		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0, p = cfg->dig_out_pins; i < cfg->dig_outs; i++, p++) {
-		err = add_jack_kctl(codec, *p, cfg, lastname, &lastidx);
+		err = add_jack_kctl(codec, *p, cfg);
 		if (err < 0)
 			return err;
 	}
-	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg, lastname, &lastidx);
+	err = add_jack_kctl(codec, cfg->dig_in_pin, cfg);
 	if (err < 0)
 		return err;
-	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg, lastname, &lastidx);
+	err = add_jack_kctl(codec, cfg->mono_out_pin, cfg);
 	if (err < 0)
 		return err;
 	return 0;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 8ae5246..a9803da 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -23,6 +23,7 @@
 	unsigned int pin_sense;		/* cached pin-sense value */
 	unsigned int jack_detect:1;	/* capable of jack-detection? */
 	unsigned int jack_dirty:1;	/* needs to update? */
+	unsigned int phantom_jack:1;    /* a fixed, always present port? */
 	struct snd_kcontrol *kctl;	/* assigned kctl for jack-detection */
 #ifdef CONFIG_SND_HDA_INPUT_JACK
 	int type;
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 9a096a8..1b4c129 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -89,7 +89,7 @@
 	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
 	  .subdevice = HDA_SUBDEV_AMP_FLAG, \
 	  .info = snd_hda_mixer_amp_switch_info, \
-	  .get = snd_hda_mixer_amp_switch_get, \
+	  .get = snd_hda_mixer_amp_switch_get_beep, \
 	  .put = snd_hda_mixer_amp_switch_put_beep, \
 	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
 #else
@@ -121,6 +121,8 @@
 int snd_hda_mixer_amp_switch_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol);
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
+int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol);
 int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 				      struct snd_ctl_elem_value *ucontrol);
 #endif
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index e59e2f0..7e46258 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -426,10 +426,10 @@
 
 static const char *get_pwr_state(u32 state)
 {
-	static const char * const buf[4] = {
-		"D0", "D1", "D2", "D3"
+	static const char * const buf[] = {
+		"D0", "D1", "D2", "D3", "D3cold"
 	};
-	if (state < 4)
+	if (state < ARRAY_SIZE(buf))
 		return buf[state];
 	return "UNKNOWN";
 }
@@ -451,14 +451,21 @@
 	int sup = snd_hda_param_read(codec, nid, AC_PAR_POWER_STATE);
 	int pwr = snd_hda_codec_read(codec, nid, 0,
 				     AC_VERB_GET_POWER_STATE, 0);
-	if (sup)
+	if (sup != -1)
 		snd_iprintf(buffer, "  Power states: %s\n",
 			    bits_names(sup, names, ARRAY_SIZE(names)));
 
-	snd_iprintf(buffer, "  Power: setting=%s, actual=%s\n",
+	snd_iprintf(buffer, "  Power: setting=%s, actual=%s",
 		    get_pwr_state(pwr & AC_PWRST_SETTING),
 		    get_pwr_state((pwr & AC_PWRST_ACTUAL) >>
 				  AC_PWRST_ACTUAL_SHIFT));
+	if (pwr & AC_PWRST_ERROR)
+		snd_iprintf(buffer, ", Error");
+	if (pwr & AC_PWRST_CLK_STOP_OK)
+		snd_iprintf(buffer, ", Clock-stop-OK");
+	if (pwr & AC_PWRST_SETTING_RESET)
+		snd_iprintf(buffer, ", Setting-reset");
+	snd_iprintf(buffer, "\n");
 }
 
 static void print_unsol_cap(struct snd_info_buffer *buffer,
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d8b2d6d..0208fa1 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -642,7 +642,7 @@
 }
 
 #ifdef CONFIG_PM
-static int ad198x_suspend(struct hda_codec *codec, pm_message_t state)
+static int ad198x_suspend(struct hda_codec *codec)
 {
 	ad198x_shutup(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 9647ed4..0c4c1a6 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -1892,7 +1892,7 @@
 	Manage PDREF, when transitioning to D3hot
 	(DAC,ADC) -> D3, PDREF=1, AFG->D3
 */
-static int cs421x_suspend(struct hda_codec *codec, pm_message_t state)
+static int cs421x_suspend(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	unsigned int coef;
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2bf99fc..1436118 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -554,7 +554,7 @@
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-static int conexant_suspend(struct hda_codec *codec, pm_message_t state)
+static int conexant_suspend(struct hda_codec *codec)
 {
 	snd_hda_shutup_pins(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index ad319d4d..641408d 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -85,7 +85,7 @@
 	 * Non-generic ATI/NVIDIA specific
 	 */
 	struct hda_multi_out multiout;
-	const struct hda_pcm_stream *pcm_playback;
+	struct hda_pcm_stream pcm_playback;
 };
 
 
@@ -787,7 +787,7 @@
 	int cp_ready = !!(res & AC_UNSOL_RES_CP_READY);
 
 	printk(KERN_INFO
-		"HDMI CP event: CODEC=%d PIN=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
+		"HDMI CP event: CODEC=%d TAG=%d SUBTAG=0x%x CP_STATE=%d CP_READY=%d\n",
 		codec->addr,
 		tag,
 		subtag,
@@ -876,7 +876,6 @@
 	struct hdmi_spec_per_pin *per_pin;
 	struct hdmi_eld *eld;
 	struct hdmi_spec_per_cvt *per_cvt = NULL;
-	int pinctl;
 
 	/* Validate hinfo */
 	pin_idx = hinfo_to_pin_index(spec, hinfo);
@@ -912,11 +911,6 @@
 	snd_hda_codec_write(codec, per_pin->pin_nid, 0,
 			    AC_VERB_SET_CONNECT_SEL,
 			    mux_idx);
-	pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0,
-				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-	snd_hda_codec_write(codec, per_pin->pin_nid, 0,
-			    AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    pinctl | PIN_OUT);
 	snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid);
 
 	/* Initially set the converter's capabilities */
@@ -1153,11 +1147,17 @@
 	struct hdmi_spec *spec = codec->spec;
 	int pin_idx = hinfo_to_pin_index(spec, hinfo);
 	hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid;
+	int pinctl;
 
 	hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels);
 
 	hdmi_setup_audio_infoframe(codec, pin_idx, substream);
 
+	pinctl = snd_hda_codec_read(codec, pin_nid, 0,
+				    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+	snd_hda_codec_write(codec, pin_nid, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT);
+
 	return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format);
 }
 
@@ -1277,6 +1277,22 @@
 	return 0;
 }
 
+static int generic_hdmi_init_per_pins(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	int pin_idx;
+
+	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
+		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
+		struct hdmi_eld *eld = &per_pin->sink_eld;
+
+		per_pin->codec = codec;
+		INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld);
+		snd_hda_eld_proc_new(codec, eld, pin_idx);
+	}
+	return 0;
+}
+
 static int generic_hdmi_init(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
@@ -1285,14 +1301,9 @@
 	for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) {
 		struct hdmi_spec_per_pin *per_pin = &spec->pins[pin_idx];
 		hda_nid_t pin_nid = per_pin->pin_nid;
-		struct hdmi_eld *eld = &per_pin->sink_eld;
 
 		hdmi_init_pin(codec, pin_nid);
 		snd_hda_jack_detect_enable(codec, pin_nid, pin_nid);
-
-		per_pin->codec = codec;
-		INIT_DELAYED_WORK(&per_pin->work, hdmi_repoll_eld);
-		snd_hda_eld_proc_new(codec, eld, pin_idx);
 	}
 	snd_hda_jack_report_sync(codec);
 	return 0;
@@ -1338,6 +1349,7 @@
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
+	generic_hdmi_init_per_pins(codec);
 
 	init_channel_allocations();
 
@@ -1352,45 +1364,65 @@
 {
 	struct hdmi_spec *spec = codec->spec;
 	struct hda_pcm *info = spec->pcm_rec;
-	int i;
+	unsigned int chans;
+	struct hda_pcm_stream *pstr;
 
-	codec->num_pcms = spec->num_cvts;
+	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
-	for (i = 0; i < codec->num_pcms; i++, info++) {
-		unsigned int chans;
-		struct hda_pcm_stream *pstr;
+	chans = get_wcaps(codec, spec->cvts[0].cvt_nid);
+	chans = get_wcaps_channels(chans);
 
-		chans = get_wcaps(codec, spec->cvts[i].cvt_nid);
-		chans = get_wcaps_channels(chans);
-
-		info->name = get_hdmi_pcm_name(i);
-		info->pcm_type = HDA_PCM_TYPE_HDMI;
-		pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
-		snd_BUG_ON(!spec->pcm_playback);
-		*pstr = *spec->pcm_playback;
-		pstr->nid = spec->cvts[i].cvt_nid;
-		if (pstr->channels_max <= 2 && chans && chans <= 16)
-			pstr->channels_max = chans;
-	}
+	info->name = get_hdmi_pcm_name(0);
+	info->pcm_type = HDA_PCM_TYPE_HDMI;
+	pstr = &info->stream[SNDRV_PCM_STREAM_PLAYBACK];
+	*pstr = spec->pcm_playback;
+	pstr->nid = spec->cvts[0].cvt_nid;
+	if (pstr->channels_max <= 2 && chans && chans <= 16)
+		pstr->channels_max = chans;
 
 	return 0;
 }
 
+/* unsolicited event for jack sensing */
+static void simple_hdmi_unsol_event(struct hda_codec *codec,
+				    unsigned int res)
+{
+	snd_hda_jack_set_dirty_all(codec);
+	snd_hda_jack_report_sync(codec);
+}
+
+/* generic_hdmi_build_jack can be used for simple_hdmi, too,
+ * as long as spec->pins[] is set correctly
+ */
+#define simple_hdmi_build_jack	generic_hdmi_build_jack
+
 static int simple_playback_build_controls(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec = codec->spec;
 	int err;
-	int i;
 
-	for (i = 0; i < codec->num_pcms; i++) {
-		err = snd_hda_create_spdif_out_ctls(codec,
-						    spec->cvts[i].cvt_nid,
-						    spec->cvts[i].cvt_nid);
-		if (err < 0)
-			return err;
-	}
+	err = snd_hda_create_spdif_out_ctls(codec,
+					    spec->cvts[0].cvt_nid,
+					    spec->cvts[0].cvt_nid);
+	if (err < 0)
+		return err;
+	return simple_hdmi_build_jack(codec, 0);
+}
 
+static int simple_playback_init(struct hda_codec *codec)
+{
+	struct hdmi_spec *spec = codec->spec;
+	hda_nid_t pin = spec->pins[0].pin_nid;
+
+	snd_hda_codec_write(codec, pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	/* some codecs require to unmute the pin */
+	if (get_wcaps(codec, pin) & AC_WCAP_OUT_AMP)
+		snd_hda_codec_write(codec, pin, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_OUT_UNMUTE);
+	snd_hda_jack_detect_enable(codec, pin, pin);
+	snd_hda_jack_report_sync(codec);
 	return 0;
 }
 
@@ -1418,7 +1450,15 @@
 	0x6, 0x8, 0xa, 0xc,
 };
 
-static const struct hda_verb nvhdmi_basic_init_7x[] = {
+static const struct hda_verb nvhdmi_basic_init_7x_2ch[] = {
+	/* set audio protect on */
+	{ 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
+	/* enable digital output on pin widget */
+	{ 0x5, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | 0x5 },
+	{} /* terminator */
+};
+
+static const struct hda_verb nvhdmi_basic_init_7x_8ch[] = {
 	/* set audio protect on */
 	{ 0x1, Nv_VERB_SET_Audio_Protection_On, 0x1},
 	/* enable digital output on pin widget */
@@ -1446,9 +1486,15 @@
 	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE)
 #endif
 
-static int nvhdmi_7x_init(struct hda_codec *codec)
+static int nvhdmi_7x_init_2ch(struct hda_codec *codec)
 {
-	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x);
+	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_2ch);
+	return 0;
+}
+
+static int nvhdmi_7x_init_8ch(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, nvhdmi_basic_init_7x_8ch);
 	return 0;
 }
 
@@ -1524,6 +1570,50 @@
 					     stream_tag, format, substream);
 }
 
+static const struct hda_pcm_stream simple_pcm_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.open = simple_playback_pcm_open,
+		.close = simple_playback_pcm_close,
+		.prepare = simple_playback_pcm_prepare
+	},
+};
+
+static const struct hda_codec_ops simple_hdmi_patch_ops = {
+	.build_controls = simple_playback_build_controls,
+	.build_pcms = simple_playback_build_pcms,
+	.init = simple_playback_init,
+	.free = simple_playback_free,
+	.unsol_event = simple_hdmi_unsol_event,
+};
+
+static int patch_simple_hdmi(struct hda_codec *codec,
+			     hda_nid_t cvt_nid, hda_nid_t pin_nid)
+{
+	struct hdmi_spec *spec;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->multiout.num_dacs = 0;  /* no analog */
+	spec->multiout.max_channels = 2;
+	spec->multiout.dig_out_nid = cvt_nid;
+	spec->num_cvts = 1;
+	spec->num_pins = 1;
+	spec->cvts[0].cvt_nid = cvt_nid;
+	spec->pins[0].pin_nid = pin_nid;
+	spec->pcm_playback = simple_pcm_playback;
+
+	codec->patch_ops = simple_hdmi_patch_ops;
+
+	return 0;
+}
+
 static void nvhdmi_8ch_7x_set_info_frame_parameters(struct hda_codec *codec,
 						    int channels)
 {
@@ -1696,54 +1786,20 @@
 	},
 };
 
-static const struct hda_pcm_stream nvhdmi_pcm_playback_2ch = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = nvhdmi_master_con_nid_7x,
-	.rates = SUPPORTED_RATES,
-	.maxbps = SUPPORTED_MAXBPS,
-	.formats = SUPPORTED_FORMATS,
-	.ops = {
-		.open = simple_playback_pcm_open,
-		.close = simple_playback_pcm_close,
-		.prepare = simple_playback_pcm_prepare
-	},
-};
-
-static const struct hda_codec_ops nvhdmi_patch_ops_8ch_7x = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = nvhdmi_7x_init,
-	.free = simple_playback_free,
-};
-
-static const struct hda_codec_ops nvhdmi_patch_ops_2ch = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = nvhdmi_7x_init,
-	.free = simple_playback_free,
-};
-
 static int patch_nvhdmi_2ch(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
+	int err = patch_simple_hdmi(codec, nvhdmi_master_con_nid_7x,
+				    nvhdmi_master_pin_nid_7x);
+	if (err < 0)
+		return err;
 
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	spec->multiout.num_dacs = 0;  /* no analog */
-	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
-	spec->num_cvts = 1;
-	spec->cvts[0].cvt_nid = nvhdmi_master_con_nid_7x;
-	spec->pcm_playback = &nvhdmi_pcm_playback_2ch;
-
-	codec->patch_ops = nvhdmi_patch_ops_2ch;
-
+	codec->patch_ops.init = nvhdmi_7x_init_2ch;
+	/* override the PCM rates, etc, as the codec doesn't give full list */
+	spec = codec->spec;
+	spec->pcm_playback.rates = SUPPORTED_RATES;
+	spec->pcm_playback.maxbps = SUPPORTED_MAXBPS;
+	spec->pcm_playback.formats = SUPPORTED_FORMATS;
 	return 0;
 }
 
@@ -1751,13 +1807,12 @@
 {
 	struct hdmi_spec *spec;
 	int err = patch_nvhdmi_2ch(codec);
-
 	if (err < 0)
 		return err;
 	spec = codec->spec;
 	spec->multiout.max_channels = 8;
-	spec->pcm_playback = &nvhdmi_pcm_playback_8ch_7x;
-	codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
+	spec->pcm_playback = nvhdmi_pcm_playback_8ch_7x;
+	codec->patch_ops.init = nvhdmi_7x_init_8ch;
 
 	/* Initialize the audio infoframe channel mask and checksum to something
 	 * valid */
@@ -1801,68 +1856,25 @@
 	return 0;
 }
 
-static const struct hda_pcm_stream atihdmi_pcm_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = ATIHDMI_CVT_NID,
-	.ops = {
-		.open = simple_playback_pcm_open,
-		.close = simple_playback_pcm_close,
-		.prepare = atihdmi_playback_pcm_prepare
-	},
-};
-
-static const struct hda_verb atihdmi_basic_init[] = {
-	/* enable digital output on pin widget */
-	{ 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{} /* terminator */
-};
-
-static int atihdmi_init(struct hda_codec *codec)
-{
-	struct hdmi_spec *spec = codec->spec;
-
-	snd_hda_sequence_write(codec, atihdmi_basic_init);
-	/* SI codec requires to unmute the pin */
-	if (get_wcaps(codec, spec->pins[0].pin_nid) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, spec->pins[0].pin_nid, 0,
-				    AC_VERB_SET_AMP_GAIN_MUTE,
-				    AMP_OUT_UNMUTE);
-	return 0;
-}
-
-static const struct hda_codec_ops atihdmi_patch_ops = {
-	.build_controls = simple_playback_build_controls,
-	.build_pcms = simple_playback_build_pcms,
-	.init = atihdmi_init,
-	.free = simple_playback_free,
-};
-
-
 static int patch_atihdmi(struct hda_codec *codec)
 {
 	struct hdmi_spec *spec;
-
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	spec->multiout.num_dacs = 0;	  /* no analog */
-	spec->multiout.max_channels = 2;
-	spec->multiout.dig_out_nid = ATIHDMI_CVT_NID;
-	spec->num_cvts = 1;
-	spec->cvts[0].cvt_nid = ATIHDMI_CVT_NID;
-	spec->pins[0].pin_nid = ATIHDMI_PIN_NID;
-	spec->pcm_playback = &atihdmi_pcm_digital_playback;
-
-	codec->patch_ops = atihdmi_patch_ops;
-
+	int err = patch_simple_hdmi(codec, ATIHDMI_CVT_NID, ATIHDMI_PIN_NID);
+	if (err < 0)
+		return err;
+	spec = codec->spec;
+	spec->pcm_playback.ops.prepare = atihdmi_playback_pcm_prepare;
 	return 0;
 }
 
+/* VIA HDMI Implementation */
+#define VIAHDMI_CVT_NID	0x02	/* audio converter1 */
+#define VIAHDMI_PIN_NID	0x03	/* HDMI output pin1 */
+
+static int patch_via_hdmi(struct hda_codec *codec)
+{
+	return patch_simple_hdmi(codec, VIAHDMI_CVT_NID, VIAHDMI_PIN_NID);
+}
 
 /*
  * patch entries
@@ -1902,8 +1914,13 @@
 { .id = 0x10de0042, .name = "GPU 42 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0043, .name = "GPU 43 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0044, .name = "GPU 44 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x10de0067, .name = "MCP67 HDMI",	.patch = patch_nvhdmi_2ch },
 { .id = 0x10de8001, .name = "MCP73 HDMI",	.patch = patch_nvhdmi_2ch },
+{ .id = 0x11069f80, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
+{ .id = 0x11069f81, .name = "VX900 HDMI/DP",	.patch = patch_via_hdmi },
+{ .id = 0x11069f84, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
+{ .id = 0x11069f85, .name = "VX11 HDMI/DP",	.patch = patch_generic_hdmi },
 { .id = 0x80860054, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862801, .name = "Bearlake HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862802, .name = "Cantiga HDMI",	.patch = patch_generic_hdmi },
@@ -1911,6 +1928,7 @@
 { .id = 0x80862804, .name = "IbexPeak HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862805, .name = "CougarPoint HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
+{ .id = 0x80862807, .name = "Haswell HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x80862880, .name = "CedarTrail HDMI",	.patch = patch_generic_hdmi },
 { .id = 0x808629fb, .name = "Crestline HDMI",	.patch = patch_generic_hdmi },
 {} /* terminator */
@@ -1948,8 +1966,13 @@
 MODULE_ALIAS("snd-hda-codec-id:10de0042");
 MODULE_ALIAS("snd-hda-codec-id:10de0043");
 MODULE_ALIAS("snd-hda-codec-id:10de0044");
+MODULE_ALIAS("snd-hda-codec-id:10de0051");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
 MODULE_ALIAS("snd-hda-codec-id:10de8001");
+MODULE_ALIAS("snd-hda-codec-id:11069f80");
+MODULE_ALIAS("snd-hda-codec-id:11069f81");
+MODULE_ALIAS("snd-hda-codec-id:11069f84");
+MODULE_ALIAS("snd-hda-codec-id:11069f85");
 MODULE_ALIAS("snd-hda-codec-id:17e80047");
 MODULE_ALIAS("snd-hda-codec-id:80860054");
 MODULE_ALIAS("snd-hda-codec-id:80862801");
@@ -1958,6 +1981,7 @@
 MODULE_ALIAS("snd-hda-codec-id:80862804");
 MODULE_ALIAS("snd-hda-codec-id:80862805");
 MODULE_ALIAS("snd-hda-codec-id:80862806");
+MODULE_ALIAS("snd-hda-codec-id:80862807");
 MODULE_ALIAS("snd-hda-codec-id:80862880");
 MODULE_ALIAS("snd-hda-codec-id:808629fb");
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index aa4c25e..f141395 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -170,10 +170,10 @@
 	hda_nid_t imux_pins[HDA_MAX_NUM_INPUTS];
 	unsigned int dyn_adc_idx[HDA_MAX_NUM_INPUTS];
 	int int_mic_idx, ext_mic_idx, dock_mic_idx; /* for auto-mic */
+	hda_nid_t inv_dmic_pin;
 
 	/* hooks */
 	void (*init_hook)(struct hda_codec *codec);
-	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	void (*power_hook)(struct hda_codec *codec);
 #endif
@@ -201,6 +201,8 @@
 	unsigned int vol_in_capsrc:1; /* use capsrc volume (ADC has no vol) */
 	unsigned int parse_flags; /* passed to snd_hda_parse_pin_defcfg() */
 	unsigned int shared_mic_hp:1; /* HP/Mic-in sharing */
+	unsigned int inv_dmic_fixup:1; /* has inverted digital-mic workaround */
+	unsigned int inv_dmic_muted:1; /* R-ch of inv d-mic is muted? */
 
 	/* auto-mute control */
 	int automute_mode;
@@ -298,6 +300,39 @@
 }
 
 static void call_update_outputs(struct hda_codec *codec);
+static void alc_inv_dmic_sync(struct hda_codec *codec, bool force);
+
+/* for shared I/O, change the pin-control accordingly */
+static void update_shared_mic_hp(struct hda_codec *codec, bool set_as_mic)
+{
+	struct alc_spec *spec = codec->spec;
+	unsigned int val;
+	hda_nid_t pin = spec->autocfg.inputs[1].pin;
+	/* NOTE: this assumes that there are only two inputs, the
+	 * first is the real internal mic and the second is HP/mic jack.
+	 */
+
+	val = snd_hda_get_default_vref(codec, pin);
+
+	/* This pin does not have vref caps - let's enable vref on pin 0x18
+	   instead, as suggested by Realtek */
+	if (val == AC_PINCTL_VREF_HIZ) {
+		const hda_nid_t vref_pin = 0x18;
+		/* Sanity check pin 0x18 */
+		if (get_wcaps_type(get_wcaps(codec, vref_pin)) == AC_WID_PIN &&
+		    get_defcfg_connect(snd_hda_codec_get_pincfg(codec, vref_pin)) == AC_JACK_PORT_NONE) {
+			unsigned int vref_val = snd_hda_get_default_vref(codec, vref_pin);
+			if (vref_val != AC_PINCTL_VREF_HIZ)
+				snd_hda_set_pin_ctl(codec, vref_pin, PIN_IN | (set_as_mic ? vref_val : 0));
+		}
+	}
+
+	val = set_as_mic ? val | PIN_IN : PIN_HP;
+	snd_hda_set_pin_ctl(codec, pin, val);
+
+	spec->automute_speaker = !set_as_mic;
+	call_update_outputs(codec);
+}
 
 /* select the given imux item; either unmute exclusively or select the route */
 static int alc_mux_select(struct hda_codec *codec, unsigned int adc_idx,
@@ -325,21 +360,8 @@
 		return 0;
 	spec->cur_mux[adc_idx] = idx;
 
-	/* for shared I/O, change the pin-control accordingly */
-	if (spec->shared_mic_hp) {
-		unsigned int val;
-		hda_nid_t pin = spec->autocfg.inputs[1].pin;
-		/* NOTE: this assumes that there are only two inputs, the
-		 * first is the real internal mic and the second is HP jack.
-		 */
-		if (spec->cur_mux[adc_idx])
-			val = snd_hda_get_default_vref(codec, pin) | PIN_IN;
-		else
-			val = PIN_HP;
-		snd_hda_set_pin_ctl(codec, pin, val);
-		spec->automute_speaker = !spec->cur_mux[adc_idx];
-		call_update_outputs(codec);
-	}
+	if (spec->shared_mic_hp)
+		update_shared_mic_hp(codec, spec->cur_mux[adc_idx]);
 
 	if (spec->dyn_adc_switch) {
 		alc_dyn_adc_pcm_resetup(codec, idx);
@@ -368,6 +390,7 @@
 					  AC_VERB_SET_CONNECT_SEL,
 					  imux->items[idx].index);
 	}
+	alc_inv_dmic_sync(codec, true);
 	return 1;
 }
 
@@ -664,7 +687,7 @@
 }
 
 /* unsolicited event for HP jack sensing */
-static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
+static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
 {
 	int action;
 
@@ -1000,11 +1023,9 @@
 	spec->automute_lo = spec->automute_lo_possible;
 	spec->automute_speaker = spec->automute_speaker_possible;
 
-	if (spec->automute_speaker_possible || spec->automute_lo_possible) {
+	if (spec->automute_speaker_possible || spec->automute_lo_possible)
 		/* create a control for automute mode */
 		alc_add_automute_mode_enum(codec);
-		spec->unsol_event = alc_sku_unsol_event;
-	}
 }
 
 /* return the position of NID in the list, or -1 if not found */
@@ -1167,7 +1188,6 @@
 
 	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x/0x%x\n",
 		    ext, fixed, dock);
-	spec->unsol_event = alc_sku_unsol_event;
 }
 
 /* check the availabilities of auto-mute and auto-mic switches */
@@ -1556,14 +1576,14 @@
 
 static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol,
-				 getput_call_t func, bool check_adc_switch)
+				 getput_call_t func, bool is_put)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct alc_spec *spec = codec->spec;
 	int i, err = 0;
 
 	mutex_lock(&codec->control_mutex);
-	if (check_adc_switch && spec->dyn_adc_switch) {
+	if (is_put && spec->dyn_adc_switch) {
 		for (i = 0; i < spec->num_adc_nids; i++) {
 			kcontrol->private_value =
 				HDA_COMPOSE_AMP_VAL(spec->adc_nids[i],
@@ -1584,6 +1604,8 @@
 						    3, 0, HDA_INPUT);
 		err = func(kcontrol, ucontrol);
 	}
+	if (err >= 0 && is_put)
+		alc_inv_dmic_sync(codec, false);
  error:
 	mutex_unlock(&codec->control_mutex);
 	return err;
@@ -1676,6 +1698,116 @@
 DEFINE_CAPMIX_NOSRC(3);
 
 /*
+ * Inverted digital-mic handling
+ *
+ * First off, it's a bit tricky.  The "Inverted Internal Mic Capture Switch"
+ * gives the additional mute only to the right channel of the digital mic
+ * capture stream.  This is a workaround for avoiding the almost silence
+ * by summing the stereo stream from some (known to be ForteMedia)
+ * digital mic unit.
+ *
+ * The logic is to call alc_inv_dmic_sync() after each action (possibly)
+ * modifying ADC amp.  When the mute flag is set, it mutes the R-channel
+ * without caching so that the cache can still keep the original value.
+ * The cached value is then restored when the flag is set off or any other
+ * than d-mic is used as the current input source.
+ */
+static void alc_inv_dmic_sync(struct hda_codec *codec, bool force)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	if (!spec->inv_dmic_fixup)
+		return;
+	if (!spec->inv_dmic_muted && !force)
+		return;
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		int src = spec->dyn_adc_switch ? 0 : i;
+		bool dmic_fixup = false;
+		hda_nid_t nid;
+		int parm, dir, v;
+
+		if (spec->inv_dmic_muted &&
+		    spec->imux_pins[spec->cur_mux[src]] == spec->inv_dmic_pin)
+			dmic_fixup = true;
+		if (!dmic_fixup && !force)
+			continue;
+		if (spec->vol_in_capsrc) {
+			nid = spec->capsrc_nids[i];
+			parm = AC_AMP_SET_RIGHT | AC_AMP_SET_OUTPUT;
+			dir = HDA_OUTPUT;
+		} else {
+			nid = spec->adc_nids[i];
+			parm = AC_AMP_SET_RIGHT | AC_AMP_SET_INPUT;
+			dir = HDA_INPUT;
+		}
+		/* we care only right channel */
+		v = snd_hda_codec_amp_read(codec, nid, 1, dir, 0);
+		if (v & 0x80) /* if already muted, we don't need to touch */
+			continue;
+		if (dmic_fixup) /* add mute for d-mic */
+			v |= 0x80;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    parm | v);
+	}
+}
+
+static int alc_inv_dmic_sw_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	ucontrol->value.integer.value[0] = !spec->inv_dmic_muted;
+	return 0;
+}
+
+static int alc_inv_dmic_sw_put(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	unsigned int val = !ucontrol->value.integer.value[0];
+
+	if (val == spec->inv_dmic_muted)
+		return 0;
+	spec->inv_dmic_muted = val;
+	alc_inv_dmic_sync(codec, true);
+	return 0;
+}
+
+static const struct snd_kcontrol_new alc_inv_dmic_sw = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.info = snd_ctl_boolean_mono_info,
+	.get = alc_inv_dmic_sw_get,
+	.put = alc_inv_dmic_sw_put,
+};
+
+static int alc_add_inv_dmic_mixer(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct alc_spec *spec = codec->spec;
+	struct snd_kcontrol_new *knew = alc_kcontrol_new(spec);
+	if (!knew)
+		return -ENOMEM;
+	*knew = alc_inv_dmic_sw;
+	knew->name = kstrdup("Inverted Internal Mic Capture Switch", GFP_KERNEL);
+	if (!knew->name)
+		return -ENOMEM;
+	spec->inv_dmic_fixup = 1;
+	spec->inv_dmic_muted = 0;
+	spec->inv_dmic_pin = nid;
+	return 0;
+}
+
+/* typically the digital mic is put at node 0x12 */
+static void alc_fixup_inv_dmic_0x12(struct hda_codec *codec,
+				    const struct alc_fixup *fix, int action)
+{
+	if (action == ALC_FIXUP_ACT_PROBE)
+		alc_add_inv_dmic_mixer(codec, 0x12);
+}
+
+/*
  * virtual master controls
  */
 
@@ -1865,13 +1997,31 @@
 	return 0;
 }
 
-static int alc_build_controls(struct hda_codec *codec)
+static int alc_build_jacks(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
+
+	if (spec->shared_mic_hp) {
+		int err;
+		int nid = spec->autocfg.inputs[1].pin;
+		err = snd_hda_jack_add_kctl(codec, nid, "Headphone Mic", 0);
+		if (err < 0)
+			return err;
+		err = snd_hda_jack_detect_enable(codec, nid, 0);
+		if (err < 0)
+			return err;
+	}
+
+	return snd_hda_jack_add_kctls(codec, &spec->autocfg);
+}
+
+static int alc_build_controls(struct hda_codec *codec)
+{
 	int err = __alc_build_controls(codec);
 	if (err < 0)
 		return err;
-	err = snd_hda_jack_add_kctls(codec, &spec->autocfg);
+
+	err = alc_build_jacks(codec);
 	if (err < 0)
 		return err;
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_BUILD);
@@ -1908,14 +2058,6 @@
 	return 0;
 }
 
-static void alc_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	struct alc_spec *spec = codec->spec;
-
-	if (spec->unsol_event)
-		spec->unsol_event(codec, res);
-}
-
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static int alc_check_power_status(struct hda_codec *codec, hda_nid_t nid)
 {
@@ -2300,7 +2442,7 @@
 	alc_auto_setup_eapd(codec, false);
 }
 
-static int alc_suspend(struct hda_codec *codec, pm_message_t state)
+static int alc_suspend(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	alc_shutup(codec);
@@ -2317,6 +2459,7 @@
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
+	alc_inv_dmic_sync(codec, true);
 	hda_call_check_power_status(codec, 0x01);
 	return 0;
 }
@@ -4116,14 +4259,12 @@
  */
 static void alc_auto_init_std(struct hda_codec *codec)
 {
-	struct alc_spec *spec = codec->spec;
 	alc_auto_init_multi_out(codec);
 	alc_auto_init_extra_out(codec);
 	alc_auto_init_analog_input(codec);
 	alc_auto_init_input_src(codec);
 	alc_auto_init_digital(codec);
-	if (spec->unsol_event)
-		alc_inithook(codec);
+	alc_inithook(codec);
 }
 
 /*
@@ -4724,7 +4865,6 @@
 		spec->automute_speaker = 1;
 		spec->autocfg.hp_pins[0] = 0x0f; /* copy it for automute */
 		snd_hda_jack_detect_enable(codec, 0x0f, ALC_HP_EVENT);
-		spec->unsol_event = alc_sku_unsol_event;
 		snd_hda_gen_add_verbs(&spec->gen, alc_gpio1_init_verbs);
 	}
 }
@@ -4909,6 +5049,7 @@
 	ALC889_FIXUP_DAC_ROUTE,
 	ALC889_FIXUP_MBP_VREF,
 	ALC889_FIXUP_IMAC91_VREF,
+	ALC882_FIXUP_INV_DMIC,
 };
 
 static void alc889_fixup_coef(struct hda_codec *codec,
@@ -5212,6 +5353,10 @@
 		.chained = true,
 		.chain_id = ALC882_FIXUP_GPIO1,
 	},
+	[ALC882_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
 };
 
 static const struct snd_pci_quirk alc882_fixup_tbl[] = {
@@ -5286,6 +5431,7 @@
 	{.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"},
 	{.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"},
 	{.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"},
+	{.id = ALC882_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{}
 };
 
@@ -5373,6 +5519,7 @@
 	ALC262_FIXUP_LENOVO_3000,
 	ALC262_FIXUP_BENQ,
 	ALC262_FIXUP_BENQ_T31,
+	ALC262_FIXUP_INV_DMIC,
 };
 
 static const struct alc_fixup alc262_fixups[] = {
@@ -5424,6 +5571,10 @@
 			{}
 		}
 	},
+	[ALC262_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
 };
 
 static const struct snd_pci_quirk alc262_fixup_tbl[] = {
@@ -5438,6 +5589,10 @@
 	{}
 };
 
+static const struct alc_model_fixup alc262_fixup_models[] = {
+	{.id = ALC262_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{}
+};
 
 /*
  */
@@ -5466,7 +5621,8 @@
 #endif
 	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
 
-	alc_pick_fixup(codec, NULL, alc262_fixup_tbl, alc262_fixups);
+	alc_pick_fixup(codec, alc262_fixup_models, alc262_fixup_tbl,
+		       alc262_fixups);
 	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
 
 	alc_auto_parse_customize_define(codec);
@@ -5522,6 +5678,22 @@
 	{ }
 };
 
+enum {
+	ALC268_FIXUP_INV_DMIC,
+};
+
+static const struct alc_fixup alc268_fixups[] = {
+	[ALC268_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
+};
+
+static const struct alc_model_fixup alc268_fixup_models[] = {
+	{.id = ALC268_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{}
+};
+
 /*
  * BIOS auto configuration
  */
@@ -5553,6 +5725,9 @@
 
 	spec = codec->spec;
 
+	alc_pick_fixup(codec, alc268_fixup_models, NULL, alc268_fixups);
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE);
+
 	/* automatic parse from the BIOS config */
 	err = alc268_parse_auto_config(codec);
 	if (err < 0)
@@ -5582,6 +5757,8 @@
 	codec->patch_ops = alc_patch_ops;
 	spec->shutup = alc_eapd_shutup;
 
+	alc_apply_fixup(codec, ALC_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
@@ -5704,6 +5881,15 @@
 }
 #endif /* CONFIG_PM */
 
+static void alc269_fixup_pincfg_no_hp_to_lineout(struct hda_codec *codec,
+						 const struct alc_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	if (action == ALC_FIXUP_ACT_PRE_PROBE)
+		spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP;
+}
+
 static void alc269_fixup_hweq(struct hda_codec *codec,
 			       const struct alc_fixup *fix, int action)
 {
@@ -5810,6 +5996,7 @@
 	}
 }
 
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -5828,6 +6015,9 @@
 	ALC269VB_FIXUP_AMIC,
 	ALC269VB_FIXUP_DMIC,
 	ALC269_FIXUP_MIC2_MUTE_LED,
+	ALC269_FIXUP_INV_DMIC,
+	ALC269_FIXUP_LENOVO_DOCK,
+	ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT,
 };
 
 static const struct alc_fixup alc269_fixups[] = {
@@ -5952,12 +6142,33 @@
 		.type = ALC_FIXUP_FUNC,
 		.v.func = alc269_fixup_mic2_mute,
 	},
+	[ALC269_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
+	[ALC269_FIXUP_LENOVO_DOCK] = {
+		.type = ALC_FIXUP_PINS,
+		.v.pins = (const struct alc_pincfg[]) {
+			{ 0x19, 0x23a11040 }, /* dock mic */
+			{ 0x1b, 0x2121103f }, /* dock headphone */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT
+	},
+	[ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc269_fixup_pincfg_no_hp_to_lineout,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW),
+	SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC),
 	SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC),
@@ -5975,6 +6186,7 @@
 	SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
@@ -6033,6 +6245,10 @@
 static const struct alc_model_fixup alc269_fixup_models[] = {
 	{.id = ALC269_FIXUP_AMIC, .name = "laptop-amic"},
 	{.id = ALC269_FIXUP_DMIC, .name = "laptop-dmic"},
+	{.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"},
+	{.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"},
+	{.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"},
+	{.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"},
 	{}
 };
 
@@ -6329,12 +6545,6 @@
 	{}
 };
 
-static const struct hda_verb alc660vd_eapd_verbs[] = {
-	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
-	{ }
-};
-
 /*
  */
 static int patch_alc861vd(struct hda_codec *codec)
@@ -6356,11 +6566,6 @@
 	if (err < 0)
 		goto error;
 
-	if (codec->vendor_id == 0x10ec0660) {
-		/* always turn on EAPD */
-		snd_hda_gen_add_verbs(&spec->gen, alc660vd_eapd_verbs);
-	}
-
 	if (!spec->no_analog) {
 		err = snd_hda_attach_beep_device(codec, 0x23);
 		if (err < 0)
@@ -6443,6 +6648,7 @@
 	ALC662_FIXUP_ASUS_MODE8,
 	ALC662_FIXUP_NO_JACK_DETECT,
 	ALC662_FIXUP_ZOTAC_Z68,
+	ALC662_FIXUP_INV_DMIC,
 };
 
 static const struct alc_fixup alc662_fixups[] = {
@@ -6599,12 +6805,17 @@
 			{ }
 		}
 	},
+	[ALC662_FIXUP_INV_DMIC] = {
+		.type = ALC_FIXUP_FUNC,
+		.v.func = alc_fixup_inv_dmic_0x12,
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE),
+	SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE),
 	SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800),
 	SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT),
@@ -6685,6 +6896,7 @@
 	{.id = ALC662_FIXUP_ASUS_MODE6, .name = "asus-mode6"},
 	{.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"},
 	{.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"},
+	{.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"},
 	{}
 };
 
@@ -6831,6 +7043,7 @@
 	{ .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
 	{ .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
 	{ .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
+	{ .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
 	{ .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
 	  .patch = patch_alc861 },
 	{ .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 0767528..a1596a3 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4997,7 +4997,7 @@
 	return 0;
 }
 
-static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
+static int stac92xx_suspend(struct hda_codec *codec)
 {
 	stac92xx_shutup(codec);
 	return 0;
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 82b3680..9064556 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1748,7 +1748,7 @@
 }
 
 #ifdef CONFIG_PM
-static int via_suspend(struct hda_codec *codec, pm_message_t state)
+static int via_suspend(struct hda_codec *codec)
 {
 	struct via_spec *spec = codec->spec;
 	vt1708_stop_hp_work(spec);
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index a01a00d..bed9f34 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -2793,9 +2793,10 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_vt1724_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_vt1724_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ice1712 *ice = card->private_data;
 
 	if (!ice->pm_suspend_enabled)
@@ -2820,13 +2821,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_vt1724_resume(struct pci_dev *pci)
+static int snd_vt1724_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ice1712 *ice = card->private_data;
 
 	if (!ice->pm_suspend_enabled)
@@ -2871,17 +2873,21 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(snd_vt1724_pm, snd_vt1724_suspend, snd_vt1724_resume);
+#define SND_VT1724_PM_OPS	&snd_vt1724_pm
+#else
+#define SND_VT1724_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static struct pci_driver vt1724_driver = {
 	.name = KBUILD_MODNAME,
 	.id_table = snd_vt1724_ids,
 	.probe = snd_vt1724_probe,
 	.remove = __devexit_p(snd_vt1724_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_vt1724_suspend,
-	.resume = snd_vt1724_resume,
-#endif
+	.driver = {
+		.pm = SND_VT1724_PM_OPS,
+	},
 };
 
 module_pci_driver(vt1724_driver);
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index f4e2dd4..cd553f5 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -2624,9 +2624,10 @@
 /*
  * power management
  */
-static int intel8x0_suspend(struct pci_dev *pci, pm_message_t state)
+static int intel8x0_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0 *chip = card->private_data;
 	int i;
 
@@ -2658,13 +2659,14 @@
 	/* The call below may disable built-in speaker on some laptops
 	 * after S2RAM.  So, don't touch it.
 	 */
-	/* pci_set_power_state(pci, pci_choose_state(pci, state)); */
+	/* pci_set_power_state(pci, PCI_D3hot); */
 	return 0;
 }
 
-static int intel8x0_resume(struct pci_dev *pci)
+static int intel8x0_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0 *chip = card->private_data;
 	int i;
 
@@ -2734,6 +2736,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(intel8x0_pm, intel8x0_suspend, intel8x0_resume);
+#define INTEL8X0_PM_OPS	&intel8x0_pm
+#else
+#define INTEL8X0_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #define INTEL8X0_TESTBUF_SIZE	32768	/* enough large for one shot */
@@ -3343,10 +3350,9 @@
 	.id_table = snd_intel8x0_ids,
 	.probe = snd_intel8x0_probe,
 	.remove = __devexit_p(snd_intel8x0_remove),
-#ifdef CONFIG_PM
-	.suspend = intel8x0_suspend,
-	.resume = intel8x0_resume,
-#endif
+	.driver = {
+		.pm = INTEL8X0_PM_OPS,
+	},
 };
 
 module_pci_driver(intel8x0_driver);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index fc27a6a..da44bb3 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -1012,9 +1012,10 @@
 /*
  * power management
  */
-static int intel8x0m_suspend(struct pci_dev *pci, pm_message_t state)
+static int intel8x0m_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0m *chip = card->private_data;
 	int i;
 
@@ -1028,13 +1029,14 @@
 	}
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int intel8x0m_resume(struct pci_dev *pci)
+static int intel8x0m_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct intel8x0m *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -1060,6 +1062,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(intel8x0m_pm, intel8x0m_suspend, intel8x0m_resume);
+#define INTEL8X0M_PM_OPS	&intel8x0m_pm
+#else
+#define INTEL8X0M_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef CONFIG_PROC_FS
@@ -1329,10 +1336,9 @@
 	.id_table = snd_intel8x0m_ids,
 	.probe = snd_intel8x0m_probe,
 	.remove = __devexit_p(snd_intel8x0m_remove),
-#ifdef CONFIG_PM
-	.suspend = intel8x0m_suspend,
-	.resume = intel8x0m_resume,
-#endif
+	.driver = {
+		.pm = INTEL8X0M_PM_OPS,
+	},
 };
 
 module_pci_driver(intel8x0m_driver);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index deef213..c85d1ff 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -361,74 +361,6 @@
 #define DSP2HOST_REQ_I2SRATE    0x02
 #define DSP2HOST_REQ_TIMER      0x04
 
-/* AC97 registers */
-/* XXX fix this crap up */
-/*#define AC97_RESET              0x00*/
-
-#define AC97_VOL_MUTE_B         0x8000
-#define AC97_VOL_M              0x1F
-#define AC97_LEFT_VOL_S         8
-
-#define AC97_MASTER_VOL         0x02
-#define AC97_LINE_LEVEL_VOL     0x04
-#define AC97_MASTER_MONO_VOL    0x06
-#define AC97_PC_BEEP_VOL        0x0A
-#define AC97_PC_BEEP_VOL_M      0x0F
-#define AC97_SROUND_MASTER_VOL  0x38
-#define AC97_PC_BEEP_VOL_S      1
-
-/*#define AC97_PHONE_VOL          0x0C
-#define AC97_MIC_VOL            0x0E*/
-#define AC97_MIC_20DB_ENABLE    0x40
-
-/*#define AC97_LINEIN_VOL         0x10
-#define AC97_CD_VOL             0x12
-#define AC97_VIDEO_VOL          0x14
-#define AC97_AUX_VOL            0x16*/
-#define AC97_PCM_OUT_VOL        0x18
-/*#define AC97_RECORD_SELECT      0x1A*/
-#define AC97_RECORD_MIC         0x00
-#define AC97_RECORD_CD          0x01
-#define AC97_RECORD_VIDEO       0x02
-#define AC97_RECORD_AUX         0x03
-#define AC97_RECORD_MONO_MUX    0x02
-#define AC97_RECORD_DIGITAL     0x03
-#define AC97_RECORD_LINE        0x04
-#define AC97_RECORD_STEREO      0x05
-#define AC97_RECORD_MONO        0x06
-#define AC97_RECORD_PHONE       0x07
-
-/*#define AC97_RECORD_GAIN        0x1C*/
-#define AC97_RECORD_VOL_M       0x0F
-
-/*#define AC97_GENERAL_PURPOSE    0x20*/
-#define AC97_POWER_DOWN_CTRL    0x26
-#define AC97_ADC_READY          0x0001
-#define AC97_DAC_READY          0x0002
-#define AC97_ANALOG_READY       0x0004
-#define AC97_VREF_ON            0x0008
-#define AC97_PR0                0x0100
-#define AC97_PR1                0x0200
-#define AC97_PR2                0x0400
-#define AC97_PR3                0x0800
-#define AC97_PR4                0x1000
-
-#define AC97_RESERVED1          0x28
-
-#define AC97_VENDOR_TEST        0x5A
-
-#define AC97_CLOCK_DELAY        0x5C
-#define AC97_LINEOUT_MUX_SEL    0x0001
-#define AC97_MONO_MUX_SEL       0x0002
-#define AC97_CLOCK_DELAY_SEL    0x1F
-#define AC97_DAC_CDS_SHIFT      6
-#define AC97_ADC_CDS_SHIFT      11
-
-#define AC97_MULTI_CHANNEL_SEL  0x74
-
-/*#define AC97_VENDOR_ID1         0x7C
-#define AC97_VENDOR_ID2         0x7E*/
-
 /*
  * ASSP control regs
  */
@@ -2459,9 +2391,10 @@
  * APM support
  */
 #ifdef CONFIG_PM
-static int m3_suspend(struct pci_dev *pci, pm_message_t state)
+static int m3_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_m3 *chip = card->private_data;
 	int i, dsp_index;
 
@@ -2489,13 +2422,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int m3_resume(struct pci_dev *pci)
+static int m3_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_m3 *chip = card->private_data;
 	int i, dsp_index;
 
@@ -2546,6 +2480,11 @@
 	chip->in_suspend = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(m3_pm, m3_suspend, m3_resume);
+#define M3_PM_OPS	&m3_pm
+#else
+#define M3_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 #ifdef CONFIG_SND_MAESTRO3_INPUT
@@ -2842,10 +2781,9 @@
 	.id_table = snd_m3_ids,
 	.probe = snd_m3_probe,
 	.remove = __devexit_p(snd_m3_remove),
-#ifdef CONFIG_PM
-	.suspend = m3_suspend,
-	.resume = m3_resume,
-#endif
+	.driver = {
+		.pm = M3_PM_OPS,
+	},
 };
 	
 module_pci_driver(m3_driver);
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 8159b05..465cff2 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1382,9 +1382,10 @@
  * APM event handler, so the card is properly reinitialized after a power
  * event.
  */
-static int nm256_suspend(struct pci_dev *pci, pm_message_t state)
+static int nm256_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct nm256 *chip = card->private_data;
 
 	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
@@ -1393,13 +1394,14 @@
 	chip->coeffs_current = 0;
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int nm256_resume(struct pci_dev *pci)
+static int nm256_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct nm256 *chip = card->private_data;
 	int i;
 
@@ -1434,6 +1436,11 @@
 	chip->in_resume = 0;
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(nm256_pm, nm256_suspend, nm256_resume);
+#define NM256_PM_OPS	&nm256_pm
+#else
+#define NM256_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_nm256_free(struct nm256 *chip)
@@ -1747,10 +1754,9 @@
 	.id_table = snd_nm256_ids,
 	.probe = snd_nm256_probe,
 	.remove = __devexit_p(snd_nm256_remove),
-#ifdef CONFIG_PM
-	.suspend = nm256_suspend,
-	.resume = nm256_resume,
-#endif
+	.driver = {
+		.pm = NM256_PM_OPS,
+	},
 };
 
 module_pci_driver(nm256_driver);
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index 610275b..37520a2b4 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -873,8 +873,9 @@
 	.probe = generic_oxygen_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
 #ifdef CONFIG_PM
-	.suspend = oxygen_pci_suspend,
-	.resume = oxygen_pci_resume,
+	.driver = {
+		.pm = &oxygen_pci_pm,
+	},
 #endif
 };
 
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index f53897a..7112a89 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -162,8 +162,7 @@
 		    );
 void oxygen_pci_remove(struct pci_dev *pci);
 #ifdef CONFIG_PM
-int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
-int oxygen_pci_resume(struct pci_dev *pci);
+extern const struct dev_pm_ops oxygen_pci_pm;
 #endif
 void oxygen_pci_shutdown(struct pci_dev *pci);
 
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 92e2d67..ab8738e 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -727,9 +727,10 @@
 EXPORT_SYMBOL(oxygen_pci_remove);
 
 #ifdef CONFIG_PM
-int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state)
+static int oxygen_pci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct oxygen *chip = card->private_data;
 	unsigned int i, saved_interrupt_mask;
 
@@ -756,10 +757,9 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
-EXPORT_SYMBOL(oxygen_pci_suspend);
 
 static const u32 registers_to_restore[OXYGEN_IO_SIZE / 32] = {
 	0xffffffff, 0x00ff077f, 0x00011d08, 0x007f00ff,
@@ -787,9 +787,10 @@
 					  chip->saved_ac97_registers[codec][i]);
 }
 
-int oxygen_pci_resume(struct pci_dev *pci)
+static int oxygen_pci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct oxygen *chip = card->private_data;
 	unsigned int i;
 
@@ -820,7 +821,9 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
-EXPORT_SYMBOL(oxygen_pci_resume);
+
+SIMPLE_DEV_PM_OPS(oxygen_pci_pm, oxygen_pci_suspend, oxygen_pci_resume);
+EXPORT_SYMBOL(oxygen_pci_pm);
 #endif /* CONFIG_PM */
 
 void oxygen_pci_shutdown(struct pci_dev *pci)
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 19962c6..d3b606b 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -94,8 +94,9 @@
 	.probe = xonar_probe,
 	.remove = __devexit_p(oxygen_pci_remove),
 #ifdef CONFIG_PM
-	.suspend = oxygen_pci_suspend,
-	.resume = oxygen_pci_resume,
+	.driver = {
+		.pm = &oxygen_pci_pm,
+	},
 #endif
 	.shutdown = oxygen_pci_shutdown,
 };
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 0435f45..e3ac1f7 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1368,6 +1368,67 @@
 	}
 }
 
+/* Access to the results of the CMD_GET_TIME_CODE RMH */
+#define TIME_CODE_VALID_MASK	0x00800000
+#define TIME_CODE_NEW_MASK	0x00400000
+#define TIME_CODE_BACK_MASK	0x00200000
+#define TIME_CODE_WAIT_MASK	0x00100000
+
+/* Values for the CMD_MANAGE_SIGNAL RMH */
+#define MANAGE_SIGNAL_TIME_CODE	0x01
+#define MANAGE_SIGNAL_MIDI	0x02
+
+/* linear time code read proc*/
+static void pcxhr_proc_ltc(struct snd_info_entry *entry,
+			   struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	struct pcxhr_rmh rmh;
+	unsigned int ltcHrs, ltcMin, ltcSec, ltcFrm;
+	int err;
+	/* commands available when embedded DSP is running */
+	if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) {
+		snd_iprintf(buffer, "no firmware loaded\n");
+		return;
+	}
+	if (!mgr->capture_ltc) {
+		pcxhr_init_rmh(&rmh, CMD_MANAGE_SIGNAL);
+		rmh.cmd[0] |= MANAGE_SIGNAL_TIME_CODE;
+		err = pcxhr_send_msg(mgr, &rmh);
+		if (err) {
+			snd_iprintf(buffer, "ltc not activated (%d)\n", err);
+			return;
+		}
+		if (mgr->is_hr_stereo)
+			hr222_manage_timecode(mgr, 1);
+		else
+			pcxhr_write_io_num_reg_cont(mgr, REG_CONT_VALSMPTE,
+						    REG_CONT_VALSMPTE, NULL);
+		mgr->capture_ltc = 1;
+	}
+	pcxhr_init_rmh(&rmh, CMD_GET_TIME_CODE);
+	err = pcxhr_send_msg(mgr, &rmh);
+	if (err) {
+		snd_iprintf(buffer, "ltc read error (err=%d)\n", err);
+		return ;
+	}
+	ltcHrs = 10*((rmh.stat[0] >> 8) & 0x3) + (rmh.stat[0] & 0xf);
+	ltcMin = 10*((rmh.stat[1] >> 16) & 0x7) + ((rmh.stat[1] >> 8) & 0xf);
+	ltcSec = 10*(rmh.stat[1] & 0x7) + ((rmh.stat[2] >> 16) & 0xf);
+	ltcFrm = 10*((rmh.stat[2] >> 8) & 0x3) + (rmh.stat[2] & 0xf);
+
+	snd_iprintf(buffer, "timecode: %02u:%02u:%02u-%02u\n",
+			    ltcHrs, ltcMin, ltcSec, ltcFrm);
+	snd_iprintf(buffer, "raw: 0x%04x%06x%06x\n", rmh.stat[0] & 0x00ffff,
+			    rmh.stat[1] & 0xffffff, rmh.stat[2] & 0xffffff);
+	/*snd_iprintf(buffer, "dsp ref time: 0x%06x%06x\n",
+			    rmh.stat[3] & 0xffffff, rmh.stat[4] & 0xffffff);*/
+	if (!(rmh.stat[0] & TIME_CODE_VALID_MASK)) {
+		snd_iprintf(buffer, "warning: linear timecode not valid\n");
+	}
+}
+
 static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 {
 	struct snd_info_entry *entry;
@@ -1383,6 +1444,8 @@
 		entry->c.text.write = pcxhr_proc_gpo_write;
 		entry->mode |= S_IWUSR;
 	}
+	if (!snd_card_proc_new(chip->card, "ltc", &entry))
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_ltc);
 }
 /* end of proc interface */
 
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index bda776c..a4c602c 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -103,6 +103,7 @@
 	unsigned int board_has_mic:1; /* if 1 the board has microphone input */
 	unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */
 	unsigned int mono_capture:1; /* if 1 the board does mono capture */
+	unsigned int capture_ltc:1; /* if 1 the board captures LTC input */
 
 	struct snd_dma_buffer hostport;
 
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 304411c..b33db1e 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -504,6 +504,8 @@
 [CMD_FORMAT_STREAM_IN] =		{ 0x870000, 0, RMH_SSIZE_FIXED },
 [CMD_STREAM_SAMPLE_COUNT] =		{ 0x902000, 2, RMH_SSIZE_FIXED },
 [CMD_AUDIO_LEVEL_ADJUST] =		{ 0xc22000, 0, RMH_SSIZE_FIXED },
+[CMD_GET_TIME_CODE] =			{ 0x060000, 5, RMH_SSIZE_FIXED },
+[CMD_MANAGE_SIGNAL] =			{ 0x0f0000, 0, RMH_SSIZE_FIXED },
 };
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
@@ -533,6 +535,8 @@
 [CMD_FORMAT_STREAM_IN] =		"CMD_FORMAT_STREAM_IN",
 [CMD_STREAM_SAMPLE_COUNT] =		"CMD_STREAM_SAMPLE_COUNT",
 [CMD_AUDIO_LEVEL_ADJUST] =		"CMD_AUDIO_LEVEL_ADJUST",
+[CMD_GET_TIME_CODE] =			"CMD_GET_TIME_CODE",
+[CMD_MANAGE_SIGNAL] =			"CMD_MANAGE_SIGNAL",
 };
 #endif
 
@@ -1133,13 +1137,12 @@
 	hw_sample_count = ((u_int64_t)rmh.stat[0]) << 24;
 	hw_sample_count += (u_int64_t)rmh.stat[1];
 
-	snd_printdd("stream %c%d : abs samples real(%ld) timer(%ld)\n",
+	snd_printdd("stream %c%d : abs samples real(%llu) timer(%llu)\n",
 		    stream->pipe->is_capture ? 'C' : 'P',
 		    stream->substream->number,
-		    (long unsigned int)hw_sample_count,
-		    (long unsigned int)(stream->timer_abs_periods +
-					stream->timer_period_frag +
-					mgr->granularity));
+		    hw_sample_count,
+		    stream->timer_abs_periods + stream->timer_period_frag +
+						mgr->granularity);
 	return hw_sample_count;
 }
 
@@ -1243,10 +1246,18 @@
 
 		if ((dsp_time_diff < 0) &&
 		    (mgr->dsp_time_last != PCXHR_DSP_TIME_INVALID)) {
-			snd_printdd("ERROR DSP TIME old(%d) new(%d) -> "
-				    "resynchronize all streams\n",
+			/* handle dsp counter wraparound without resync */
+			int tmp_diff = dsp_time_diff + PCXHR_DSP_TIME_MASK + 1;
+			snd_printdd("WARNING DSP timestamp old(%d) new(%d)",
 				    mgr->dsp_time_last, dsp_time_new);
-			mgr->dsp_time_err++;
+			if (tmp_diff > 0 && tmp_diff <= (2*mgr->granularity)) {
+				snd_printdd("-> timestamp wraparound OK: "
+					    "diff=%d\n", tmp_diff);
+				dsp_time_diff = tmp_diff;
+			} else {
+				snd_printdd("-> resynchronize all streams\n");
+				mgr->dsp_time_err++;
+			}
 		}
 #ifdef CONFIG_SND_DEBUG_VERBOSE
 		if (dsp_time_diff == 0)
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index be01737..a81ab6b 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -79,6 +79,8 @@
 	CMD_FORMAT_STREAM_IN,		/* cmd_len >= 4	stat_len = 0 */
 	CMD_STREAM_SAMPLE_COUNT,	/* cmd_len = 2	stat_len = (2 * nb_stream) */
 	CMD_AUDIO_LEVEL_ADJUST,		/* cmd_len = 3	stat_len = 0 */
+	CMD_GET_TIME_CODE,		/* cmd_len = 1  stat_len = 5 */
+	CMD_MANAGE_SIGNAL,		/* cmd_len = 1  stat_len = 0 */
 	CMD_LAST_INDEX
 };
 
@@ -116,7 +118,7 @@
 #define IO_NUM_REG_OUT_ANA_LEVEL	20
 #define IO_NUM_REG_IN_ANA_LEVEL		21
 
-
+#define REG_CONT_VALSMPTE		0x000800
 #define REG_CONT_UNMUTE_INPUTS		0x020000
 
 /* parameters used with register IO_NUM_REG_STATUS */
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index 1cb82c0..84fe576 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -53,6 +53,7 @@
 #define PCXHR_DSP_RESET_DSP	0x01
 #define PCXHR_DSP_RESET_MUTE	0x02
 #define PCXHR_DSP_RESET_CODEC	0x08
+#define PCXHR_DSP_RESET_SMPTE	0x10
 #define PCXHR_DSP_RESET_GPO_OFFSET	5
 #define PCXHR_DSP_RESET_GPO_MASK	0x60
 
@@ -527,6 +528,16 @@
 	return 0;
 }
 
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable)
+{
+	if (enable)
+		mgr->dsp_reset |= PCXHR_DSP_RESET_SMPTE;
+	else
+		mgr->dsp_reset &= ~PCXHR_DSP_RESET_SMPTE;
+
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset);
+	return 0;
+}
 
 int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
 				    int is_capture, int channel)
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
index 5a37a00..5971b99 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -34,6 +34,7 @@
 
 int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value);
 int hr222_write_gpo(struct pcxhr_mgr *mgr, int value);
+int hr222_manage_timecode(struct pcxhr_mgr *mgr, int enable);
 
 #define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
 #define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index cbeb3f7..760ee46 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -1151,9 +1151,10 @@
 }
 
 #ifdef CONFIG_PM
-static int riptide_suspend(struct pci_dev *pci, pm_message_t state)
+static int riptide_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_riptide *chip = card->private_data;
 
 	chip->in_suspend = 1;
@@ -1162,13 +1163,14 @@
 	snd_ac97_suspend(chip->ac97);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int riptide_resume(struct pci_dev *pci)
+static int riptide_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_riptide *chip = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -1186,7 +1188,12 @@
 	chip->in_suspend = 0;
 	return 0;
 }
-#endif
+
+static SIMPLE_DEV_PM_OPS(riptide_pm, riptide_suspend, riptide_resume);
+#define RIPTIDE_PM_OPS	&riptide_pm
+#else
+#define RIPTIDE_PM_OPS	NULL
+#endif /* CONFIG_PM */
 
 static int try_to_load_firmware(struct cmdif *cif, struct snd_riptide *chip)
 {
@@ -2180,10 +2187,9 @@
 	.id_table = snd_riptide_ids,
 	.probe = snd_card_riptide_probe,
 	.remove = __devexit_p(snd_card_riptide_remove),
-#ifdef CONFIG_PM
-	.suspend = riptide_suspend,
-	.resume = riptide_resume,
-#endif
+	.driver = {
+		.pm = RIPTIDE_PM_OPS,
+	},
 };
 
 #ifdef SUPPORT_JOYSTICK
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index 1552642..512434e 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1209,9 +1209,10 @@
 }
 
 #ifdef CONFIG_PM
-static int sis_suspend(struct pci_dev *pci, pm_message_t state)
+static int sis_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct sis7019 *sis = card->private_data;
 	void __iomem *ioaddr = sis->ioaddr;
 	int i;
@@ -1241,13 +1242,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int sis_resume(struct pci_dev *pci)
+static int sis_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct sis7019 *sis = card->private_data;
 	void __iomem *ioaddr = sis->ioaddr;
 	int i;
@@ -1298,6 +1300,11 @@
 	snd_card_disconnect(card);
 	return -EIO;
 }
+
+static SIMPLE_DEV_PM_OPS(sis_pm, sis_suspend, sis_resume);
+#define SIS_PM_OPS	&sis_pm
+#else
+#define SIS_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int sis_alloc_suspend(struct sis7019 *sis)
@@ -1481,11 +1488,9 @@
 	.id_table = snd_sis7019_ids,
 	.probe = snd_sis7019_probe,
 	.remove = __devexit_p(snd_sis7019_remove),
-
-#ifdef CONFIG_PM
-	.suspend = sis_suspend,
-	.resume = sis_resume,
-#endif
+	.driver = {
+		.pm = SIS_PM_OPS,
+	},
 };
 
 module_pci_driver(sis7019_driver);
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index 611983e..d36e6ca 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -26,7 +26,7 @@
 #include <linux/time.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/trident.h>
+#include "trident.h"
 #include <sound/initval.h>
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>, <audio@tridentmicro.com>");
@@ -178,8 +178,9 @@
 	.probe = snd_trident_probe,
 	.remove = __devexit_p(snd_trident_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_trident_suspend,
-	.resume = snd_trident_resume,
+	.driver = {
+		.pm = &snd_trident_pm,
+	},
 #endif
 };
 
diff --git a/include/sound/trident.h b/sound/pci/trident/trident.h
similarity index 98%
rename from include/sound/trident.h
rename to sound/pci/trident/trident.h
index 9f191a0..5f110eb 100644
--- a/include/sound/trident.h
+++ b/sound/pci/trident/trident.h
@@ -23,10 +23,10 @@
  *
  */
 
-#include "pcm.h"
-#include "mpu401.h"
-#include "ac97_codec.h"
-#include "util_mem.h"
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+#include <sound/ac97_codec.h>
+#include <sound/util_mem.h>
 
 #define TRIDENT_DEVICE_ID_DX		((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_DX)
 #define TRIDENT_DEVICE_ID_NX		((PCI_VENDOR_ID_TRIDENT<<16)|PCI_DEVICE_ID_TRIDENT_4DWAVE_NX)
@@ -430,8 +430,7 @@
 void snd_trident_start_voice(struct snd_trident * trident, unsigned int voice);
 void snd_trident_stop_voice(struct snd_trident * trident, unsigned int voice);
 void snd_trident_write_voice_regs(struct snd_trident * trident, struct snd_trident_voice *voice);
-int snd_trident_suspend(struct pci_dev *pci, pm_message_t state);
-int snd_trident_resume(struct pci_dev *pci);
+extern const struct dev_pm_ops snd_trident_pm;
 
 /* TLB memory allocation */
 struct snd_util_memblk *snd_trident_alloc_pages(struct snd_trident *trident,
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index 61d3c0e..94011dc 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -41,7 +41,7 @@
 #include <sound/info.h>
 #include <sound/control.h>
 #include <sound/tlv.h>
-#include <sound/trident.h>
+#include "trident.h"
 #include <sound/asoundef.h>
 
 #include <asm/io.h>
@@ -3920,9 +3920,10 @@
 }
 
 #ifdef CONFIG_PM
-int snd_trident_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_trident_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_trident *trident = card->private_data;
 
 	trident->in_suspend = 1;
@@ -3936,13 +3937,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_trident_resume(struct pci_dev *pci)
+static int snd_trident_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_trident *trident = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -3979,4 +3981,6 @@
 	trident->in_suspend = 0;
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_trident_pm, snd_trident_suspend, snd_trident_resume);
 #endif /* CONFIG_PM */
diff --git a/sound/pci/trident/trident_memory.c b/sound/pci/trident/trident_memory.c
index f9779e2..3102a57 100644
--- a/sound/pci/trident/trident_memory.c
+++ b/sound/pci/trident/trident_memory.c
@@ -29,7 +29,7 @@
 #include <linux/mutex.h>
 
 #include <sound/core.h>
-#include <sound/trident.h>
+#include "trident.h"
 
 /* page arguments of these two macros are Trident page (4096 bytes), not like
  * aligned pages in others
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index b5afab4..0eb7245 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -2242,9 +2242,10 @@
 /*
  * power management
  */
-static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_via82xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx *chip = card->private_data;
 	int i;
 
@@ -2265,13 +2266,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_via82xx_resume(struct pci_dev *pci)
+static int snd_via82xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx *chip = card->private_data;
 	int i;
 
@@ -2306,6 +2308,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume);
+#define SND_VIA82XX_PM_OPS	&snd_via82xx_pm
+#else
+#define SND_VIA82XX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_via82xx_free(struct via82xx *chip)
@@ -2624,10 +2631,9 @@
 	.id_table = snd_via82xx_ids,
 	.probe = snd_via82xx_probe,
 	.remove = __devexit_p(snd_via82xx_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_via82xx_suspend,
-	.resume = snd_via82xx_resume,
-#endif
+	.driver = {
+		.pm = SND_VIA82XX_PM_OPS,
+	},
 };
 
 module_pci_driver(via82xx_driver);
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 59fd47e..e886bc1 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -1023,9 +1023,10 @@
 /*
  * power management
  */
-static int snd_via82xx_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_via82xx_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx_modem *chip = card->private_data;
 	int i;
 
@@ -1039,13 +1040,14 @@
 
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-static int snd_via82xx_resume(struct pci_dev *pci)
+static int snd_via82xx_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct via82xx_modem *chip = card->private_data;
 	int i;
 
@@ -1069,6 +1071,11 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_via82xx_pm, snd_via82xx_suspend, snd_via82xx_resume);
+#define SND_VIA82XX_PM_OPS	&snd_via82xx_pm
+#else
+#define SND_VIA82XX_PM_OPS	NULL
 #endif /* CONFIG_PM */
 
 static int snd_via82xx_free(struct via82xx_modem *chip)
@@ -1228,10 +1235,9 @@
 	.id_table = snd_via82xx_modem_ids,
 	.probe = snd_via82xx_probe,
 	.remove = __devexit_p(snd_via82xx_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_via82xx_suspend,
-	.resume = snd_via82xx_resume,
-#endif
+	.driver = {
+		.pm = SND_VIA82XX_PM_OPS,
+	},
 };
 
 module_pci_driver(via82xx_modem_driver);
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index 1ea1f65..b89e7a8 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -258,22 +258,24 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_vx222_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_vx222_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_vx222 *vx = card->private_data;
 	int err;
 
-	err = snd_vx_suspend(&vx->core, state);
+	err = snd_vx_suspend(&vx->core);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return err;
 }
 
-static int snd_vx222_resume(struct pci_dev *pci)
+static int snd_vx222_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_vx222 *vx = card->private_data;
 
 	pci_set_power_state(pci, PCI_D0);
@@ -287,6 +289,11 @@
 	pci_set_master(pci);
 	return snd_vx_resume(&vx->core);
 }
+
+static SIMPLE_DEV_PM_OPS(snd_vx222_pm, snd_vx222_suspend, snd_vx222_resume);
+#define SND_VX222_PM_OPS	&snd_vx222_pm
+#else
+#define SND_VX222_PM_OPS	NULL
 #endif
 
 static struct pci_driver vx222_driver = {
@@ -294,10 +301,9 @@
 	.id_table = snd_vx222_ids,
 	.probe = snd_vx222_probe,
 	.remove = __devexit_p(snd_vx222_remove),
-#ifdef CONFIG_PM
-	.suspend = snd_vx222_suspend,
-	.resume = snd_vx222_resume,
-#endif
+	.driver = {
+		.pm = SND_VX222_PM_OPS,
+	},
 };
 
 module_pci_driver(vx222_driver);
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 9a1d01d..4810356 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -24,7 +24,7 @@
 #include <linux/time.h>
 #include <linux/module.h>
 #include <sound/core.h>
-#include <sound/ymfpci.h>
+#include "ymfpci.h"
 #include <sound/mpu401.h>
 #include <sound/opl3.h>
 #include <sound/initval.h>
@@ -356,8 +356,9 @@
 	.probe = snd_card_ymfpci_probe,
 	.remove = __devexit_p(snd_card_ymfpci_remove),
 #ifdef CONFIG_PM
-	.suspend = snd_ymfpci_suspend,
-	.resume = snd_ymfpci_resume,
+	.driver = {
+		.pm = &snd_ymfpci_pm,
+	},
 #endif
 };
 
diff --git a/include/sound/ymfpci.h b/sound/pci/ymfpci/ymfpci.h
similarity index 97%
rename from include/sound/ymfpci.h
rename to sound/pci/ymfpci/ymfpci.h
index 4119966..bddc405 100644
--- a/include/sound/ymfpci.h
+++ b/sound/pci/ymfpci/ymfpci.h
@@ -22,10 +22,10 @@
  *
  */
 
-#include "pcm.h"
-#include "rawmidi.h"
-#include "ac97_codec.h"
-#include "timer.h"
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+#include <sound/ac97_codec.h>
+#include <sound/timer.h>
 #include <linux/gameport.h>
 
 /*
@@ -377,8 +377,7 @@
 		      struct snd_ymfpci ** rcodec);
 void snd_ymfpci_free_gameport(struct snd_ymfpci *chip);
 
-int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state);
-int snd_ymfpci_resume(struct pci_dev *pci);
+extern const struct dev_pm_ops snd_ymfpci_pm;
 
 int snd_ymfpci_pcm(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
 int snd_ymfpci_pcm2(struct snd_ymfpci *chip, int device, struct snd_pcm **rpcm);
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index a8159b81..62b2363 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -33,7 +33,7 @@
 #include <sound/control.h>
 #include <sound/info.h>
 #include <sound/tlv.h>
-#include <sound/ymfpci.h>
+#include "ymfpci.h"
 #include <sound/asoundef.h>
 #include <sound/mpu401.h>
 
@@ -2302,9 +2302,10 @@
 };
 #define YDSXGR_NUM_SAVED_REGS	ARRAY_SIZE(saved_regs_index)
 
-int snd_ymfpci_suspend(struct pci_dev *pci, pm_message_t state)
+static int snd_ymfpci_suspend(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ymfpci *chip = card->private_data;
 	unsigned int i;
 	
@@ -2326,13 +2327,14 @@
 	snd_ymfpci_disable_dsp(chip);
 	pci_disable_device(pci);
 	pci_save_state(pci);
-	pci_set_power_state(pci, pci_choose_state(pci, state));
+	pci_set_power_state(pci, PCI_D3hot);
 	return 0;
 }
 
-int snd_ymfpci_resume(struct pci_dev *pci)
+static int snd_ymfpci_resume(struct device *dev)
 {
-	struct snd_card *card = pci_get_drvdata(pci);
+	struct pci_dev *pci = to_pci_dev(dev);
+	struct snd_card *card = dev_get_drvdata(dev);
 	struct snd_ymfpci *chip = card->private_data;
 	unsigned int i;
 
@@ -2370,6 +2372,8 @@
 	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
 	return 0;
 }
+
+SIMPLE_DEV_PM_OPS(snd_ymfpci_pm, snd_ymfpci_suspend, snd_ymfpci_resume);
 #endif /* CONFIG_PM */
 
 int __devinit snd_ymfpci_create(struct snd_card *card,
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 830839a..f9b5229 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -251,7 +251,7 @@
 	snd_printdd(KERN_DEBUG "SUSPEND\n");
 	if (chip) {
 		snd_printdd(KERN_DEBUG "snd_pdacf_suspend calling\n");
-		snd_pdacf_suspend(chip, PMSG_SUSPEND);
+		snd_pdacf_suspend(chip);
 	}
 
 	return 0;
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.h b/sound/pcmcia/pdaudiocf/pdaudiocf.h
index 6ce9ad7..ea41e57 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.h
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.h
@@ -131,7 +131,7 @@
 int snd_pdacf_ak4117_create(struct snd_pdacf *pdacf);
 void snd_pdacf_powerdown(struct snd_pdacf *chip);
 #ifdef CONFIG_PM
-int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state);
+int snd_pdacf_suspend(struct snd_pdacf *chip);
 int snd_pdacf_resume(struct snd_pdacf *chip);
 #endif
 int snd_pdacf_pcm_new(struct snd_pdacf *chip);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
index 9dce0bd..ea0adfb 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
@@ -262,7 +262,7 @@
 
 #ifdef CONFIG_PM
 
-int snd_pdacf_suspend(struct snd_pdacf *chip, pm_message_t state)
+int snd_pdacf_suspend(struct snd_pdacf *chip)
 {
 	u16 val;
 	
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 512f0b4..8f93504 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -260,7 +260,7 @@
 	snd_printdd(KERN_DEBUG "SUSPEND\n");
 	if (chip) {
 		snd_printdd(KERN_DEBUG "snd_vx_suspend calling\n");
-		snd_vx_suspend(chip, PMSG_SUSPEND);
+		snd_vx_suspend(chip);
 	}
 
 	return 0;
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index 5a4e263..f5ceb6f 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -144,19 +144,24 @@
 }
 
 #ifdef CONFIG_PM
-static int snd_pmac_driver_suspend(struct platform_device *devptr, pm_message_t state)
+static int snd_pmac_driver_suspend(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(devptr);
+	struct snd_card *card = dev_get_drvdata(dev);
 	snd_pmac_suspend(card->private_data);
 	return 0;
 }
 
-static int snd_pmac_driver_resume(struct platform_device *devptr)
+static int snd_pmac_driver_resume(struct device *dev)
 {
-	struct snd_card *card = platform_get_drvdata(devptr);
+	struct snd_card *card = dev_get_drvdata(dev);
 	snd_pmac_resume(card->private_data);
 	return 0;
 }
+
+static SIMPLE_DEV_PM_OPS(snd_pmac_pm, snd_pmac_driver_suspend, snd_pmac_driver_resume);
+#define SND_PMAC_PM_OPS	&snd_pmac_pm
+#else
+#define SND_PMAC_PM_OPS	NULL
 #endif
 
 #define SND_PMAC_DRIVER		"snd_powermac"
@@ -164,12 +169,10 @@
 static struct platform_driver snd_pmac_driver = {
 	.probe		= snd_pmac_probe,
 	.remove		= __devexit_p(snd_pmac_remove),
-#ifdef CONFIG_PM
-	.suspend	= snd_pmac_driver_suspend,
-	.resume		= snd_pmac_driver_resume,
-#endif
 	.driver		= {
-		.name	= SND_PMAC_DRIVER
+		.name	= SND_PMAC_DRIVER,
+		.owner	= THIS_MODULE,
+		.pm	= SND_PMAC_PM_OPS,
 	},
 };
 
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 391a38c..d48b523 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -654,7 +654,9 @@
 	.probe = snd_aica_probe,
 	.remove = __devexit_p(snd_aica_remove),
 	.driver = {
-		   .name = SND_AICA_DRIVER},
+		.name = SND_AICA_DRIVER,
+		.owner	= THIS_MODULE,
+	},
 };
 
 static int __init aica_init(void)
diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c
index f8b01c7..0a33947 100644
--- a/sound/sh/sh_dac_audio.c
+++ b/sound/sh/sh_dac_audio.c
@@ -438,6 +438,7 @@
 	.remove = snd_sh_dac_remove,
 	.driver = {
 		.name = "dac_audio",
+		.owner	= THIS_MODULE,
 	},
 };
 
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 40b2ad1..c5de0a8 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -33,6 +33,7 @@
 source "sound/soc/au1x/Kconfig"
 source "sound/soc/blackfin/Kconfig"
 source "sound/soc/davinci/Kconfig"
+source "sound/soc/dwc/Kconfig"
 source "sound/soc/ep93xx/Kconfig"
 source "sound/soc/fsl/Kconfig"
 source "sound/soc/jz4740/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 70990f4..00a555a 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -11,6 +11,7 @@
 obj-$(CONFIG_SND_SOC)	+= au1x/
 obj-$(CONFIG_SND_SOC)	+= blackfin/
 obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= dwc/
 obj-$(CONFIG_SND_SOC)	+= ep93xx/
 obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= jz4740/
diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index 9f6bc55..16b88f5 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -1,7 +1,8 @@
 config SND_BF5XX_I2S
-	tristate "SoC I2S Audio for the ADI BF5xx chip"
+	tristate "SoC I2S Audio for the ADI Blackfin chip"
 	depends on BLACKFIN
-	select SND_BF5XX_SOC_SPORT
+	select SND_BF5XX_SOC_SPORT if !BF60x
+	select SND_BF6XX_SOC_SPORT if BF60x
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -9,12 +10,14 @@
 	  You will also need to select the audio interfaces to support below.
 
 config SND_BF5XX_SOC_SSM2602
-	tristate "SoC SSM2602 Audio support for BF52x ezkit"
+	tristate "SoC SSM2602 Audio Codec Add-On Card support"
 	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
-	select SND_BF5XX_SOC_I2S
+	select SND_BF5XX_SOC_I2S if !BF60x
+	select SND_BF6XX_SOC_I2S if BF60x
 	select SND_SOC_SSM2602
 	help
-	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
+	  Say Y if you want to add support for the Analog Devices
+	  SSM2602 Audio Codec Add-On Card.
 
 config SND_SOC_BFIN_EVAL_ADAU1701
 	tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards"
@@ -162,9 +165,15 @@
 config SND_BF5XX_SOC_SPORT
 	tristate
 
+config SND_BF6XX_SOC_SPORT
+	tristate
+
 config SND_BF5XX_SOC_I2S
 	tristate
 
+config SND_BF6XX_SOC_I2S
+	tristate
+
 config SND_BF5XX_SOC_TDM
 	tristate
 
@@ -173,7 +182,7 @@
 
 config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
-	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM)
+	depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT)
 	range 0 3 if BF54x
 	range 0 1 if !BF54x
 	default 0
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index 1bf86cc..6fea1f4 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -3,16 +3,20 @@
 snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o
 snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o
 snd-soc-bf5xx-sport-objs := bf5xx-sport.o
+snd-soc-bf6xx-sport-objs := bf6xx-sport.o
 snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o
 snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o
+snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o
 snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o
 
 obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o
 obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o
 obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o
 obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o
+obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o
 obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o
 obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
+obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o
 obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o
 
 # Blackfin Machine Support
diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c
new file mode 100644
index 0000000..c3c2466
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-i2s.c
@@ -0,0 +1,234 @@
+/*
+ * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+
+#include "bf6xx-sport.h"
+
+struct sport_params param;
+
+static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+		unsigned int fmt)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret = 0;
+
+	param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR
+			| SPORT_CTL_LFS | SPORT_CTL_LAFS);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE
+			| SPORT_CTL_LFS;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		param.spctl |= SPORT_CTL_FSR;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS
+			| SPORT_CTL_LAFS;
+		break;
+	default:
+		dev_err(dev, "%s: Unknown DAI format type\n", __func__);
+		ret = -EINVAL;
+		break;
+	}
+
+	param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS);
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+	case SND_SOC_DAIFMT_CBM_CFS:
+	case SND_SOC_DAIFMT_CBS_CFM:
+		ret = -EINVAL;
+		break;
+	default:
+		dev_err(dev, "%s: Unknown DAI master type\n", __func__);
+		ret = -EINVAL;
+		break;
+	}
+
+	return ret;
+}
+
+static int bfin_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret = 0;
+
+	param.spctl &= ~SPORT_CTL_SLEN;
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		param.spctl |= 0x70;
+		sport->wdsize = 1;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		param.spctl |= 0xf0;
+		sport->wdsize = 2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		param.spctl |= 0x170;
+		sport->wdsize = 3;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		param.spctl |= 0x1f0;
+		sport->wdsize = 4;
+		break;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ret = sport_set_tx_params(sport, &param);
+		if (ret) {
+			dev_err(dev, "SPORT tx is busy!\n");
+			return ret;
+		}
+	} else {
+		ret = sport_set_rx_params(sport, &param);
+		if (ret) {
+			dev_err(dev, "SPORT rx is busy!\n");
+			return ret;
+		}
+	}
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int bfin_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+
+	if (dai->capture_active)
+		sport_rx_stop(sport);
+	if (dai->playback_active)
+		sport_tx_stop(sport);
+	return 0;
+}
+
+static int bfin_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct sport_device *sport = snd_soc_dai_get_drvdata(dai);
+	struct device *dev = &sport->pdev->dev;
+	int ret;
+
+	ret = sport_set_tx_params(sport, &param);
+	if (ret) {
+		dev_err(dev, "SPORT tx is busy!\n");
+		return ret;
+	}
+	ret = sport_set_rx_params(sport, &param);
+	if (ret) {
+		dev_err(dev, "SPORT rx is busy!\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+#else
+#define bfin_i2s_suspend NULL
+#define bfin_i2s_resume NULL
+#endif
+
+#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+		SNDRV_PCM_RATE_96000)
+
+#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops bfin_i2s_dai_ops = {
+	.hw_params	= bfin_i2s_hw_params,
+	.set_fmt	= bfin_i2s_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver bfin_i2s_dai = {
+	.suspend = bfin_i2s_suspend,
+	.resume = bfin_i2s_resume,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = BFIN_I2S_RATES,
+		.formats = BFIN_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = BFIN_I2S_RATES,
+		.formats = BFIN_I2S_FORMATS,
+	},
+	.ops = &bfin_i2s_dai_ops,
+};
+
+static int __devinit bfin_i2s_probe(struct platform_device *pdev)
+{
+	struct sport_device *sport;
+	struct device *dev = &pdev->dev;
+	int ret;
+
+	sport = sport_create(pdev);
+	if (!sport)
+		return -ENODEV;
+
+	/* register with the ASoC layers */
+	ret = snd_soc_register_dai(dev, &bfin_i2s_dai);
+	if (ret) {
+		dev_err(dev, "Failed to register DAI: %d\n", ret);
+		sport_delete(sport);
+		return ret;
+	}
+	platform_set_drvdata(pdev, sport);
+
+	return 0;
+}
+
+static int __devexit bfin_i2s_remove(struct platform_device *pdev)
+{
+	struct sport_device *sport = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	sport_delete(sport);
+
+	return 0;
+}
+
+static struct platform_driver bfin_i2s_driver = {
+	.probe  = bfin_i2s_probe,
+	.remove = __devexit_p(bfin_i2s_remove),
+	.driver = {
+		.name = "bfin-i2s",
+		.owner = THIS_MODULE,
+	},
+};
+
+module_platform_driver(bfin_i2s_driver);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c
new file mode 100644
index 0000000..318c5ba5
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.c
@@ -0,0 +1,422 @@
+/*
+ * bf6xx_sport.c Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#include <linux/device.h>
+#include <linux/dma-mapping.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+
+#include <asm/blackfin.h>
+#include <asm/dma.h>
+#include <asm/portmux.h>
+
+#include "bf6xx-sport.h"
+
+int sport_set_tx_params(struct sport_device *sport,
+			struct sport_params *params)
+{
+	if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI)
+		return -EBUSY;
+	sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN;
+	sport->tx_regs->div = params->div;
+	SSYNC();
+	return 0;
+}
+EXPORT_SYMBOL(sport_set_tx_params);
+
+int sport_set_rx_params(struct sport_device *sport,
+			struct sport_params *params)
+{
+	if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI)
+		return -EBUSY;
+	sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN;
+	sport->rx_regs->div = params->div;
+	SSYNC();
+	return 0;
+}
+EXPORT_SYMBOL(sport_set_rx_params);
+
+static int compute_wdsize(size_t wdsize)
+{
+	switch (wdsize) {
+	case 1:
+		return WDSIZE_8 | PSIZE_8;
+	case 2:
+		return WDSIZE_16 | PSIZE_16;
+	default:
+		return WDSIZE_32 | PSIZE_32;
+	}
+}
+
+void sport_tx_start(struct sport_device *sport)
+{
+	set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc);
+	set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN
+			| compute_wdsize(sport->wdsize) | NDSIZE_6);
+	enable_dma(sport->tx_dma_chan);
+	sport->tx_regs->spctl |= SPORT_CTL_SPENPRI;
+	SSYNC();
+}
+EXPORT_SYMBOL(sport_tx_start);
+
+void sport_rx_start(struct sport_device *sport)
+{
+	set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc);
+	set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR
+			| compute_wdsize(sport->wdsize) | NDSIZE_6);
+	enable_dma(sport->rx_dma_chan);
+	sport->rx_regs->spctl |= SPORT_CTL_SPENPRI;
+	SSYNC();
+}
+EXPORT_SYMBOL(sport_rx_start);
+
+void sport_tx_stop(struct sport_device *sport)
+{
+	sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+	SSYNC();
+	disable_dma(sport->tx_dma_chan);
+}
+EXPORT_SYMBOL(sport_tx_stop);
+
+void sport_rx_stop(struct sport_device *sport)
+{
+	sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI;
+	SSYNC();
+	disable_dma(sport->rx_dma_chan);
+}
+EXPORT_SYMBOL(sport_rx_stop);
+
+void sport_set_tx_callback(struct sport_device *sport,
+		void (*tx_callback)(void *), void *tx_data)
+{
+	sport->tx_callback = tx_callback;
+	sport->tx_data = tx_data;
+}
+EXPORT_SYMBOL(sport_set_tx_callback);
+
+void sport_set_rx_callback(struct sport_device *sport,
+		void (*rx_callback)(void *), void *rx_data)
+{
+	sport->rx_callback = rx_callback;
+	sport->rx_data = rx_data;
+}
+EXPORT_SYMBOL(sport_set_rx_callback);
+
+static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
+		size_t fragsize, unsigned int cfg,
+		unsigned int count, size_t wdsize)
+{
+
+	int i;
+
+	for (i = 0; i < fragcount; ++i) {
+		desc[i].next_desc_addr  = &(desc[i + 1]);
+		desc[i].start_addr = (unsigned long)buf + i*fragsize;
+		desc[i].cfg = cfg;
+		desc[i].x_count = count;
+		desc[i].x_modify = wdsize;
+		desc[i].y_count = 0;
+		desc[i].y_modify = 0;
+	}
+
+	/* make circular */
+	desc[fragcount-1].next_desc_addr = desc;
+}
+
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+		int fragcount, size_t fragsize)
+{
+	unsigned int count;
+	unsigned int cfg;
+	dma_addr_t addr;
+
+	count = fragsize/sport->wdsize;
+
+	if (sport->tx_desc)
+		dma_free_coherent(NULL, sport->tx_desc_size,
+				sport->tx_desc, 0);
+
+	sport->tx_desc = dma_alloc_coherent(NULL,
+			fragcount * sizeof(struct dmasg), &addr, 0);
+	sport->tx_desc_size = fragcount * sizeof(struct dmasg);
+	if (!sport->tx_desc)
+		return -ENOMEM;
+
+	sport->tx_buf = buf;
+	sport->tx_fragsize = fragsize;
+	sport->tx_frags = fragcount;
+	cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6;
+
+	setup_desc(sport->tx_desc, buf, fragcount, fragsize,
+			cfg|DMAEN, count, sport->wdsize);
+
+	return 0;
+}
+EXPORT_SYMBOL(sport_config_tx_dma);
+
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+		int fragcount, size_t fragsize)
+{
+	unsigned int count;
+	unsigned int cfg;
+	dma_addr_t addr;
+
+	count = fragsize/sport->wdsize;
+
+	if (sport->rx_desc)
+		dma_free_coherent(NULL, sport->rx_desc_size,
+				sport->rx_desc, 0);
+
+	sport->rx_desc = dma_alloc_coherent(NULL,
+			fragcount * sizeof(struct dmasg), &addr, 0);
+	sport->rx_desc_size = fragcount * sizeof(struct dmasg);
+	if (!sport->rx_desc)
+		return -ENOMEM;
+
+	sport->rx_buf = buf;
+	sport->rx_fragsize = fragsize;
+	sport->rx_frags = fragcount;
+	cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize)
+		| WNR | NDSIZE_6;
+
+	setup_desc(sport->rx_desc, buf, fragcount, fragsize,
+			cfg|DMAEN, count, sport->wdsize);
+
+	return 0;
+}
+EXPORT_SYMBOL(sport_config_rx_dma);
+
+unsigned long sport_curr_offset_tx(struct sport_device *sport)
+{
+	unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan);
+
+	return (unsigned char *)curr - sport->tx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_tx);
+
+unsigned long sport_curr_offset_rx(struct sport_device *sport)
+{
+	unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan);
+
+	return (unsigned char *)curr - sport->rx_buf;
+}
+EXPORT_SYMBOL(sport_curr_offset_rx);
+
+static irqreturn_t sport_tx_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	static unsigned long status;
+
+	status = get_dma_curr_irqstat(sport->tx_dma_chan);
+	if (status & (DMA_DONE|DMA_ERR)) {
+		clear_dma_irqstat(sport->tx_dma_chan);
+		SSYNC();
+	}
+	if (sport->tx_callback)
+		sport->tx_callback(sport->tx_data);
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_rx_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	unsigned long status;
+
+	status = get_dma_curr_irqstat(sport->rx_dma_chan);
+	if (status & (DMA_DONE|DMA_ERR)) {
+		clear_dma_irqstat(sport->rx_dma_chan);
+		SSYNC();
+	}
+	if (sport->rx_callback)
+		sport->rx_callback(sport->rx_data);
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t sport_err_irq(int irq, void *dev_id)
+{
+	struct sport_device *sport = dev_id;
+	struct device *dev = &sport->pdev->dev;
+
+	if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI)
+		dev_err(dev, "sport error: TUVF\n");
+	if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI)
+		dev_err(dev, "sport error: ROVF\n");
+
+	return IRQ_HANDLED;
+}
+
+static int sport_get_resource(struct sport_device *sport)
+{
+	struct platform_device *pdev = sport->pdev;
+	struct device *dev = &pdev->dev;
+	struct bfin_snd_platform_data *pdata = dev->platform_data;
+	struct resource *res;
+
+	if (!pdata) {
+		dev_err(dev, "No platform data\n");
+		return -ENODEV;
+	}
+	sport->pin_req = pdata->pin_req;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(dev, "No tx MEM resource\n");
+		return -ENODEV;
+	}
+	sport->tx_regs = (struct sport_register *)res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 1);
+	if (!res) {
+		dev_err(dev, "No rx MEM resource\n");
+		return -ENODEV;
+	}
+	sport->rx_regs = (struct sport_register *)res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!res) {
+		dev_err(dev, "No tx DMA resource\n");
+		return -ENODEV;
+	}
+	sport->tx_dma_chan = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (!res) {
+		dev_err(dev, "No rx DMA resource\n");
+		return -ENODEV;
+	}
+	sport->rx_dma_chan = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_IRQ, 0);
+	if (!res) {
+		dev_err(dev, "No tx error irq resource\n");
+		return -ENODEV;
+	}
+	sport->tx_err_irq = res->start;
+
+	res = platform_get_resource(pdev, IORESOURCE_IRQ, 1);
+	if (!res) {
+		dev_err(dev, "No rx error irq resource\n");
+		return -ENODEV;
+	}
+	sport->rx_err_irq = res->start;
+
+	return 0;
+}
+
+static int sport_request_resource(struct sport_device *sport)
+{
+	struct device *dev = &sport->pdev->dev;
+	int ret;
+
+	ret = peripheral_request_list(sport->pin_req, "soc-audio");
+	if (ret) {
+		dev_err(dev, "Unable to request sport pin\n");
+		return ret;
+	}
+
+	ret = request_dma(sport->tx_dma_chan, "SPORT TX Data");
+	if (ret) {
+		dev_err(dev, "Unable to allocate DMA channel for sport tx\n");
+		goto err_tx_dma;
+	}
+	set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport);
+
+	ret = request_dma(sport->rx_dma_chan, "SPORT RX Data");
+	if (ret) {
+		dev_err(dev, "Unable to allocate DMA channel for sport rx\n");
+		goto err_rx_dma;
+	}
+	set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport);
+
+	ret = request_irq(sport->tx_err_irq, sport_err_irq,
+			0, "SPORT TX ERROR", sport);
+	if (ret) {
+		dev_err(dev, "Unable to allocate tx error IRQ for sport\n");
+		goto err_tx_irq;
+	}
+
+	ret = request_irq(sport->rx_err_irq, sport_err_irq,
+			0, "SPORT RX ERROR", sport);
+	if (ret) {
+		dev_err(dev, "Unable to allocate rx error IRQ for sport\n");
+		goto err_rx_irq;
+	}
+
+	return 0;
+err_rx_irq:
+	free_irq(sport->tx_err_irq, sport);
+err_tx_irq:
+	free_dma(sport->rx_dma_chan);
+err_rx_dma:
+	free_dma(sport->tx_dma_chan);
+err_tx_dma:
+	peripheral_free_list(sport->pin_req);
+	return ret;
+}
+
+static void sport_free_resource(struct sport_device *sport)
+{
+	free_irq(sport->rx_err_irq, sport);
+	free_irq(sport->tx_err_irq, sport);
+	free_dma(sport->rx_dma_chan);
+	free_dma(sport->tx_dma_chan);
+	peripheral_free_list(sport->pin_req);
+}
+
+struct sport_device *sport_create(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct sport_device *sport;
+	int ret;
+
+	sport = kzalloc(sizeof(*sport), GFP_KERNEL);
+	if (!sport) {
+		dev_err(dev, "Unable to allocate memory for sport device\n");
+		return NULL;
+	}
+	sport->pdev = pdev;
+
+	ret = sport_get_resource(sport);
+	if (ret) {
+		kfree(sport);
+		return NULL;
+	}
+
+	ret = sport_request_resource(sport);
+	if (ret) {
+		kfree(sport);
+		return NULL;
+	}
+
+	dev_dbg(dev, "SPORT create success\n");
+	return sport;
+}
+EXPORT_SYMBOL(sport_create);
+
+void sport_delete(struct sport_device *sport)
+{
+	sport_free_resource(sport);
+}
+EXPORT_SYMBOL(sport_delete);
+
+MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver");
+MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux@gmail.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h
new file mode 100644
index 0000000..307d193
--- /dev/null
+++ b/sound/soc/blackfin/bf6xx-sport.h
@@ -0,0 +1,82 @@
+/*
+ * bf6xx_sport - Analog Devices BF6XX SPORT driver
+ *
+ * Copyright (c) 2012 Analog Devices Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+#ifndef _BF6XX_SPORT_H_
+#define _BF6XX_SPORT_H_
+
+#include <linux/platform_device.h>
+#include <asm/bfin_sport3.h>
+
+struct sport_device {
+	struct platform_device *pdev;
+	const unsigned short *pin_req;
+	struct sport_register *tx_regs;
+	struct sport_register *rx_regs;
+	int tx_dma_chan;
+	int rx_dma_chan;
+	int tx_err_irq;
+	int rx_err_irq;
+
+	void (*tx_callback)(void *data);
+	void *tx_data;
+	void (*rx_callback)(void *data);
+	void *rx_data;
+
+	struct dmasg *tx_desc;
+	struct dmasg *rx_desc;
+	unsigned int tx_desc_size;
+	unsigned int rx_desc_size;
+	unsigned char *tx_buf;
+	unsigned char *rx_buf;
+	unsigned int tx_fragsize;
+	unsigned int rx_fragsize;
+	unsigned int tx_frags;
+	unsigned int rx_frags;
+	unsigned int wdsize;
+};
+
+struct sport_params {
+	u32 spctl;
+	u32 div;
+};
+
+struct sport_device *sport_create(struct platform_device *pdev);
+void sport_delete(struct sport_device *sport);
+int sport_set_tx_params(struct sport_device *sport,
+		struct sport_params *params);
+int sport_set_rx_params(struct sport_device *sport,
+		struct sport_params *params);
+void sport_tx_start(struct sport_device *sport);
+void sport_rx_start(struct sport_device *sport);
+void sport_tx_stop(struct sport_device *sport);
+void sport_rx_stop(struct sport_device *sport);
+void sport_set_tx_callback(struct sport_device *sport,
+	void (*tx_callback)(void *), void *tx_data);
+void sport_set_rx_callback(struct sport_device *sport,
+	void (*rx_callback)(void *), void *rx_data);
+int sport_config_tx_dma(struct sport_device *sport, void *buf,
+	int fragcount, size_t fragsize);
+int sport_config_rx_dma(struct sport_device *sport, void *buf,
+	int fragcount, size_t fragsize);
+unsigned long sport_curr_offset_tx(struct sport_device *sport);
+unsigned long sport_curr_offset_rx(struct sport_device *sport);
+
+
+
+#endif
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 1e1613a..9f8e859 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -12,6 +12,7 @@
 	tristate "Build all ASoC CODEC drivers"
 	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
+	select SND_SOC_AB8500_CODEC if ABX500_CORE
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
@@ -35,7 +36,9 @@
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
+	select SND_SOC_DA732X if I2C
 	select SND_SOC_DFBMCS320
+	select SND_SOC_ISABELLE if I2C
 	select SND_SOC_JZ4740_CODEC
 	select SND_SOC_LM4857 if I2C
 	select SND_SOC_LM49453 if I2C
@@ -54,6 +57,7 @@
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_STA32X if I2C
+	select SND_SOC_STA529 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
@@ -70,6 +74,8 @@
 	select SND_SOC_WM2000 if I2C
 	select SND_SOC_WM2200 if I2C
 	select SND_SOC_WM5100 if I2C
+	select SND_SOC_WM5102 if MFD_WM5102
+	select SND_SOC_WM5110 if MFD_WM5110
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
@@ -126,11 +132,21 @@
 config SND_SOC_88PM860X
 	tristate
 
+config SND_SOC_ARIZONA
+	tristate
+	default y if SND_SOC_WM5102=y
+	default y if SND_SOC_WM5110=y
+	default m if SND_SOC_WM5102=m
+	default m if SND_SOC_WM5110=m
+
 config SND_SOC_WM_HUBS
 	tristate
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
 	default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m
 
+config SND_SOC_AB8500_CODEC
+	tristate
+
 config SND_SOC_AC97_CODEC
 	tristate
 	select SND_AC97_CODEC
@@ -219,12 +235,18 @@
 config SND_SOC_DA7210
         tristate
 
+config SND_SOC_DA732X
+        tristate
+
 config SND_SOC_DFBMCS320
 	tristate
 
 config SND_SOC_DMIC
 	tristate
 
+config SND_SOC_ISABELLE
+        tristate
+
 config SND_SOC_LM49453
 	tristate
 
@@ -266,6 +288,9 @@
 config SND_SOC_STA32X
 	tristate
 
+config SND_SOC_STA529
+	tristate
+
 config SND_SOC_STAC9766
 	tristate
 
@@ -313,6 +338,12 @@
 config SND_SOC_WM5100
 	tristate
 
+config SND_SOC_WM5102
+	tristate
+
+config SND_SOC_WM5110
+	tristate
+
 config SND_SOC_WM8350
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index fc27fec..34148bb 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,4 +1,5 @@
 snd-soc-88pm860x-objs := 88pm860x-codec.o
+snd-soc-ab8500-codec-objs := ab8500-codec.o
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
@@ -13,6 +14,7 @@
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
+snd-soc-arizona-objs := arizona.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
 snd-soc-cs42l52-objs := cs42l52.o
@@ -21,8 +23,10 @@
 snd-soc-cs4271-objs := cs4271.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
+snd-soc-da732x-objs := da732x.o
 snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
+snd-soc-isabelle-objs := isabelle.o
 snd-soc-jz4740-codec-objs := jz4740.o
 snd-soc-l3-objs := l3.o
 snd-soc-lm4857-objs := lm4857.o
@@ -41,9 +45,11 @@
 snd-soc-alc5632-objs := alc5632.o
 snd-soc-sigmadsp-objs := sigmadsp.o
 snd-soc-sn95031-objs := sn95031.o
-snd-soc-spdif-objs := spdif_transciever.o
+snd-soc-spdif-tx-objs := spdif_transciever.o
+snd-soc-spdif-rx-objs := spdif_receiver.o
 snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
+snd-soc-sta529-objs := sta529.o
 snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
@@ -59,6 +65,8 @@
 snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm2200-objs := wm2200.o
 snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
+snd-soc-wm5102-objs := wm5102.o
+snd-soc-wm5110-objs := wm5110.o
 snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
@@ -108,6 +116,7 @@
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 
 obj-$(CONFIG_SND_SOC_88PM860X)	+= snd-soc-88pm860x.o
+obj-$(CONFIG_SND_SOC_AB8500_CODEC)	+= snd-soc-ab8500-codec.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
@@ -124,6 +133,7 @@
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
+obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS42L52)	+= snd-soc-cs42l52.o
@@ -132,8 +142,10 @@
 obj-$(CONFIG_SND_SOC_CS4271)	+= snd-soc-cs4271.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
 obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
+obj-$(CONFIG_SND_SOC_DA732X)	+= snd-soc-da732x.o
 obj-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
+obj-$(CONFIG_SND_SOC_ISABELLE)	+= snd-soc-isabelle.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
@@ -150,9 +162,10 @@
 obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SIGMADSP)	+= snd-soc-sigmadsp.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
-obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
+obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif-rx.o snd-soc-spdif-tx.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
+obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
@@ -168,6 +181,8 @@
 obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
 obj-$(CONFIG_SND_SOC_WM2200)	+= snd-soc-wm2200.o
 obj-$(CONFIG_SND_SOC_WM5100)	+= snd-soc-wm5100.o
+obj-$(CONFIG_SND_SOC_WM5102)	+= snd-soc-wm5102.o
+obj-$(CONFIG_SND_SOC_WM5110)	+= snd-soc-wm5110.o
 obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c
new file mode 100644
index 0000000..3c79592
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.c
@@ -0,0 +1,2522 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ *         for ST-Ericsson.
+ *
+ *         Based on the early work done by:
+ *         Mikko J. Lehto <mikko.lehto@symbio.com>,
+ *         Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ *         Jarmo K. Kuronen <jarmo.kuronen@symbio.com>,
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <linux/mfd/abx500/ab8500.h>
+#include <linux/mfd/abx500.h>
+#include <linux/mfd/abx500/ab8500-sysctrl.h>
+#include <linux/mfd/abx500/ab8500-codec.h>
+#include <linux/regulator/consumer.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+#include "ab8500-codec.h"
+
+/* Macrocell value definitions */
+#define CLK_32K_OUT2_DISABLE			0x01
+#define INACTIVE_RESET_AUDIO			0x02
+#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK		0x10
+#define ENABLE_VINTCORE12_SUPPLY		0x04
+#define GPIO27_DIR_OUTPUT			0x04
+#define GPIO29_DIR_OUTPUT			0x10
+#define GPIO31_DIR_OUTPUT			0x40
+
+/* Macrocell register definitions */
+#define AB8500_CTRL3_REG			0x0200
+#define AB8500_GPIO_DIR4_REG			0x1013
+
+/* Nr of FIR/IIR-coeff banks in ANC-block */
+#define AB8500_NR_OF_ANC_COEFF_BANKS		2
+
+/* Minimum duration to keep ANC IIR Init bit high or
+low before proceeding with the configuration sequence */
+#define AB8500_ANC_SM_DELAY			2000
+
+#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.info = filter_control_info, \
+	.get = filter_control_get, .put = filter_control_put, \
+	.private_value = (unsigned long)&(struct filter_control) \
+		{.count = xcount, .min = xmin, .max = xmax} }
+
+struct filter_control {
+	long min, max;
+	unsigned int count;
+	long value[128];
+};
+
+/* Sidetone states */
+static const char * const enum_sid_state[] = {
+	"Unconfigured",
+	"Apply FIR",
+	"FIR is configured",
+};
+enum sid_state {
+	SID_UNCONFIGURED = 0,
+	SID_APPLY_FIR = 1,
+	SID_FIR_CONFIGURED = 2,
+};
+
+static const char * const enum_anc_state[] = {
+	"Unconfigured",
+	"Apply FIR and IIR",
+	"FIR and IIR are configured",
+	"Apply FIR",
+	"FIR is configured",
+	"Apply IIR",
+	"IIR is configured"
+};
+enum anc_state {
+	ANC_UNCONFIGURED = 0,
+	ANC_APPLY_FIR_IIR = 1,
+	ANC_FIR_IIR_CONFIGURED = 2,
+	ANC_APPLY_FIR = 3,
+	ANC_FIR_CONFIGURED = 4,
+	ANC_APPLY_IIR = 5,
+	ANC_IIR_CONFIGURED = 6
+};
+
+/* Analog microphones */
+enum amic_idx {
+	AMIC_IDX_1A,
+	AMIC_IDX_1B,
+	AMIC_IDX_2
+};
+
+struct ab8500_codec_drvdata_dbg {
+	struct regulator *vaud;
+	struct regulator *vamic1;
+	struct regulator *vamic2;
+	struct regulator *vdmic;
+};
+
+/* Private data for AB8500 device-driver */
+struct ab8500_codec_drvdata {
+	/* Sidetone */
+	long *sid_fir_values;
+	enum sid_state sid_status;
+
+	/* ANC */
+	struct mutex anc_lock;
+	long *anc_fir_values;
+	long *anc_iir_values;
+	enum anc_state anc_status;
+};
+
+static inline const char *amic_micbias_str(enum amic_micbias micbias)
+{
+	switch (micbias) {
+	case AMIC_MICBIAS_VAMIC1:
+		return "VAMIC1";
+	case AMIC_MICBIAS_VAMIC2:
+		return "VAMIC2";
+	default:
+		return "Unknown";
+	}
+}
+
+static inline const char *amic_type_str(enum amic_type type)
+{
+	switch (type) {
+	case AMIC_TYPE_DIFFERENTIAL:
+		return "DIFFERENTIAL";
+	case AMIC_TYPE_SINGLE_ENDED:
+		return "SINGLE ENDED";
+	default:
+		return "Unknown";
+	}
+}
+
+/*
+ * Read'n'write functions
+ */
+
+/* Read a register from the audio-bank of AB8500 */
+static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	int status;
+	unsigned int value = 0;
+
+	u8 value8;
+	status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO,
+						reg, &value8);
+	if (status < 0) {
+		dev_err(codec->dev,
+			"%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n",
+			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
+	} else {
+		dev_dbg(codec->dev,
+			"%s: Read 0x%02x from register 0x%02x:0x%02x\n",
+			__func__, value8, (u8)AB8500_AUDIO, (u8)reg);
+		value = (unsigned int)value8;
+	}
+
+	return value;
+}
+
+/* Write to a register in the audio-bank of AB8500 */
+static int ab8500_codec_write_reg(struct snd_soc_codec *codec,
+				unsigned int reg, unsigned int value)
+{
+	int status;
+
+	status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO,
+						reg, value);
+	if (status < 0)
+		dev_err(codec->dev,
+			"%s: ERROR: Register (%02x:%02x) write failed (%d).\n",
+			__func__, (u8)AB8500_AUDIO, (u8)reg, status);
+	else
+		dev_dbg(codec->dev,
+			"%s: Wrote 0x%02x into register %02x:%02x\n",
+			__func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg);
+
+	return status;
+}
+
+/*
+ * Controls - DAPM
+ */
+
+/* Earpiece */
+
+/* Earpiece source selector */
+static const char * const enum_ear_lineout_source[] = {"Headset Left",
+						"Speaker Left"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source);
+static const struct snd_kcontrol_new dapm_ear_lineout_source =
+	SOC_DAPM_ENUM("Earpiece or LineOut Mono Source",
+		dapm_enum_ear_lineout_source);
+
+/* LineOut */
+
+/* LineOut source selector */
+static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"};
+static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5,
+			AB8500_ANACONF5_HSLDACTOLOL,
+			AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source);
+static const struct snd_kcontrol_new dapm_lineout_source[] = {
+	SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source),
+};
+
+/* Handsfree */
+
+/* Speaker Left - ANC selector */
+static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFl_select[] = {
+	SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel),
+};
+
+/* Speaker Right - ANC selector */
+static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel);
+static const struct snd_kcontrol_new dapm_HFr_select[] = {
+	SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel),
+};
+
+/* Mic 1 */
+
+/* Mic 1 - Mic 1a or 1b selector */
+static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3,
+			AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel);
+static const struct snd_kcontrol_new dapm_mic1ab_mux[] = {
+	SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel),
+};
+
+/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */
+static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel);
+static const struct snd_kcontrol_new dapm_ad3_select[] = {
+	SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel),
+};
+
+/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */
+static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel);
+static const struct snd_kcontrol_new dapm_ad6_select[] = {
+	SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel),
+};
+
+/* Mic 2 */
+
+/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */
+static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel);
+static const struct snd_kcontrol_new dapm_ad5_select[] = {
+	SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel),
+};
+
+/* LineIn */
+
+/* LineIn left - AD1 - LineIn Left or DMic 1 selector */
+static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel);
+static const struct snd_kcontrol_new dapm_ad1_select[] = {
+	SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel),
+};
+
+/* LineIn right - Mic 2 or LineIn Right selector */
+static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3,
+			AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel);
+static const struct snd_kcontrol_new dapm_mic2lr_select[] = {
+	SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel),
+};
+
+/* LineIn right - AD2 - LineIn Right or DMic2 selector */
+static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel);
+static const struct snd_kcontrol_new dapm_ad2_select[] = {
+	SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel),
+};
+
+
+/* ANC */
+
+static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6",
+					"Mic 2 / DMic 5"};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel);
+static const struct snd_kcontrol_new dapm_anc_in_select[] = {
+	SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel),
+};
+
+/* ANC - Enable/Disable */
+static const struct snd_kcontrol_new dapm_anc_enable[] = {
+	SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1,
+			AB8500_ANCCONF1_ENANC, 0, 0),
+};
+
+/* ANC to Earpiece - Mute */
+static const struct snd_kcontrol_new dapm_anc_ear_mute[] = {
+	SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_ANCSEL, 1, 0),
+};
+
+
+
+/* Sidetone left */
+
+/* Sidetone left - Input selector */
+static const char * const enum_stfir1_in_sel[] = {
+	"LineIn Left", "LineIn Right", "Mic 1", "Headset Left"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel);
+static const struct snd_kcontrol_new dapm_stfir1_in_select[] = {
+	SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel),
+};
+
+/* Sidetone right path */
+
+/* Sidetone right - Input selector */
+static const char * const enum_stfir2_in_sel[] = {
+	"LineIn Right", "Mic 1", "DMic 4", "Headset Right"
+};
+static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2,
+			AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel);
+static const struct snd_kcontrol_new dapm_stfir2_in_select[] = {
+	SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel),
+};
+
+/* Vibra */
+
+static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1,
+			AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib1[] = {
+	SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1),
+};
+
+static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1,
+			AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx);
+
+static const struct snd_kcontrol_new dapm_pwm2vib2[] = {
+	SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2),
+};
+
+/*
+ * DAPM-widgets
+ */
+
+static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = {
+
+	/* Clocks */
+	SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"),
+
+	/* Regulators */
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0),
+	SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0),
+
+	/* Power */
+	SND_SOC_DAPM_SUPPLY("Audio Power",
+			AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("Audio Analog Power",
+			AB8500_POWERUP, AB8500_POWERUP_ENANA, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* Main supply node */
+	SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0,
+			NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* DA/AD */
+
+	SND_SOC_DAPM_INPUT("ADC Input"),
+	SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_OUTPUT("DAC Output"),
+
+	SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0),
+
+	/* Headset path */
+
+	SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5,
+			AB8500_ANACONF5_ENCPHS, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0),
+	SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0),
+
+	SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0),
+	SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0),
+	SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF,
+			AB8500_MUTECONF_MUTDACHSL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF,
+			AB8500_MUTECONF_MUTDACHSR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0),
+	SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0),
+
+	SND_SOC_DAPM_MIXER("HSL Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSL Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HSR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSL Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_PGA("HSR Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Headset Left"),
+	SND_SOC_DAPM_OUTPUT("Headset Right"),
+
+	/* LineOut path */
+
+	SND_SOC_DAPM_MUX("LineOut Source",
+			SND_SOC_NOPM, 0, 0, dapm_lineout_source),
+
+	SND_SOC_DAPM_MIXER("LOL Disable HFL",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Disable HFR",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1,
+			NULL, 0),
+
+	SND_SOC_DAPM_MIXER("LOL Enable",
+			AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Enable",
+			AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LineOut Left"),
+	SND_SOC_DAPM_OUTPUT("LineOut Right"),
+
+	/* Earpiece path */
+
+	SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source",
+			SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source),
+	SND_SOC_DAPM_MIXER("EAR DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("EAR Mute",
+			AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("EAR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Earpiece"),
+
+	/* Handsfree path */
+
+	SND_SOC_DAPM_MIXER("DA3 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA4 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("Speaker Left Source",
+			SND_SOC_NOPM, 0, 0, dapm_HFl_select),
+	SND_SOC_DAPM_MUX("Speaker Right Source",
+			SND_SOC_NOPM, 0, 0, dapm_HFr_select),
+	SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF,
+			AB8500_DAPATHCONF_ENDACHFL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFR DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR",
+			AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL",
+			AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFL Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("HFR Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Speaker Left"),
+	SND_SOC_DAPM_OUTPUT("Speaker Right"),
+
+	/* Vibrator path */
+
+	SND_SOC_DAPM_INPUT("PWMGEN1"),
+	SND_SOC_DAPM_INPUT("PWMGEN2"),
+
+	SND_SOC_DAPM_MIXER("DA5 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DA6 Channel Volume",
+			AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB1 DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB2 DAC",
+			AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("Vibra 1 Controller",
+			SND_SOC_NOPM, 0, 0, dapm_pwm2vib1),
+	SND_SOC_DAPM_MUX("Vibra 2 Controller",
+			SND_SOC_NOPM, 0, 0, dapm_pwm2vib2),
+	SND_SOC_DAPM_MIXER("VIB1 Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("VIB2 Enable",
+			AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("Vibra 1"),
+	SND_SOC_DAPM_OUTPUT("Vibra 2"),
+
+	/* Mic 1 */
+
+	SND_SOC_DAPM_INPUT("Mic 1"),
+
+	SND_SOC_DAPM_MUX("Mic 1a or 1b Select",
+			SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux),
+	SND_SOC_DAPM_MIXER("MIC1 Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable",
+			AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable",
+			AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC1 ADC",
+			AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("AD3 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad3_select),
+	SND_SOC_DAPM_MIXER("AD3 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD3 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0,
+			NULL, 0),
+
+	/* Mic 2 */
+
+	SND_SOC_DAPM_INPUT("Mic 2"),
+
+	SND_SOC_DAPM_MIXER("MIC2 Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENMIC2, 0,
+			NULL, 0),
+
+	/* LineIn */
+
+	SND_SOC_DAPM_INPUT("LineIn Left"),
+	SND_SOC_DAPM_INPUT("LineIn Right"),
+
+	SND_SOC_DAPM_MIXER("LINL Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR Mute",
+			AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENLINL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2,
+			AB8500_ANACONF2_ENLINR, 0,
+			NULL, 0),
+
+	/* LineIn Bypass path */
+	SND_SOC_DAPM_MIXER("LINL to HSL Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR to HSR Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	/* LineIn, Mic 2 */
+	SND_SOC_DAPM_MUX("Mic 2 or LINR Select",
+			SND_SOC_NOPM, 0, 0, dapm_mic2lr_select),
+	SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3,
+			AB8500_ANACONF3_ENADCLINL, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3,
+			AB8500_ANACONF3_ENADCLINR, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MUX("AD1 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad1_select),
+	SND_SOC_DAPM_MUX("AD2 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad2_select),
+	SND_SOC_DAPM_MIXER("AD1 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD2 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+
+	SND_SOC_DAPM_MIXER("AD12 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0,
+			NULL, 0),
+
+	/* HD Capture path */
+
+	SND_SOC_DAPM_MUX("AD5 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad5_select),
+	SND_SOC_DAPM_MUX("AD6 Source Select",
+			SND_SOC_NOPM, 0, 0, dapm_ad6_select),
+	SND_SOC_DAPM_MIXER("AD5 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD6 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD57 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD68 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0,
+			NULL, 0),
+
+	/* Digital Microphone path */
+
+	SND_SOC_DAPM_INPUT("DMic 1"),
+	SND_SOC_DAPM_INPUT("DMic 2"),
+	SND_SOC_DAPM_INPUT("DMic 3"),
+	SND_SOC_DAPM_INPUT("DMic 4"),
+	SND_SOC_DAPM_INPUT("DMic 5"),
+	SND_SOC_DAPM_INPUT("DMic 6"),
+
+	SND_SOC_DAPM_MIXER("DMIC1",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC2",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC3",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC4",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC5",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("DMIC6",
+			AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD4 Channel Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("AD4 Enable",
+			AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34,
+			0, NULL, 0),
+
+	/* Acoustical Noise Cancellation path */
+
+	SND_SOC_DAPM_INPUT("ANC Configure Input"),
+	SND_SOC_DAPM_OUTPUT("ANC Configure Output"),
+
+	SND_SOC_DAPM_MUX("ANC Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_in_select),
+	SND_SOC_DAPM_SWITCH("ANC",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_enable),
+	SND_SOC_DAPM_SWITCH("ANC to Earpiece",
+			SND_SOC_NOPM, 0, 0,
+			dapm_anc_ear_mute),
+
+	/* Sidetone Filter path */
+
+	SND_SOC_DAPM_MUX("Sidetone Left Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_stfir1_in_select),
+	SND_SOC_DAPM_MUX("Sidetone Right Source",
+			SND_SOC_NOPM, 0, 0,
+			dapm_stfir2_in_select),
+	SND_SOC_DAPM_MIXER("STFIR1 Control",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR2 Control",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR1 Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+	SND_SOC_DAPM_MIXER("STFIR2 Volume",
+			SND_SOC_NOPM, 0, 0,
+			NULL, 0),
+};
+
+/*
+ * DAPM-routes
+ */
+static const struct snd_soc_dapm_route ab8500_dapm_routes[] = {
+	/* Power AB8500 audio-block when AD/DA is active */
+	{"Main Supply", NULL, "V-AUD"},
+	{"Main Supply", NULL, "audioclk"},
+	{"Main Supply", NULL, "Audio Power"},
+	{"Main Supply", NULL, "Audio Analog Power"},
+
+	{"DAC", NULL, "ab8500_0p"},
+	{"DAC", NULL, "Main Supply"},
+	{"ADC", NULL, "ab8500_0c"},
+	{"ADC", NULL, "Main Supply"},
+
+	/* ANC Configure */
+	{"ANC Configure Input", NULL, "Main Supply"},
+	{"ANC Configure Output", NULL, "ANC Configure Input"},
+
+	/* AD/DA */
+	{"ADC", NULL, "ADC Input"},
+	{"DAC Output", NULL, "DAC"},
+
+	/* Powerup charge pump if DA1/2 is in use */
+
+	{"DA_IN1", NULL, "ab8500_0p"},
+	{"DA_IN1", NULL, "Charge Pump"},
+	{"DA_IN2", NULL, "ab8500_0p"},
+	{"DA_IN2", NULL, "Charge Pump"},
+
+	/* Headset path */
+
+	{"DA1 Enable", NULL, "DA_IN1"},
+	{"DA2 Enable", NULL, "DA_IN2"},
+
+	{"HSL Digital Volume", NULL, "DA1 Enable"},
+	{"HSR Digital Volume", NULL, "DA2 Enable"},
+
+	{"HSL DAC", NULL, "HSL Digital Volume"},
+	{"HSR DAC", NULL, "HSR Digital Volume"},
+
+	{"HSL DAC Mute", NULL, "HSL DAC"},
+	{"HSR DAC Mute", NULL, "HSR DAC"},
+
+	{"HSL DAC Driver", NULL, "HSL DAC Mute"},
+	{"HSR DAC Driver", NULL, "HSR DAC Mute"},
+
+	{"HSL Mute", NULL, "HSL DAC Driver"},
+	{"HSR Mute", NULL, "HSR DAC Driver"},
+
+	{"HSL Enable", NULL, "HSL Mute"},
+	{"HSR Enable", NULL, "HSR Mute"},
+
+	{"HSL Volume", NULL, "HSL Enable"},
+	{"HSR Volume", NULL, "HSR Enable"},
+
+	{"Headset Left", NULL, "HSL Volume"},
+	{"Headset Right", NULL, "HSR Volume"},
+
+	/* HF or LineOut path */
+
+	{"DA_IN3", NULL, "ab8500_0p"},
+	{"DA3 Channel Volume", NULL, "DA_IN3"},
+	{"DA_IN4", NULL, "ab8500_0p"},
+	{"DA4 Channel Volume", NULL, "DA_IN4"},
+
+	{"Speaker Left Source", "Audio Path", "DA3 Channel Volume"},
+	{"Speaker Right Source", "Audio Path", "DA4 Channel Volume"},
+
+	{"DA3 or ANC path to HfL", NULL, "Speaker Left Source"},
+	{"DA4 or ANC path to HfR", NULL, "Speaker Right Source"},
+
+	/* HF path */
+
+	{"HFL DAC", NULL, "DA3 or ANC path to HfL"},
+	{"HFR DAC", NULL, "DA4 or ANC path to HfR"},
+
+	{"HFL Enable", NULL, "HFL DAC"},
+	{"HFR Enable", NULL, "HFR DAC"},
+
+	{"Speaker Left", NULL, "HFL Enable"},
+	{"Speaker Right", NULL, "HFR Enable"},
+
+	/* Earpiece path */
+
+	{"Earpiece or LineOut Mono Source", "Headset Left",
+		"HSL Digital Volume"},
+	{"Earpiece or LineOut Mono Source", "Speaker Left",
+		"DA3 or ANC path to HfL"},
+
+	{"EAR DAC", NULL, "Earpiece or LineOut Mono Source"},
+
+	{"EAR Mute", NULL, "EAR DAC"},
+
+	{"EAR Enable", NULL, "EAR Mute"},
+
+	{"Earpiece", NULL, "EAR Enable"},
+
+	/* LineOut path stereo */
+
+	{"LineOut Source", "Stereo Path", "HSL DAC Driver"},
+	{"LineOut Source", "Stereo Path", "HSR DAC Driver"},
+
+	/* LineOut path mono */
+
+	{"LineOut Source", "Mono Path", "EAR DAC"},
+
+	/* LineOut path */
+
+	{"LOL Disable HFL", NULL, "LineOut Source"},
+	{"LOR Disable HFR", NULL, "LineOut Source"},
+
+	{"LOL Enable", NULL, "LOL Disable HFL"},
+	{"LOR Enable", NULL, "LOR Disable HFR"},
+
+	{"LineOut Left", NULL, "LOL Enable"},
+	{"LineOut Right", NULL, "LOR Enable"},
+
+	/* Vibrator path */
+
+	{"DA_IN5", NULL, "ab8500_0p"},
+	{"DA5 Channel Volume", NULL, "DA_IN5"},
+	{"DA_IN6", NULL, "ab8500_0p"},
+	{"DA6 Channel Volume", NULL, "DA_IN6"},
+
+	{"VIB1 DAC", NULL, "DA5 Channel Volume"},
+	{"VIB2 DAC", NULL, "DA6 Channel Volume"},
+
+	{"Vibra 1 Controller", "Audio Path", "VIB1 DAC"},
+	{"Vibra 2 Controller", "Audio Path", "VIB2 DAC"},
+	{"Vibra 1 Controller", "PWM Generator", "PWMGEN1"},
+	{"Vibra 2 Controller", "PWM Generator", "PWMGEN2"},
+
+	{"VIB1 Enable", NULL, "Vibra 1 Controller"},
+	{"VIB2 Enable", NULL, "Vibra 2 Controller"},
+
+	{"Vibra 1", NULL, "VIB1 Enable"},
+	{"Vibra 2", NULL, "VIB2 Enable"},
+
+
+	/* Mic 2 */
+
+	{"MIC2 V-AMICx Enable", NULL, "Mic 2"},
+
+	/* LineIn */
+	{"LINL Mute", NULL, "LineIn Left"},
+	{"LINR Mute", NULL, "LineIn Right"},
+
+	{"LINL Enable", NULL, "LINL Mute"},
+	{"LINR Enable", NULL, "LINR Mute"},
+
+	/* LineIn, Mic 2 */
+	{"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"},
+	{"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"},
+
+	{"LINL ADC", NULL, "LINL Enable"},
+	{"LINR ADC", NULL, "Mic 2 or LINR Select"},
+
+	{"AD1 Source Select", "LineIn Left", "LINL ADC"},
+	{"AD2 Source Select", "LineIn Right", "LINR ADC"},
+
+	{"AD1 Channel Volume", NULL, "AD1 Source Select"},
+	{"AD2 Channel Volume", NULL, "AD2 Source Select"},
+
+	{"AD12 Enable", NULL, "AD1 Channel Volume"},
+	{"AD12 Enable", NULL, "AD2 Channel Volume"},
+
+	{"AD_OUT1", NULL, "ab8500_0c"},
+	{"AD_OUT1", NULL, "AD12 Enable"},
+	{"AD_OUT2", NULL, "ab8500_0c"},
+	{"AD_OUT2", NULL, "AD12 Enable"},
+
+	/* Mic 1 */
+
+	{"MIC1 Mute", NULL, "Mic 1"},
+
+	{"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"},
+	{"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"},
+
+	{"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"},
+	{"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"},
+
+	{"MIC1 ADC", NULL, "Mic 1a or 1b Select"},
+
+	{"AD3 Source Select", "Mic 1", "MIC1 ADC"},
+
+	{"AD3 Channel Volume", NULL, "AD3 Source Select"},
+
+	{"AD3 Enable", NULL, "AD3 Channel Volume"},
+
+	{"AD_OUT3", NULL, "ab8500_0c"},
+	{"AD_OUT3", NULL, "AD3 Enable"},
+
+	/* HD Capture path */
+
+	{"AD5 Source Select", "Mic 2", "LINR ADC"},
+	{"AD6 Source Select", "Mic 1", "MIC1 ADC"},
+
+	{"AD5 Channel Volume", NULL, "AD5 Source Select"},
+	{"AD6 Channel Volume", NULL, "AD6 Source Select"},
+
+	{"AD57 Enable", NULL, "AD5 Channel Volume"},
+	{"AD68 Enable", NULL, "AD6 Channel Volume"},
+
+	{"AD_OUT57", NULL, "ab8500_0c"},
+	{"AD_OUT57", NULL, "AD57 Enable"},
+	{"AD_OUT68", NULL, "ab8500_0c"},
+	{"AD_OUT68", NULL, "AD68 Enable"},
+
+	/* Digital Microphone path */
+
+	{"DMic 1", NULL, "V-DMIC"},
+	{"DMic 2", NULL, "V-DMIC"},
+	{"DMic 3", NULL, "V-DMIC"},
+	{"DMic 4", NULL, "V-DMIC"},
+	{"DMic 5", NULL, "V-DMIC"},
+	{"DMic 6", NULL, "V-DMIC"},
+
+	{"AD1 Source Select", NULL, "DMic 1"},
+	{"AD2 Source Select", NULL, "DMic 2"},
+	{"AD3 Source Select", NULL, "DMic 3"},
+	{"AD5 Source Select", NULL, "DMic 5"},
+	{"AD6 Source Select", NULL, "DMic 6"},
+
+	{"AD4 Channel Volume", NULL, "DMic 4"},
+	{"AD4 Enable", NULL, "AD4 Channel Volume"},
+
+	{"AD_OUT4", NULL, "ab8500_0c"},
+	{"AD_OUT4", NULL, "AD4 Enable"},
+
+	/* LineIn Bypass path */
+
+	{"LINL to HSL Volume", NULL, "LINL Enable"},
+	{"LINR to HSR Volume", NULL, "LINR Enable"},
+
+	{"HSL DAC Driver", NULL, "LINL to HSL Volume"},
+	{"HSR DAC Driver", NULL, "LINR to HSR Volume"},
+
+	/* ANC path (Acoustic Noise Cancellation) */
+
+	{"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"},
+	{"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"},
+
+	{"ANC", "Switch", "ANC Source"},
+
+	{"Speaker Left Source", "ANC", "ANC"},
+	{"Speaker Right Source", "ANC", "ANC"},
+	{"ANC to Earpiece", "Switch", "ANC"},
+
+	{"HSL Digital Volume", NULL, "ANC to Earpiece"},
+
+	/* Sidetone Filter path */
+
+	{"Sidetone Left Source", "LineIn Left", "AD12 Enable"},
+	{"Sidetone Left Source", "LineIn Right", "AD12 Enable"},
+	{"Sidetone Left Source", "Mic 1", "AD3 Enable"},
+	{"Sidetone Left Source", "Headset Left", "DA_IN1"},
+	{"Sidetone Right Source", "LineIn Right", "AD12 Enable"},
+	{"Sidetone Right Source", "Mic 1", "AD3 Enable"},
+	{"Sidetone Right Source", "DMic 4", "AD4 Enable"},
+	{"Sidetone Right Source", "Headset Right", "DA_IN2"},
+
+	{"STFIR1 Control", NULL, "Sidetone Left Source"},
+	{"STFIR2 Control", NULL, "Sidetone Right Source"},
+
+	{"STFIR1 Volume", NULL, "STFIR1 Control"},
+	{"STFIR2 Volume", NULL, "STFIR2 Control"},
+
+	{"DA1 Enable", NULL, "STFIR1 Volume"},
+	{"DA2 Enable", NULL, "STFIR2 Volume"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = {
+	{"MIC1A V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC1A V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = {
+	{"MIC1B V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC1B V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = {
+	{"MIC2 V-AMICx Enable", NULL, "V-AMIC1"},
+	{"MIC2 V-AMICx Enable", NULL, "V-AMIC2"},
+};
+
+/* ANC FIR-coefficients configuration sequence */
+static void anc_fir(struct snd_soc_codec *codec,
+		unsigned int bnk, unsigned int par, unsigned int val)
+{
+	if (par == 0 && bnk == 0)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE),
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE));
+
+	snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff);
+	snd_soc_write(codec, AB8500_ANCCONF6, val &  0xff);
+
+	if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0);
+}
+
+/* ANC IIR-coefficients configuration sequence */
+static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk,
+		unsigned int par, unsigned int val)
+{
+	if (par == 0) {
+		if (bnk == 0) {
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRINIT),
+					BIT(AB8500_ANCCONF1_ANCIIRINIT));
+			usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRINIT), 0);
+			usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY);
+		} else {
+			snd_soc_update_bits(codec, AB8500_ANCCONF1,
+					BIT(AB8500_ANCCONF1_ANCIIRUPDATE),
+					BIT(AB8500_ANCCONF1_ANCIIRUPDATE));
+		}
+	} else if (par > 3) {
+		snd_soc_write(codec, AB8500_ANCCONF7, 0);
+		snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff);
+	}
+
+	snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff);
+	snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff);
+
+	if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0);
+}
+
+/* ANC IIR-/FIR-coefficients configuration sequence */
+static void anc_configure(struct snd_soc_codec *codec,
+			bool apply_fir, bool apply_iir)
+{
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	unsigned int bnk, par, val;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	if (apply_fir)
+		snd_soc_update_bits(codec, AB8500_ANCCONF1,
+			BIT(AB8500_ANCCONF1_ENANC), 0);
+
+	snd_soc_update_bits(codec, AB8500_ANCCONF1,
+		BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC));
+
+	if (apply_fir)
+		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+			for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) {
+				val = snd_soc_read(codec,
+						drvdata->anc_fir_values[par]);
+				anc_fir(codec, bnk, par, val);
+			}
+
+	if (apply_iir)
+		for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++)
+			for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) {
+				val = snd_soc_read(codec,
+						drvdata->anc_iir_values[par]);
+				anc_iir(codec, bnk, par, val);
+			}
+
+	dev_dbg(codec->dev, "%s: Exit.\n", __func__);
+}
+
+/*
+ * Control-events
+ */
+
+static int sid_status_control_get(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+	mutex_lock(&codec->mutex);
+	ucontrol->value.integer.value[0] = drvdata->sid_status;
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+/* Write sidetone FIR-coefficients configuration sequence */
+static int sid_status_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	unsigned int param, sidconf, val;
+	int status = 1;
+
+	dev_dbg(codec->dev, "%s: Enter\n", __func__);
+
+	if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) {
+		dev_err(codec->dev,
+			"%s: ERROR: This control supports '%s' only!\n",
+			__func__, enum_sid_state[SID_APPLY_FIR]);
+		return -EIO;
+	}
+
+	mutex_lock(&codec->mutex);
+
+	sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF);
+	if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) {
+		if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) {
+			dev_err(codec->dev, "%s: Sidetone busy while off!\n",
+				__func__);
+			status = -EPERM;
+		} else {
+			status = -EBUSY;
+		}
+		goto out;
+	}
+
+	snd_soc_write(codec, AB8500_SIDFIRADR, 0);
+
+	for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) {
+		val = snd_soc_read(codec, drvdata->sid_fir_values[param]);
+		snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff);
+		snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff);
+	}
+
+	snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+		BIT(AB8500_SIDFIRADR_FIRSIDSET),
+		BIT(AB8500_SIDFIRADR_FIRSIDSET));
+	snd_soc_update_bits(codec, AB8500_SIDFIRADR,
+		BIT(AB8500_SIDFIRADR_FIRSIDSET), 0);
+
+	drvdata->sid_status = SID_FIR_CONFIGURED;
+
+out:
+	mutex_unlock(&codec->mutex);
+
+	dev_dbg(codec->dev, "%s: Exit\n", __func__);
+
+	return status;
+}
+
+static int anc_status_control_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+
+	mutex_lock(&codec->mutex);
+	ucontrol->value.integer.value[0] = drvdata->anc_status;
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+static int anc_status_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev);
+	struct device *dev = codec->dev;
+	bool apply_fir, apply_iir;
+	int req, status;
+
+	dev_dbg(dev, "%s: Enter.\n", __func__);
+
+	mutex_lock(&drvdata->anc_lock);
+
+	req = ucontrol->value.integer.value[0];
+	if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR &&
+		req != ANC_APPLY_IIR) {
+		dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n",
+			__func__, enum_anc_state[req]);
+		status = -EINVAL;
+		goto cleanup;
+	}
+	apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR;
+	apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR;
+
+	status = snd_soc_dapm_force_enable_pin(&codec->dapm,
+					"ANC Configure Input");
+	if (status < 0) {
+		dev_err(dev,
+			"%s: ERROR: Failed to enable power (status = %d)!\n",
+			__func__, status);
+		goto cleanup;
+	}
+	snd_soc_dapm_sync(&codec->dapm);
+
+	mutex_lock(&codec->mutex);
+	anc_configure(codec, apply_fir, apply_iir);
+	mutex_unlock(&codec->mutex);
+
+	if (apply_fir) {
+		if (drvdata->anc_status == ANC_IIR_CONFIGURED)
+			drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+		else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+			drvdata->anc_status =  ANC_FIR_CONFIGURED;
+	}
+	if (apply_iir) {
+		if (drvdata->anc_status == ANC_FIR_CONFIGURED)
+			drvdata->anc_status = ANC_FIR_IIR_CONFIGURED;
+		else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED)
+			drvdata->anc_status =  ANC_IIR_CONFIGURED;
+	}
+
+	status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+	snd_soc_dapm_sync(&codec->dapm);
+
+cleanup:
+	mutex_unlock(&drvdata->anc_lock);
+
+	if (status < 0)
+		dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n",
+			__func__, status);
+
+	dev_dbg(dev, "%s: Exit.\n", __func__);
+
+	return (status < 0) ? status : 1;
+}
+
+static int filter_control_info(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = fc->count;
+	uinfo->value.integer.min = fc->min;
+	uinfo->value.integer.max = fc->max;
+
+	return 0;
+}
+
+static int filter_control_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+	unsigned int i;
+
+	mutex_lock(&codec->mutex);
+	for (i = 0; i < fc->count; i++)
+		ucontrol->value.integer.value[i] = fc->value[i];
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+static int filter_control_put(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct filter_control *fc =
+			(struct filter_control *)kcontrol->private_value;
+	unsigned int i;
+
+	mutex_lock(&codec->mutex);
+	for (i = 0; i < fc->count; i++)
+		fc->value[i] = ucontrol->value.integer.value[i];
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+
+/*
+ * Controls - Non-DAPM ASoC
+ */
+
+static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1);
+/* -32dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1);
+/* -63dB = Mute */
+
+static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1);
+/* -1dB = Mute */
+
+static const unsigned int hs_gain_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0),
+	4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0),
+};
+
+static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0);
+
+static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0);
+
+static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1);
+/* -38dB = Mute */
+
+static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms",
+					"5ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed,
+	AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed);
+
+static const char * const enum_envdetthre[] = {
+	"250mV", "300mV", "350mV", "400mV",
+	"450mV", "500mV", "550mV", "600mV",
+	"650mV", "700mV", "750mV", "800mV",
+	"850mV", "900mV", "950mV", "1.00V" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre,
+	AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre);
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre,
+	AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre);
+static const char * const enum_envdettime[] = {
+	"26.6us", "53.2us", "106us",  "213us",
+	"426us",  "851us",  "1.70ms", "3.40ms",
+	"6.81ms", "13.6ms", "27.2ms", "54.5ms",
+	"109ms",  "218ms",  "436ms",  "872ms" };
+static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime,
+	AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime);
+
+static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN,
+			AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31);
+
+static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN,
+			AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed);
+
+/* Earpiece */
+
+static const char * const enum_lowpow[] = {"Normal", "Low Power"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1,
+			AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow);
+static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1,
+			AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow);
+
+static const char * const enum_av_mode[] = {"Audio", "Voice"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF,
+	AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF,
+	AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode);
+
+/* DA */
+
+static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice,
+			AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE,
+			enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice,
+			AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE,
+			enum_av_mode);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice,
+			AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE,
+			enum_av_mode);
+
+static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1,
+			AB8500_DIGMULTCONF1_DATOHSLEN,
+			AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr);
+
+static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"};
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC1SINC3,
+			AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC3SINC3,
+			AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53);
+static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF,
+			AB8500_DMICFILTCONF_DMIC5SINC3,
+			AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53);
+
+/* Digital interface - DA from slot mapping */
+static const char * const enum_da_from_slot_map[] = {"SLOT0",
+					"SLOT1",
+					"SLOT2",
+					"SLOT3",
+					"SLOT4",
+					"SLOT5",
+					"SLOT6",
+					"SLOT7",
+					"SLOT8",
+					"SLOT9",
+					"SLOT10",
+					"SLOT11",
+					"SLOT12",
+					"SLOT13",
+					"SLOT14",
+					"SLOT15",
+					"SLOT16",
+					"SLOT17",
+					"SLOT18",
+					"SLOT19",
+					"SLOT20",
+					"SLOT21",
+					"SLOT22",
+					"SLOT23",
+					"SLOT24",
+					"SLOT25",
+					"SLOT26",
+					"SLOT27",
+					"SLOT28",
+					"SLOT29",
+					"SLOT30",
+					"SLOT31"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap,
+			AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap,
+			AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap,
+			AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap,
+			AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap,
+			AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap,
+			AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap,
+			AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap,
+			AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT,
+			enum_da_from_slot_map);
+
+/* Digital interface - AD to slot mapping */
+static const char * const enum_ad_to_slot_map[] = {"AD_OUT1",
+					"AD_OUT2",
+					"AD_OUT3",
+					"AD_OUT4",
+					"AD_OUT5",
+					"AD_OUT6",
+					"AD_OUT7",
+					"AD_OUT8",
+					"zeroes",
+					"tristate"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map,
+			AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map,
+			AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map,
+			AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map,
+			AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map,
+			AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map,
+			AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map,
+			AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map,
+			AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map,
+			AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map,
+			AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map,
+			AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map,
+			AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map,
+			AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map,
+			AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map,
+			AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map,
+			AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map,
+			AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map,
+			AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map,
+			AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map,
+			AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map,
+			AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map,
+			AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map,
+			AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map,
+			AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map,
+			AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map,
+			AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map,
+			AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map,
+			AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map,
+			AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map,
+			AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map,
+			AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT,
+			enum_ad_to_slot_map);
+static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map,
+			AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT,
+			enum_ad_to_slot_map);
+
+/* Digital interface - Burst mode */
+static const char * const enum_mask[] = {"Unmasked", "Masked"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask,
+			AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK,
+			enum_mask);
+static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2,
+			AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2,
+			enum_bitclk0);
+static const char * const enum_slavemaster[] = {"Slave", "Master"};
+static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast,
+			AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT,
+			enum_slavemaster);
+
+/* Sidetone */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state);
+
+/* ANC */
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state);
+
+static struct snd_kcontrol_new ab8500_ctrls[] = {
+	/* Charge pump */
+	SOC_ENUM("Charge Pump High Threshold For Low Voltage",
+		soc_enum_envdeththre),
+	SOC_ENUM("Charge Pump Low Threshold For Low Voltage",
+		soc_enum_envdetlthre),
+	SOC_SINGLE("Charge Pump Envelope Detection Switch",
+		AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN,
+		1, 0),
+	SOC_ENUM("Charge Pump Envelope Detection Decay Time",
+		soc_enum_envdettime),
+
+	/* Headset */
+	SOC_ENUM("Headset Mode", soc_enum_da12voice),
+	SOC_SINGLE("Headset High Pass Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN,
+		1, 0),
+	SOC_SINGLE("Headset Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW,
+		1, 0),
+	SOC_SINGLE("Headset DAC Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1,
+		1, 0),
+	SOC_SINGLE("Headset DAC Drv Low Power Switch",
+		AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0,
+		1, 0),
+	SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed),
+	SOC_ENUM("Headset Source", soc_enum_da2hslr),
+	SOC_ENUM("Headset Filter", soc_enum_hsesinc),
+	SOC_DOUBLE_R_TLV("Headset Master Volume",
+		AB8500_DADIGGAIN1, AB8500_DADIGGAIN2,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+	SOC_DOUBLE_R_TLV("Headset Digital Volume",
+		AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN,
+		0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv),
+	SOC_DOUBLE_TLV("Headset Volume",
+		AB8500_ANAGAIN3,
+		AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN,
+		AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv),
+
+	/* Earpiece */
+	SOC_ENUM("Earpiece DAC Mode",
+		soc_enum_eardaclowpow),
+	SOC_ENUM("Earpiece DAC Drv Mode",
+		soc_enum_eardrvlowpow),
+
+	/* HandsFree */
+	SOC_ENUM("HF Mode", soc_enum_da34voice),
+	SOC_SINGLE("HF and Headset Swap Switch",
+		AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34,
+		1, 0),
+	SOC_DOUBLE("HF Low EMI Mode Switch",
+		AB8500_CLASSDCONF1,
+		AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN,
+		1, 0),
+	SOC_DOUBLE("HF FIR Bypass Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1,
+		1, 0),
+	SOC_DOUBLE("HF High Volume Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1,
+		1, 0),
+	SOC_SINGLE("HF L and R Bridge Switch",
+		AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF,
+		1, 0),
+	SOC_DOUBLE_R_TLV("HF Master Volume",
+		AB8500_DADIGGAIN3, AB8500_DADIGGAIN4,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* Vibra */
+	SOC_DOUBLE("Vibra High Volume Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3,
+		1, 0),
+	SOC_DOUBLE("Vibra Low EMI Mode Switch",
+		AB8500_CLASSDCONF1,
+		AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN,
+		1, 0),
+	SOC_DOUBLE("Vibra FIR Bypass Switch",
+		AB8500_CLASSDCONF2,
+		AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3,
+		1, 0),
+	SOC_ENUM("Vibra Mode", soc_enum_da56voice),
+	SOC_DOUBLE_R("Vibra PWM Duty Cycle N",
+		AB8500_PWMGENCONF3, AB8500_PWMGENCONF5,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+	SOC_DOUBLE_R("Vibra PWM Duty Cycle P",
+		AB8500_PWMGENCONF2, AB8500_PWMGENCONF4,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC,
+		AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0),
+	SOC_SINGLE("Vibra 1 and 2 Bridge Switch",
+		AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB,
+		1, 0),
+	SOC_DOUBLE_R_TLV("Vibra Master Volume",
+		AB8500_DADIGGAIN5, AB8500_DADIGGAIN6,
+		0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* HandsFree, Vibra */
+	SOC_SINGLE("ClassD High Pass Volume",
+		AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN,
+		AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0),
+	SOC_SINGLE("ClassD White Volume",
+		AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN,
+		AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0),
+
+	/* Mic 1, Mic 2, LineIn */
+	SOC_DOUBLE_R_TLV("Mic Master Volume",
+		AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+	/* Mic 1 */
+	SOC_SINGLE_TLV("Mic 1",
+		AB8500_ANAGAIN1,
+		AB8500_ANAGAINX_MICXGAIN,
+		AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+	SOC_SINGLE("Mic 1 Low Power Switch",
+		AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX,
+		1, 0),
+
+	/* Mic 2 */
+	SOC_DOUBLE("Mic High Pass Switch",
+		AB8500_ADFILTCONF,
+		AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH,
+		1, 1),
+	SOC_ENUM("Mic Mode", soc_enum_ad34voice),
+	SOC_ENUM("Mic Filter", soc_enum_dmic34sinc),
+	SOC_SINGLE_TLV("Mic 2",
+		AB8500_ANAGAIN2,
+		AB8500_ANAGAINX_MICXGAIN,
+		AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv),
+	SOC_SINGLE("Mic 2 Low Power Switch",
+		AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX,
+		1, 0),
+
+	/* LineIn */
+	SOC_DOUBLE("LineIn High Pass Switch",
+		AB8500_ADFILTCONF,
+		AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH,
+		1, 1),
+	SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc),
+	SOC_ENUM("LineIn Mode", soc_enum_ad12voice),
+	SOC_DOUBLE_R_TLV("LineIn Master Volume",
+		AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+	SOC_DOUBLE_TLV("LineIn",
+		AB8500_ANAGAIN4,
+		AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN,
+		AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv),
+	SOC_DOUBLE_R_TLV("LineIn to Headset Volume",
+		AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN,
+		AB8500_DIGLINHSXGAIN_LINTOHSXGAIN,
+		AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX,
+		1, lin2hs_gain_tlv),
+
+	/* DMic */
+	SOC_ENUM("DMic Filter", soc_enum_dmic56sinc),
+	SOC_DOUBLE_R_TLV("DMic Master Volume",
+		AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6,
+		0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv),
+
+	/* Digital gains */
+	SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed),
+
+	/* Analog loopback */
+	SOC_DOUBLE_R_TLV("Analog Loopback Volume",
+		AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2,
+		0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv),
+
+	/* Digital interface - DA from slot mapping */
+	SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap),
+	SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap),
+	SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap),
+	SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap),
+	SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap),
+	SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap),
+	SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap),
+	SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap),
+
+	/* Digital interface - AD to slot mapping */
+	SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map),
+	SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map),
+	SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map),
+	SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map),
+	SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map),
+	SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map),
+	SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map),
+	SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map),
+	SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map),
+	SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map),
+	SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map),
+	SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map),
+	SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map),
+	SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map),
+	SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map),
+	SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map),
+	SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map),
+	SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map),
+	SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map),
+	SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map),
+	SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map),
+	SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map),
+	SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map),
+	SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map),
+	SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map),
+	SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map),
+	SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map),
+	SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map),
+	SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map),
+	SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map),
+	SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map),
+	SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map),
+
+	/* Digital interface - Loopback */
+	SOC_SINGLE("Digital Interface AD 1 Loopback Switch",
+		AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 2 Loopback Switch",
+		AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 3 Loopback Switch",
+		AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 4 Loopback Switch",
+		AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 5 Loopback Switch",
+		AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 6 Loopback Switch",
+		AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 7 Loopback Switch",
+		AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7,
+		1, 0),
+	SOC_SINGLE("Digital Interface AD 8 Loopback Switch",
+		AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8,
+		1, 0),
+
+	/* Digital interface - Burst FIFO */
+	SOC_SINGLE("Digital Interface 0 FIFO Enable Switch",
+		AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN,
+		1, 0),
+	SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask),
+	SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2),
+	SOC_SINGLE("Burst FIFO Threshold",
+		AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT,
+		AB8500_FIFOCONF1_BFIFOINT_MAX, 0),
+	SOC_SINGLE("Burst FIFO Length",
+		AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT,
+		AB8500_FIFOCONF2_BFIFOTX_MAX, 0),
+	SOC_SINGLE("Burst FIFO EOS Extra Slots",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT,
+		AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0),
+	SOC_SINGLE("Burst FIFO FS Extra Bit-clocks",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT,
+		AB8500_FIFOCONF3_PREBITCLK0_MAX, 0),
+	SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast),
+
+	SOC_SINGLE("Burst FIFO Interface Switch",
+		AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT,
+		1, 0),
+	SOC_SINGLE("Burst FIFO Switch Frame Number",
+		AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT,
+		AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0),
+	SOC_SINGLE("Burst FIFO Wake Up Delay",
+		AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT,
+		AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0),
+	SOC_SINGLE("Burst FIFO Samples In FIFO",
+		AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT,
+		AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0),
+
+	/* ANC */
+	SOC_ENUM_EXT("ANC Status", soc_enum_ancstate,
+		anc_status_control_get, anc_status_control_put),
+	SOC_SINGLE_XR_SX("ANC Warp Delay Shift",
+		AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT,
+		AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC FIR Output Shift",
+		AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT,
+		AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC IIR Output Shift",
+		AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT,
+		AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0),
+	SOC_SINGLE_XR_SX("ANC Warp Delay",
+		AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT,
+		AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0),
+
+	/* Sidetone */
+	SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate,
+		sid_status_control_get, sid_status_control_put),
+	SOC_SINGLE_STROBE("Sidetone Reset",
+		AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0),
+};
+
+static struct snd_kcontrol_new ab8500_filter_controls[] = {
+	AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS,
+		AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX),
+	AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS,
+		AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX),
+	AB8500_FILTER_CONTROL("Sidetone FIR Coefficients",
+			AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN,
+			AB8500_SID_FIR_COEFF_MAX)
+};
+enum ab8500_filter {
+	AB8500_FILTER_ANC_FIR = 0,
+	AB8500_FILTER_ANC_IIR = 1,
+	AB8500_FILTER_SID_FIR = 2,
+};
+
+/*
+ * Extended interface for codec-driver
+ */
+
+static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec)
+{
+	int status;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	/* Reset audio-registers and disable 32kHz-clock output 2 */
+	status = ab8500_sysctrl_write(AB8500_STW4500CTRL3,
+				AB8500_STW4500CTRL3_CLK32KOUT2DIS |
+					AB8500_STW4500CTRL3_RESETAUDN,
+				AB8500_STW4500CTRL3_RESETAUDN);
+	if (status < 0)
+		return status;
+
+	return 0;
+}
+
+static int ab8500_audio_setup_mics(struct snd_soc_codec *codec,
+			struct amic_settings *amics)
+{
+	u8 value8;
+	unsigned int value;
+	int status;
+	const struct snd_soc_dapm_route *route;
+
+	dev_dbg(codec->dev, "%s: Enter.\n", __func__);
+
+	/* Set DMic-clocks to outputs */
+	status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC,
+						(u8)AB8500_GPIO_DIR4_REG,
+						&value8);
+	if (status < 0)
+		return status;
+	value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT |
+		GPIO31_DIR_OUTPUT;
+	status = abx500_set_register_interruptible(codec->dev,
+						(u8)AB8500_MISC,
+						(u8)AB8500_GPIO_DIR4_REG,
+						value);
+	if (status < 0)
+		return status;
+
+	/* Attach regulators to AMic DAPM-paths */
+	dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic1a_micbias));
+	route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias];
+	status = snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic1b_micbias));
+	route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias];
+	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__,
+		amic_micbias_str(amics->mic2_micbias));
+	route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias];
+	status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1);
+	if (status < 0) {
+		dev_err(codec->dev,
+			"%s: Failed to add AMic-regulator DAPM-routes (%d).\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Set AMic-configuration */
+	dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__,
+		amic_type_str(amics->mic1_type));
+	snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX,
+			amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ?
+				0 : AB8500_ANAGAINX_ENSEMICX);
+	dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__,
+		amic_type_str(amics->mic2_type));
+	snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX,
+			amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ?
+				0 : AB8500_ANAGAINX_ENSEMICX);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics);
+
+static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec,
+				enum ear_cm_voltage ear_cmv)
+{
+	char *cmv_str;
+
+	switch (ear_cmv) {
+	case EAR_CMV_0_95V:
+		cmv_str = "0.95V";
+		break;
+	case EAR_CMV_1_10V:
+		cmv_str = "1.10V";
+		break;
+	case EAR_CMV_1_27V:
+		cmv_str = "1.27V";
+		break;
+	case EAR_CMV_1_58V:
+		cmv_str = "1.58V";
+		break;
+	default:
+		dev_err(codec->dev,
+			"%s: Unknown earpiece CM-voltage (%d)!\n",
+			__func__, (int)ear_cmv);
+		return -EINVAL;
+	}
+	dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__,
+		cmv_str);
+	snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM,
+			ear_cmv);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv);
+
+static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai,
+				unsigned int delay)
+{
+	unsigned int mask, val;
+	struct snd_soc_codec *codec = dai->codec;
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0DEL);
+	val = 0;
+
+	switch (delay) {
+	case 0:
+		break;
+	case 1:
+		val |= BIT(AB8500_DIGIFCONF2_IF0DEL);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported bit-delay (0x%x)!\n",
+			__func__, delay);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n",
+		__func__, delay);
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	return 0;
+}
+
+/* Gates clocking according format mask */
+static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec,
+					unsigned int fmt)
+{
+	unsigned int mask;
+	unsigned int val;
+
+	mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) |
+			BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+
+	val = BIT(AB8500_DIGIFCONF1_ENMASTGEN);
+
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) {
+	case SND_SOC_DAIFMT_CONT: /* continuous clock */
+		dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0);
+		break;
+	case SND_SOC_DAIFMT_GATED: /* clock is gated */
+		dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n",
+			__func__);
+		break;
+	default:
+		dev_err(codec->dev,
+			"%s: ERROR: Unsupported clock mask (0x%x)!\n",
+			__func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+	return 0;
+}
+
+static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	unsigned int mask;
+	unsigned int val;
+	struct snd_soc_codec *codec = dai->codec;
+	int status;
+
+	dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt);
+
+	mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) |
+			BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) |
+			BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) |
+			BIT(AB8500_DIGIFCONF3_IF0MASTER);
+	val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Master-mode: AB8500 master.\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF3_IF0MASTER);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Master-mode: AB8500 slave.\n", __func__);
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */
+	case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */
+		dev_err(dai->codec->dev,
+			"%s: ERROR: The device is either a master or a slave.\n",
+			__func__);
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupporter master mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+		break;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val);
+
+	/* Set clock gating */
+	status = ab8500_codec_set_dai_clock_gate(codec, fmt);
+	if (status) {
+		dev_err(dai->codec->dev,
+			"%s: ERRROR: Failed to set clock gate (%d).\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Setting data transfer format */
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) |
+		BIT(AB8500_DIGIFCONF2_IF0FORMAT1) |
+		BIT(AB8500_DIGIFCONF2_FSYNC0P) |
+		BIT(AB8500_DIGIFCONF2_BITCLK0P);
+	val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S: /* I2S mode */
+		dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1);
+		ab8500_audio_set_bit_delay(dai, 0);
+		break;
+
+	case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Protocol: DSP A (TDM)\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+		ab8500_audio_set_bit_delay(dai, 1);
+		break;
+
+	case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0 Protocol: DSP B (TDM)\n", __func__);
+		val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0);
+		ab8500_audio_set_bit_delay(dai, 0);
+		break;
+
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported format (0x%x)!\n",
+			__func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Normal bit clock, normal frame\n",
+			__func__);
+		break;
+	case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Normal bit clock, inverted frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+		break;
+	case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Inverted bit clock, normal frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+		break;
+	case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */
+		dev_dbg(dai->codec->dev,
+			"%s: IF0: Inverted bit clock, inverted frame\n",
+			__func__);
+		val |= BIT(AB8500_DIGIFCONF2_FSYNC0P);
+		val |= BIT(AB8500_DIGIFCONF2_BITCLK0P);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported INV mask 0x%x\n",
+			__func__, fmt & SND_SOC_DAIFMT_INV_MASK);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	return 0;
+}
+
+static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai,
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned int val, mask, slots_active;
+
+	mask = BIT(AB8500_DIGIFCONF2_IF0WL0) |
+		BIT(AB8500_DIGIFCONF2_IF0WL1);
+	val = 0;
+
+	switch (slot_width) {
+	case 16:
+		break;
+	case 20:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL0);
+		break;
+	case 24:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL1);
+		break;
+	case 32:
+		val |= BIT(AB8500_DIGIFCONF2_IF0WL1) |
+			BIT(AB8500_DIGIFCONF2_IF0WL0);
+		break;
+	default:
+		dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n",
+			__func__, slot_width);
+		return -EINVAL;
+	}
+
+	dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n",
+		__func__, slot_width);
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val);
+
+	/* Setup TDM clocking according to slot count */
+	dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots);
+	mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+			BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+	switch (slots) {
+	case 2:
+		val = AB8500_MASK_NONE;
+		break;
+	case 4:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0);
+		break;
+	case 8:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+		break;
+	case 16:
+		val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) |
+			BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: ERROR: Unsupported number of slots (%d)!\n",
+			__func__, slots);
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val);
+
+	/* Setup TDM DA according to active tx slots */
+	mask = AB8500_DASLOTCONFX_SLTODAX_MASK;
+	slots_active = hweight32(tx_mask);
+	dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__,
+		slots_active);
+	switch (slots_active) {
+	case 0:
+		break;
+	case 1:
+		/* Slot 9 -> DA_IN1 & DA_IN3 */
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+		break;
+	case 2:
+		/* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11);
+		snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11);
+
+		break;
+	case 8:
+		dev_dbg(dai->codec->dev,
+			"%s: In 8-channel mode DA-from-slot mapping is set manually.",
+			__func__);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: Unsupported number of active TX-slots (%d)!\n",
+			__func__, slots_active);
+		return -EINVAL;
+	}
+
+	/* Setup TDM AD according to active RX-slots */
+	slots_active = hweight32(rx_mask);
+	dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__,
+		slots_active);
+	switch (slots_active) {
+	case 0:
+		break;
+	case 1:
+		/* AD_OUT3 -> slot 0 & 1 */
+		snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL,
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD);
+		break;
+	case 2:
+		/* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */
+		snd_soc_update_bits(codec,
+				AB8500_ADSLOTSEL1,
+				AB8500_MASK_ALL,
+				AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN |
+				AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD);
+		break;
+	case 8:
+		dev_dbg(dai->codec->dev,
+			"%s: In 8-channel mode AD-to-slot mapping is set manually.",
+			__func__);
+		break;
+	default:
+		dev_err(dai->codec->dev,
+			"%s: Unsupported number of active RX-slots (%d)!\n",
+			__func__, slots_active);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+struct snd_soc_dai_driver ab8500_codec_dai[] = {
+	{
+		.name = "ab8500-codec-dai.0",
+		.id = 0,
+		.playback = {
+			.stream_name = "ab8500_0p",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = AB8500_SUPPORTED_RATE,
+			.formats = AB8500_SUPPORTED_FMT,
+		},
+		.ops = (struct snd_soc_dai_ops[]) {
+			{
+				.set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+				.set_fmt = ab8500_codec_set_dai_fmt,
+			}
+		},
+		.symmetric_rates = 1
+	},
+	{
+		.name = "ab8500-codec-dai.1",
+		.id = 1,
+		.capture = {
+			.stream_name = "ab8500_0c",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = AB8500_SUPPORTED_RATE,
+			.formats = AB8500_SUPPORTED_FMT,
+		},
+		.ops = (struct snd_soc_dai_ops[]) {
+			{
+				.set_tdm_slot = ab8500_codec_set_dai_tdm_slot,
+				.set_fmt = ab8500_codec_set_dai_fmt,
+			}
+		},
+		.symmetric_rates = 1
+	}
+};
+
+static int ab8500_codec_probe(struct snd_soc_codec *codec)
+{
+	struct device *dev = codec->dev;
+	struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev);
+	struct ab8500_platform_data *pdata;
+	struct filter_control *fc;
+	int status;
+
+	dev_dbg(dev, "%s: Enter.\n", __func__);
+
+	/* Setup AB8500 according to board-settings */
+	pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent);
+	status = ab8500_audio_setup_mics(codec, &pdata->codec->amics);
+	if (status < 0) {
+		pr_err("%s: Failed to setup mics (%d)!\n", __func__, status);
+		return status;
+	}
+	status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv);
+	if (status < 0) {
+		pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n",
+			__func__, status);
+		return status;
+	}
+
+	status = ab8500_audio_init_audioblock(codec);
+	if (status < 0) {
+		dev_err(dev, "%s: failed to init audio-block (%d)!\n",
+			__func__, status);
+		return status;
+	}
+
+	/* Override HW-defaults */
+	ab8500_codec_write_reg(codec,
+				AB8500_ANACONF5,
+				BIT(AB8500_ANACONF5_HSAUTOEN));
+	ab8500_codec_write_reg(codec,
+				AB8500_SHORTCIRCONF,
+				BIT(AB8500_SHORTCIRCONF_HSZCDDIS));
+
+	/* Add filter controls */
+	status = snd_soc_add_codec_controls(codec, ab8500_filter_controls,
+				ARRAY_SIZE(ab8500_filter_controls));
+	if (status < 0) {
+		dev_err(dev,
+			"%s: failed to add ab8500 filter controls (%d).\n",
+			__func__, status);
+		return status;
+	}
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value;
+	drvdata->anc_fir_values = (long *)fc->value;
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value;
+	drvdata->anc_iir_values = (long *)fc->value;
+	fc = (struct filter_control *)
+		&ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value;
+	drvdata->sid_fir_values = (long *)fc->value;
+
+	(void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input");
+
+	mutex_init(&drvdata->anc_lock);
+
+	return status;
+}
+
+static struct snd_soc_codec_driver ab8500_codec_driver = {
+	.probe =		ab8500_codec_probe,
+	.read =			ab8500_codec_read_reg,
+	.write =		ab8500_codec_write_reg,
+	.reg_word_size =	sizeof(u8),
+	.controls =		ab8500_ctrls,
+	.num_controls =		ARRAY_SIZE(ab8500_ctrls),
+	.dapm_widgets =		ab8500_dapm_widgets,
+	.num_dapm_widgets =	ARRAY_SIZE(ab8500_dapm_widgets),
+	.dapm_routes =		ab8500_dapm_routes,
+	.num_dapm_routes =	ARRAY_SIZE(ab8500_dapm_routes),
+};
+
+static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev)
+{
+	int status;
+	struct ab8500_codec_drvdata *drvdata;
+
+	dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+	/* Create driver private-data struct */
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata),
+			GFP_KERNEL);
+	drvdata->sid_status = SID_UNCONFIGURED;
+	drvdata->anc_status = ANC_UNCONFIGURED;
+	dev_set_drvdata(&pdev->dev, drvdata);
+
+	dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__);
+	status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver,
+				ab8500_codec_dai,
+				ARRAY_SIZE(ab8500_codec_dai));
+	if (status < 0)
+		dev_err(&pdev->dev,
+			"%s: Error: Failed to register codec (%d).\n",
+			__func__, status);
+
+	return status;
+}
+
+static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev)
+{
+	dev_info(&pdev->dev, "%s Enter.\n", __func__);
+
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver ab8500_codec_platform_driver = {
+	.driver	= {
+		.name	= "ab8500-codec",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ab8500_codec_driver_probe,
+	.remove		= __devexit_p(ab8500_codec_driver_remove),
+	.suspend	= NULL,
+	.resume		= NULL,
+};
+module_platform_driver(ab8500_codec_platform_driver);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h
new file mode 100644
index 0000000..114f69a
--- /dev/null
+++ b/sound/soc/codecs/ab8500-codec.h
@@ -0,0 +1,590 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>,
+ *         for ST-Ericsson.
+ *
+ *         Based on the early work done by:
+ *         Mikko J. Lehto <mikko.lehto@symbio.com>,
+ *         Mikko Sarmanne <mikko.sarmanne@symbio.com>,
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef AB8500_CODEC_REGISTERS_H
+#define AB8500_CODEC_REGISTERS_H
+
+#define AB8500_SUPPORTED_RATE			(SNDRV_PCM_RATE_48000)
+#define AB8500_SUPPORTED_FMT			(SNDRV_PCM_FMTBIT_S16_LE)
+
+/* AB8500 audio bank (0x0d) register definitions */
+
+#define AB8500_POWERUP				0x00
+#define AB8500_AUDSWRESET			0x01
+#define AB8500_ADPATHENA			0x02
+#define AB8500_DAPATHENA			0x03
+#define AB8500_ANACONF1				0x04
+#define AB8500_ANACONF2				0x05
+#define AB8500_DIGMICCONF			0x06
+#define AB8500_ANACONF3				0x07
+#define AB8500_ANACONF4				0x08
+#define AB8500_DAPATHCONF			0x09
+#define AB8500_MUTECONF				0x0A
+#define AB8500_SHORTCIRCONF			0x0B
+#define AB8500_ANACONF5				0x0C
+#define AB8500_ENVCPCONF			0x0D
+#define AB8500_SIGENVCONF			0x0E
+#define AB8500_PWMGENCONF1			0x0F
+#define AB8500_PWMGENCONF2			0x10
+#define AB8500_PWMGENCONF3			0x11
+#define AB8500_PWMGENCONF4			0x12
+#define AB8500_PWMGENCONF5			0x13
+#define AB8500_ANAGAIN1				0x14
+#define AB8500_ANAGAIN2				0x15
+#define AB8500_ANAGAIN3				0x16
+#define AB8500_ANAGAIN4				0x17
+#define AB8500_DIGLINHSLGAIN			0x18
+#define AB8500_DIGLINHSRGAIN			0x19
+#define AB8500_ADFILTCONF			0x1A
+#define AB8500_DIGIFCONF1			0x1B
+#define AB8500_DIGIFCONF2			0x1C
+#define AB8500_DIGIFCONF3			0x1D
+#define AB8500_DIGIFCONF4			0x1E
+#define AB8500_ADSLOTSEL1			0x1F
+#define AB8500_ADSLOTSEL2			0x20
+#define AB8500_ADSLOTSEL3			0x21
+#define AB8500_ADSLOTSEL4			0x22
+#define AB8500_ADSLOTSEL5			0x23
+#define AB8500_ADSLOTSEL6			0x24
+#define AB8500_ADSLOTSEL7			0x25
+#define AB8500_ADSLOTSEL8			0x26
+#define AB8500_ADSLOTSEL9			0x27
+#define AB8500_ADSLOTSEL10			0x28
+#define AB8500_ADSLOTSEL11			0x29
+#define AB8500_ADSLOTSEL12			0x2A
+#define AB8500_ADSLOTSEL13			0x2B
+#define AB8500_ADSLOTSEL14			0x2C
+#define AB8500_ADSLOTSEL15			0x2D
+#define AB8500_ADSLOTSEL16			0x2E
+#define AB8500_ADSLOTHIZCTRL1			0x2F
+#define AB8500_ADSLOTHIZCTRL2			0x30
+#define AB8500_ADSLOTHIZCTRL3			0x31
+#define AB8500_ADSLOTHIZCTRL4			0x32
+#define AB8500_DASLOTCONF1			0x33
+#define AB8500_DASLOTCONF2			0x34
+#define AB8500_DASLOTCONF3			0x35
+#define AB8500_DASLOTCONF4			0x36
+#define AB8500_DASLOTCONF5			0x37
+#define AB8500_DASLOTCONF6			0x38
+#define AB8500_DASLOTCONF7			0x39
+#define AB8500_DASLOTCONF8			0x3A
+#define AB8500_CLASSDCONF1			0x3B
+#define AB8500_CLASSDCONF2			0x3C
+#define AB8500_CLASSDCONF3			0x3D
+#define AB8500_DMICFILTCONF			0x3E
+#define AB8500_DIGMULTCONF1			0x3F
+#define AB8500_DIGMULTCONF2			0x40
+#define AB8500_ADDIGGAIN1			0x41
+#define AB8500_ADDIGGAIN2			0x42
+#define AB8500_ADDIGGAIN3			0x43
+#define AB8500_ADDIGGAIN4			0x44
+#define AB8500_ADDIGGAIN5			0x45
+#define AB8500_ADDIGGAIN6			0x46
+#define AB8500_DADIGGAIN1			0x47
+#define AB8500_DADIGGAIN2			0x48
+#define AB8500_DADIGGAIN3			0x49
+#define AB8500_DADIGGAIN4			0x4A
+#define AB8500_DADIGGAIN5			0x4B
+#define AB8500_DADIGGAIN6			0x4C
+#define AB8500_ADDIGLOOPGAIN1			0x4D
+#define AB8500_ADDIGLOOPGAIN2			0x4E
+#define AB8500_HSLEARDIGGAIN			0x4F
+#define AB8500_HSRDIGGAIN			0x50
+#define AB8500_SIDFIRGAIN1			0x51
+#define AB8500_SIDFIRGAIN2			0x52
+#define AB8500_ANCCONF1				0x53
+#define AB8500_ANCCONF2				0x54
+#define AB8500_ANCCONF3				0x55
+#define AB8500_ANCCONF4				0x56
+#define AB8500_ANCCONF5				0x57
+#define AB8500_ANCCONF6				0x58
+#define AB8500_ANCCONF7				0x59
+#define AB8500_ANCCONF8				0x5A
+#define AB8500_ANCCONF9				0x5B
+#define AB8500_ANCCONF10			0x5C
+#define AB8500_ANCCONF11			0x5D
+#define AB8500_ANCCONF12			0x5E
+#define AB8500_ANCCONF13			0x5F
+#define AB8500_ANCCONF14			0x60
+#define AB8500_SIDFIRADR			0x61
+#define AB8500_SIDFIRCOEF1			0x62
+#define AB8500_SIDFIRCOEF2			0x63
+#define AB8500_SIDFIRCONF			0x64
+#define AB8500_AUDINTMASK1			0x65
+#define AB8500_AUDINTSOURCE1			0x66
+#define AB8500_AUDINTMASK2			0x67
+#define AB8500_AUDINTSOURCE2			0x68
+#define AB8500_FIFOCONF1			0x69
+#define AB8500_FIFOCONF2			0x6A
+#define AB8500_FIFOCONF3			0x6B
+#define AB8500_FIFOCONF4			0x6C
+#define AB8500_FIFOCONF5			0x6D
+#define AB8500_FIFOCONF6			0x6E
+#define AB8500_AUDREV				0x6F
+
+#define AB8500_FIRST_REG			AB8500_POWERUP
+#define AB8500_LAST_REG				AB8500_AUDREV
+#define AB8500_CACHEREGNUM			(AB8500_LAST_REG + 1)
+
+#define AB8500_MASK_ALL				0xFF
+#define AB8500_MASK_NONE			0x00
+
+/* AB8500_POWERUP */
+#define AB8500_POWERUP_POWERUP			7
+#define AB8500_POWERUP_ENANA			3
+
+/* AB8500_AUDSWRESET */
+#define AB8500_AUDSWRESET_SWRESET		7
+
+/* AB8500_ADPATHENA */
+#define AB8500_ADPATHENA_ENAD12			7
+#define AB8500_ADPATHENA_ENAD34			5
+#define AB8500_ADPATHENA_ENAD5768		3
+
+/* AB8500_DAPATHENA */
+#define AB8500_DAPATHENA_ENDA1			7
+#define AB8500_DAPATHENA_ENDA2			6
+#define AB8500_DAPATHENA_ENDA3			5
+#define AB8500_DAPATHENA_ENDA4			4
+#define AB8500_DAPATHENA_ENDA5			3
+#define AB8500_DAPATHENA_ENDA6			2
+
+/* AB8500_ANACONF1 */
+#define AB8500_ANACONF1_HSLOWPOW		7
+#define AB8500_ANACONF1_DACLOWPOW1		6
+#define AB8500_ANACONF1_DACLOWPOW0		5
+#define AB8500_ANACONF1_EARDACLOWPOW		4
+#define AB8500_ANACONF1_EARSELCM		2
+#define AB8500_ANACONF1_HSHPEN			1
+#define AB8500_ANACONF1_EARDRVLOWPOW		0
+
+/* AB8500_ANACONF2 */
+#define AB8500_ANACONF2_ENMIC1			7
+#define AB8500_ANACONF2_ENMIC2			6
+#define AB8500_ANACONF2_ENLINL			5
+#define AB8500_ANACONF2_ENLINR			4
+#define AB8500_ANACONF2_MUTMIC1			3
+#define AB8500_ANACONF2_MUTMIC2			2
+#define AB8500_ANACONF2_MUTLINL			1
+#define AB8500_ANACONF2_MUTLINR			0
+
+/* AB8500_DIGMICCONF */
+#define AB8500_DIGMICCONF_ENDMIC1		7
+#define AB8500_DIGMICCONF_ENDMIC2		6
+#define AB8500_DIGMICCONF_ENDMIC3		5
+#define AB8500_DIGMICCONF_ENDMIC4		4
+#define AB8500_DIGMICCONF_ENDMIC5		3
+#define AB8500_DIGMICCONF_ENDMIC6		2
+#define AB8500_DIGMICCONF_HSFADSPEED		0
+
+/* AB8500_ANACONF3 */
+#define AB8500_ANACONF3_MIC1SEL			7
+#define AB8500_ANACONF3_LINRSEL			6
+#define AB8500_ANACONF3_ENDRVHSL		5
+#define AB8500_ANACONF3_ENDRVHSR		4
+#define AB8500_ANACONF3_ENADCMIC		2
+#define AB8500_ANACONF3_ENADCLINL		1
+#define AB8500_ANACONF3_ENADCLINR		0
+
+/* AB8500_ANACONF4 */
+#define AB8500_ANACONF4_DISPDVSS		7
+#define AB8500_ANACONF4_ENEAR			6
+#define AB8500_ANACONF4_ENHSL			5
+#define AB8500_ANACONF4_ENHSR			4
+#define AB8500_ANACONF4_ENHFL			3
+#define AB8500_ANACONF4_ENHFR			2
+#define AB8500_ANACONF4_ENVIB1			1
+#define AB8500_ANACONF4_ENVIB2			0
+
+/* AB8500_DAPATHCONF */
+#define AB8500_DAPATHCONF_ENDACEAR		6
+#define AB8500_DAPATHCONF_ENDACHSL		5
+#define AB8500_DAPATHCONF_ENDACHSR		4
+#define AB8500_DAPATHCONF_ENDACHFL		3
+#define AB8500_DAPATHCONF_ENDACHFR		2
+#define AB8500_DAPATHCONF_ENDACVIB1		1
+#define AB8500_DAPATHCONF_ENDACVIB2		0
+
+/* AB8500_MUTECONF */
+#define AB8500_MUTECONF_MUTEAR			6
+#define AB8500_MUTECONF_MUTHSL			5
+#define AB8500_MUTECONF_MUTHSR			4
+#define AB8500_MUTECONF_MUTDACEAR		2
+#define AB8500_MUTECONF_MUTDACHSL		1
+#define AB8500_MUTECONF_MUTDACHSR		0
+
+/* AB8500_SHORTCIRCONF */
+#define AB8500_SHORTCIRCONF_ENSHORTPWD		7
+#define AB8500_SHORTCIRCONF_EARSHORTDIS		6
+#define AB8500_SHORTCIRCONF_HSSHORTDIS		5
+#define AB8500_SHORTCIRCONF_HSPULLDEN		4
+#define AB8500_SHORTCIRCONF_HSOSCEN		2
+#define AB8500_SHORTCIRCONF_HSFADDIS		1
+#define AB8500_SHORTCIRCONF_HSZCDDIS		0
+/* Zero cross should be disabled */
+
+/* AB8500_ANACONF5 */
+#define AB8500_ANACONF5_ENCPHS			7
+#define AB8500_ANACONF5_HSLDACTOLOL		5
+#define AB8500_ANACONF5_HSRDACTOLOR		4
+#define AB8500_ANACONF5_ENLOL			3
+#define AB8500_ANACONF5_ENLOR			2
+#define AB8500_ANACONF5_HSAUTOEN		0
+
+/* AB8500_ENVCPCONF */
+#define AB8500_ENVCPCONF_ENVDETHTHRE		4
+#define AB8500_ENVCPCONF_ENVDETLTHRE		0
+#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX	0x0F
+#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX	0x0F
+
+/* AB8500_SIGENVCONF */
+#define AB8500_SIGENVCONF_CPLVEN		5
+#define AB8500_SIGENVCONF_ENVDETCPEN		4
+#define AB8500_SIGENVCONF_ENVDETTIME		0
+#define AB8500_SIGENVCONF_ENVDETTIME_MAX	0x0F
+
+/* AB8500_PWMGENCONF1 */
+#define AB8500_PWMGENCONF1_PWMTOVIB1		7
+#define AB8500_PWMGENCONF1_PWMTOVIB2		6
+#define AB8500_PWMGENCONF1_PWM1CTRL		5
+#define AB8500_PWMGENCONF1_PWM2CTRL		4
+#define AB8500_PWMGENCONF1_PWM1NCTRL		3
+#define AB8500_PWMGENCONF1_PWM1PCTRL		2
+#define AB8500_PWMGENCONF1_PWM2NCTRL		1
+#define AB8500_PWMGENCONF1_PWM2PCTRL		0
+
+/* AB8500_PWMGENCONF2 */
+/* AB8500_PWMGENCONF3 */
+/* AB8500_PWMGENCONF4 */
+/* AB8500_PWMGENCONF5 */
+#define AB8500_PWMGENCONFX_PWMVIBXPOL		7
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC	0
+#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX	0x64
+
+/* AB8500_ANAGAIN1 */
+/* AB8500_ANAGAIN2 */
+#define AB8500_ANAGAINX_ENSEMICX		7
+#define AB8500_ANAGAINX_LOWPOWMICX		6
+#define AB8500_ANAGAINX_MICXGAIN		0
+#define AB8500_ANAGAINX_MICXGAIN_MAX		0x1F
+
+/* AB8500_ANAGAIN3 */
+#define AB8500_ANAGAIN3_HSLGAIN			4
+#define AB8500_ANAGAIN3_HSRGAIN			0
+#define AB8500_ANAGAIN3_HSXGAIN_MAX		0x0F
+
+/* AB8500_ANAGAIN4 */
+#define AB8500_ANAGAIN4_LINLGAIN		4
+#define AB8500_ANAGAIN4_LINRGAIN		0
+#define AB8500_ANAGAIN4_LINXGAIN_MAX		0x0F
+
+/* AB8500_DIGLINHSLGAIN */
+/* AB8500_DIGLINHSRGAIN */
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN	0
+#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX	0x13
+
+/* AB8500_ADFILTCONF */
+#define AB8500_ADFILTCONF_AD1NH			7
+#define AB8500_ADFILTCONF_AD2NH			6
+#define AB8500_ADFILTCONF_AD3NH			5
+#define AB8500_ADFILTCONF_AD4NH			4
+#define AB8500_ADFILTCONF_AD1VOICE		3
+#define AB8500_ADFILTCONF_AD2VOICE		2
+#define AB8500_ADFILTCONF_AD3VOICE		1
+#define AB8500_ADFILTCONF_AD4VOICE		0
+
+/* AB8500_DIGIFCONF1 */
+#define AB8500_DIGIFCONF1_ENMASTGEN		7
+#define AB8500_DIGIFCONF1_IF1BITCLKOS1		6
+#define AB8500_DIGIFCONF1_IF1BITCLKOS0		5
+#define AB8500_DIGIFCONF1_ENFSBITCLK1		4
+#define AB8500_DIGIFCONF1_IF0BITCLKOS1		2
+#define AB8500_DIGIFCONF1_IF0BITCLKOS0		1
+#define AB8500_DIGIFCONF1_ENFSBITCLK0		0
+
+/* AB8500_DIGIFCONF2 */
+#define AB8500_DIGIFCONF2_FSYNC0P		6
+#define AB8500_DIGIFCONF2_BITCLK0P		5
+#define AB8500_DIGIFCONF2_IF0DEL		4
+#define AB8500_DIGIFCONF2_IF0FORMAT1		3
+#define AB8500_DIGIFCONF2_IF0FORMAT0		2
+#define AB8500_DIGIFCONF2_IF0WL1		1
+#define AB8500_DIGIFCONF2_IF0WL0		0
+
+/* AB8500_DIGIFCONF3 */
+#define AB8500_DIGIFCONF3_IF0DATOIF1AD		7
+#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK	6
+#define AB8500_DIGIFCONF3_IF1MASTER		5
+#define AB8500_DIGIFCONF3_IF1DATOIF0AD		3
+#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK	2
+#define AB8500_DIGIFCONF3_IF0MASTER		1
+#define AB8500_DIGIFCONF3_IF0BFIFOEN		0
+
+/* AB8500_DIGIFCONF4 */
+#define AB8500_DIGIFCONF4_FSYNC1P		6
+#define AB8500_DIGIFCONF4_BITCLK1P		5
+#define AB8500_DIGIFCONF4_IF1DEL		4
+#define AB8500_DIGIFCONF4_IF1FORMAT1		3
+#define AB8500_DIGIFCONF4_IF1FORMAT0		2
+#define AB8500_DIGIFCONF4_IF1WL1		1
+#define AB8500_DIGIFCONF4_IF1WL0		0
+
+/* AB8500_ADSLOTSELX */
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD	0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD	0x01
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD	0x02
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD	0x03
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD	0x04
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD	0x05
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD	0x06
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD	0x07
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD	0x08
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD	0x0F
+#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN	0x00
+#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN	0x10
+#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN	0x20
+#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN	0x30
+#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN	0x40
+#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN	0x50
+#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN	0x60
+#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN	0x70
+#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN	0x80
+#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN	0xF0
+#define AB8500_ADSLOTSELX_EVEN_SHIFT		0
+#define AB8500_ADSLOTSELX_ODD_SHIFT		4
+
+/* AB8500_ADSLOTHIZCTRL1 */
+/* AB8500_ADSLOTHIZCTRL2 */
+/* AB8500_ADSLOTHIZCTRL3 */
+/* AB8500_ADSLOTHIZCTRL4 */
+/* AB8500_DASLOTCONF1 */
+#define AB8500_DASLOTCONF1_DA12VOICE		7
+#define AB8500_DASLOTCONF1_SWAPDA12_34		6
+#define AB8500_DASLOTCONF1_DAI7TOADO1		5
+
+/* AB8500_DASLOTCONF2 */
+#define AB8500_DASLOTCONF2_DAI8TOADO2		5
+
+/* AB8500_DASLOTCONF3 */
+#define AB8500_DASLOTCONF3_DA34VOICE		7
+#define AB8500_DASLOTCONF3_DAI7TOADO3		5
+
+/* AB8500_DASLOTCONF4 */
+#define AB8500_DASLOTCONF4_DAI8TOADO4		5
+
+/* AB8500_DASLOTCONF5 */
+#define AB8500_DASLOTCONF5_DA56VOICE		7
+#define AB8500_DASLOTCONF5_DAI7TOADO5		5
+
+/* AB8500_DASLOTCONF6 */
+#define AB8500_DASLOTCONF6_DAI8TOADO6		5
+
+/* AB8500_DASLOTCONF7 */
+#define AB8500_DASLOTCONF7_DAI8TOADO7		5
+
+/* AB8500_DASLOTCONF8 */
+#define AB8500_DASLOTCONF8_DAI7TOADO8		5
+
+#define AB8500_DASLOTCONFX_SLTODAX_SHIFT	0
+#define AB8500_DASLOTCONFX_SLTODAX_MASK		0x1F
+
+/* AB8500_CLASSDCONF1 */
+#define AB8500_CLASSDCONF1_PARLHF		7
+#define AB8500_CLASSDCONF1_PARLVIB		6
+#define AB8500_CLASSDCONF1_VIB1SWAPEN		3
+#define AB8500_CLASSDCONF1_VIB2SWAPEN		2
+#define AB8500_CLASSDCONF1_HFLSWAPEN		1
+#define AB8500_CLASSDCONF1_HFRSWAPEN		0
+
+/* AB8500_CLASSDCONF2 */
+#define AB8500_CLASSDCONF2_FIRBYP3		7
+#define AB8500_CLASSDCONF2_FIRBYP2		6
+#define AB8500_CLASSDCONF2_FIRBYP1		5
+#define AB8500_CLASSDCONF2_FIRBYP0		4
+#define AB8500_CLASSDCONF2_HIGHVOLEN3		3
+#define AB8500_CLASSDCONF2_HIGHVOLEN2		2
+#define AB8500_CLASSDCONF2_HIGHVOLEN1		1
+#define AB8500_CLASSDCONF2_HIGHVOLEN0		0
+
+/* AB8500_CLASSDCONF3 */
+#define AB8500_CLASSDCONF3_DITHHPGAIN		4
+#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX	0x0A
+#define AB8500_CLASSDCONF3_DITHWGAIN		0
+#define AB8500_CLASSDCONF3_DITHWGAIN_MAX	0x0A
+
+/* AB8500_DMICFILTCONF */
+#define AB8500_DMICFILTCONF_ANCINSEL		7
+#define AB8500_DMICFILTCONF_DA3TOEAR		6
+#define AB8500_DMICFILTCONF_DMIC1SINC3		5
+#define AB8500_DMICFILTCONF_DMIC2SINC3		4
+#define AB8500_DMICFILTCONF_DMIC3SINC3		3
+#define AB8500_DMICFILTCONF_DMIC4SINC3		2
+#define AB8500_DMICFILTCONF_DMIC5SINC3		1
+#define AB8500_DMICFILTCONF_DMIC6SINC3		0
+
+/* AB8500_DIGMULTCONF1 */
+#define AB8500_DIGMULTCONF1_DATOHSLEN		7
+#define AB8500_DIGMULTCONF1_DATOHSREN		6
+#define AB8500_DIGMULTCONF1_AD1SEL		5
+#define AB8500_DIGMULTCONF1_AD2SEL		4
+#define AB8500_DIGMULTCONF1_AD3SEL		3
+#define AB8500_DIGMULTCONF1_AD5SEL		2
+#define AB8500_DIGMULTCONF1_AD6SEL		1
+#define AB8500_DIGMULTCONF1_ANCSEL		0
+
+/* AB8500_DIGMULTCONF2 */
+#define AB8500_DIGMULTCONF2_DATOHFREN		7
+#define AB8500_DIGMULTCONF2_DATOHFLEN		6
+#define AB8500_DIGMULTCONF2_HFRSEL		5
+#define AB8500_DIGMULTCONF2_HFLSEL		4
+#define AB8500_DIGMULTCONF2_FIRSID1SEL		2
+#define AB8500_DIGMULTCONF2_FIRSID2SEL		0
+
+/* AB8500_ADDIGGAIN1 */
+/* AB8500_ADDIGGAIN2 */
+/* AB8500_ADDIGGAIN3 */
+/* AB8500_ADDIGGAIN4 */
+/* AB8500_ADDIGGAIN5 */
+/* AB8500_ADDIGGAIN6 */
+#define AB8500_ADDIGGAINX_FADEDISADX		6
+#define AB8500_ADDIGGAINX_ADXGAIN_MAX		0x3F
+
+/* AB8500_DADIGGAIN1 */
+/* AB8500_DADIGGAIN2 */
+/* AB8500_DADIGGAIN3 */
+/* AB8500_DADIGGAIN4 */
+/* AB8500_DADIGGAIN5 */
+/* AB8500_DADIGGAIN6 */
+#define AB8500_DADIGGAINX_FADEDISDAX		6
+#define AB8500_DADIGGAINX_DAXGAIN_MAX		0x3F
+
+/* AB8500_ADDIGLOOPGAIN1 */
+/* AB8500_ADDIGLOOPGAIN2 */
+#define AB8500_ADDIGLOOPGAINX_FADEDISADXL	6
+#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX	0x3F
+
+/* AB8500_HSLEARDIGGAIN */
+#define AB8500_HSLEARDIGGAIN_HSSINC1		7
+#define AB8500_HSLEARDIGGAIN_FADEDISHSL		4
+#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX	0x09
+
+/* AB8500_HSRDIGGAIN */
+#define AB8500_HSRDIGGAIN_FADESPEED		6
+#define AB8500_HSRDIGGAIN_FADEDISHSR		4
+#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX		0x09
+
+/* AB8500_SIDFIRGAIN1 */
+/* AB8500_SIDFIRGAIN2 */
+#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX	0x1F
+
+/* AB8500_ANCCONF1 */
+#define AB8500_ANCCONF1_ANCIIRUPDATE		3
+#define AB8500_ANCCONF1_ENANC			2
+#define AB8500_ANCCONF1_ANCIIRINIT		1
+#define AB8500_ANCCONF1_ANCFIRUPDATE		0
+
+/* AB8500_ANCCONF2 */
+#define AB8500_ANCCONF2_SHIFT			5
+#define AB8500_ANCCONF2_MIN			-0x10
+#define AB8500_ANCCONF2_MAX			0xF
+
+/* AB8500_ANCCONF3 */
+#define AB8500_ANCCONF3_SHIFT			5
+#define AB8500_ANCCONF3_MIN			-0x10
+#define AB8500_ANCCONF3_MAX			0xF
+
+/* AB8500_ANCCONF4 */
+#define AB8500_ANCCONF4_SHIFT			5
+#define AB8500_ANCCONF4_MIN			-0x10
+#define AB8500_ANCCONF4_MAX			0xF
+
+/* AB8500_ANC_FIR_COEFFS */
+#define AB8500_ANC_FIR_COEFF_MIN		-0x8000
+#define AB8500_ANC_FIR_COEFF_MAX		0x7FFF
+#define AB8500_ANC_FIR_COEFFS			15
+
+/* AB8500_ANC_IIR_COEFFS */
+#define AB8500_ANC_IIR_COEFF_MIN		-0x800000
+#define AB8500_ANC_IIR_COEFF_MAX		0x7FFFFF
+#define AB8500_ANC_IIR_COEFFS			24
+/* AB8500_ANC_WARP_DELAY */
+#define AB8500_ANC_WARP_DELAY_SHIFT		16
+#define AB8500_ANC_WARP_DELAY_MIN		0x0000
+#define AB8500_ANC_WARP_DELAY_MAX		0xFFFF
+
+/* AB8500_ANCCONF11 */
+/* AB8500_ANCCONF12 */
+/* AB8500_ANCCONF13 */
+/* AB8500_ANCCONF14 */
+
+/* AB8500_SIDFIRADR */
+#define AB8500_SIDFIRADR_FIRSIDSET		7
+#define AB8500_SIDFIRADR_ADDRESS_SHIFT		0
+#define AB8500_SIDFIRADR_ADDRESS_MAX		0x7F
+
+/* AB8500_SIDFIRCOEF1 */
+/* AB8500_SIDFIRCOEF2 */
+#define AB8500_SID_FIR_COEFF_MIN		0
+#define AB8500_SID_FIR_COEFF_MAX		0xFFFF
+#define AB8500_SID_FIR_COEFFS			128
+
+/* AB8500_SIDFIRCONF */
+#define AB8500_SIDFIRCONF_ENFIRSIDS		2
+#define AB8500_SIDFIRCONF_FIRSIDSTOIF1		1
+#define AB8500_SIDFIRCONF_FIRSIDBUSY		0
+
+/* AB8500_AUDINTMASK1 */
+/* AB8500_AUDINTSOURCE1 */
+/* AB8500_AUDINTMASK2 */
+/* AB8500_AUDINTSOURCE2 */
+
+/* AB8500_FIFOCONF1 */
+#define AB8500_FIFOCONF1_BFIFOMASK		0x80
+#define AB8500_FIFOCONF1_BFIFO19M2		0x40
+#define AB8500_FIFOCONF1_BFIFOINT_SHIFT		0
+#define AB8500_FIFOCONF1_BFIFOINT_MAX		0x3F
+
+/* AB8500_FIFOCONF2 */
+#define AB8500_FIFOCONF2_BFIFOTX_SHIFT		0
+#define AB8500_FIFOCONF2_BFIFOTX_MAX		0xFF
+
+/* AB8500_FIFOCONF3 */
+#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT	5
+#define AB8500_FIFOCONF3_BFIFOEXSL_MAX		0x5
+#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT	2
+#define AB8500_FIFOCONF3_PREBITCLK0_MAX		0x7
+#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT	1
+#define AB8500_FIFOCONF3_BFIFORUN_SHIFT		0
+
+/* AB8500_FIFOCONF4 */
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT	0
+#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX	0xFF
+
+/* AB8500_FIFOCONF5 */
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT	0
+#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX	0xFF
+
+/* AB8500_FIFOCONF6 */
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT	0
+#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX	0xFF
+
+/* AB8500_AUDREV */
+
+#endif
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 2023c74..ea06b83 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -91,11 +91,6 @@
 	return 0;
 }
 
-static int ac97_soc_remove(struct snd_soc_codec *codec)
-{
-	return 0;
-}
-
 #ifdef CONFIG_PM
 static int ac97_soc_suspend(struct snd_soc_codec *codec)
 {
@@ -119,7 +114,6 @@
 	.write =	ac97_write,
 	.read =		ac97_read,
 	.probe = 	ac97_soc_probe,
-	.remove = 	ac97_soc_remove,
 	.suspend =	ac97_soc_suspend,
 	.resume =	ac97_soc_resume,
 };
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
new file mode 100644
index 0000000..5c9caca
--- /dev/null
+++ b/sound/soc/codecs/arizona.c
@@ -0,0 +1,937 @@
+/*
+ * arizona.c - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/gcd.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+
+#define ARIZONA_AIF_BCLK_CTRL                   0x00
+#define ARIZONA_AIF_TX_PIN_CTRL                 0x01
+#define ARIZONA_AIF_RX_PIN_CTRL                 0x02
+#define ARIZONA_AIF_RATE_CTRL                   0x03
+#define ARIZONA_AIF_FORMAT                      0x04
+#define ARIZONA_AIF_TX_BCLK_RATE                0x05
+#define ARIZONA_AIF_RX_BCLK_RATE                0x06
+#define ARIZONA_AIF_FRAME_CTRL_1                0x07
+#define ARIZONA_AIF_FRAME_CTRL_2                0x08
+#define ARIZONA_AIF_FRAME_CTRL_3                0x09
+#define ARIZONA_AIF_FRAME_CTRL_4                0x0A
+#define ARIZONA_AIF_FRAME_CTRL_5                0x0B
+#define ARIZONA_AIF_FRAME_CTRL_6                0x0C
+#define ARIZONA_AIF_FRAME_CTRL_7                0x0D
+#define ARIZONA_AIF_FRAME_CTRL_8                0x0E
+#define ARIZONA_AIF_FRAME_CTRL_9                0x0F
+#define ARIZONA_AIF_FRAME_CTRL_10               0x10
+#define ARIZONA_AIF_FRAME_CTRL_11               0x11
+#define ARIZONA_AIF_FRAME_CTRL_12               0x12
+#define ARIZONA_AIF_FRAME_CTRL_13               0x13
+#define ARIZONA_AIF_FRAME_CTRL_14               0x14
+#define ARIZONA_AIF_FRAME_CTRL_15               0x15
+#define ARIZONA_AIF_FRAME_CTRL_16               0x16
+#define ARIZONA_AIF_FRAME_CTRL_17               0x17
+#define ARIZONA_AIF_FRAME_CTRL_18               0x18
+#define ARIZONA_AIF_TX_ENABLES                  0x19
+#define ARIZONA_AIF_RX_ENABLES                  0x1A
+#define ARIZONA_AIF_FORCE_WRITE                 0x1B
+
+#define arizona_fll_err(_fll, fmt, ...) \
+	dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_warn(_fll, fmt, ...) \
+	dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+#define arizona_fll_dbg(_fll, fmt, ...) \
+	dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__)
+
+#define arizona_aif_err(_dai, fmt, ...) \
+	dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_warn(_dai, fmt, ...) \
+	dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+#define arizona_aif_dbg(_dai, fmt, ...) \
+	dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
+
+const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
+	"None",
+	"Tone Generator 1",
+	"Tone Generator 2",
+	"Haptics",
+	"AEC",
+	"Mic Mute Mixer",
+	"Noise Generator",
+	"IN1L",
+	"IN1R",
+	"IN2L",
+	"IN2R",
+	"IN3L",
+	"IN3R",
+	"IN4L",
+	"IN4R",
+	"AIF1RX1",
+	"AIF1RX2",
+	"AIF1RX3",
+	"AIF1RX4",
+	"AIF1RX5",
+	"AIF1RX6",
+	"AIF1RX7",
+	"AIF1RX8",
+	"AIF2RX1",
+	"AIF2RX2",
+	"AIF3RX1",
+	"AIF3RX2",
+	"SLIMRX1",
+	"SLIMRX2",
+	"SLIMRX3",
+	"SLIMRX4",
+	"SLIMRX5",
+	"SLIMRX6",
+	"SLIMRX7",
+	"SLIMRX8",
+	"EQ1",
+	"EQ2",
+	"EQ3",
+	"EQ4",
+	"DRC1L",
+	"DRC1R",
+	"DRC2L",
+	"DRC2R",
+	"LHPF1",
+	"LHPF2",
+	"LHPF3",
+	"LHPF4",
+	"DSP1.1",
+	"DSP1.2",
+	"DSP1.3",
+	"DSP1.4",
+	"DSP1.5",
+	"DSP1.6",
+	"ASRC1L",
+	"ASRC1R",
+	"ASRC2L",
+	"ASRC2R",
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_texts);
+
+int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
+	0x00,  /* None */
+	0x04,  /* Tone */
+	0x05,
+	0x06,  /* Haptics */
+	0x08,  /* AEC */
+	0x0c,  /* Noise mixer */
+	0x0d,  /* Comfort noise */
+	0x10,  /* IN1L */
+	0x11,
+	0x12,
+	0x13,
+	0x14,
+	0x15,
+	0x16,
+	0x17,
+	0x20,  /* AIF1RX1 */
+	0x21,
+	0x22,
+	0x23,
+	0x24,
+	0x25,
+	0x26,
+	0x27,
+	0x28,  /* AIF2RX1 */
+	0x29,
+	0x30,  /* AIF3RX1 */
+	0x31,
+	0x38,  /* SLIMRX1 */
+	0x39,
+	0x3a,
+	0x3b,
+	0x3c,
+	0x3d,
+	0x3e,
+	0x3f,
+	0x50,  /* EQ1 */
+	0x51,
+	0x52,
+	0x53,
+	0x58,  /* DRC1L */
+	0x59,
+	0x5a,
+	0x5b,
+	0x60,  /* LHPF1 */
+	0x61,
+	0x62,
+	0x63,
+	0x68,  /* DSP1.1 */
+	0x69,
+	0x6a,
+	0x6b,
+	0x6c,
+	0x6d,
+	0x90,  /* ASRC1L */
+	0x91,
+	0x92,
+	0x93,
+};
+EXPORT_SYMBOL_GPL(arizona_mixer_values);
+
+const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
+EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+
+static const char *arizona_lhpf_mode_text[] = {
+	"Low-pass", "High-pass"
+};
+
+const struct soc_enum arizona_lhpf1_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf1_mode);
+
+const struct soc_enum arizona_lhpf2_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf2_mode);
+
+const struct soc_enum arizona_lhpf3_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf3_mode);
+
+const struct soc_enum arizona_lhpf4_mode =
+	SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2,
+			arizona_lhpf_mode_text);
+EXPORT_SYMBOL_GPL(arizona_lhpf4_mode);
+
+int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
+		  int event)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_in_ev);
+
+int arizona_out_ev(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol,
+		   int event)
+{
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_out_ev);
+
+int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+		       int source, unsigned int freq, int dir)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	char *name;
+	unsigned int reg;
+	unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK;
+	unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT;
+	unsigned int *clk;
+
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+		name = "SYSCLK";
+		reg = ARIZONA_SYSTEM_CLOCK_1;
+		clk = &priv->sysclk;
+		mask |= ARIZONA_SYSCLK_FRAC;
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		name = "ASYNCCLK";
+		reg = ARIZONA_ASYNC_CLOCK_1;
+		clk = &priv->asyncclk;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (freq) {
+	case  5644800:
+	case  6144000:
+		break;
+	case 11289600:
+	case 12288000:
+		val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	case 22579200:
+	case 24576000:
+		val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	case 45158400:
+	case 49152000:
+		val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	*clk = freq;
+
+	if (freq % 6144000)
+		val |= ARIZONA_SYSCLK_FRAC;
+
+	dev_dbg(arizona->dev, "%s set to %uHz", name, freq);
+
+	return regmap_update_bits(arizona->regmap, reg, mask, val);
+}
+EXPORT_SYMBOL_GPL(arizona_set_sysclk);
+
+static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	int lrclk, bclk, mode, base;
+
+	base = dai->driver->base;
+
+	lrclk = 0;
+	bclk = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		mode = 0;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		mode = 1;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		mode = 2;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = 3;
+		break;
+	default:
+		arizona_aif_err(dai, "Unsupported DAI format %d\n",
+				fmt & SND_SOC_DAIFMT_FORMAT_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBS_CFM:
+		lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		bclk |= ARIZONA_AIF1_BCLK_MSTR;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		bclk |= ARIZONA_AIF1_BCLK_MSTR;
+		lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR;
+		break;
+	default:
+		arizona_aif_err(dai, "Unsupported master mode %d\n",
+				fmt & SND_SOC_DAIFMT_MASTER_MASK);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		bclk |= ARIZONA_AIF1_BCLK_INV;
+		lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		bclk |= ARIZONA_AIF1_BCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		lrclk |= ARIZONA_AIF1TX_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+			    ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR,
+			    bclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL,
+			    ARIZONA_AIF1TX_LRCLK_INV |
+			    ARIZONA_AIF1TX_LRCLK_MSTR, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL,
+			    ARIZONA_AIF1RX_LRCLK_INV |
+			    ARIZONA_AIF1RX_LRCLK_MSTR, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT,
+			    ARIZONA_AIF1_FMT_MASK, mode);
+
+	return 0;
+}
+
+static const int arizona_48k_bclk_rates[] = {
+	-1,
+	48000,
+	64000,
+	96000,
+	128000,
+	192000,
+	256000,
+	384000,
+	512000,
+	768000,
+	1024000,
+	1536000,
+	2048000,
+	3072000,
+	4096000,
+	6144000,
+	8192000,
+	12288000,
+	24576000,
+};
+
+static const unsigned int arizona_48k_rates[] = {
+	12000,
+	24000,
+	48000,
+	96000,
+	192000,
+	384000,
+	768000,
+	4000,
+	8000,
+	16000,
+	32000,
+	64000,
+	128000,
+	256000,
+	512000,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = {
+	.count	= ARRAY_SIZE(arizona_48k_rates),
+	.list	= arizona_48k_rates,
+};
+
+static const int arizona_44k1_bclk_rates[] = {
+	-1,
+	44100,
+	58800,
+	88200,
+	117600,
+	177640,
+	235200,
+	352800,
+	470400,
+	705600,
+	940800,
+	1411200,
+	1881600,
+	2882400,
+	3763200,
+	5644800,
+	7526400,
+	11289600,
+	22579200,
+};
+
+static const unsigned int arizona_44k1_rates[] = {
+	11025,
+	22050,
+	44100,
+	88200,
+	176400,
+	352800,
+	705600,
+};
+
+static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = {
+	.count	= ARRAY_SIZE(arizona_44k1_rates),
+	.list	= arizona_44k1_rates,
+};
+
+static int arizona_sr_vals[] = {
+	0,
+	12000,
+	24000,
+	48000,
+	96000,
+	192000,
+	384000,
+	768000,
+	0,
+	11025,
+	22050,
+	44100,
+	88200,
+	176400,
+	352800,
+	705600,
+	4000,
+	8000,
+	16000,
+	32000,
+	64000,
+	128000,
+	256000,
+	512000,
+};
+
+static int arizona_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	const struct snd_pcm_hw_constraint_list *constraint;
+	unsigned int base_rate;
+
+	switch (dai_priv->clk) {
+	case ARIZONA_CLK_SYSCLK:
+		base_rate = priv->sysclk;
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		base_rate = priv->asyncclk;
+		break;
+	default:
+		return 0;
+	}
+
+	if (base_rate % 8000)
+		constraint = &arizona_44k1_constraint;
+	else
+		constraint = &arizona_48k_constraint;
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  constraint);
+}
+
+static int arizona_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *params,
+			     struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	int base = dai->driver->base;
+	const int *rates;
+	int i;
+	int bclk, lrclk, wl, frame, sr_val;
+
+	if (params_rate(params) % 8000)
+		rates = &arizona_44k1_bclk_rates[0];
+	else
+		rates = &arizona_48k_bclk_rates[0];
+
+	for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
+		if (rates[i] >= snd_soc_params_to_bclk(params) &&
+		    rates[i] % params_rate(params) == 0) {
+			bclk = i;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) {
+		arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+				params_rate(params));
+		return -EINVAL;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++)
+		if (arizona_sr_vals[i] == params_rate(params))
+			break;
+	if (i == ARRAY_SIZE(arizona_sr_vals)) {
+		arizona_aif_err(dai, "Unsupported sample rate %dHz\n",
+				params_rate(params));
+		return -EINVAL;
+	}
+	sr_val = i;
+
+	lrclk = snd_soc_params_to_bclk(params) / params_rate(params);
+
+	arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n",
+			rates[bclk], rates[bclk] / lrclk);
+
+	wl = snd_pcm_format_width(params_format(params));
+	frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl;
+
+	/*
+	 * We will need to be more flexible than this in future,
+	 * currently we use a single sample rate for SYSCLK.
+	 */
+	switch (dai_priv->clk) {
+	case ARIZONA_CLK_SYSCLK:
+		snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1,
+				    ARIZONA_SAMPLE_RATE_1_MASK, sr_val);
+		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+				    ARIZONA_AIF1_RATE_MASK, 0);
+		break;
+	case ARIZONA_CLK_ASYNCCLK:
+		snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1,
+				    ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val);
+		snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL,
+				    ARIZONA_AIF1_RATE_MASK, 8);
+		break;
+	default:
+		arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL,
+			    ARIZONA_AIF1_BCLK_FREQ_MASK, bclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE,
+			    ARIZONA_AIF1TX_BCPF_MASK, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE,
+			    ARIZONA_AIF1RX_BCPF_MASK, lrclk);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1,
+			    ARIZONA_AIF1TX_WL_MASK |
+			    ARIZONA_AIF1TX_SLOT_LEN_MASK, frame);
+	snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2,
+			    ARIZONA_AIF1RX_WL_MASK |
+			    ARIZONA_AIF1RX_SLOT_LEN_MASK, frame);
+
+	return 0;
+}
+
+static const char *arizona_dai_clk_str(int clk_id)
+{
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+		return "SYSCLK";
+	case ARIZONA_CLK_ASYNCCLK:
+		return "ASYNCCLK";
+	default:
+		return "Unknown clock";
+	}
+}
+
+static int arizona_dai_set_sysclk(struct snd_soc_dai *dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1];
+	struct snd_soc_dapm_route routes[2];
+
+	switch (clk_id) {
+	case ARIZONA_CLK_SYSCLK:
+	case ARIZONA_CLK_ASYNCCLK:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (clk_id == dai_priv->clk)
+		return 0;
+
+	if (dai->active) {
+		dev_err(codec->dev, "Can't change clock on active DAI %d\n",
+			dai->id);
+		return -EBUSY;
+	}
+
+	memset(&routes, 0, sizeof(routes));
+	routes[0].sink = dai->driver->capture.stream_name;
+	routes[1].sink = dai->driver->playback.stream_name;
+
+	routes[0].source = arizona_dai_clk_str(dai_priv->clk);
+	routes[1].source = arizona_dai_clk_str(dai_priv->clk);
+	snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+	routes[0].source = arizona_dai_clk_str(clk_id);
+	routes[1].source = arizona_dai_clk_str(clk_id);
+	snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes));
+
+	return snd_soc_dapm_sync(&codec->dapm);
+}
+
+const struct snd_soc_dai_ops arizona_dai_ops = {
+	.startup = arizona_startup,
+	.set_fmt = arizona_set_fmt,
+	.hw_params = arizona_hw_params,
+	.set_sysclk = arizona_dai_set_sysclk,
+};
+EXPORT_SYMBOL_GPL(arizona_dai_ops);
+
+int arizona_init_dai(struct arizona_priv *priv, int id)
+{
+	struct arizona_dai_priv *dai_priv = &priv->dai[id];
+
+	dai_priv->clk = ARIZONA_CLK_SYSCLK;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_dai);
+
+static irqreturn_t arizona_fll_lock(int irq, void *data)
+{
+	struct arizona_fll *fll = data;
+
+	arizona_fll_dbg(fll, "Locked\n");
+
+	complete(&fll->lock);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t arizona_fll_clock_ok(int irq, void *data)
+{
+	struct arizona_fll *fll = data;
+
+	arizona_fll_dbg(fll, "clock OK\n");
+
+	complete(&fll->ok);
+
+	return IRQ_HANDLED;
+}
+
+static struct {
+	unsigned int min;
+	unsigned int max;
+	u16 fratio;
+	int ratio;
+} fll_fratios[] = {
+	{       0,    64000, 4, 16 },
+	{   64000,   128000, 3,  8 },
+	{  128000,   256000, 2,  4 },
+	{  256000,  1000000, 1,  2 },
+	{ 1000000, 13500000, 0,  1 },
+};
+
+struct arizona_fll_cfg {
+	int n;
+	int theta;
+	int lambda;
+	int refdiv;
+	int outdiv;
+	int fratio;
+};
+
+static int arizona_calc_fll(struct arizona_fll *fll,
+			    struct arizona_fll_cfg *cfg,
+			    unsigned int Fref,
+			    unsigned int Fout)
+{
+	unsigned int target, div, gcd_fll;
+	int i, ratio;
+
+	arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout);
+
+	/* Fref must be <=13.5MHz */
+	div = 1;
+	cfg->refdiv = 0;
+	while ((Fref / div) > 13500000) {
+		div *= 2;
+		cfg->refdiv++;
+
+		if (div > 8) {
+			arizona_fll_err(fll,
+					"Can't scale %dMHz in to <=13.5MHz\n",
+					Fref);
+			return -EINVAL;
+		}
+	}
+
+	/* Apply the division for our remaining calculations */
+	Fref /= div;
+
+	/* Fvco should be over the targt; don't check the upper bound */
+	div = 1;
+	while (Fout * div < 90000000 * fll->vco_mult) {
+		div++;
+		if (div > 7) {
+			arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n",
+					Fout);
+			return -EINVAL;
+		}
+	}
+	target = Fout * div / fll->vco_mult;
+	cfg->outdiv = div;
+
+	arizona_fll_dbg(fll, "Fvco=%dHz\n", target);
+
+	/* Find an appropraite FLL_FRATIO and factor it out of the target */
+	for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) {
+		if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) {
+			cfg->fratio = fll_fratios[i].fratio;
+			ratio = fll_fratios[i].ratio;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(fll_fratios)) {
+		arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n",
+				Fref);
+		return -EINVAL;
+	}
+
+	cfg->n = target / (ratio * Fref);
+
+	if (target % Fref) {
+		gcd_fll = gcd(target, ratio * Fref);
+		arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll);
+
+		cfg->theta = (target - (cfg->n * ratio * Fref))
+			/ gcd_fll;
+		cfg->lambda = (ratio * Fref) / gcd_fll;
+	} else {
+		cfg->theta = 0;
+		cfg->lambda = 0;
+	}
+
+	arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n",
+			cfg->n, cfg->theta, cfg->lambda);
+	arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
+			cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+
+	return 0;
+
+}
+
+static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
+			      struct arizona_fll_cfg *cfg, int source)
+{
+	regmap_update_bits(arizona->regmap, base + 3,
+			   ARIZONA_FLL1_THETA_MASK, cfg->theta);
+	regmap_update_bits(arizona->regmap, base + 4,
+			   ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda);
+	regmap_update_bits(arizona->regmap, base + 5,
+			   ARIZONA_FLL1_FRATIO_MASK,
+			   cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT);
+	regmap_update_bits(arizona->regmap, base + 6,
+			   ARIZONA_FLL1_CLK_REF_DIV_MASK |
+			   ARIZONA_FLL1_CLK_REF_SRC_MASK,
+			   cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
+			   source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
+
+	regmap_update_bits(arizona->regmap, base + 2,
+			   ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
+			   ARIZONA_FLL1_CTRL_UPD | cfg->n);
+}
+
+int arizona_set_fll(struct arizona_fll *fll, int source,
+		    unsigned int Fref, unsigned int Fout)
+{
+	struct arizona *arizona = fll->arizona;
+	struct arizona_fll_cfg cfg, sync;
+	unsigned int reg, val;
+	int syncsrc;
+	bool ena;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
+	if (ret != 0) {
+		arizona_fll_err(fll, "Failed to read current state: %d\n",
+				ret);
+		return ret;
+	}
+	ena = reg & ARIZONA_FLL1_ENA;
+
+	if (Fout) {
+		/* Do we have a 32kHz reference? */
+		regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
+		switch (val & ARIZONA_CLK_32K_SRC_MASK) {
+		case ARIZONA_CLK_SRC_MCLK1:
+		case ARIZONA_CLK_SRC_MCLK2:
+			syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
+			break;
+		default:
+			syncsrc = -1;
+		}
+
+		if (source == syncsrc)
+			syncsrc = -1;
+
+		if (syncsrc >= 0) {
+			ret = arizona_calc_fll(fll, &sync, Fref, Fout);
+			if (ret != 0)
+				return ret;
+
+			ret = arizona_calc_fll(fll, &cfg, 32768, Fout);
+			if (ret != 0)
+				return ret;
+		} else {
+			ret = arizona_calc_fll(fll, &cfg, Fref, Fout);
+			if (ret != 0)
+				return ret;
+		}
+	} else {
+		regmap_update_bits(arizona->regmap, fll->base + 1,
+				   ARIZONA_FLL1_ENA, 0);
+		regmap_update_bits(arizona->regmap, fll->base + 0x11,
+				   ARIZONA_FLL1_SYNC_ENA, 0);
+
+		if (ena)
+			pm_runtime_put_autosuspend(arizona->dev);
+
+		return 0;
+	}
+
+	regmap_update_bits(arizona->regmap, fll->base + 5,
+			   ARIZONA_FLL1_OUTDIV_MASK,
+			   cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT);
+
+	if (syncsrc >= 0) {
+		arizona_apply_fll(arizona, fll->base, &cfg, syncsrc);
+		arizona_apply_fll(arizona, fll->base + 0x10, &sync, source);
+	} else {
+		arizona_apply_fll(arizona, fll->base, &cfg, source);
+	}
+
+	if (!ena)
+		pm_runtime_get(arizona->dev);
+
+	/* Clear any pending completions */
+	try_wait_for_completion(&fll->ok);
+
+	regmap_update_bits(arizona->regmap, fll->base + 1,
+			   ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA);
+	if (syncsrc >= 0)
+		regmap_update_bits(arizona->regmap, fll->base + 0x11,
+				   ARIZONA_FLL1_SYNC_ENA,
+				   ARIZONA_FLL1_SYNC_ENA);
+
+	ret = wait_for_completion_timeout(&fll->ok,
+					  msecs_to_jiffies(25));
+	if (ret == 0)
+		arizona_fll_warn(fll, "Timed out waiting for lock\n");
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_set_fll);
+
+int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq,
+		     int ok_irq, struct arizona_fll *fll)
+{
+	int ret;
+
+	init_completion(&fll->lock);
+	init_completion(&fll->ok);
+
+	fll->id = id;
+	fll->base = base;
+	fll->arizona = arizona;
+
+	snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id);
+	snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name),
+		 "FLL%d clock OK", id);
+
+	ret = arizona_request_irq(arizona, lock_irq, fll->lock_name,
+				  arizona_fll_lock, fll);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n",
+			id, ret);
+	}
+
+	ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name,
+				  arizona_fll_clock_ok, fll);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n",
+			id, ret);
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_fll);
+
+MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
new file mode 100644
index 0000000..59caca8
--- /dev/null
+++ b/sound/soc/codecs/arizona.h
@@ -0,0 +1,159 @@
+/*
+ * arizona.h - Wolfson Arizona class device shared support
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _ASOC_ARIZONA_H
+#define _ASOC_ARIZONA_H
+
+#include <linux/completion.h>
+
+#include <sound/soc.h>
+
+#define ARIZONA_CLK_SYSCLK   1
+#define ARIZONA_CLK_ASYNCCLK 2
+
+#define ARIZONA_CLK_SRC_MCLK1    0x0
+#define ARIZONA_CLK_SRC_MCLK2    0x1
+#define ARIZONA_CLK_SRC_FLL1     0x4
+#define ARIZONA_CLK_SRC_FLL2     0x5
+#define ARIZONA_CLK_SRC_AIF1BCLK 0x8
+#define ARIZONA_CLK_SRC_AIF2BCLK 0x9
+#define ARIZONA_CLK_SRC_AIF3BCLK 0xa
+
+#define ARIZONA_FLL_SRC_MCLK1      0
+#define ARIZONA_FLL_SRC_MCLK2      1
+#define ARIZONA_FLL_SRC_SLIMCLK    2
+#define ARIZONA_FLL_SRC_FLL1       3
+#define ARIZONA_FLL_SRC_FLL2       4
+#define ARIZONA_FLL_SRC_AIF1BCLK   5
+#define ARIZONA_FLL_SRC_AIF2BCLK   6
+#define ARIZONA_FLL_SRC_AIF3BCLK   7
+#define ARIZONA_FLL_SRC_AIF1LRCLK  8
+#define ARIZONA_FLL_SRC_AIF2LRCLK  9
+#define ARIZONA_FLL_SRC_AIF3LRCLK 10
+
+#define ARIZONA_MIXER_VOL_MASK             0x00FE
+#define ARIZONA_MIXER_VOL_SHIFT                 1
+#define ARIZONA_MIXER_VOL_WIDTH                 7
+
+#define ARIZONA_MAX_DAI 3
+
+struct arizona;
+
+struct arizona_dai_priv {
+	int clk;
+};
+
+struct arizona_priv {
+	struct arizona *arizona;
+	int sysclk;
+	int asyncclk;
+	struct arizona_dai_priv dai[ARIZONA_MAX_DAI];
+};
+
+#define ARIZONA_NUM_MIXER_INPUTS 57
+
+extern const unsigned int arizona_mixer_tlv[];
+extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
+extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+
+#define ARIZONA_MIXER_CONTROLS(name, base) \
+	SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv),			\
+	SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7,		\
+			     ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0,	\
+			     arizona_mixer_tlv)
+
+#define ARIZONA_MUX_ENUM_DECL(name, reg) \
+	SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff,			\
+				   arizona_mixer_texts, arizona_mixer_values)
+
+#define ARIZONA_MUX_CTL_DECL(name) \
+	const struct snd_kcontrol_new name##_mux =	\
+		SOC_DAPM_VALUE_ENUM("Route", name##_enum)
+
+#define ARIZONA_MIXER_ENUMS(name, base_reg) \
+	static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg);      \
+	static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2);  \
+	static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4);  \
+	static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6);  \
+	static ARIZONA_MUX_CTL_DECL(name##_in1); \
+	static ARIZONA_MUX_CTL_DECL(name##_in2); \
+	static ARIZONA_MUX_CTL_DECL(name##_in3); \
+	static ARIZONA_MUX_CTL_DECL(name##_in4)
+
+#define ARIZONA_MUX(name, ctrl) \
+	SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+#define ARIZONA_MIXER_WIDGETS(name, name_str)	\
+	ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \
+	ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \
+	ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \
+	ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \
+	SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0)
+
+#define ARIZONA_MIXER_ROUTES(widget, name) \
+	{ widget, NULL, name " Mixer" },         \
+	{ name " Mixer", NULL, name " Input 1" }, \
+	{ name " Mixer", NULL, name " Input 2" }, \
+	{ name " Mixer", NULL, name " Input 3" }, \
+	{ name " Mixer", NULL, name " Input 4" }, \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \
+	ARIZONA_MIXER_INPUT_ROUTES(name " Input 4")
+
+extern const struct soc_enum arizona_lhpf1_mode;
+extern const struct soc_enum arizona_lhpf2_mode;
+extern const struct soc_enum arizona_lhpf3_mode;
+extern const struct soc_enum arizona_lhpf4_mode;
+
+extern int arizona_in_ev(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *kcontrol,
+			 int event);
+extern int arizona_out_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event);
+
+extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+			      int source, unsigned int freq, int dir);
+
+extern const struct snd_soc_dai_ops arizona_dai_ops;
+
+#define ARIZONA_FLL_NAME_LEN 20
+
+struct arizona_fll {
+	struct arizona *arizona;
+	int id;
+	unsigned int base;
+	unsigned int vco_mult;
+	struct completion lock;
+	struct completion ok;
+
+	char lock_name[ARIZONA_FLL_NAME_LEN];
+	char clock_ok_name[ARIZONA_FLL_NAME_LEN];
+};
+
+extern int arizona_init_fll(struct arizona *arizona, int id, int base,
+			    int lock_irq, int ok_irq, struct arizona_fll *fll);
+extern int arizona_set_fll(struct arizona_fll *fll, int source,
+			   unsigned int Fref, unsigned int Fout);
+
+extern int arizona_init_dai(struct arizona_priv *priv, int dai);
+
+#endif
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c
index a710941..628daf6 100644
--- a/sound/soc/codecs/cs42l52.c
+++ b/sound/soc/codecs/cs42l52.c
@@ -14,7 +14,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -1217,11 +1216,11 @@
 		return -ENOMEM;
 	cs42l52->dev = &i2c_client->dev;
 
-	cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap);
+	cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap);
 	if (IS_ERR(cs42l52->regmap)) {
 		ret = PTR_ERR(cs42l52->regmap);
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
-		goto err;
+		return ret;
 	}
 
 	i2c_set_clientdata(i2c_client, cs42l52);
@@ -1243,7 +1242,7 @@
 		dev_err(&i2c_client->dev,
 			"CS42L52 Device ID (%X). Expected %X\n",
 			devid, CS42L52_CHIP_ID);
-		goto err_regmap;
+		return ret;
 	}
 
 	regcache_cache_only(cs42l52->regmap, true);
@@ -1251,23 +1250,13 @@
 	ret =  snd_soc_register_codec(&i2c_client->dev,
 			&soc_codec_dev_cs42l52, &cs42l52_dai, 1);
 	if (ret < 0)
-		goto err_regmap;
+		return ret;
 	return 0;
-
-err_regmap:
-	regmap_exit(cs42l52->regmap);
-
-err:
-	return ret;
 }
 
 static int cs42l52_i2c_remove(struct i2c_client *client)
 {
-	struct cs42l52_private *cs42l52 = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(cs42l52->regmap);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
index e0d45fd..2c08c4c 100644
--- a/sound/soc/codecs/cs42l73.c
+++ b/sound/soc/codecs/cs42l73.c
@@ -1362,11 +1362,11 @@
 
 	i2c_set_clientdata(i2c_client, cs42l73);
 
-	cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap);
+	cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap);
 	if (IS_ERR(cs42l73->regmap)) {
 		ret = PTR_ERR(cs42l73->regmap);
 		dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret);
-		goto err;
+		return ret;
 	}
 	/* initialize codec */
 	ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, &reg);
@@ -1384,13 +1384,13 @@
 		dev_err(&i2c_client->dev,
 			"CS42L73 Device ID (%X). Expected %X\n",
 			devid, CS42L73_DEVID);
-		goto err_regmap;
+		return ret;
 	}
 
 	ret = regmap_read(cs42l73->regmap, CS42L73_REVID, &reg);
 	if (ret < 0) {
 		dev_err(&i2c_client->dev, "Get Revision ID failed\n");
-		goto err_regmap;
+		return ret;;
 	}
 
 	dev_info(&i2c_client->dev,
@@ -1402,23 +1402,13 @@
 			&soc_codec_dev_cs42l73, cs42l73_dai,
 			ARRAY_SIZE(cs42l73_dai));
 	if (ret < 0)
-		goto err_regmap;
+		return ret;
 	return 0;
-
-err_regmap:
-	regmap_exit(cs42l73->regmap);
-
-err:
-	return ret;
 }
 
 static __devexit int cs42l73_i2c_remove(struct i2c_client *client)
 {
-	struct cs42l73_private *cs42l73 = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(cs42l73->regmap);
-
 	return 0;
 }
 
diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c
new file mode 100644
index 0000000..01be2a3
--- /dev/null
+++ b/sound/soc/codecs/da732x.c
@@ -0,0 +1,1627 @@
+/*
+ * da732x.c --- Dialog DA732X ALSA SoC Audio Driver
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/regmap.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/sysfs.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include "da732x.h"
+#include "da732x_reg.h"
+
+
+struct da732x_priv {
+	struct regmap *regmap;
+	struct snd_soc_codec *codec;
+
+	unsigned int sysclk;
+	bool pll_en;
+};
+
+/*
+ * da732x register cache - default settings
+ */
+static struct reg_default da732x_reg_cache[] = {
+	{ DA732X_REG_REF1		, 0x02 },
+	{ DA732X_REG_BIAS_EN		, 0x80 },
+	{ DA732X_REG_BIAS1		, 0x00 },
+	{ DA732X_REG_BIAS2		, 0x00 },
+	{ DA732X_REG_BIAS3		, 0x00 },
+	{ DA732X_REG_BIAS4		, 0x00 },
+	{ DA732X_REG_MICBIAS2		, 0x00 },
+	{ DA732X_REG_MICBIAS1		, 0x00 },
+	{ DA732X_REG_MICDET		, 0x00 },
+	{ DA732X_REG_MIC1_PRE		, 0x01 },
+	{ DA732X_REG_MIC1		, 0x40 },
+	{ DA732X_REG_MIC2_PRE		, 0x01 },
+	{ DA732X_REG_MIC2		, 0x40 },
+	{ DA732X_REG_AUX1L		, 0x75 },
+	{ DA732X_REG_AUX1R		, 0x75 },
+	{ DA732X_REG_MIC3_PRE		, 0x01 },
+	{ DA732X_REG_MIC3		, 0x40 },
+	{ DA732X_REG_INP_PINBIAS	, 0x00 },
+	{ DA732X_REG_INP_ZC_EN		, 0x00 },
+	{ DA732X_REG_INP_MUX		, 0x50 },
+	{ DA732X_REG_HP_DET		, 0x00 },
+	{ DA732X_REG_HPL_DAC_OFFSET	, 0x00 },
+	{ DA732X_REG_HPL_DAC_OFF_CNTL	, 0x00 },
+	{ DA732X_REG_HPL_OUT_OFFSET	, 0x00 },
+	{ DA732X_REG_HPL		, 0x40 },
+	{ DA732X_REG_HPL_VOL		, 0x0F },
+	{ DA732X_REG_HPR_DAC_OFFSET	, 0x00 },
+	{ DA732X_REG_HPR_DAC_OFF_CNTL	, 0x00 },
+	{ DA732X_REG_HPR_OUT_OFFSET	, 0x00 },
+	{ DA732X_REG_HPR		, 0x40 },
+	{ DA732X_REG_HPR_VOL		, 0x0F },
+	{ DA732X_REG_LIN2		, 0x4F },
+	{ DA732X_REG_LIN3		, 0x4F },
+	{ DA732X_REG_LIN4		, 0x4F },
+	{ DA732X_REG_OUT_ZC_EN		, 0x00 },
+	{ DA732X_REG_HP_LIN1_GNDSEL	, 0x00 },
+	{ DA732X_REG_CP_HP1		, 0x0C },
+	{ DA732X_REG_CP_HP2		, 0x03 },
+	{ DA732X_REG_CP_CTRL1		, 0x00 },
+	{ DA732X_REG_CP_CTRL2		, 0x99 },
+	{ DA732X_REG_CP_CTRL3		, 0x25 },
+	{ DA732X_REG_CP_LEVEL_MASK	, 0x3F },
+	{ DA732X_REG_CP_DET		, 0x00 },
+	{ DA732X_REG_CP_STATUS		, 0x00 },
+	{ DA732X_REG_CP_THRESH1		, 0x00 },
+	{ DA732X_REG_CP_THRESH2		, 0x00 },
+	{ DA732X_REG_CP_THRESH3		, 0x00 },
+	{ DA732X_REG_CP_THRESH4		, 0x00 },
+	{ DA732X_REG_CP_THRESH5		, 0x00 },
+	{ DA732X_REG_CP_THRESH6		, 0x00 },
+	{ DA732X_REG_CP_THRESH7		, 0x00 },
+	{ DA732X_REG_CP_THRESH8		, 0x00 },
+	{ DA732X_REG_PLL_DIV_LO		, 0x00 },
+	{ DA732X_REG_PLL_DIV_MID	, 0x00 },
+	{ DA732X_REG_PLL_DIV_HI		, 0x00 },
+	{ DA732X_REG_PLL_CTRL		, 0x02 },
+	{ DA732X_REG_CLK_CTRL		, 0xaa },
+	{ DA732X_REG_CLK_DSP		, 0x07 },
+	{ DA732X_REG_CLK_EN1		, 0x00 },
+	{ DA732X_REG_CLK_EN2		, 0x00 },
+	{ DA732X_REG_CLK_EN3		, 0x00 },
+	{ DA732X_REG_CLK_EN4		, 0x00 },
+	{ DA732X_REG_CLK_EN5		, 0x00 },
+	{ DA732X_REG_AIF_MCLK		, 0x00 },
+	{ DA732X_REG_AIFA1		, 0x02 },
+	{ DA732X_REG_AIFA2		, 0x00 },
+	{ DA732X_REG_AIFA3		, 0x08 },
+	{ DA732X_REG_AIFB1		, 0x02 },
+	{ DA732X_REG_AIFB2		, 0x00 },
+	{ DA732X_REG_AIFB3		, 0x08 },
+	{ DA732X_REG_PC_CTRL		, 0xC0 },
+	{ DA732X_REG_DATA_ROUTE		, 0x00 },
+	{ DA732X_REG_DSP_CTRL		, 0x00 },
+	{ DA732X_REG_CIF_CTRL2		, 0x00 },
+	{ DA732X_REG_HANDSHAKE		, 0x00 },
+	{ DA732X_REG_SPARE1_OUT		, 0x00 },
+	{ DA732X_REG_SPARE2_OUT		, 0x00 },
+	{ DA732X_REG_SPARE1_IN		, 0x00 },
+	{ DA732X_REG_ADC1_PD		, 0x00 },
+	{ DA732X_REG_ADC1_HPF		, 0x00 },
+	{ DA732X_REG_ADC1_SEL		, 0x00 },
+	{ DA732X_REG_ADC1_EQ12		, 0x00 },
+	{ DA732X_REG_ADC1_EQ34		, 0x00 },
+	{ DA732X_REG_ADC1_EQ5		, 0x00 },
+	{ DA732X_REG_ADC2_PD		, 0x00 },
+	{ DA732X_REG_ADC2_HPF		, 0x00 },
+	{ DA732X_REG_ADC2_SEL		, 0x00 },
+	{ DA732X_REG_ADC2_EQ12		, 0x00 },
+	{ DA732X_REG_ADC2_EQ34		, 0x00 },
+	{ DA732X_REG_ADC2_EQ5		, 0x00 },
+	{ DA732X_REG_DAC1_HPF		, 0x00 },
+	{ DA732X_REG_DAC1_L_VOL		, 0x00 },
+	{ DA732X_REG_DAC1_R_VOL		, 0x00 },
+	{ DA732X_REG_DAC1_SEL		, 0x00 },
+	{ DA732X_REG_DAC1_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC1_EQ12		, 0x00 },
+	{ DA732X_REG_DAC1_EQ34		, 0x00 },
+	{ DA732X_REG_DAC1_EQ5		, 0x00 },
+	{ DA732X_REG_DAC2_HPF		, 0x00 },
+	{ DA732X_REG_DAC2_L_VOL		, 0x00 },
+	{ DA732X_REG_DAC2_R_VOL		, 0x00 },
+	{ DA732X_REG_DAC2_SEL		, 0x00 },
+	{ DA732X_REG_DAC2_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC2_EQ12		, 0x00 },
+	{ DA732X_REG_DAC2_EQ34		, 0x00 },
+	{ DA732X_REG_DAC2_EQ5		, 0x00 },
+	{ DA732X_REG_DAC3_HPF		, 0x00 },
+	{ DA732X_REG_DAC3_VOL		, 0x00 },
+	{ DA732X_REG_DAC3_SEL		, 0x00 },
+	{ DA732X_REG_DAC3_SOFTMUTE	, 0x00 },
+	{ DA732X_REG_DAC3_EQ12		, 0x00 },
+	{ DA732X_REG_DAC3_EQ34		, 0x00 },
+	{ DA732X_REG_DAC3_EQ5		, 0x00 },
+	{ DA732X_REG_BIQ_BYP		, 0x00 },
+	{ DA732X_REG_DMA_CMD		, 0x00 },
+	{ DA732X_REG_DMA_ADDR0		, 0x00 },
+	{ DA732X_REG_DMA_ADDR1		, 0x00 },
+	{ DA732X_REG_DMA_DATA0		, 0x00 },
+	{ DA732X_REG_DMA_DATA1		, 0x00 },
+	{ DA732X_REG_DMA_DATA2		, 0x00 },
+	{ DA732X_REG_DMA_DATA3		, 0x00 },
+	{ DA732X_REG_UNLOCK		, 0x00 },
+};
+
+static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk)
+{
+	int val;
+	int ret;
+
+	if (sysclk < DA732X_MCLK_10MHZ) {
+		val = DA732X_MCLK_RET_0_10MHZ;
+		ret = DA732X_MCLK_VAL_0_10MHZ;
+	} else if ((sysclk >= DA732X_MCLK_10MHZ) &&
+	    (sysclk < DA732X_MCLK_20MHZ)) {
+		val = DA732X_MCLK_RET_10_20MHZ;
+		ret = DA732X_MCLK_VAL_10_20MHZ;
+	} else if ((sysclk >= DA732X_MCLK_20MHZ) &&
+	    (sysclk < DA732X_MCLK_40MHZ)) {
+		val = DA732X_MCLK_RET_20_40MHZ;
+		ret = DA732X_MCLK_VAL_20_40MHZ;
+	} else if ((sysclk >= DA732X_MCLK_40MHZ) &&
+	    (sysclk <= DA732X_MCLK_54MHZ)) {
+		val = DA732X_MCLK_RET_40_54MHZ;
+		ret = DA732X_MCLK_VAL_40_54MHZ;
+	} else {
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, DA732X_REG_PLL_CTRL, val);
+
+	return ret;
+}
+
+static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state)
+{
+	switch (state) {
+	case DA732X_ENABLE_CP:
+		snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN);
+		snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN |
+			      DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN |
+			      DA732X_CP_CTRL_CPVDD1);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL2,
+			      DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ);
+		break;
+	case DA732X_DISABLE_CP:
+		snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS);
+		snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS);
+		snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS);
+		break;
+	default:
+		pr_err(KERN_ERR "Wrong charge pump state\n");
+		break;
+	}
+}
+
+static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN,
+				  DA732X_MIC_PRE_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN,
+				  DA732X_MIC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN,
+				  DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN,
+				  DA732X_AUX_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN,
+				  DA732X_LIN2_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN,
+				  DA732X_LIN3_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN,
+				  DA732X_LIN4_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN,
+				  DA732X_ADC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN,
+				  DA732X_DAC_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN,
+				  DA732X_EQ_BAND_VOL_DB_INC, 0);
+
+static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN,
+				  DA732X_EQ_OVERALL_VOL_DB_INC, 0);
+
+/* High Pass Filter */
+static const char *da732x_hpf_mode[] = {
+	"Disable", "Music", "Voice",
+};
+
+static const char *da732x_hpf_music[] = {
+	"1.8Hz", "3.75Hz", "7.5Hz", "15Hz",
+};
+
+static const char *da732x_hpf_voice[] = {
+	"2.5Hz", "25Hz", "50Hz", "100Hz",
+	"150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da732x_dac1_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac2_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac3_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc1_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_adc2_hpf_mode_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT,
+			DA732X_HPF_MODE_MAX, da732x_hpf_mode)
+};
+
+static const struct soc_enum da732x_dac1_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac2_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac3_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc1_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_adc2_hp_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT,
+			DA732X_HPF_MUSIC_MAX, da732x_hpf_music)
+};
+
+static const struct soc_enum da732x_dac1_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac2_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_dac3_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc1_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+static const struct soc_enum da732x_adc2_voice_filter_enum[] = {
+	SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT,
+			DA732X_HPF_VOICE_MAX, da732x_hpf_voice)
+};
+
+
+static int da732x_hpf_set(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+	unsigned int reg = enum_ctrl->reg;
+	unsigned int sel = ucontrol->value.integer.value[0];
+	unsigned int bits;
+
+	switch (sel) {
+	case DA732X_HPF_DISABLED:
+		bits = DA732X_HPF_DIS;
+		break;
+	case DA732X_HPF_VOICE:
+		bits = DA732X_HPF_VOICE_EN;
+		break;
+	case DA732X_HPF_MUSIC:
+		bits = DA732X_HPF_MUSIC_EN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits);
+
+	return 0;
+}
+
+static int da732x_hpf_get(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value;
+	unsigned int reg = enum_ctrl->reg;
+	int val;
+
+	val = snd_soc_read(codec, reg) & DA732X_HPF_MASK;
+
+	switch (val) {
+	case DA732X_HPF_VOICE_EN:
+		ucontrol->value.integer.value[0] = DA732X_HPF_VOICE;
+		break;
+	case DA732X_HPF_MUSIC_EN:
+		ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC;
+		break;
+	default:
+		ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED;
+		break;
+	}
+
+	return 0;
+}
+
+static const struct snd_kcontrol_new da732x_snd_controls[] = {
+	/* Input PGAs */
+	SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+	SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+	SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE,
+			     DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN,
+			     DA732X_MICBOOST_MAX, 0, mic_boost_tlv),
+
+	/* MICs */
+	SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+	SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+	SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3,
+			     DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN,
+			     DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv),
+
+	/* AUXs */
+	SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L,
+		       DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, aux_pga_tlv),
+	SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R,
+		       DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, aux_pga_tlv),
+
+	/* ADCs */
+	SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL,
+		       DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+		       DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+	SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL,
+		       DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT,
+		       DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv),
+
+	/* DACs */
+	SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL,
+			 DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT,
+			 DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL,
+			DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+			DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL,
+		   DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL,
+		       DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+		       DA732X_INVERT, dac_pga_tlv),
+	SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL,
+		   DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL,
+		       DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX,
+		       DA732X_INVERT, dac_pga_tlv),
+
+	/* High Pass Filters */
+	SOC_ENUM_EXT("DAC1 High Pass Filter Mode",
+		     da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum),
+	SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum),
+
+	SOC_ENUM_EXT("DAC2 High Pass Filter Mode",
+		     da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum),
+	SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum),
+
+	SOC_ENUM_EXT("DAC3 High Pass Filter Mode",
+		     da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum),
+	SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum),
+
+	SOC_ENUM_EXT("ADC1 High Pass Filter Mode",
+		     da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum),
+	SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum),
+
+	SOC_ENUM_EXT("ADC2 High Pass Filter Mode",
+		     da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set),
+	SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum),
+	SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum),
+
+	/* Equalizers */
+	SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_overall_tlv),
+
+	SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5,
+		       DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_overall_tlv),
+
+	SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5,
+		   DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT),
+	SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12,
+		       DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12,
+		       DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34,
+		       DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34,
+		       DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+	SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5,
+		       DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX,
+		       DA732X_INVERT, eq_band_pga_tlv),
+
+	/* Lineout 2 Reciever*/
+	SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin2_pga_tlv),
+
+	/* Lineout 3 SPEAKER*/
+	SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin3_pga_tlv),
+
+	/* Lineout 4 */
+	SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT,
+		   DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4,
+		       DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX,
+		       DA732X_NO_INVERT, lin4_pga_tlv),
+
+	/* Headphones */
+	SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL,
+		     DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT),
+	SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL,
+			 DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT,
+			 DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv),
+};
+
+static int da732x_adc_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		switch (w->reg) {
+		case DA732X_REG_ADC1_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCA_BB_CLK_EN,
+					    DA732X_ADCA_BB_CLK_EN);
+			break;
+		case DA732X_REG_ADC2_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCC_BB_CLK_EN,
+					    DA732X_ADCC_BB_CLK_EN);
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+				    DA732X_ADC_SET_ACT);
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+				    DA732X_ADC_ON);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK,
+				    DA732X_ADC_OFF);
+		snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK,
+				    DA732X_ADC_SET_RST);
+
+		switch (w->reg) {
+		case DA732X_REG_ADC1_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCA_BB_CLK_EN, 0);
+			break;
+		case DA732X_REG_ADC2_PD:
+			snd_soc_update_bits(codec, DA732X_REG_CLK_EN3,
+					    DA732X_ADCC_BB_CLK_EN, 0);
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int da732x_out_pga_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec, w->reg,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		snd_soc_update_bits(codec, w->reg,
+				    (1 << w->shift) | DA732X_OUT_HIZ_EN,
+				    (1 << w->shift) | DA732X_OUT_HIZ_DIS);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const char *adcl_text[] = {
+	"AUX1L", "MIC1"
+};
+
+static const char *adcr_text[] = {
+	"AUX1R", "MIC2", "MIC3"
+};
+
+static const char *enable_text[] = {
+	"Disabled",
+	"Enabled"
+};
+
+/* ADC1LMUX */
+static const struct soc_enum adc1l_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT,
+			DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc1l_mux =
+	SOC_DAPM_ENUM("ADC Route", adc1l_enum);
+
+/* ADC1RMUX */
+static const struct soc_enum adc1r_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT,
+			DA732X_ADCR_MUX_MAX, adcr_text);
+static const struct snd_kcontrol_new adc1r_mux =
+	SOC_DAPM_ENUM("ADC Route", adc1r_enum);
+
+/* ADC2LMUX */
+static const struct soc_enum adc2l_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT,
+			DA732X_ADCL_MUX_MAX, adcl_text);
+static const struct snd_kcontrol_new adc2l_mux =
+	SOC_DAPM_ENUM("ADC Route", adc2l_enum);
+
+/* ADC2RMUX */
+static const struct soc_enum adc2r_enum =
+	SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT,
+			DA732X_ADCR_MUX_MAX, adcr_text);
+
+static const struct snd_kcontrol_new adc2r_mux =
+	SOC_DAPM_ENUM("ADC Route", adc2r_enum);
+
+static const struct soc_enum da732x_hp_left_output =
+	SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpl_mux =
+	SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output);
+
+static const struct soc_enum da732x_hp_right_output =
+	SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new hpr_mux =
+	SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output);
+
+static const struct soc_enum da732x_speaker_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new spk_mux =
+	SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output);
+
+static const struct soc_enum da732x_lout4_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout4_mux =
+	SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output);
+
+static const struct soc_enum da732x_lout2_output =
+	SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT,
+			DA732X_DAC_EN_MAX, enable_text);
+
+static const struct snd_kcontrol_new lout2_mux =
+	SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output);
+
+static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = {
+	/* Supplies */
+	SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0,
+			    DA732X_NO_INVERT, da732x_adc_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0,
+			    DA732X_NO_INVERT, da732x_adc_event,
+			    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4,
+			    DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4,
+			    DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+	SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5,
+			    DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT,
+			    NULL, 0),
+
+	/* Micbias */
+	SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1,
+			    DA732X_MICBIAS_EN_SHIFT,
+			    DA732X_NO_INVERT, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2,
+			    DA732X_MICBIAS_EN_SHIFT,
+			    DA732X_NO_INVERT, NULL, 0),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("MIC2"),
+	SND_SOC_DAPM_INPUT("MIC3"),
+	SND_SOC_DAPM_INPUT("AUX1L"),
+	SND_SOC_DAPM_INPUT("AUX1R"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("LOUTL"),
+	SND_SOC_DAPM_OUTPUT("LOUTR"),
+	SND_SOC_DAPM_OUTPUT("ClassD"),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL,
+			 DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL,
+			 DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL,
+			 DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL,
+			 DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL,
+			 DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL,
+			 DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT),
+	SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL,
+			 DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT),
+
+	/* Input Pgas */
+	SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT,
+			 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT,
+			 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT,
+			   0, NULL, 0, da732x_out_pga_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+	/* MUXs */
+	SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux),
+	SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux),
+	SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux),
+	SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux),
+
+	SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux),
+	SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux),
+	SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux),
+	SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux),
+	SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux),
+
+	/* AIF interfaces */
+	SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3,
+			     DA732X_AIF_EN_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3,
+			    DA732X_AIF_EN_SHIFT, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3,
+			     DA732X_AIF_EN_SHIFT, 0),
+	SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3,
+			    DA732X_AIF_EN_SHIFT, 0),
+};
+
+static const struct snd_soc_dapm_route da732x_dapm_routes[] = {
+	/* Inputs */
+	{"AUX1L PGA", "NULL", "AUX1L"},
+	{"AUX1R PGA", "NULL", "AUX1R"},
+	{"MIC1 PGA", NULL, "MIC1"},
+	{"MIC2 PGA", "NULL", "MIC2"},
+	{"MIC3 PGA", "NULL", "MIC3"},
+
+	/* Capture Path */
+	{"ADC1 Left MUX", "MIC1", "MIC1 PGA"},
+	{"ADC1 Left MUX", "AUX1L", "AUX1L PGA"},
+
+	{"ADC1 Right MUX", "AUX1R", "AUX1R PGA"},
+	{"ADC1 Right MUX", "MIC2", "MIC2 PGA"},
+	{"ADC1 Right MUX", "MIC3", "MIC3 PGA"},
+
+	{"ADC2 Left MUX", "AUX1L", "AUX1L PGA"},
+	{"ADC2 Left MUX", "MIC1", "MIC1 PGA"},
+
+	{"ADC2 Right MUX", "AUX1R", "AUX1R PGA"},
+	{"ADC2 Right MUX", "MIC2", "MIC2 PGA"},
+	{"ADC2 Right MUX", "MIC3", "MIC3 PGA"},
+
+	{"ADC1L", NULL, "ADC1 Supply"},
+	{"ADC1R", NULL, "ADC1 Supply"},
+	{"ADC2L", NULL, "ADC2 Supply"},
+	{"ADC2R", NULL, "ADC2 Supply"},
+
+	{"ADC1L", NULL, "ADC1 Left MUX"},
+	{"ADC1R", NULL, "ADC1 Right MUX"},
+	{"ADC2L", NULL, "ADC2 Left MUX"},
+	{"ADC2R", NULL, "ADC2 Right MUX"},
+
+	{"AIFA Output", NULL, "ADC1L"},
+	{"AIFA Output", NULL, "ADC1R"},
+	{"AIFB Output", NULL, "ADC2L"},
+	{"AIFB Output", NULL, "ADC2R"},
+
+	{"HP Left MUX", "Enabled", "AIFA Input"},
+	{"HP Right MUX", "Enabled", "AIFA Input"},
+	{"Speaker MUX", "Enabled", "AIFB Input"},
+	{"LOUT2 MUX", "Enabled", "AIFB Input"},
+	{"LOUT4 MUX", "Enabled", "AIFB Input"},
+
+	{"DAC1L", NULL, "DAC1 CLK"},
+	{"DAC1R", NULL, "DAC1 CLK"},
+	{"DAC2L", NULL, "DAC2 CLK"},
+	{"DAC2R", NULL, "DAC2 CLK"},
+	{"DAC3", NULL, "DAC3 CLK"},
+
+	{"DAC1L", NULL, "HP Left MUX"},
+	{"DAC1R", NULL, "HP Right MUX"},
+	{"DAC2L", NULL, "Speaker MUX"},
+	{"DAC2R", NULL, "LOUT4 MUX"},
+	{"DAC3", NULL, "LOUT2 MUX"},
+
+	/* Output Pgas */
+	{"HP Left", NULL, "DAC1L"},
+	{"HP Right", NULL, "DAC1R"},
+	{"LIN3", NULL, "DAC2L"},
+	{"LIN4", NULL, "DAC2R"},
+	{"LIN2", NULL, "DAC3"},
+
+	/* Outputs */
+	{"ClassD", NULL, "LIN3"},
+	{"LOUTL", NULL, "LIN2"},
+	{"LOUTR", NULL, "LIN4"},
+	{"HPL", NULL, "HP Left"},
+	{"HPR", NULL, "HP Right"},
+};
+
+static int da732x_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u32 aif = 0;
+	u32 reg_aif;
+	u32 fs;
+
+	reg_aif = dai->driver->base;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		aif |= DA732X_AIF_WORD_16;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		aif |= DA732X_AIF_WORD_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		aif |= DA732X_AIF_WORD_24;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		aif |= DA732X_AIF_WORD_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs = DA732X_SR_8KHZ;
+		break;
+	case 11025:
+		fs = DA732X_SR_11_025KHZ;
+		break;
+	case 12000:
+		fs = DA732X_SR_12KHZ;
+		break;
+	case 16000:
+		fs = DA732X_SR_16KHZ;
+		break;
+	case 22050:
+		fs = DA732X_SR_22_05KHZ;
+		break;
+	case 24000:
+		fs = DA732X_SR_24KHZ;
+		break;
+	case 32000:
+		fs = DA732X_SR_32KHZ;
+		break;
+	case 44100:
+		fs = DA732X_SR_44_1KHZ;
+		break;
+	case 48000:
+		fs = DA732X_SR_48KHZ;
+		break;
+	case 88100:
+		fs = DA732X_SR_88_1KHZ;
+		break;
+	case 96000:
+		fs = DA732X_SR_96KHZ;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif);
+	snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs);
+
+	return 0;
+}
+
+static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u32 aif_mclk, pc_count;
+	u32 reg_aif1, aif1;
+	u32 reg_aif3, aif3;
+
+	switch (dai->id) {
+	case DA732X_DAI_ID1:
+		reg_aif1 = DA732X_REG_AIFA1;
+		reg_aif3 = DA732X_REG_AIFA3;
+		pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT |
+			   DA732X_PC_SAME;
+		break;
+	case DA732X_DAI_ID2:
+		reg_aif1 = DA732X_REG_AIFB1;
+		reg_aif3 = DA732X_REG_AIFB3;
+		pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT |
+			   DA732X_PC_SAME;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aif1 = DA732X_AIF_SLAVE;
+		aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif1 = DA732X_AIF_CLK_FROM_SRC;
+		aif_mclk = DA732X_CLK_GENERATION_AIF_A;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		aif3 = DA732X_AIF_I2S_MODE;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		aif3 = DA732X_AIF_RIGHT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aif3 = DA732X_AIF_LEFT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		aif3 = DA732X_AIF_DSP_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_B:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif3 |= DA732X_AIF_BCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+		case SND_SOC_DAIFMT_NB_NF:
+			break;
+		case SND_SOC_DAIFMT_IB_IF:
+			aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_IB_NF:
+			aif3 |= DA732X_AIF_BCLK_INV;
+			break;
+		case SND_SOC_DAIFMT_NB_IF:
+			aif3 |= DA732X_AIF_WCLK_INV;
+			break;
+		default:
+			return -EINVAL;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk);
+	snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1);
+	snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV |
+			    DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3);
+	snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count);
+
+	return 0;
+}
+
+
+
+static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id,
+			      int source, unsigned int freq_in,
+			      unsigned int freq_out)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+	int fref, indiv;
+	u8 div_lo, div_mid, div_hi;
+	u64 frac_div;
+
+	/* Disable PLL */
+	if (freq_out == 0) {
+		snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+				    DA732X_PLL_EN, 0);
+		da732x->pll_en = false;
+		return 0;
+	}
+
+	if (da732x->pll_en)
+		return -EBUSY;
+
+	if (source == DA732X_SRCCLK_MCLK) {
+		/* Validate Sysclk rate */
+		switch (da732x->sysclk) {
+		case 11290000:
+		case 12288000:
+		case 22580000:
+		case 24576000:
+		case 45160000:
+		case 49152000:
+			snd_soc_write(codec, DA732X_REG_PLL_CTRL,
+				      DA732X_PLL_BYPASS);
+			return 0;
+		default:
+			dev_err(codec->dev,
+				"Cannot use PLL Bypass, invalid SYSCLK rate\n");
+			return -EINVAL;
+		}
+	}
+
+	indiv = da732x_get_input_div(codec, da732x->sysclk);
+	if (indiv < 0)
+		return indiv;
+
+	fref = (da732x->sysclk / indiv);
+	div_hi = freq_out / fref;
+	frac_div = (u64)(freq_out % fref) * 8192ULL;
+	do_div(frac_div, fref);
+	div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK;
+	div_lo = (frac_div) & DA732X_U8_MASK;
+
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo);
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid);
+	snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi);
+
+	snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN,
+			    DA732X_PLL_EN);
+
+	da732x->pll_en = true;
+
+	return 0;
+}
+
+static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				 unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+	da732x->sysclk = freq;
+
+	return 0;
+}
+
+#define DA732X_RATES	SNDRV_PCM_RATE_8000_96000
+
+#define	DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops da732x_dai1_ops = {
+	.hw_params	= da732x_hw_params,
+	.set_fmt	= da732x_set_dai_fmt,
+	.set_sysclk	= da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops da732x_dai2_ops = {
+	.hw_params	= da732x_hw_params,
+	.set_fmt	= da732x_set_dai_fmt,
+	.set_sysclk	= da732x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver da732x_dai[] = {
+	{
+		.name	= "DA732X_AIFA",
+		.id	= DA732X_DAI_ID1,
+		.base	= DA732X_REG_AIFA1,
+		.playback = {
+			.stream_name = "AIFA Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIFA Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.ops = &da732x_dai1_ops,
+	},
+	{
+		.name	= "DA732X_AIFB",
+		.id	= DA732X_DAI_ID2,
+		.base	= DA732X_REG_AIFB1,
+		.playback = {
+			.stream_name = "AIFB Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "AIFB Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = DA732X_RATES,
+			.formats = DA732X_FORMATS,
+		},
+		.ops = &da732x_dai2_ops,
+	},
+};
+
+static const struct regmap_config da732x_regmap = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+
+	.max_register		= DA732X_MAX_REG,
+	.reg_defaults		= da732x_reg_cache,
+	.num_reg_defaults	= ARRAY_SIZE(da732x_reg_cache),
+	.cache_type		= REGCACHE_RBTREE,
+};
+
+
+static void da732x_dac_offset_adjust(struct snd_soc_codec *codec)
+{
+	u8 offset[DA732X_HP_DACS];
+	u8 sign[DA732X_HP_DACS];
+	u8 step = DA732X_DAC_OFFSET_STEP;
+
+	/* Initialize DAC offset calibration circuits and registers */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+		      DA732X_HP_DAC_OFFSET_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+		      DA732X_HP_DAC_OFFSET_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+		      DA732X_HP_DAC_OFF_CALIBRATION |
+		      DA732X_HP_DAC_OFF_SCALE_STEPS);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+		      DA732X_HP_DAC_OFF_CALIBRATION |
+		      DA732X_HP_DAC_OFF_SCALE_STEPS);
+
+	/* Wait for voltage stabilization */
+	msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+	/* Check DAC offset sign */
+	sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+				DA732X_HP_DAC_OFF_CNTL_COMPO);
+	sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+				DA732X_HP_DAC_OFF_CNTL_COMPO);
+
+	/* Binary search DAC offset values (both channels at once) */
+	offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+	offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT;
+
+	do {
+		offset[DA732X_HPL_DAC] |= step;
+		offset[DA732X_HPR_DAC] |= step;
+		snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+			      ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK);
+		snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+			      ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK);
+
+		msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+		if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) &
+		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC])
+			offset[DA732X_HPL_DAC] &= ~step;
+		if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) &
+		     DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC])
+			offset[DA732X_HPR_DAC] &= ~step;
+
+		step >>= 1;
+	} while (step);
+
+	/* Write final DAC offsets to registers */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET,
+		      ~offset[DA732X_HPL_DAC] &	DA732X_HP_DAC_OFF_MASK);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET,
+		      ~offset[DA732X_HPR_DAC] &	DA732X_HP_DAC_OFF_MASK);
+
+	/* End DAC calibration mode */
+	snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL,
+		DA732X_HP_DAC_OFF_SCALE_STEPS);
+	snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL,
+		DA732X_HP_DAC_OFF_SCALE_STEPS);
+}
+
+static void da732x_output_offset_adjust(struct snd_soc_codec *codec)
+{
+	u8 offset[DA732X_HP_AMPS];
+	u8 sign[DA732X_HP_AMPS];
+	u8 step = DA732X_OUTPUT_OFFSET_STEP;
+
+	offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL;
+	offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL;
+
+	/* Initialize output offset calibration circuits and registers  */
+	snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL);
+	snd_soc_write(codec, DA732X_REG_HPL,
+		      DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR,
+		      DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN);
+
+	/* Wait for voltage stabilization */
+	msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+	/* Check output offset sign */
+	sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) &
+			       DA732X_HP_OUT_COMPO;
+	sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) &
+			       DA732X_HP_OUT_COMPO;
+
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP |
+		      (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+		      DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP |
+		      (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) |
+		      DA732X_HP_OUT_EN);
+
+	/* Binary search output offset values (both channels at once) */
+	do {
+		offset[DA732X_HPL_AMP] |= step;
+		offset[DA732X_HPR_AMP] |= step;
+		snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET,
+			      offset[DA732X_HPL_AMP]);
+		snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET,
+			      offset[DA732X_HPR_AMP]);
+
+		msleep(DA732X_WAIT_FOR_STABILIZATION);
+
+		if ((codec->hw_read(codec, DA732X_REG_HPL) &
+		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP])
+			offset[DA732X_HPL_AMP] &= ~step;
+		if ((codec->hw_read(codec, DA732X_REG_HPR) &
+		     DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP])
+			offset[DA732X_HPR_AMP] &= ~step;
+
+		step >>= 1;
+	} while (step);
+
+	/* Write final DAC offsets to registers */
+	snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]);
+	snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]);
+}
+
+static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec)
+{
+	/* Make sure that we have Soft Mute enabled */
+	snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN |
+		      DA732X_GAIN_RAMPED | DA732X_16_SAMPLES);
+	snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN |
+		      DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM |
+		      DA732X_DACL_MUTE | DA732X_DACR_MUTE);
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN |
+		      DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN |
+		      DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN);
+
+	da732x_dac_offset_adjust(codec);
+	da732x_output_offset_adjust(codec);
+
+	snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS);
+	snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS);
+	snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS);
+}
+
+static int da732x_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+				    DA732X_BIAS_BOOST_MASK,
+				    DA732X_BIAS_BOOST_100PC);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			/* Init Codec */
+			snd_soc_write(codec, DA732X_REG_REF1,
+				      DA732X_VMID_FASTCHG);
+			snd_soc_write(codec, DA732X_REG_BIAS_EN,
+				      DA732X_BIAS_EN);
+
+			mdelay(DA732X_STARTUP_DELAY);
+
+			/* Disable Fast Charge and enable DAC ref voltage */
+			snd_soc_write(codec, DA732X_REG_REF1,
+				      DA732X_REFBUFX2_EN);
+
+			/* Enable bypass DSP routing */
+			snd_soc_write(codec, DA732X_REG_DATA_ROUTE,
+				      DA732X_BYPASS_DSP);
+
+			/* Enable Digital subsystem */
+			snd_soc_write(codec, DA732X_REG_DSP_CTRL,
+				      DA732X_DIGITAL_EN);
+
+			snd_soc_write(codec, DA732X_REG_SPARE1_OUT,
+				      DA732X_HP_DRIVER_EN |
+				      DA732X_HP_GATE_LOW |
+				      DA732X_HP_LOOP_GAIN_CTRL);
+			snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL,
+				      DA732X_HP_OUT_GNDSEL);
+
+			da732x_set_charge_pump(codec, DA732X_ENABLE_CP);
+
+			snd_soc_write(codec, DA732X_REG_CLK_EN1,
+			      DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN);
+
+			/* Enable Zero Crossing */
+			snd_soc_write(codec, DA732X_REG_INP_ZC_EN,
+				      DA732X_MIC1_PRE_ZC_EN |
+				      DA732X_MIC1_ZC_EN |
+				      DA732X_MIC2_PRE_ZC_EN |
+				      DA732X_MIC2_ZC_EN |
+				      DA732X_AUXL_ZC_EN |
+				      DA732X_AUXR_ZC_EN |
+				      DA732X_MIC3_PRE_ZC_EN |
+				      DA732X_MIC3_ZC_EN);
+			snd_soc_write(codec, DA732X_REG_OUT_ZC_EN,
+				      DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN |
+				      DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN |
+				      DA732X_LIN4_ZC_EN);
+
+			da732x_hp_dc_offset_cancellation(codec);
+
+			regcache_cache_only(codec->control_data, false);
+			regcache_sync(codec->control_data);
+		} else {
+			snd_soc_update_bits(codec, DA732X_REG_BIAS_EN,
+					    DA732X_BIAS_BOOST_MASK,
+					    DA732X_BIAS_BOOST_50PC);
+			snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL,
+					    DA732X_PLL_EN, 0);
+			da732x->pll_en = false;
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		regcache_cache_only(codec->control_data, true);
+		da732x_set_charge_pump(codec, DA732X_DISABLE_CP);
+		snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN,
+				    DA732X_BIAS_DIS);
+		da732x->pll_en = false;
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int da732x_probe(struct snd_soc_codec *codec)
+{
+	struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret = 0;
+
+	da732x->codec = codec;
+
+	dapm->idle_bias_off = false;
+
+	codec->control_data = da732x->regmap;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec.\n");
+		goto err;
+	}
+
+	da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+err:
+	return ret;
+}
+
+static int da732x_remove(struct snd_soc_codec *codec)
+{
+
+	da732x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_da732x = {
+	.probe			= da732x_probe,
+	.remove			= da732x_remove,
+	.set_bias_level		= da732x_set_bias_level,
+	.controls		= da732x_snd_controls,
+	.num_controls		= ARRAY_SIZE(da732x_snd_controls),
+	.dapm_widgets		= da732x_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(da732x_dapm_widgets),
+	.dapm_routes		= da732x_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(da732x_dapm_routes),
+	.set_pll		= da732x_set_dai_pll,
+	.reg_cache_size		= ARRAY_SIZE(da732x_reg_cache),
+};
+
+static __devinit int da732x_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct da732x_priv *da732x;
+	unsigned int reg;
+	int ret;
+
+	da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv),
+			      GFP_KERNEL);
+	if (!da732x)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, da732x);
+
+	da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap);
+	if (IS_ERR(da732x->regmap)) {
+		ret = PTR_ERR(da732x->regmap);
+		dev_err(&i2c->dev, "Failed to initialize regmap\n");
+		goto err;
+	}
+
+	ret = regmap_read(da732x->regmap, DA732X_REG_ID, &reg);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+		goto err;
+	}
+
+	dev_info(&i2c->dev, "Revision: %d.%d\n",
+		 (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK));
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x,
+				     da732x_dai, ARRAY_SIZE(da732x_dai));
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to register codec.\n");
+
+err:
+	return ret;
+}
+
+static __devexit int da732x_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+
+	return 0;
+}
+
+static const struct i2c_device_id da732x_i2c_id[] = {
+	{ "da7320", 0},
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, da732x_i2c_id);
+
+static struct i2c_driver da732x_i2c_driver = {
+	.driver		= {
+		.name	= "da7320",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= da732x_i2c_probe,
+	.remove		= __devexit_p(da732x_i2c_remove),
+	.id_table	= da732x_i2c_id,
+};
+
+module_i2c_driver(da732x_i2c_driver);
+
+
+MODULE_DESCRIPTION("ASoC DA732X driver");
+MODULE_AUTHOR("Michal Hajduk <michal.hajduk@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h
new file mode 100644
index 0000000..c8ce547
--- /dev/null
+++ b/sound/soc/codecs/da732x.h
@@ -0,0 +1,133 @@
+/*
+ * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __DA732X_H_
+#define __DA732X_H
+
+#include <sound/soc.h>
+
+/* General */
+#define	DA732X_U8_MASK			0xFF
+#define	DA732X_4BYTES			4
+#define	DA732X_3BYTES			3
+#define	DA732X_2BYTES			2
+#define	DA732X_1BYTE			1
+#define	DA732X_1BYTE_SHIFT		8
+#define	DA732X_2BYTES_SHIFT		16
+#define	DA732X_3BYTES_SHIFT		24
+#define	DA732X_4BYTES_SHIFT		32
+
+#define	DA732X_DACS_DIS			0x0
+#define	DA732X_HP_DIS			0x0
+#define	DA732X_CLEAR_REG		0x0
+
+/* Calibration */
+#define	DA732X_DAC_OFFSET_STEP		0x20
+#define	DA732X_OUTPUT_OFFSET_STEP	0x80
+#define	DA732X_HP_OUT_TRIM_VAL		0x0
+#define	DA732X_WAIT_FOR_STABILIZATION	1
+#define	DA732X_HPL_DAC			0
+#define	DA732X_HPR_DAC			1
+#define	DA732X_HP_DACS			2
+#define	DA732X_HPL_AMP			0
+#define	DA732X_HPR_AMP			1
+#define	DA732X_HP_AMPS			2
+
+/* Clock settings */
+#define DA732X_STARTUP_DELAY		100
+#define	DA732X_PLL_OUT_196608		196608000
+#define	DA732X_PLL_OUT_180634		180633600
+#define	DA732X_PLL_OUT_SRM		188620800
+#define	DA732X_MCLK_10MHZ		10000000
+#define	DA732X_MCLK_20MHZ		20000000
+#define	DA732X_MCLK_40MHZ		40000000
+#define	DA732X_MCLK_54MHZ		54000000
+#define	DA732X_MCLK_RET_0_10MHZ		0
+#define	DA732X_MCLK_VAL_0_10MHZ		1
+#define	DA732X_MCLK_RET_10_20MHZ	1
+#define	DA732X_MCLK_VAL_10_20MHZ	2
+#define	DA732X_MCLK_RET_20_40MHZ	2
+#define	DA732X_MCLK_VAL_20_40MHZ	4
+#define	DA732X_MCLK_RET_40_54MHZ	3
+#define	DA732X_MCLK_VAL_40_54MHZ	8
+#define	DA732X_DAI_ID1			0
+#define	DA732X_DAI_ID2			1
+#define	DA732X_SRCCLK_PLL		0
+#define	DA732X_SRCCLK_MCLK		1
+
+#define	DA732X_LIN_LP_VOL		0x4F
+#define	DA732X_LP_VOL			0x40
+
+/* Kcontrols */
+#define	DA732X_DAC_EN_MAX		2
+#define	DA732X_ADCL_MUX_MAX		2
+#define	DA732X_ADCR_MUX_MAX		3
+#define	DA732X_HPF_MODE_MAX		3
+#define	DA732X_HPF_MODE_SHIFT		4
+#define	DA732X_HPF_MUSIC_SHIFT		0
+#define	DA732X_HPF_MUSIC_MAX		4
+#define	DA732X_HPF_VOICE_SHIFT		4
+#define	DA732X_HPF_VOICE_MAX		8
+#define	DA732X_EQ_EN_MAX		1
+#define	DA732X_HPF_VOICE		1
+#define	DA732X_HPF_MUSIC		2
+#define	DA732X_HPF_DISABLED		0
+#define	DA732X_NO_INVERT		0
+#define	DA732X_INVERT			1
+#define	DA732X_SWITCH_MAX		1
+#define	DA732X_ENABLE_CP		1
+#define	DA732X_DISABLE_CP		0
+#define	DA732X_DISABLE_ALL_CLKS		0
+#define	DA732X_RESET_ADCS		0
+
+/* dB values */
+#define DA732X_MIC_VOL_DB_MIN		0
+#define DA732X_MIC_VOL_DB_INC		50
+#define DA732X_MIC_PRE_VOL_DB_MIN	0
+#define DA732X_MIC_PRE_VOL_DB_INC	600
+#define DA732X_AUX_VOL_DB_MIN		-6000
+#define DA732X_AUX_VOL_DB_INC		150
+#define DA732X_HP_VOL_DB_MIN		-2250
+#define DA732X_HP_VOL_DB_INC		150
+#define	DA732X_LIN2_VOL_DB_MIN		-1650
+#define	DA732X_LIN2_VOL_DB_INC		150
+#define	DA732X_LIN3_VOL_DB_MIN		-1650
+#define DA732X_LIN3_VOL_DB_INC		150
+#define	DA732X_LIN4_VOL_DB_MIN		-2250
+#define DA732X_LIN4_VOL_DB_INC		150
+#define	DA732X_EQ_BAND_VOL_DB_MIN	-1050
+#define	DA732X_EQ_BAND_VOL_DB_INC	150
+#define DA732X_DAC_VOL_DB_MIN		-7725
+#define DA732X_DAC_VOL_DB_INC		75
+#define DA732X_ADC_VOL_DB_MIN		0
+#define DA732X_ADC_VOL_DB_INC		-1
+#define	DA732X_EQ_OVERALL_VOL_DB_MIN	-1800
+#define	DA732X_EQ_OVERALL_VOL_DB_INC	600
+
+#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \
+	{.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext}
+
+enum da732x_sysctl {
+	DA732X_SR_8KHZ		= 0x1,
+	DA732X_SR_11_025KHZ	= 0x2,
+	DA732X_SR_12KHZ		= 0x3,
+	DA732X_SR_16KHZ		= 0x5,
+	DA732X_SR_22_05KHZ	= 0x6,
+	DA732X_SR_24KHZ		= 0x7,
+	DA732X_SR_32KHZ		= 0x9,
+	DA732X_SR_44_1KHZ	= 0xA,
+	DA732X_SR_48KHZ		= 0xB,
+	DA732X_SR_88_1KHZ	= 0xE,
+	DA732X_SR_96KHZ		= 0xF,
+};
+
+#endif /* __DA732X_H_ */
diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h
new file mode 100644
index 0000000..bdd03ca
--- /dev/null
+++ b/sound/soc/codecs/da732x_reg.h
@@ -0,0 +1,654 @@
+/*
+ * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File
+ *
+ * Copyright (C) 2012 Dialog Semiconductor GmbH
+ *
+ * Author: Michal Hajduk <Michal.Hajduk@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __DA732X_REG_H_
+#define __DA732X_REG_H_
+
+/* DA732X registers */
+#define	DA732X_REG_STATUS_EXT		0x00
+#define DA732X_REG_STATUS		0x01
+#define DA732X_REG_REF1			0x02
+#define DA732X_REG_BIAS_EN		0x03
+#define DA732X_REG_BIAS1		0x04
+#define DA732X_REG_BIAS2		0x05
+#define DA732X_REG_BIAS3		0x06
+#define DA732X_REG_BIAS4		0x07
+#define DA732X_REG_MICBIAS2		0x0F
+#define DA732X_REG_MICBIAS1		0x10
+#define DA732X_REG_MICDET		0x11
+#define DA732X_REG_MIC1_PRE		0x12
+#define DA732X_REG_MIC1			0x13
+#define DA732X_REG_MIC2_PRE		0x14
+#define DA732X_REG_MIC2			0x15
+#define DA732X_REG_AUX1L		0x16
+#define DA732X_REG_AUX1R		0x17
+#define DA732X_REG_MIC3_PRE		0x18
+#define DA732X_REG_MIC3			0x19
+#define DA732X_REG_INP_PINBIAS		0x1A
+#define DA732X_REG_INP_ZC_EN		0x1B
+#define DA732X_REG_INP_MUX		0x1D
+#define DA732X_REG_HP_DET		0x20
+#define DA732X_REG_HPL_DAC_OFFSET	0x21
+#define DA732X_REG_HPL_DAC_OFF_CNTL	0x22
+#define DA732X_REG_HPL_OUT_OFFSET	0x23
+#define DA732X_REG_HPL			0x24
+#define DA732X_REG_HPL_VOL		0x25
+#define DA732X_REG_HPR_DAC_OFFSET	0x26
+#define DA732X_REG_HPR_DAC_OFF_CNTL	0x27
+#define DA732X_REG_HPR_OUT_OFFSET	0x28
+#define DA732X_REG_HPR			0x29
+#define DA732X_REG_HPR_VOL		0x2A
+#define DA732X_REG_LIN2			0x2B
+#define DA732X_REG_LIN3			0x2C
+#define DA732X_REG_LIN4			0x2D
+#define DA732X_REG_OUT_ZC_EN		0x2E
+#define DA732X_REG_HP_LIN1_GNDSEL	0x37
+#define DA732X_REG_CP_HP1		0x3A
+#define DA732X_REG_CP_HP2		0x3B
+#define DA732X_REG_CP_CTRL1		0x40
+#define DA732X_REG_CP_CTRL2		0x41
+#define DA732X_REG_CP_CTRL3		0x42
+#define DA732X_REG_CP_LEVEL_MASK	0x43
+#define DA732X_REG_CP_DET		0x44
+#define DA732X_REG_CP_STATUS		0x45
+#define DA732X_REG_CP_THRESH1		0x46
+#define DA732X_REG_CP_THRESH2		0x47
+#define DA732X_REG_CP_THRESH3		0x48
+#define DA732X_REG_CP_THRESH4		0x49
+#define DA732X_REG_CP_THRESH5		0x4A
+#define DA732X_REG_CP_THRESH6		0x4B
+#define DA732X_REG_CP_THRESH7		0x4C
+#define DA732X_REG_CP_THRESH8		0x4D
+#define DA732X_REG_PLL_DIV_LO		0x50
+#define DA732X_REG_PLL_DIV_MID		0x51
+#define DA732X_REG_PLL_DIV_HI		0x52
+#define DA732X_REG_PLL_CTRL		0x53
+#define DA732X_REG_CLK_CTRL		0x54
+#define DA732X_REG_CLK_DSP		0x5A
+#define DA732X_REG_CLK_EN1		0x5B
+#define DA732X_REG_CLK_EN2		0x5C
+#define DA732X_REG_CLK_EN3		0x5D
+#define DA732X_REG_CLK_EN4		0x5E
+#define DA732X_REG_CLK_EN5		0x5F
+#define DA732X_REG_AIF_MCLK		0x60
+#define DA732X_REG_AIFA1		0x61
+#define DA732X_REG_AIFA2		0x62
+#define DA732X_REG_AIFA3		0x63
+#define DA732X_REG_AIFB1		0x64
+#define DA732X_REG_AIFB2		0x65
+#define DA732X_REG_AIFB3		0x66
+#define DA732X_REG_PC_CTRL		0x6A
+#define DA732X_REG_DATA_ROUTE		0x70
+#define DA732X_REG_DSP_CTRL		0x71
+#define DA732X_REG_CIF_CTRL2		0x74
+#define DA732X_REG_HANDSHAKE		0x75
+#define DA732X_REG_MBOX0		0x76
+#define DA732X_REG_MBOX1		0x77
+#define DA732X_REG_MBOX2		0x78
+#define DA732X_REG_MBOX_STATUS		0x79
+#define DA732X_REG_SPARE1_OUT		0x7D
+#define DA732X_REG_SPARE2_OUT		0x7E
+#define DA732X_REG_SPARE1_IN		0x7F
+#define DA732X_REG_ID			0x81
+#define DA732X_REG_ADC1_PD		0x90
+#define DA732X_REG_ADC1_HPF		0x93
+#define DA732X_REG_ADC1_SEL		0x94
+#define DA732X_REG_ADC1_EQ12		0x95
+#define DA732X_REG_ADC1_EQ34		0x96
+#define DA732X_REG_ADC1_EQ5		0x97
+#define DA732X_REG_ADC2_PD		0x98
+#define DA732X_REG_ADC2_HPF		0x9B
+#define DA732X_REG_ADC2_SEL		0x9C
+#define DA732X_REG_ADC2_EQ12		0x9D
+#define DA732X_REG_ADC2_EQ34		0x9E
+#define DA732X_REG_ADC2_EQ5		0x9F
+#define DA732X_REG_DAC1_HPF		0xA0
+#define DA732X_REG_DAC1_L_VOL		0xA1
+#define DA732X_REG_DAC1_R_VOL		0xA2
+#define DA732X_REG_DAC1_SEL		0xA3
+#define DA732X_REG_DAC1_SOFTMUTE	0xA4
+#define DA732X_REG_DAC1_EQ12		0xA5
+#define DA732X_REG_DAC1_EQ34		0xA6
+#define DA732X_REG_DAC1_EQ5		0xA7
+#define DA732X_REG_DAC2_HPF		0xB0
+#define DA732X_REG_DAC2_L_VOL		0xB1
+#define DA732X_REG_DAC2_R_VOL		0xB2
+#define DA732X_REG_DAC2_SEL		0xB3
+#define DA732X_REG_DAC2_SOFTMUTE	0xB4
+#define DA732X_REG_DAC2_EQ12		0xB5
+#define DA732X_REG_DAC2_EQ34		0xB6
+#define DA732X_REG_DAC2_EQ5		0xB7
+#define DA732X_REG_DAC3_HPF		0xC0
+#define DA732X_REG_DAC3_VOL		0xC1
+#define DA732X_REG_DAC3_SEL		0xC3
+#define DA732X_REG_DAC3_SOFTMUTE	0xC4
+#define DA732X_REG_DAC3_EQ12		0xC5
+#define DA732X_REG_DAC3_EQ34		0xC6
+#define DA732X_REG_DAC3_EQ5		0xC7
+#define DA732X_REG_BIQ_BYP		0xD2
+#define DA732X_REG_DMA_CMD		0xD3
+#define DA732X_REG_DMA_ADDR0		0xD4
+#define DA732X_REG_DMA_ADDR1		0xD5
+#define DA732X_REG_DMA_DATA0		0xD6
+#define DA732X_REG_DMA_DATA1		0xD7
+#define DA732X_REG_DMA_DATA2		0xD8
+#define DA732X_REG_DMA_DATA3		0xD9
+#define DA732X_REG_DMA_STATUS		0xDA
+#define DA732X_REG_BROWNOUT		0xDF
+#define DA732X_REG_UNLOCK		0xE0
+
+#define	DA732X_MAX_REG			DA732X_REG_UNLOCK
+/*
+ * Bits
+ */
+
+/* DA732X_REG_STATUS_EXT (addr=0x00) */
+#define	DA732X_STATUS_EXT_DSP			(1 << 4)
+#define	DA732X_STATUS_EXT_CLEAR			(0 << 0)
+
+/* DA732X_REG_STATUS	(addr=0x01) */
+#define DA732X_STATUS_PLL_LOCK			(1 << 0)
+#define DA732X_STATUS_PLL_MCLK_DET		(1 << 1)
+#define DA732X_STATUS_HPDET_OUT			(1 << 2)
+#define DA732X_STATUS_INP_MIXDET_1		(1 << 3)
+#define DA732X_STATUS_INP_MIXDET_2		(1 << 4)
+#define DA732X_STATUS_BO_STATUS			(1 << 5)
+
+/* DA732X_REG_REF1	(addr=0x02) */
+#define DA732X_VMID_FASTCHG			(1 << 1)
+#define DA732X_VMID_FASTDISCHG			(1 << 2)
+#define DA732X_REFBUFX2_EN			(1 << 6)
+#define DA732X_REFBUFX2_DIS			(0 << 6)
+
+/* DA732X_REG_BIAS_EN	(addr=0x03) */
+#define DA732X_BIAS_BOOST_MASK			(3 << 0)
+#define DA732X_BIAS_BOOST_100PC			(0 << 0)
+#define DA732X_BIAS_BOOST_133PC			(1 << 0)
+#define DA732X_BIAS_BOOST_88PC			(2 << 0)
+#define DA732X_BIAS_BOOST_50PC			(3 << 0)
+#define DA732X_BIAS_EN				(1 << 7)
+#define DA732X_BIAS_DIS				(0 << 7)
+
+/* DA732X_REG_BIAS1	(addr=0x04) */
+#define DA732X_BIAS1_HP_DAC_BIAS_MASK		(3 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_100PC		(0 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_150PC		(1 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_50PC		(2 << 0)
+#define DA732X_BIAS1_HP_DAC_BIAS_75PC		(3 << 0)
+#define DA732X_BIAS1_HP_OUT_BIAS_MASK		(7 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_100PC		(0 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_125PC		(1 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_150PC		(2 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_175PC		(3 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_200PC		(4 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_250PC		(5 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_300PC		(6 << 4)
+#define DA732X_BIAS1_HP_OUT_BIAS_350PC		(7 << 4)
+
+/* DA732X_REG_BIAS2	(addr=0x05) */
+#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_BIAS3	(addr=0x06) */
+#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_BIAS4	(addr=0x07) */
+#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK	(3 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC	(0 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC	(1 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC	(2 << 0)
+#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC	(3 << 0)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK	(7 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC	(0 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC	(1 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC	(2 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC	(3 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC	(4 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC	(5 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC	(6 << 4)
+#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC	(7 << 4)
+
+/* DA732X_REG_SIF_VDD_SEL	(addr=0x08) */
+#define DA732X_SIF_VDD_SEL_AIFA_VDD2		(1 << 0)
+#define DA732X_SIF_VDD_SEL_AIFB_VDD2		(1 << 1)
+#define DA732X_SIF_VDD_SEL_CIFA_VDD2		(1 << 4)
+
+/* DA732X_REG_MICBIAS2/1	(addr=0x0F/0x10) */
+#define DA732X_MICBIAS_VOLTAGE_MASK		(0x0F << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V		(0x00 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V05		(0x01 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V1		(0x02 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V15		(0x03 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V2		(0x04 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V25		(0x05 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V3		(0x06 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V35		(0x07 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V4		(0x08 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V45		(0x09 << 0)
+#define DA732X_MICBIAS_VOLTAGE_2V5		(0x0A << 0)
+#define DA732X_MICBIAS_EN			(1 << 7)
+#define DA732X_MICBIAS_EN_SHIFT			7
+#define DA732X_MICBIAS_VOLTAGE_SHIFT		0
+#define	DA732X_MICBIAS_VOLTAGE_MAX		0x0B
+
+/* DA732X_REG_MICDET	(addr=0x11) */
+#define DA732X_MICDET_INP_MICRES		(1 << 0)
+#define DA732X_MICDET_INP_MICHOOK		(1 << 1)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS	(0 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS	(1 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS	(2 << 0)
+#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS	(3 << 0)
+#define DA732X_MICDET_INP_MICDET_EN		(1 << 7)
+
+/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */
+#define	DA732X_MICBOOST_MASK			0x7
+#define	DA732X_MICBOOST_SHIFT			0
+#define	DA732X_MICBOOST_MIN			0x1
+#define	DA732X_MICBOOST_MAX			DA732X_MICBOOST_MASK
+
+/* DA732X_REG_MIC1/2/3	(addr=0x13/0x15/0x19) */
+#define	DA732X_MIC_VOL_SHIFT			0
+#define	DA732X_MIC_VOL_VAL_MASK			0x1F
+#define DA732X_MIC_MUTE_SHIFT			6
+#define DA732X_MIC_EN_SHIFT			7
+#define DA732X_MIC_VOL_VAL_MIN			0x7
+#define	DA732X_MIC_VOL_VAL_MAX			DA732X_MIC_VOL_VAL_MASK
+
+/* DA732X_REG_AUX1L/R	(addr=0x16/0x17) */
+#define	DA732X_AUX_VOL_SHIFT			0
+#define	DA732X_AUX_VOL_MASK			0x7
+#define DA732X_AUX_MUTE_SHIFT			6
+#define DA732X_AUX_EN_SHIFT			7
+#define	DA732X_AUX_VOL_VAL_MAX			DA732X_AUX_VOL_MASK
+
+/* DA732X_REG_INP_PINBIAS	(addr=0x1A) */
+#define DA732X_INP_MICL_PINBIAS_EN		(1 << 0)
+#define DA732X_INP_MICR_PINBIAS_EN		(1 << 1)
+#define DA732X_INP_AUX1L_PINBIAS_EN		(1 << 2)
+#define DA732X_INP_AUX1R_PINBIAS_EN		(1 << 3)
+#define DA732X_INP_AUX2_PINBIAS_EN		(1 << 4)
+
+/* DA732X_REG_INP_ZC_EN	(addr=0x1B) */
+#define	DA732X_MIC1_PRE_ZC_EN			(1 << 0)
+#define	DA732X_MIC1_ZC_EN			(1 << 1)
+#define	DA732X_MIC2_PRE_ZC_EN			(1 << 2)
+#define	DA732X_MIC2_ZC_EN			(1 << 3)
+#define	DA732X_AUXL_ZC_EN			(1 << 4)
+#define	DA732X_AUXR_ZC_EN			(1 << 5)
+#define	DA732X_MIC3_PRE_ZC_EN			(1 << 6)
+#define	DA732X_MIC3_ZC_EN			(1 << 7)
+
+/* DA732X_REG_INP_MUX	(addr=0x1D) */
+#define DA732X_INP_ADC1L_MUX_SEL_AUX1L		(0 << 0)
+#define DA732X_INP_ADC1L_MUX_SEL_MIC1		(1 << 0)
+#define DA732X_INP_ADC1R_MUX_SEL_MASK		(3 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_AUX1R		(0 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC2		(1 << 2)
+#define DA732X_INP_ADC1R_MUX_SEL_MIC3		(2 << 2)
+#define DA732X_INP_ADC2L_MUX_SEL_AUX1L		(0 << 4)
+#define DA732X_INP_ADC2L_MUX_SEL_MICL		(1 << 4)
+#define DA732X_INP_ADC2R_MUX_SEL_MASK		(3 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX1R		(0 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_MICR		(1 << 6)
+#define DA732X_INP_ADC2R_MUX_SEL_AUX2		(2 << 6)
+#define	DA732X_ADC1L_MUX_SEL_SHIFT		0
+#define	DA732X_ADC1R_MUX_SEL_SHIFT		2
+#define	DA732X_ADC2L_MUX_SEL_SHIFT		4
+#define	DA732X_ADC2R_MUX_SEL_SHIFT		6
+
+/* DA732X_REG_HP_DET		(addr=0x20) */
+#define DA732X_HP_DET_AZ			(1 << 0)
+#define DA732X_HP_DET_SEL1			(1 << 1)
+#define DA732X_HP_DET_IS_MASK			(3 << 2)
+#define DA732X_HP_DET_IS_0_5UA			(0 << 2)
+#define DA732X_HP_DET_IS_1UA			(1 << 2)
+#define DA732X_HP_DET_IS_2UA			(2 << 2)
+#define DA732X_HP_DET_IS_4UA			(3 << 2)
+#define DA732X_HP_DET_RS_MASK			(3 << 4)
+#define DA732X_HP_DET_RS_INFINITE		(0 << 4)
+#define DA732X_HP_DET_RS_100KOHM		(1 << 4)
+#define DA732X_HP_DET_RS_10KOHM			(2 << 4)
+#define DA732X_HP_DET_RS_1KOHM			(3 << 4)
+#define DA732X_HP_DET_EN			(1 << 7)
+
+/* DA732X_REG_HPL_DAC_OFFSET	(addr=0x21/0x26) */
+#define DA732X_HP_DAC_OFFSET_TRIM_MASK		(0x3F << 0)
+#define DA732X_HP_DAC_OFFSET_DAC_SIGN		(1 << 6)
+
+/* DA732X_REG_HPL_DAC_OFF_CNTL	(addr=0x22/0x27) */
+#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK	(7 << 0)
+#define DA732X_HP_DAC_OFF_CNTL_COMPO		(1 << 3)
+#define	DA732X_HP_DAC_OFF_CALIBRATION		(1 << 0)
+#define	DA732X_HP_DAC_OFF_SCALE_STEPS		(1 << 1)
+#define	DA732X_HP_DAC_OFF_MASK			0x7F
+#define DA732X_HP_DAC_COMPO_SHIFT		3
+
+/* DA732X_REG_HPL_OUT_OFFSET	(addr=0x23/0x28) */
+#define DA732X_HP_OUT_OFFSET_MASK		(0xFF << 0)
+#define	DA732X_HP_DAC_OFFSET_TRIM_VAL		0x7F
+
+/* DA732X_REG_HPL/R	(addr=0x24/0x29) */
+#define DA732X_HP_OUT_SIGN			(1 << 0)
+#define DA732X_HP_OUT_COMP			(1 << 1)
+#define DA732X_HP_OUT_RESERVED			(1 << 2)
+#define DA732X_HP_OUT_COMPO			(1 << 3)
+#define DA732X_HP_OUT_DAC_EN			(1 << 4)
+#define DA732X_HP_OUT_HIZ_EN			(1 << 5)
+#define	DA732X_HP_OUT_HIZ_DIS			(0 << 5)
+#define DA732X_HP_OUT_MUTE			(1 << 6)
+#define DA732X_HP_OUT_EN			(1 << 7)
+#define	DA732X_HP_OUT_COMPO_SHIFT		3
+#define	DA732X_HP_OUT_DAC_EN_SHIFT		4
+#define	DA732X_HP_HIZ_SHIFT			5
+#define	DA732X_HP_MUTE_SHIFT			6
+#define DA732X_HP_OUT_EN_SHIFT			7
+
+#define DA732X_OUT_HIZ_EN			(1 << 5)
+#define	DA732X_OUT_HIZ_DIS			(0 << 5)
+
+/* DA732X_REG_HPL/R_VOL	(addr=0x25/0x2A) */
+#define	DA732X_HP_VOL_VAL_MASK			0xF
+#define	DA732X_HP_VOL_SHIFT			0
+#define	DA732X_HP_VOL_VAL_MAX			DA732X_HP_VOL_VAL_MASK
+
+/* DA732X_REG_LIN2/3/4	(addr=0x2B/0x2C/0x2D) */
+#define DA732X_LOUT_VOL_SHIFT			0
+#define DA732X_LOUT_VOL_MASK			0x0F
+#define DA732X_LOUT_DAC_OFF			(0 << 4)
+#define DA732X_LOUT_DAC_EN			(1 << 4)
+#define DA732X_LOUT_HIZ_N_DIS			(0 << 5)
+#define DA732X_LOUT_HIZ_N_EN			(1 << 5)
+#define DA732X_LOUT_UNMUTED			(0 << 6)
+#define DA732X_LOUT_MUTED			(1 << 6)
+#define DA732X_LOUT_EN				(0 << 7)
+#define DA732X_LOUT_DIS				(1 << 7)
+#define DA732X_LOUT_DAC_EN_SHIFT		4
+#define	DA732X_LOUT_MUTE_SHIFT			6
+#define DA732X_LIN_OUT_EN_SHIFT			7
+#define DA732X_LOUT_VOL_VAL_MAX			DA732X_LOUT_VOL_MASK
+
+/* DA732X_REG_OUT_ZC_EN		(addr=0x2E) */
+#define	DA732X_HPL_ZC_EN_SHIFT			0
+#define DA732X_HPR_ZC_EN_SHIFT			1
+#define DA732X_HPL_ZC_EN			(1 << 0)
+#define DA732X_HPL_ZC_DIS			(0 << 0)
+#define DA732X_HPR_ZC_EN			(1 << 1)
+#define DA732X_HPR_ZC_DIS			(0 << 1)
+#define DA732X_LIN2_ZC_EN			(1 << 2)
+#define DA732X_LIN2_ZC_DIS			(0 << 2)
+#define DA732X_LIN3_ZC_EN			(1 << 3)
+#define DA732X_LIN3_ZC_DIS			(0 << 3)
+#define DA732X_LIN4_ZC_EN			(1 << 4)
+#define DA732X_LIN4_ZC_DIS			(0 << 4)
+
+/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */
+#define	DA732X_HP_OUT_GNDSEL			(1 << 0)
+
+/* DA732X_REG_CP_HP2 (addr=0x3a) */
+#define	DA732X_HP_CP_PULSESKIP			(1 << 0)
+#define	DA732X_HP_CP_REG			(1 << 1)
+#define DA732X_HP_CP_EN				(1 << 3)
+#define DA732X_HP_CP_DIS			(0 << 3)
+
+/* DA732X_REG_CP_CTRL1 (addr=0x40) */
+#define	DA732X_CP_MODE_MASK			(7 << 1)
+#define	DA732X_CP_CTRL_STANDBY			(0 << 1)
+#define	DA732X_CP_CTRL_CPVDD6			(2 << 1)
+#define	DA732X_CP_CTRL_CPVDD5			(3 << 1)
+#define	DA732X_CP_CTRL_CPVDD4			(4 << 1)
+#define	DA732X_CP_CTRL_CPVDD3			(5 << 1)
+#define	DA732X_CP_CTRL_CPVDD2			(6 << 1)
+#define	DA732X_CP_CTRL_CPVDD1			(7 << 1)
+#define	DA723X_CP_DIS				(0 << 7)
+#define	DA732X_CP_EN				(1 << 7)
+
+/* DA732X_REG_CP_CTRL2 (addr=0x41) */
+#define	DA732X_CP_BOOST				(1 << 0)
+#define	DA732X_CP_MANAGE_MAGNITUDE		(2 << 2)
+
+/* DA732X_REG_CP_CTRL3 (addr=0x42) */
+#define	DA732X_CP_1MHZ				(0 << 0)
+#define	DA732X_CP_500KHZ			(1 << 0)
+#define	DA732X_CP_250KHZ			(2 << 0)
+#define	DA732X_CP_125KHZ			(3 << 0)
+#define	DA732X_CP_63KHZ				(4 << 0)
+#define	DA732X_CP_0KHZ				(5 << 0)
+
+/* DA732X_REG_PLL_CTRL (addr=0x53) */
+#define	DA732X_PLL_INDIV_MASK			(3 << 0)
+#define	DA732X_PLL_SRM_EN			(1 << 2)
+#define	DA732X_PLL_EN				(1 << 7)
+#define	DA732X_PLL_BYPASS			(0 << 0)
+
+/* DA732X_REG_CLK_CTRL (addr=0x54) */
+#define	DA732X_SR1_MASK				(0xF)
+#define	DA732X_SR2_MASK				(0xF0)
+
+/* DA732X_REG_CLK_DSP (addr=0x5A) */
+#define	DA732X_DSP_FREQ_MASK			(7 << 0)
+#define	DA732X_DSP_FREQ_12MHZ			(0 << 0)
+#define	DA732X_DSP_FREQ_24MHZ			(1 << 0)
+#define	DA732X_DSP_FREQ_36MHZ			(2 << 0)
+#define	DA732X_DSP_FREQ_48MHZ			(3 << 0)
+#define	DA732X_DSP_FREQ_60MHZ			(4 << 0)
+#define	DA732X_DSP_FREQ_72MHZ			(5 << 0)
+#define	DA732X_DSP_FREQ_84MHZ			(6 << 0)
+#define	DA732X_DSP_FREQ_96MHZ			(7 << 0)
+
+/* DA732X_REG_CLK_EN1 (addr=0x5B) */
+#define	DA732X_DSP_CLK_EN			(1 << 0)
+#define	DA732X_SYS3_CLK_EN			(1 << 1)
+#define	DA732X_DSP12_CLK_EN			(1 << 2)
+#define	DA732X_PC_CLK_EN			(1 << 3)
+#define	DA732X_MCLK_SQR_EN			(1 << 7)
+
+/* DA732X_REG_CLK_EN2 (addr=0x5C) */
+#define	DA732X_UART_CLK_EN			(1 << 1)
+#define	DA732X_CP_CLK_EN			(1 << 2)
+#define	DA732X_CP_CLK_DIS			(0 << 2)
+
+/* DA732X_REG_CLK_EN3 (addr=0x5D) */
+#define	DA732X_ADCA_BB_CLK_EN			(1 << 0)
+#define	DA732X_ADCC_BB_CLK_EN			(1 << 4)
+
+/* DA732X_REG_CLK_EN4 (addr=0x5E) */
+#define	DA732X_DACA_BB_CLK_EN			(1 << 0)
+#define	DA732X_DACC_BB_CLK_EN			(1 << 4)
+#define DA732X_DACA_BB_CLK_SHIFT		0
+#define DA732X_DACC_BB_CLK_SHIFT		4
+
+/* DA732X_REG_CLK_EN5 (addr=0x5F) */
+#define	DA732X_DACE_BB_CLK_EN			(1 << 0)
+#define DA732X_DACE_BB_CLK_SHIFT		0
+
+/* DA732X_REG_AIF_MCLK (addr=0x60) */
+#define DA732X_AIFM_FRAME_64			(1 << 2)
+#define	DA732X_AIFM_SRC_SEL_AIFA		(1 << 6)
+#define	DA732X_CLK_GENERATION_AIF_A		(1 << 4)
+#define	DA732X_NO_CLK_GENERATION		0x0
+
+/* DA732X_REG_AIFA1 (addr=0x61) */
+#define	DA732X_AIF_WORD_MASK			(0x3 << 0)
+#define	DA732X_AIF_WORD_16			(0 << 0)
+#define	DA732X_AIF_WORD_20			(1 << 0)
+#define	DA732X_AIF_WORD_24			(2 << 0)
+#define	DA732X_AIF_WORD_32			(3 << 0)
+#define	DA732X_AIF_TDM_MONO_SHIFT		(1 << 6)
+#define	DA732X_AIF1_CLK_MASK			(1 << 7)
+#define	DA732X_AIF_SLAVE			(0 << 7)
+#define DA732X_AIF_CLK_FROM_SRC			(1 << 7)
+
+/* DA732X_REG_AIFA3 (addr=0x63) */
+#define	DA732X_AIF_MODE_SHIFT			0
+#define	DA732X_AIF_MODE_MASK			0x3
+#define	DA732X_AIF_I2S_MODE			(0 << 0)
+#define	DA732X_AIF_LEFT_J_MODE			(1 << 0)
+#define	DA732X_AIF_RIGHT_J_MODE			(2 << 0)
+#define	DA732X_AIF_DSP_MODE			(3 << 0)
+#define DA732X_AIF_WCLK_INV			(1 << 4)
+#define DA732X_AIF_BCLK_INV			(1 << 5)
+#define	DA732X_AIF_EN				(1 << 7)
+#define	DA732X_AIF_EN_SHIFT			7
+
+/* DA732X_REG_PC_CTRL (addr=0x6a) */
+#define	DA732X_PC_PULSE_AIFA			(0 << 0)
+#define	DA732X_PC_PULSE_AIFB			(1 << 0)
+#define	DA732X_PC_RESYNC_AUT			(1 << 6)
+#define	DA732X_PC_RESYNC_NOT_AUT		(0 << 6)
+#define	DA732X_PC_SAME				(1 << 7)
+
+/* DA732X_REG_DATA_ROUTE (addr=0x70) */
+#define DA732X_ADC1_TO_AIFA			(0 << 0)
+#define DA732X_DSP_TO_AIFA			(1 << 0)
+#define DA732X_ADC2_TO_AIFB			(0 << 1)
+#define DA732X_DSP_TO_AIFB			(1 << 1)
+#define DA732X_AIFA_TO_DAC1L			(0 << 2)
+#define DA732X_DSP_TO_DAC1L			(1 << 2)
+#define DA732X_AIFA_TO_DAC1R			(0 << 3)
+#define DA732X_DSP_TO_DAC1R			(1 << 3)
+#define DA732X_AIFB_TO_DAC2L			(0 << 4)
+#define DA732X_DSP_TO_DAC2L			(1 << 4)
+#define DA732X_AIFB_TO_DAC2R			(0 << 5)
+#define DA732X_DSP_TO_DAC2R			(1 << 5)
+#define DA732X_AIFB_TO_DAC3			(0 << 6)
+#define DA732X_DSP_TO_DAC3			(1 << 6)
+#define	DA732X_BYPASS_DSP			(0 << 0)
+#define	DA732X_ALL_TO_DSP			(0x7F << 0)
+
+/* DA732X_REG_DSP_CTRL (addr=0x71) */
+#define	DA732X_DIGITAL_EN			(1 << 0)
+#define	DA732X_DIGITAL_RESET			(0 << 0)
+#define	DA732X_DSP_CORE_EN			(1 << 1)
+#define	DA732X_DSP_CORE_RESET			(0 << 1)
+
+/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/
+#define	DA732X_HP_DRIVER_EN			(1 << 0)
+#define	DA732X_HP_GATE_LOW			(1 << 2)
+#define DA732X_HP_LOOP_GAIN_CTRL		(1 << 3)
+
+/* DA732X_REG_ID (addr=0x81)*/
+#define DA732X_ID_MINOR_MASK			(0xF << 0)
+#define DA732X_ID_MAJOR_MASK			(0xF << 4)
+
+/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */
+#define	DA732X_ADC_RST_MASK			(0x3 << 0)
+#define	DA732X_ADC_PD_MASK			(0x3 << 2)
+#define	DA732X_ADC_SET_ACT			(0x3 << 0)
+#define	DA732X_ADC_SET_RST			(0x0 << 0)
+#define	DA732X_ADC_ON				(0x3 << 2)
+#define	DA732X_ADC_OFF				(0x0 << 2)
+
+/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */
+#define	DA732X_ADC_VOL_VAL_MASK			0x7
+#define	DA732X_ADCL_VOL_SHIFT			0
+#define	DA732X_ADCR_VOL_SHIFT			4
+#define DA732X_ADCL_EN_SHIFT			2
+#define DA732X_ADCR_EN_SHIFT			3
+#define	DA732X_ADCL_EN				(1 << 2)
+#define	DA732X_ADCR_EN				(1 << 3)
+#define	DA732X_ADC_VOL_VAL_MAX			DA732X_ADC_VOL_VAL_MASK
+
+/*
+ * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b)
+ * DA732x_REG_DAC1/2/3_HPG	(addr=0xA5/0xB5/0xC5)
+ */
+#define	DA732X_HPF_MUSIC_EN			(1 << 3)
+#define	DA732X_HPF_VOICE_EN			((1 << 3) | (1 << 7))
+#define	DA732X_HPF_MASK				((1 << 3) | (1 << 7))
+#define DA732X_HPF_DIS				((0 << 3) | (0 << 7))
+
+/* DA732X_REG_DAC1/2/3_VOL */
+#define DA732X_DAC_VOL_VAL_MASK			0x7F
+#define DA732X_DAC_VOL_SHIFT			0
+#define DA732X_DAC_VOL_VAL_MAX			DA732X_DAC_VOL_VAL_MASK
+
+/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */
+#define DA732X_DACL_EN_SHIFT			3
+#define	DA732X_DACR_EN_SHIFT			7
+#define DA732X_DACL_MUTE_SHIFT			2
+#define	DA732X_DACR_MUTE_SHIFT			6
+#define DA732X_DACL_EN				(1 << 3)
+#define	DA732X_DACR_EN				(1 << 7)
+#define	DA732X_DACL_SDM				(1 << 0)
+#define	DA732X_DACR_SDM				(1 << 4)
+#define	DA732X_DACL_MUTE			(1 << 2)
+#define	DA732X_DACR_MUTE			(1 << 6)
+
+/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */
+#define	DA732X_SOFTMUTE_EN			(1 << 7)
+#define	DA732X_GAIN_RAMPED			(1 << 6)
+#define	DA732X_16_SAMPLES			(4 << 0)
+#define	DA732X_SOFTMUTE_MASK			(1 << 7)
+#define	DA732X_SOFTMUTE_SHIFT			7
+
+/*
+ * DA732x_REG_ADC1/2_EQ12	(addr=0x95/0x9D)
+ * DA732x_REG_ADC1/2_EQ34	(addr=0x96/0x9E)
+ * DA732x_REG_ADC1/2_EQ5	(addr=0x97/0x9F)
+ * DA732x_REG_DAC1/2/3_EQ12	(addr=0xA5/0xB5/0xC5)
+ * DA732x_REG_DAC1/2/3_EQ34	(addr=0xA6/0xB6/0xC6)
+ * DA732x_REG_DAC1/2/3_EQ5	(addr=0xA7/0xB7/0xB7)
+ */
+#define	DA732X_EQ_VOL_VAL_MASK			0xF
+#define	DA732X_EQ_BAND1_SHIFT			0
+#define	DA732X_EQ_BAND2_SHIFT			4
+#define	DA732X_EQ_BAND3_SHIFT			0
+#define	DA732X_EQ_BAND4_SHIFT			4
+#define	DA732X_EQ_BAND5_SHIFT			0
+#define	DA732X_EQ_OVERALL_SHIFT			4
+#define	DA732X_EQ_OVERALL_VOL_VAL_MASK		0x3
+#define	DA732X_EQ_DIS				(0 << 7)
+#define	DA732X_EQ_EN				(1 << 7)
+#define	DA732X_EQ_EN_SHIFT			7
+#define	DA732X_EQ_VOL_VAL_MAX			DA732X_EQ_VOL_VAL_MASK
+#define	DA732X_EQ_OVERALL_VOL_VAL_MAX		DA732X_EQ_OVERALL_VOL_VAL_MASK
+
+/* DA732X_REG_DMA_CMD (addr=0xD3) */
+#define	DA732X_SEL_DSP_DMA_MASK			(3 << 0)
+#define	DA732X_SEL_DSP_DMA_DIS			(0 << 0)
+#define	DA732X_SEL_DSP_DMA_PMEM			(1 << 0)
+#define	DA732X_SEL_DSP_DMA_XMEM			(2 << 0)
+#define	DA732X_SEL_DSP_DMA_YMEM			(3 << 0)
+#define	DA732X_DSP_RW_MASK			(1 << 4)
+#define	DA732X_DSP_DMA_WRITE			(0 << 4)
+#define	DA732X_DSP_DMA_READ			(1 << 4)
+
+/* DA732X_REG_DMA_STATUS (addr=0xDA) */
+#define	DA732X_DSP_DMA_FREE			(0 << 0)
+#define	DA732X_DSP_DMA_BUSY			(1 << 0)
+
+#endif /* __DA732X_REG_H_ */
diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c
new file mode 100644
index 0000000..5d8f39e
--- /dev/null
+++ b/sound/soc/codecs/isabelle.c
@@ -0,0 +1,1176 @@
+/*
+ * isabelle.c - Low power high fidelity audio codec driver
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ *
+ * Initially based on sound/soc/codecs/twl6040.c
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+#include "isabelle.h"
+
+
+/* Register default values for ISABELLE driver. */
+static struct reg_default isabelle_reg_defs[] = {
+	{ 0, 0x00 },
+	{ 1, 0x00 },
+	{ 2, 0x00 },
+	{ 3, 0x00 },
+	{ 4, 0x00 },
+	{ 5, 0x00 },
+	{ 6, 0x00 },
+	{ 7, 0x00 },
+	{ 8, 0x00 },
+	{ 9, 0x00 },
+	{ 10, 0x00 },
+	{ 11, 0x00 },
+	{ 12, 0x00 },
+	{ 13, 0x00 },
+	{ 14, 0x00 },
+	{ 15, 0x00 },
+	{ 16, 0x00 },
+	{ 17, 0x00 },
+	{ 18, 0x00 },
+	{ 19, 0x00 },
+	{ 20, 0x00 },
+	{ 21, 0x02 },
+	{ 22, 0x02 },
+	{ 23, 0x02 },
+	{ 24, 0x02 },
+	{ 25, 0x0F },
+	{ 26, 0x8F },
+	{ 27, 0x0F },
+	{ 28, 0x8F },
+	{ 29, 0x00 },
+	{ 30, 0x00 },
+	{ 31, 0x00 },
+	{ 32, 0x00 },
+	{ 33, 0x00 },
+	{ 34, 0x00 },
+	{ 35, 0x00 },
+	{ 36, 0x00 },
+	{ 37, 0x00 },
+	{ 38, 0x00 },
+	{ 39, 0x00 },
+	{ 40, 0x00 },
+	{ 41, 0x00 },
+	{ 42, 0x00 },
+	{ 43, 0x00 },
+	{ 44, 0x00 },
+	{ 45, 0x00 },
+	{ 46, 0x00 },
+	{ 47, 0x00 },
+	{ 48, 0x00 },
+	{ 49, 0x00 },
+	{ 50, 0x00 },
+	{ 51, 0x00 },
+	{ 52, 0x00 },
+	{ 53, 0x00 },
+	{ 54, 0x00 },
+	{ 55, 0x00 },
+	{ 56, 0x00 },
+	{ 57, 0x00 },
+	{ 58, 0x00 },
+	{ 59, 0x00 },
+	{ 60, 0x00 },
+	{ 61, 0x00 },
+	{ 62, 0x00 },
+	{ 63, 0x00 },
+	{ 64, 0x00 },
+	{ 65, 0x00 },
+	{ 66, 0x00 },
+	{ 67, 0x00 },
+	{ 68, 0x00 },
+	{ 69, 0x90 },
+	{ 70, 0x90 },
+	{ 71, 0x90 },
+	{ 72, 0x00 },
+	{ 73, 0x00 },
+	{ 74, 0x00 },
+	{ 75, 0x00 },
+	{ 76, 0x00 },
+	{ 77, 0x00 },
+	{ 78, 0x00 },
+	{ 79, 0x00 },
+	{ 80, 0x00 },
+	{ 81, 0x00 },
+	{ 82, 0x00 },
+	{ 83, 0x00 },
+	{ 84, 0x00 },
+	{ 85, 0x07 },
+	{ 86, 0x00 },
+	{ 87, 0x00 },
+	{ 88, 0x00 },
+	{ 89, 0x07 },
+	{ 90, 0x80 },
+	{ 91, 0x07 },
+	{ 92, 0x07 },
+	{ 93, 0x00 },
+	{ 94, 0x00 },
+	{ 95, 0x00 },
+	{ 96, 0x00 },
+	{ 97, 0x00 },
+	{ 98, 0x00 },
+	{ 99, 0x00 },
+};
+
+static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"};
+static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"};
+
+static const struct soc_enum isabelle_rx1_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts),
+};
+
+static const struct soc_enum isabelle_rx2_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts),
+	SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts),
+};
+
+/* Headset DAC playback switches */
+static const struct snd_kcontrol_new rx1_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_rx1_enum);
+
+static const struct snd_kcontrol_new rx2_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_rx2_enum);
+
+/* TX input selection */
+static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"};
+static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"};
+
+static const struct soc_enum isabelle_atx_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts),
+};
+
+static const struct soc_enum isabelle_vtx_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts),
+	SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts),
+};
+
+static const struct snd_kcontrol_new atx_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_atx_enum);
+
+static const struct snd_kcontrol_new vtx_mux_controls =
+	SOC_DAPM_ENUM("Route", isabelle_vtx_enum);
+
+/* Left analog microphone selection */
+static const char *isabelle_amic1_texts[] = {
+	"Main Mic", "Headset Mic", "Aux/FM Left"};
+
+/* Left analog microphone selection */
+static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"};
+
+static const struct soc_enum isabelle_amic1_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5,
+			ARRAY_SIZE(isabelle_amic1_texts),
+			isabelle_amic1_texts),
+};
+
+static const struct soc_enum isabelle_amic2_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4,
+			ARRAY_SIZE(isabelle_amic2_texts),
+			isabelle_amic2_texts),
+};
+
+static const struct snd_kcontrol_new amic1_control =
+	SOC_DAPM_ENUM("Route", isabelle_amic1_enum);
+
+static const struct snd_kcontrol_new amic2_control =
+	SOC_DAPM_ENUM("Route", isabelle_amic2_enum);
+
+static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"};
+
+static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"};
+
+static const struct soc_enum isabelle_st_audio_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1,
+			isabelle_st_audio_texts),
+	SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1,
+			isabelle_st_audio_texts),
+};
+
+static const struct soc_enum isabelle_st_voice_enum[] = {
+	SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1,
+			isabelle_st_voice_texts),
+	SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1,
+			isabelle_st_voice_texts),
+};
+
+static const struct snd_kcontrol_new st_audio_control =
+	SOC_DAPM_ENUM("Route", isabelle_st_audio_enum);
+
+static const struct snd_kcontrol_new st_voice_control =
+	SOC_DAPM_ENUM("Route", isabelle_st_voice_enum);
+
+/* Mixer controls */
+static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0),
+SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = {
+SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0),
+SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0),
+SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0),
+SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0),
+SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0),
+};
+
+static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = {
+SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0),
+SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0),
+};
+
+static const struct snd_kcontrol_new ep_path_enable_control =
+	SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0);
+
+/* TLV Declarations */
+static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0);
+static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0);
+static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0);
+
+/* from -63 to 0 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1);
+
+/* from -63 to 9 dB in 1 dB steps */
+static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1);
+
+static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1);
+static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0);
+
+static const struct snd_kcontrol_new isabelle_snd_controls[] = {
+	SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG,
+			4, 0, 0xF, 0, dac_tlv),
+	SOC_DOUBLE_R_TLV("Handsfree Playback Volume",
+			ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG,
+			0, 0x1F, 0, hf_tlv),
+	SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG,
+			4, 0, 0xF, 0, dac_tlv),
+	SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG,
+			0, 0xF, 0, dac_tlv),
+
+	SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0,
+			afm_amp_tlv),
+	SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F,
+			0, mic_amp_tlv),
+	SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F,
+			0, mic_amp_tlv),
+
+	SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG,
+			ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+	SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG,
+			ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+	SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG,
+			ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv),
+
+	SOC_SINGLE_TLV("Sidetone Audio TX1 Volume",
+			ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Audio TX2 Volume",
+			ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Voice TX1 Volume",
+			ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv),
+	SOC_SINGLE_TLV("Sidetone Voice TX2 Volume",
+			ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv),
+
+	SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+	SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0,
+			tx_tlv),
+
+	SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+	SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0,
+			rx_tlv),
+
+	SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0),
+	SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0),
+
+	SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		6, 1, 0),
+	SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		5, 1, 0),
+	SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		4, 1, 0),
+	SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		3, 1, 0),
+	SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		2, 1, 0),
+	SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		1, 1, 0),
+	SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		0, 1, 0),
+	SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		3, 1, 0),
+	SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG,
+		2, 1, 0),
+
+	SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		7, 1, 0),
+	SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		6, 1, 0),
+	SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		5, 1, 0),
+	SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG,
+		4, 1, 0),
+	SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		5, 1, 0),
+	SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		4, 1, 0),
+	SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		3, 1, 0),
+	SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		2, 1, 0),
+	SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		1, 1, 0),
+	SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG,
+		0, 1, 0),
+
+	SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG,
+		7, 1, 0),
+
+	SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG,
+		7, 1, 0),
+	SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG,
+		7, 1, 0),
+
+	/* DMIC Switch */
+	SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = {
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("MAINMIC"),
+	SND_SOC_DAPM_INPUT("HSMIC"),
+	SND_SOC_DAPM_INPUT("SUBMIC"),
+	SND_SOC_DAPM_INPUT("LINEIN1"),
+	SND_SOC_DAPM_INPUT("LINEIN2"),
+	SND_SOC_DAPM_INPUT("DMICDAT"),
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("HSOL"),
+	SND_SOC_DAPM_OUTPUT("HSOR"),
+	SND_SOC_DAPM_OUTPUT("HFL"),
+	SND_SOC_DAPM_OUTPUT("HFR"),
+	SND_SOC_DAPM_OUTPUT("EP"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+
+	SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog input muxes for the capture amplifiers */
+	SND_SOC_DAPM_MUX("Analog Left Capture Route",
+			SND_SOC_NOPM, 0, 0, &amic1_control),
+	SND_SOC_DAPM_MUX("Analog Right Capture Route",
+			SND_SOC_NOPM, 0, 0, &amic2_control),
+
+	SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0,
+			&st_audio_control),
+	SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0,
+			&st_voice_control),
+
+	/* AIF */
+	SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0),
+	SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0),
+
+	SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0),
+	SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0),
+
+	SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog Capture PGAs */
+	SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0),
+
+	/* Auxiliary FM PGAs */
+	SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC1", "Left Front Capture",
+			ISABELLE_AMIC_CFG_REG, 7, 0),
+	SND_SOC_DAPM_ADC("ADC2", "Right Front Capture",
+			ISABELLE_AMIC_CFG_REG, 6, 0),
+
+	/* Microphone Bias */
+	SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG,
+			3, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG,
+			2, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias",
+			ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias",
+			ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0),
+
+	/* Mixers */
+	SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hs_left_mixer_controls,
+			ARRAY_SIZE(isabelle_hs_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hs_right_mixer_controls,
+			ARRAY_SIZE(isabelle_hs_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hf_left_mixer_controls,
+			ARRAY_SIZE(isabelle_hf_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_hf_right_mixer_controls,
+			ARRAY_SIZE(isabelle_hf_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_aux_left_mixer_controls,
+			ARRAY_SIZE(isabelle_aux_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_aux_right_mixer_controls,
+			ARRAY_SIZE(isabelle_aux_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_ep_mixer_controls,
+			ARRAY_SIZE(isabelle_ep_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga1_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga1_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga2_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga2_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga3_left_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_dpga3_right_mixer_controls,
+			ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)),
+
+	SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx1_mixer_controls,
+			ARRAY_SIZE(isabelle_rx1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx2_mixer_controls,
+			ARRAY_SIZE(isabelle_rx2_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx3_mixer_controls,
+			ARRAY_SIZE(isabelle_rx3_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx4_mixer_controls,
+			ARRAY_SIZE(isabelle_rx4_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx5_mixer_controls,
+			ARRAY_SIZE(isabelle_rx5_mixer_controls)),
+	SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0,
+			isabelle_rx6_mixer_controls,
+			ARRAY_SIZE(isabelle_rx6_mixer_controls)),
+
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG,
+			5, 0),
+	SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG,
+			4, 0),
+	SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+			3, 0),
+	SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG,
+			2, 0),
+	SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+			1, 0),
+	SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG,
+			0, 0),
+
+	/* Analog Playback PGAs */
+	SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Analog Playback Mux */
+	SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0,
+			&rx1_mux_controls),
+	SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0,
+			&rx2_mux_controls),
+
+	/* TX Select */
+	SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG,
+			7, 0, &atx_mux_controls),
+	SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG,
+			6, 0, &vtx_mux_controls),
+
+	SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0,
+			&ep_path_enable_control),
+
+	/* Output Drivers */
+	SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG,
+			1, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG,
+			0, 0, NULL, 0),
+};
+
+static const struct snd_soc_dapm_route isabelle_intercon[] = {
+	/* Interface mapping */
+	{ "DL1", "DL12 Playback Switch", "INTF1_SDI" },
+	{ "DL2", "DL12 Playback Switch", "INTF1_SDI" },
+	{ "DL3", "DL34 Playback Switch", "INTF1_SDI" },
+	{ "DL4", "DL34 Playback Switch", "INTF1_SDI" },
+	{ "DL5", "DL56 Playback Switch", "INTF1_SDI" },
+	{ "DL6", "DL56 Playback Switch", "INTF1_SDI" },
+
+	{ "DL1", "DL12 Playback Switch", "INTF2_SDI" },
+	{ "DL2", "DL12 Playback Switch", "INTF2_SDI" },
+	{ "DL3", "DL34 Playback Switch", "INTF2_SDI" },
+	{ "DL4", "DL34 Playback Switch", "INTF2_SDI" },
+	{ "DL5", "DL56 Playback Switch", "INTF2_SDI" },
+	{ "DL6", "DL56 Playback Switch", "INTF2_SDI" },
+
+	/* Input side mapping */
+	{ "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" },
+	{ "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" },
+
+	{ "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" },
+
+	{ "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX1 Mixer", "DL1 Playback Switch", "DL1" },
+
+	{ "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" },
+
+	{ "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX2 Mixer", "DL2 Playback Switch", "DL2" },
+
+	{ "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX3 Mixer", "DL3 Playback Switch", "DL3" },
+
+	{ "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX4 Mixer", "DL4 Playback Switch", "DL4" },
+
+	{ "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX5 Mixer", "DL5 Playback Switch", "DL5" },
+
+	{ "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" },
+	{ "RX6 Mixer", "DL6 Playback Switch", "DL6" },
+
+	/* Capture path */
+	{ "Analog Left Capture Route", "Headset Mic", "HSMIC" },
+	{ "Analog Left Capture Route", "Main Mic", "MAINMIC" },
+	{ "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" },
+
+	{ "Analog Right Capture Route", "Sub Mic", "SUBMIC" },
+	{ "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" },
+
+	{ "MicAmp1", NULL, "Analog Left Capture Route" },
+	{ "MicAmp2", NULL, "Analog Right Capture Route" },
+
+	{ "ADC1", NULL, "MicAmp1" },
+	{ "ADC2", NULL, "MicAmp2" },
+
+	{ "ATX Select", "AMIC1", "ADC1" },
+	{ "ATX Select", "DMIC", "DMICDAT" },
+	{ "ATX Select", "AMIC2", "ADC2" },
+
+	{ "VTX Select", "AMIC1", "ADC1" },
+	{ "VTX Select", "DMIC", "DMICDAT" },
+	{ "VTX Select", "AMIC2", "ADC2" },
+
+	{ "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" },
+	{ "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" },
+	{ "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" },
+	{ "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" },
+
+	{ "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" },
+	{ "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" },
+	{ "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" },
+	{ "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" },
+
+	{ "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" },
+	{ "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" },
+	{ "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" },
+	{ "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" },
+
+	{ "INTF1_SDO", NULL, "ULVTX1" },
+	{ "INTF1_SDO", NULL, "ULVTX2" },
+	{ "INTF2_SDO", NULL, "ULVTX1" },
+	{ "INTF2_SDO", NULL, "ULVTX2" },
+
+	/* AFM Path */
+	{ "APGA1", NULL, "LINEIN1" },
+	{ "APGA2", NULL, "LINEIN2" },
+
+	{ "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" },
+	{ "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" },
+	{ "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" },
+
+	{ "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" },
+	{ "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" },
+	{ "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" },
+
+	{ "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" },
+	{ "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" },
+
+	{ "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" },
+	{ "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" },
+
+	{ "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" },
+	{ "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" },
+
+	{ "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" },
+	{ "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" },
+
+	{ "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+	{ "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA1L", NULL, "DPGA1L Mixer" },
+	{ "DPGA1R", NULL, "DPGA1R Mixer" },
+
+	{ "DAC1L", NULL, "DPGA1L" },
+	{ "DAC1R", NULL, "DPGA1R" },
+
+	{ "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+	{ "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA2L", NULL, "DPGA2L Mixer" },
+	{ "DPGA2R", NULL, "DPGA2R Mixer" },
+
+	{ "DAC2L", NULL, "DPGA2L" },
+	{ "DAC2R", NULL, "DPGA2R" },
+
+	{ "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" },
+	{ "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" },
+	{ "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" },
+
+	{ "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" },
+	{ "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" },
+	{ "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" },
+
+	{ "DPGA3L", NULL, "DPGA3L Mixer" },
+	{ "DPGA3R", NULL, "DPGA3R Mixer" },
+
+	{ "DAC3L", NULL, "DPGA3L" },
+	{ "DAC3R", NULL, "DPGA3R" },
+
+	{ "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" },
+	{ "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" },
+	{ "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "HS Left Driver", NULL, "Headset Left Mixer" },
+	{ "HS Right Driver", NULL, "Headset Right Mixer" },
+
+	{ "HSOL", NULL, "HS Left Driver" },
+	{ "HSOR", NULL, "HS Right Driver" },
+
+	/* Earphone playback path */
+	{ "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" },
+	{ "Earphone Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Earphone Playback", "Switch", "Earphone Mixer" },
+	{ "Earphone Driver", NULL, "Earphone Playback" },
+	{ "EP", NULL, "Earphone Driver" },
+
+	{ "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" },
+	{ "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" },
+	{ "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "HF Left PGA", NULL, "Handsfree Left Mixer" },
+	{ "HF Right PGA", NULL, "Handsfree Right Mixer" },
+
+	{ "HF Left Driver", NULL, "HF Left PGA" },
+	{ "HF Right Driver", NULL, "HF Right PGA" },
+
+	{ "HFL", NULL, "HF Left Driver" },
+	{ "HFR", NULL, "HF Right Driver" },
+
+	{ "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" },
+	{ "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" },
+
+	{ "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" },
+	{ "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" },
+
+	{ "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" },
+	{ "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" },
+
+	{ "LINEOUT1", NULL, "LINEOUT1 Driver" },
+	{ "LINEOUT2", NULL, "LINEOUT2 Driver" },
+};
+
+static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_line_mute(struct snd_soc_dai *dai, int mute)
+{
+	snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG,
+			BIT(4), (mute ? BIT(4) : 0));
+
+	return 0;
+}
+
+static int isabelle_set_bias_level(struct snd_soc_codec *codec,
+				enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+				ISABELLE_CHIP_EN, BIT(0));
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG,
+				ISABELLE_CHIP_EN, 0);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static int isabelle_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	u16 aif = 0;
+	unsigned int fs_val = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+		fs_val = ISABELLE_FS_RATE_8;
+		break;
+	case 11025:
+		fs_val = ISABELLE_FS_RATE_11;
+		break;
+	case 12000:
+		fs_val = ISABELLE_FS_RATE_12;
+		break;
+	case 16000:
+		fs_val = ISABELLE_FS_RATE_16;
+		break;
+	case 22050:
+		fs_val = ISABELLE_FS_RATE_22;
+		break;
+	case 24000:
+		fs_val = ISABELLE_FS_RATE_24;
+		break;
+	case 32000:
+		fs_val = ISABELLE_FS_RATE_32;
+		break;
+	case 44100:
+		fs_val = ISABELLE_FS_RATE_44;
+		break;
+	case 48000:
+		fs_val = ISABELLE_FS_RATE_48;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG,
+			ISABELLE_FS_RATE_MASK, fs_val);
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		aif |= ISABELLE_AIF_LENGTH_20;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		aif |= ISABELLE_AIF_LENGTH_32;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+			ISABELLE_AIF_LENGTH_MASK, aif);
+
+	return 0;
+}
+
+static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	unsigned int aif_val = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aif_val &= ~ISABELLE_AIF_MS;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aif_val |= ISABELLE_AIF_MS;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		aif_val |= ISABELLE_I2S_MODE;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		aif_val |= ISABELLE_LEFT_J_MODE;
+		break;
+	case SND_SOC_DAIFMT_PDM:
+		aif_val |= ISABELLE_PDM_MODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG,
+			(ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val);
+
+	return 0;
+}
+
+/* Rates supported by Isabelle driver */
+#define ISABELLE_RATES		SNDRV_PCM_RATE_8000_48000
+
+/* Formates supported by Isabelle driver. */
+#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops isabelle_hs_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_hs_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_hf_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_hf_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_line_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+	.digital_mute	= isabelle_line_mute,
+};
+
+static struct snd_soc_dai_ops isabelle_ul_dai_ops = {
+	.hw_params	= isabelle_hw_params,
+	.set_fmt	= isabelle_set_dai_fmt,
+};
+
+/* ISABELLE dai structure */
+static struct snd_soc_dai_driver isabelle_dai[] = {
+	{
+		.name = "isabelle-dl1",
+		.playback = {
+			.stream_name = "Headset Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_hs_dai_ops,
+	},
+	{
+		.name = "isabelle-dl2",
+		.playback = {
+			.stream_name = "Handsfree Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_hf_dai_ops,
+	},
+	{
+		.name = "isabelle-lineout",
+		.playback = {
+			.stream_name = "Lineout Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_line_dai_ops,
+	},
+	{
+		.name = "isabelle-ul",
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = ISABELLE_RATES,
+			.formats = ISABELLE_FORMATS,
+		},
+		.ops = &isabelle_ul_dai_ops,
+	},
+};
+
+static int isabelle_probe(struct snd_soc_codec *codec)
+{
+	int ret = 0;
+
+	codec->control_data = dev_get_regmap(codec->dev, NULL);
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_isabelle = {
+	.probe = isabelle_probe,
+	.set_bias_level = isabelle_set_bias_level,
+	.controls = isabelle_snd_controls,
+	.num_controls = ARRAY_SIZE(isabelle_snd_controls),
+	.dapm_widgets = isabelle_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets),
+	.dapm_routes = isabelle_intercon,
+	.num_dapm_routes = ARRAY_SIZE(isabelle_intercon),
+	.idle_bias_off = true,
+};
+
+static const struct regmap_config isabelle_regmap_config = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = ISABELLE_MAX_REGISTER,
+	.reg_defaults = isabelle_reg_defs,
+	.num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs),
+	.cache_type = REGCACHE_RBTREE,
+};
+
+static int __devinit isabelle_i2c_probe(struct i2c_client *i2c,
+					const struct i2c_device_id *id)
+{
+	struct regmap *isabelle_regmap;
+	int ret = 0;
+
+	isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config);
+	if (IS_ERR(isabelle_regmap)) {
+		ret = PTR_ERR(isabelle_regmap);
+		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
+			ret);
+		return ret;
+	}
+	i2c_set_clientdata(i2c, isabelle_regmap);
+
+	ret =  snd_soc_register_codec(&i2c->dev,
+				&soc_codec_dev_isabelle, isabelle_dai,
+				ARRAY_SIZE(isabelle_dai));
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	return ret;
+}
+
+static int __devexit isabelle_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static const struct i2c_device_id isabelle_i2c_id[] = {
+	{ "isabelle", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id);
+
+static struct i2c_driver isabelle_i2c_driver = {
+	.driver = {
+		.name = "isabelle",
+		.owner = THIS_MODULE,
+	},
+	.probe = isabelle_i2c_probe,
+	.remove = __devexit_p(isabelle_i2c_remove),
+	.id_table = isabelle_i2c_id,
+};
+
+module_i2c_driver(isabelle_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC ISABELLE driver");
+MODULE_AUTHOR("Vishwas A Deshpande <vishwas.a.deshpande@ti.com>");
+MODULE_AUTHOR("M R Swami Reddy <MR.Swami.Reddy@ti.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h
new file mode 100644
index 0000000..96d839a
--- /dev/null
+++ b/sound/soc/codecs/isabelle.h
@@ -0,0 +1,143 @@
+/*
+ * isabelle.h - Low power high fidelity audio codec driver header file
+ *
+ * Copyright (c) 2012 Texas Instruments, Inc
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; version 2 of the License.
+ *
+ */
+
+#ifndef _ISABELLE_H
+#define _ISABELLE_H
+
+#include <linux/bitops.h>
+
+/* ISABELLE REGISTERS */
+
+#define ISABELLE_PWR_CFG_REG		0x01
+#define ISABELLE_PWR_EN_REG		0x02
+#define ISABELLE_PS_EN1_REG		0x03
+#define ISABELLE_INT1_STATUS_REG	0x04
+#define ISABELLE_INT1_MASK_REG		0x05
+#define ISABELLE_INT2_STATUS_REG	0x06
+#define ISABELLE_INT2_MASK_REG		0x07
+#define ISABELLE_HKCTL1_REG		0x08
+#define ISABELLE_HKCTL2_REG		0x09
+#define ISABELLE_HKCTL3_REG		0x0A
+#define ISABELLE_ACCDET_STATUS_REG	0x0B
+#define ISABELLE_BUTTON_ID_REG		0x0C
+#define ISABELLE_PLL_CFG_REG		0x10
+#define ISABELLE_PLL_EN_REG		0x11
+#define ISABELLE_FS_RATE_CFG_REG	0x12
+#define ISABELLE_INTF_CFG_REG		0x13
+#define ISABELLE_INTF_EN_REG		0x14
+#define ISABELLE_ULATX12_INTF_CFG_REG	0x15
+#define ISABELLE_DL12_INTF_CFG_REG	0x16
+#define ISABELLE_DL34_INTF_CFG_REG	0x17
+#define ISABELLE_DL56_INTF_CFG_REG	0x18
+#define ISABELLE_ATX_STPGA1_CFG_REG	0x19
+#define ISABELLE_ATX_STPGA2_CFG_REG	0x1A
+#define ISABELLE_VTX_STPGA1_CFG_REG	0x1B
+#define ISABELLE_VTX2_STPGA2_CFG_REG	0x1C
+#define ISABELLE_ATX1_DPGA_REG		0x1D
+#define ISABELLE_ATX2_DPGA_REG		0x1E
+#define ISABELLE_VTX1_DPGA_REG		0x1F
+#define ISABELLE_VTX2_DPGA_REG		0x20
+#define ISABELLE_TX_INPUT_CFG_REG	0x21
+#define ISABELLE_RX_INPUT_CFG_REG	0x22
+#define ISABELLE_RX_INPUT_CFG2_REG	0x23
+#define ISABELLE_VOICE_HPF_CFG_REG	0x24
+#define ISABELLE_AUDIO_HPF_CFG_REG	0x25
+#define ISABELLE_RX1_DPGA_REG		0x26
+#define ISABELLE_RX2_DPGA_REG		0x27
+#define ISABELLE_RX3_DPGA_REG		0x28
+#define ISABELLE_RX4_DPGA_REG		0x29
+#define ISABELLE_RX5_DPGA_REG		0x2A
+#define ISABELLE_RX6_DPGA_REG		0x2B
+#define ISABELLE_ALU_TX_EN_REG		0x2C
+#define ISABELLE_ALU_RX_EN_REG		0x2D
+#define ISABELLE_IIR_RESYNC_REG		0x2E
+#define ISABELLE_ABIAS_CFG_REG		0x30
+#define ISABELLE_DBIAS_CFG_REG		0x31
+#define ISABELLE_MIC1_GAIN_REG		0x32
+#define ISABELLE_MIC2_GAIN_REG		0x33
+#define ISABELLE_AMIC_CFG_REG		0x34
+#define ISABELLE_DMIC_CFG_REG		0x35
+#define ISABELLE_APGA_GAIN_REG		0x36
+#define ISABELLE_APGA_CFG_REG		0x37
+#define ISABELLE_TX_GAIN_DLY_REG	0x38
+#define ISABELLE_RX_GAIN_DLY_REG	0x39
+#define ISABELLE_RX_PWR_CTRL_REG	0x3A
+#define ISABELLE_DPGA1LR_IN_SEL_REG	0x3B
+#define ISABELLE_DPGA1L_GAIN_REG	0x3C
+#define ISABELLE_DPGA1R_GAIN_REG	0x3D
+#define ISABELLE_DPGA2L_IN_SEL_REG	0x3E
+#define ISABELLE_DPGA2R_IN_SEL_REG	0x3F
+#define ISABELLE_DPGA2L_GAIN_REG	0x40
+#define ISABELLE_DPGA2R_GAIN_REG	0x41
+#define ISABELLE_DPGA3LR_IN_SEL_REG	0x42
+#define ISABELLE_DPGA3L_GAIN_REG	0x43
+#define ISABELLE_DPGA3R_GAIN_REG	0x44
+#define ISABELLE_DAC1_SOFTRAMP_REG	0x45
+#define ISABELLE_DAC2_SOFTRAMP_REG	0x46
+#define ISABELLE_DAC3_SOFTRAMP_REG	0x47
+#define ISABELLE_DAC_CFG_REG		0x48
+#define ISABELLE_EARDRV_CFG1_REG	0x49
+#define ISABELLE_EARDRV_CFG2_REG	0x4A
+#define ISABELLE_HSDRV_GAIN_REG		0x4B
+#define ISABELLE_HSDRV_CFG1_REG		0x4C
+#define ISABELLE_HSDRV_CFG2_REG		0x4D
+#define ISABELLE_HS_NG_CFG1_REG		0x4E
+#define ISABELLE_HS_NG_CFG2_REG		0x4F
+#define ISABELLE_LINEAMP_GAIN_REG	0x50
+#define ISABELLE_LINEAMP_CFG_REG	0x51
+#define ISABELLE_HFL_VOL_CTRL_REG	0x52
+#define ISABELLE_HFL_SFTVOL_CTRL_REG	0x53
+#define ISABELLE_HFL_LIM_CTRL_1_REG	0x54
+#define ISABELLE_HFL_LIM_CTRL_2_REG	0x55
+#define ISABELLE_HFR_VOL_CTRL_REG	0x56
+#define ISABELLE_HFR_SFTVOL_CTRL_REG	0x57
+#define ISABELLE_HFR_LIM_CTRL_1_REG	0x58
+#define ISABELLE_HFR_LIM_CTRL_2_REG	0x59
+#define ISABELLE_HF_MODE_REG		0x5A
+#define ISABELLE_HFLPGA_CFG_REG		0x5B
+#define ISABELLE_HFRPGA_CFG_REG		0x5C
+#define ISABELLE_HFDRV_CFG_REG		0x5D
+#define ISABELLE_PDMOUT_CFG1_REG	0x5E
+#define ISABELLE_PDMOUT_CFG2_REG	0x5F
+#define ISABELLE_PDMOUT_L_WM_REG	0x60
+#define ISABELLE_PDMOUT_R_WM_REG	0x61
+#define ISABELLE_HF_NG_CFG1_REG		0x62
+#define ISABELLE_HF_NG_CFG2_REG		0x63
+
+/* ISABELLE_PWR_EN_REG (0x02h) */
+#define ISABELLE_CHIP_EN		BIT(0)
+
+/* ISABELLE DAI FORMATS */
+#define ISABELLE_AIF_FMT_MASK		0x70
+#define ISABELLE_I2S_MODE		0x0
+#define ISABELLE_LEFT_J_MODE		0x1
+#define ISABELLE_PDM_MODE		0x2
+
+#define ISABELLE_AIF_LENGTH_MASK	0x30
+#define ISABELLE_AIF_LENGTH_20		0x00
+#define ISABELLE_AIF_LENGTH_32		0x10
+
+#define ISABELLE_AIF_MS			0x80
+
+#define ISABELLE_FS_RATE_MASK		0xF
+#define ISABELLE_FS_RATE_8		0x0
+#define ISABELLE_FS_RATE_11		0x1
+#define ISABELLE_FS_RATE_12		0x2
+#define ISABELLE_FS_RATE_16		0x4
+#define ISABELLE_FS_RATE_22		0x5
+#define ISABELLE_FS_RATE_24		0x6
+#define ISABELLE_FS_RATE_32		0x8
+#define ISABELLE_FS_RATE_44		0x9
+#define ISABELLE_FS_RATE_48		0xA
+
+#define ISABELLE_MAX_REGISTER		0xFF
+
+#endif
diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c
index 802b9f1..99b0a9d 100644
--- a/sound/soc/codecs/lm49453.c
+++ b/sound/soc/codecs/lm49453.c
@@ -12,7 +12,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -1358,7 +1357,7 @@
 };
 
 /* LM49453 dai structure. */
-static const struct snd_soc_dai_driver lm49453_dai[] = {
+static struct snd_soc_dai_driver lm49453_dai[] = {
 	{
 		.name = "LM49453 Headset",
 		.playback = {
diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c
index 35179e2..7cd508e 100644
--- a/sound/soc/codecs/max98095.c
+++ b/sound/soc/codecs/max98095.c
@@ -2216,7 +2216,7 @@
 	return IRQ_HANDLED;
 }
 
-int max98095_jack_detect_enable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_enable(struct snd_soc_codec *codec)
 {
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	int ret = 0;
@@ -2245,7 +2245,7 @@
 	return ret;
 }
 
-int max98095_jack_detect_disable(struct snd_soc_codec *codec)
+static int max98095_jack_detect_disable(struct snd_soc_codec *codec)
 {
 	int ret = 0;
 
@@ -2286,6 +2286,7 @@
 	max98095_report_jack(client->irq, codec);
 	return 0;
 }
+EXPORT_SYMBOL_GPL(max98095_jack_detect);
 
 #ifdef CONFIG_PM
 static int max98095_suspend(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c
index 22cb5bf..96aa5fa 100644
--- a/sound/soc/codecs/ml26124.c
+++ b/sound/soc/codecs/ml26124.c
@@ -638,7 +638,7 @@
 
 	i2c_set_clientdata(i2c, priv);
 
-	priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap);
+	priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap);
 	if (IS_ERR(priv->regmap)) {
 		ret = PTR_ERR(priv->regmap);
 		dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret);
@@ -651,10 +651,7 @@
 
 static __devexit int ml26124_i2c_remove(struct i2c_client *client)
 {
-	struct ml26124_priv *priv = i2c_get_clientdata(client);
-
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(priv->regmap);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c
new file mode 100644
index 0000000..dd8d856
--- /dev/null
+++ b/sound/soc/codecs/spdif_receiver.c
@@ -0,0 +1,67 @@
+/*
+ * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver
+ *
+ * Based on ALSA SoC SPDIF DIT driver
+ *
+ *  This driver is used by controllers which can operate in DIR (SPDI/F) where
+ *  no codec is needed.  This file provides stub codec that can be used
+ *  in these configurations. SPEAr SPDIF IN Audio controller uses this driver.
+ *
+ * Author:      Vipin Kumar,  <vipin.kumar@st.com>
+ * Copyright:   (C) 2012  ST Microelectronics
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_192000
+#define STUB_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+
+static struct snd_soc_codec_driver soc_codec_spdif_dir;
+
+static struct snd_soc_dai_driver dir_stub_dai = {
+	.name		= "dir-hifi",
+	.capture	= {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+};
+
+static int spdif_dir_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir,
+			&dir_stub_dai, 1);
+}
+
+static int spdif_dir_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver spdif_dir_driver = {
+	.probe		= spdif_dir_probe,
+	.remove		= spdif_dir_remove,
+	.driver		= {
+		.name	= "spdif-dir",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(spdif_dir_driver);
+
+MODULE_DESCRIPTION("ASoC SPDIF DIR driver");
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c
new file mode 100644
index 0000000..0c225cd
--- /dev/null
+++ b/sound/soc/codecs/sta529.c
@@ -0,0 +1,442 @@
+/*
+ * ASoC codec driver for spear platform
+ *
+ * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/init.h>
+#include <linux/i2c.h>
+#include <linux/io.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/pm.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+
+/* STA529 Register offsets */
+#define	 STA529_FFXCFG0		0x00
+#define	 STA529_FFXCFG1		0x01
+#define	 STA529_MVOL		0x02
+#define	 STA529_LVOL		0x03
+#define	 STA529_RVOL		0x04
+#define	 STA529_TTF0		0x05
+#define	 STA529_TTF1		0x06
+#define	 STA529_TTP0		0x07
+#define	 STA529_TTP1		0x08
+#define	 STA529_S2PCFG0		0x0A
+#define	 STA529_S2PCFG1		0x0B
+#define	 STA529_P2SCFG0		0x0C
+#define	 STA529_P2SCFG1		0x0D
+#define	 STA529_PLLCFG0		0x14
+#define	 STA529_PLLCFG1		0x15
+#define	 STA529_PLLCFG2		0x16
+#define	 STA529_PLLCFG3		0x17
+#define	 STA529_PLLPFE		0x18
+#define	 STA529_PLLST		0x19
+#define	 STA529_ADCCFG		0x1E /*mic_select*/
+#define	 STA529_CKOCFG		0x1F
+#define	 STA529_MISC		0x20
+#define	 STA529_PADST0		0x21
+#define	 STA529_PADST1		0x22
+#define	 STA529_FFXST		0x23
+#define	 STA529_PWMIN1		0x2D
+#define	 STA529_PWMIN2		0x2E
+#define	 STA529_POWST		0x32
+
+#define STA529_MAX_REGISTER	0x32
+
+#define STA529_RATES		(SNDRV_PCM_RATE_8000 | \
+				SNDRV_PCM_RATE_11025 | \
+				SNDRV_PCM_RATE_16000 | \
+				SNDRV_PCM_RATE_22050 | \
+				SNDRV_PCM_RATE_32000 | \
+				SNDRV_PCM_RATE_44100 | \
+				SNDRV_PCM_RATE_48000)
+
+#define STA529_FORMAT		(SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_S24_LE | \
+				SNDRV_PCM_FMTBIT_S32_LE)
+#define	S2PC_VALUE		0x98
+#define CLOCK_OUT		0x60
+#define LEFT_J_DATA_FORMAT	0x10
+#define I2S_DATA_FORMAT		0x12
+#define RIGHT_J_DATA_FORMAT	0x14
+#define CODEC_MUTE_VAL		0x80
+
+#define POWER_CNTLMSAK		0x40
+#define POWER_STDBY		0x40
+#define FFX_MASK		0x80
+#define FFX_OFF			0x80
+#define POWER_UP		0x00
+#define FFX_CLK_ENB		0x01
+#define FFX_CLK_DIS		0x00
+#define FFX_CLK_MSK		0x01
+#define PLAY_FREQ_RANGE_MSK	0x70
+#define CAP_FREQ_RANGE_MSK	0x0C
+#define PDATA_LEN_MSK		0xC0
+#define BCLK_TO_FS_MSK		0x30
+#define AUDIO_MUTE_MSK		0x80
+
+static const struct reg_default sta529_reg_defaults[] = {
+	{ 0,  0x35 },     /* R0   - FFX Configuration reg 0 */
+	{ 1,  0xc8 },     /* R1   - FFX Configuration reg 1 */
+	{ 2,  0x50 },     /* R2   - Master Volume */
+	{ 3,  0x00 },     /* R3   - Left Volume */
+	{ 4,  0x00 },     /* R4  -  Right Volume */
+	{ 10, 0xb2 },     /* R10  - S2P Config Reg 0 */
+	{ 11, 0x41 },     /* R11  - S2P Config Reg 1 */
+	{ 12, 0x92 },     /* R12  - P2S Config Reg 0 */
+	{ 13, 0x41 },     /* R13  - P2S Config Reg 1 */
+	{ 30, 0xd2 },     /* R30  - ADC Config Reg */
+	{ 31, 0x40 },     /* R31  - clock Out Reg */
+	{ 32, 0x21 },     /* R32  - Misc Register */
+};
+
+struct sta529 {
+	struct regmap *regmap;
+};
+
+static bool sta529_readable(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+
+	case STA529_FFXCFG0:
+	case STA529_FFXCFG1:
+	case STA529_MVOL:
+	case STA529_LVOL:
+	case STA529_RVOL:
+	case STA529_S2PCFG0:
+	case STA529_S2PCFG1:
+	case STA529_P2SCFG0:
+	case STA529_P2SCFG1:
+	case STA529_ADCCFG:
+	case STA529_CKOCFG:
+	case STA529_MISC:
+		return true;
+	default:
+		return false;
+	}
+}
+
+
+static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary",
+	"Phase-shift"};
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0);
+static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0);
+static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text);
+
+static const struct snd_kcontrol_new sta529_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0,
+			127, 0, out_gain_tlv),
+	SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1,
+			master_vol_tlv),
+	SOC_ENUM("PWM Select", pwm_src),
+};
+
+static int sta529_set_bias_level(struct snd_soc_codec *codec, enum
+		snd_soc_bias_level level)
+{
+	struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK,
+				POWER_UP);
+		snd_soc_update_bits(codec, STA529_MISC,	FFX_CLK_MSK,
+				FFX_CLK_ENB);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			regcache_sync(sta529->regmap);
+		snd_soc_update_bits(codec, STA529_FFXCFG0,
+					POWER_CNTLMSAK, POWER_STDBY);
+		/* Making FFX output to zero */
+		snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK,
+				FFX_OFF);
+		snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK,
+				FFX_CLK_DIS);
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+
+	/*
+	 * store the label for powers down audio subsystem for suspend.This is
+	 * used by soc core layer
+	 */
+	codec->dapm.bias_level = level;
+
+	return 0;
+
+}
+
+static int sta529_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	int pdata, play_freq_val, record_freq_val;
+	int bclk_to_fs_ratio;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		pdata = 1;
+		bclk_to_fs_ratio = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		pdata = 2;
+		bclk_to_fs_ratio = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		pdata = 3;
+		bclk_to_fs_ratio = 2;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported format\n");
+		return -EINVAL;
+	}
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 11025:
+		play_freq_val = 0;
+		record_freq_val = 2;
+		break;
+	case 16000:
+	case 22050:
+		play_freq_val = 1;
+		record_freq_val = 0;
+		break;
+
+	case 32000:
+	case 44100:
+	case 48000:
+		play_freq_val = 2;
+		record_freq_val = 0;
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported rate\n");
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK,
+				pdata << 6);
+		snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK,
+				bclk_to_fs_ratio << 4);
+		snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK,
+				play_freq_val << 4);
+	} else {
+		snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK,
+				pdata << 6);
+		snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK,
+				bclk_to_fs_ratio << 4);
+		snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK,
+				record_freq_val << 2);
+	}
+
+	return 0;
+}
+
+static int sta529_mute(struct snd_soc_dai *dai, int mute)
+{
+	u8 val = 0;
+
+	if (mute)
+		val |= CODEC_MUTE_VAL;
+
+	snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val);
+
+	return 0;
+}
+
+static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 mode = 0;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode = LEFT_J_DATA_FORMAT;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		mode = I2S_DATA_FORMAT;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		mode = RIGHT_J_DATA_FORMAT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops sta529_dai_ops = {
+	.hw_params	=	sta529_hw_params,
+	.set_fmt	=	sta529_set_dai_fmt,
+	.digital_mute	=	sta529_mute,
+};
+
+static struct snd_soc_dai_driver sta529_dai = {
+	.name = "sta529-audio",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = STA529_RATES,
+		.formats = STA529_FORMAT,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = STA529_RATES,
+		.formats = STA529_FORMAT,
+	},
+	.ops	= &sta529_dai_ops,
+};
+
+static int sta529_probe(struct snd_soc_codec *codec)
+{
+	struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec);
+	int ret;
+
+	codec->control_data = sta529->regmap;
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
+
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+	sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+/* power down chip */
+static int sta529_remove(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int sta529_suspend(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int sta529_resume(struct snd_soc_codec *codec)
+{
+	sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+struct snd_soc_codec_driver sta529_codec_driver = {
+	.probe = sta529_probe,
+	.remove = sta529_remove,
+	.set_bias_level = sta529_set_bias_level,
+	.suspend = sta529_suspend,
+	.resume = sta529_resume,
+	.controls = sta529_snd_controls,
+	.num_controls = ARRAY_SIZE(sta529_snd_controls),
+};
+
+static const struct regmap_config sta529_regmap = {
+	.reg_bits = 8,
+	.val_bits = 8,
+
+	.max_register = STA529_MAX_REGISTER,
+	.readable_reg = sta529_readable,
+
+	.cache_type = REGCACHE_RBTREE,
+	.reg_defaults = sta529_reg_defaults,
+	.num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults),
+};
+
+static __devinit int sta529_i2c_probe(struct i2c_client *i2c,
+		const struct i2c_device_id *id)
+{
+	struct sta529 *sta529;
+	int ret;
+
+	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+		return -EINVAL;
+
+	sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL);
+	if (sta529 == NULL) {
+		dev_err(&i2c->dev, "Can not allocate memory\n");
+		return -ENOMEM;
+	}
+
+	sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap);
+	if (IS_ERR(sta529->regmap)) {
+		ret = PTR_ERR(sta529->regmap);
+		dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret);
+		return ret;
+	}
+
+	i2c_set_clientdata(i2c, sta529);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&sta529_codec_driver, &sta529_dai, 1);
+	if (ret != 0)
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+
+	return ret;
+}
+
+static int __devexit sta529_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+
+	return 0;
+}
+
+static const struct i2c_device_id sta529_i2c_id[] = {
+	{ "sta529", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, sta529_i2c_id);
+
+static struct i2c_driver sta529_i2c_driver = {
+	.driver = {
+		.name = "sta529",
+		.owner = THIS_MODULE,
+	},
+	.probe		= sta529_i2c_probe,
+	.remove		= __devexit_p(sta529_i2c_remove),
+	.id_table	= sta529_i2c_id,
+};
+
+module_i2c_driver(sta529_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC STA529 codec driver");
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index e9b62b5..dc78f5a 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -118,7 +118,9 @@
 	0x00, 0x00, 0x00, 0x00,	/* 88 */
 	0x00, 0x00, 0x00, 0x00,	/* 92 */
 	0x00, 0x00, 0x00, 0x00,	/* 96 */
-	0x00, 0x00, 0x02,	/* 100 */
+	0x00, 0x00, 0x02, 0x00,	/* 100 */
+	0x00, 0x00, 0x00, 0x00,	/* 104 */
+	0x00, 0x00,            	/* 108 */
 };
 
 #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \
@@ -229,6 +231,25 @@
 	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
+static const char *aic3x_agc_level[] =
+	{ "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" };
+static const struct soc_enum aic3x_agc_level_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level),
+};
+
+static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" };
+static const struct soc_enum aic3x_agc_attack_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack),
+};
+
+static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" };
+static const struct soc_enum aic3x_agc_decay_enum[] = {
+	SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+	SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay),
+};
+
 /*
  * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
  */
@@ -353,6 +374,15 @@
 	 * adjust PGA to max value when ADC is on and will never go back.
 	*/
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
+	SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]),
+	SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]),
+	SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]),
+	SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]),
+	SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]),
+	SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]),
+
+	/* De-emphasis */
+	SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0),
 
 	/* Input */
 	SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
@@ -368,7 +398,7 @@
 static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
 
 static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
-	SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
+	SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
 
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
@@ -970,6 +1000,12 @@
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 
+	/* set clock on MCLK or GPIO2 or BCLK */
+	snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK,
+				clk_id << PLLCLK_IN_SHIFT);
+	snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK,
+				clk_id << CLKDIV_IN_SHIFT);
+
 	aic3x->sysclk = freq;
 	return 0;
 }
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index 08c7f66..6db3c41 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -13,7 +13,7 @@
 #define _AIC3X_H
 
 /* AIC3X register space */
-#define AIC3X_CACHEREGNUM		103
+#define AIC3X_CACHEREGNUM		110
 
 /* Page select register */
 #define AIC3X_PAGE_SELECT		0
@@ -74,6 +74,8 @@
 #define HPLCOM_CFG			37
 /* Right High Power Output control registers */
 #define HPRCOM_CFG			38
+/* High Power Output Stage Control Register */
+#define HPOUT_SC			40
 /* DAC Output Switching control registers */
 #define DAC_LINE_MUX			41
 /* High Power Output Driver Pop Reduction registers */
@@ -148,6 +150,17 @@
 #define AIC3X_GPIOB_REG			101
 /* Clock generation control register */
 #define AIC3X_CLKGEN_CTRL_REG		102
+/* New AGC registers */
+#define LAGCN_ATTACK			103
+#define LAGCN_DECAY			104
+#define RAGCN_ATTACK			105
+#define RAGCN_DECAY			106
+/* New Programmable ADC Digital Path and I2C Bus Condition Register */
+#define NEW_ADC_DIGITALPATH		107
+/* Passive Analog Signal Bypass Selection During Powerdown Register */
+#define PASSIVE_BYPASS			108
+/* DAC Quiescent Current Adjustment Register */
+#define DAC_ICC_ADJ			109
 
 /* Page select register bits */
 #define PAGE0_SELECT		0
@@ -163,6 +176,10 @@
 #define DUAL_RATE_MODE		((1 << 5) | (1 << 6))
 #define LDAC2LCH		(0x1 << 3)
 #define RDAC2RCH		(0x1 << 1)
+#define LDAC2RCH		(0x2 << 3)
+#define RDAC2LCH		(0x2 << 1)
+#define LDAC2MONOMIX		(0x3 << 3)
+#define RDAC2MONOMIX		(0x3 << 1)
 
 /* PLL registers bitfields */
 #define PLLP_SHIFT		0
@@ -179,6 +196,14 @@
 #define PLL_CLKIN_SHIFT		4
 #define MCLK_SOURCE		0x0
 #define PLL_CLKDIV_SHIFT	0
+#define PLLCLK_IN_MASK		0x30
+#define PLLCLK_IN_SHIFT		4
+#define CLKDIV_IN_MASK		0xc0
+#define CLKDIV_IN_SHIFT		6
+/* clock in source */
+#define CLKIN_MCLK		0
+#define CLKIN_GPIO2		1
+#define CLKIN_BCLK		2
 
 /* Software reset register bits */
 #define SOFT_RESET		0x80
diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c
index a36e9fc..0ff1e70 100644
--- a/sound/soc/codecs/twl6040.c
+++ b/sound/soc/codecs/twl6040.c
@@ -553,7 +553,7 @@
 
 /* Headset power mode */
 static const char *twl6040_power_mode_texts[] = {
-	"Low-Power", "High-Perfomance",
+	"Low-Power", "High-Performance",
 };
 
 static const struct soc_enum twl6040_power_mode_enum =
diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c
index e0b51e9..951d7b4 100644
--- a/sound/soc/codecs/wm1250-ev1.c
+++ b/sound/soc/codecs/wm1250-ev1.c
@@ -121,20 +121,23 @@
 	.hw_params = wm1250_ev1_hw_params,
 };
 
+#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
+			  SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000)
+
 static struct snd_soc_dai_driver wm1250_ev1_dai = {
 	.name = "wm1250-ev1",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
+		.rates = WM1250_EV1_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_8000,
+		.rates = WM1250_EV1_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.ops = &wm1250_ev1_ops,
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 0418fa1..3fd5b29 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -1,7 +1,7 @@
 /*
  * wm2000.c  --  WM2000 ALSA Soc Audio driver
  *
- * Copyright 2008-2010 Wolfson Microelectronics PLC.
+ * Copyright 2008-2011 Wolfson Microelectronics PLC.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -674,9 +674,39 @@
 #define wm2000_resume NULL
 #endif
 
+static bool wm2000_readable_reg(struct device *dev, unsigned int reg)
+{
+	switch (reg) {
+	case WM2000_REG_SYS_START:
+	case WM2000_REG_SPEECH_CLARITY:
+	case WM2000_REG_SYS_WATCHDOG:
+	case WM2000_REG_ANA_VMID_PD_TIME:
+	case WM2000_REG_ANA_VMID_PU_TIME:
+	case WM2000_REG_CAT_FLTR_INDX:
+	case WM2000_REG_CAT_GAIN_0:
+	case WM2000_REG_SYS_STATUS:
+	case WM2000_REG_SYS_MODE_CNTRL:
+	case WM2000_REG_SYS_START0:
+	case WM2000_REG_SYS_START1:
+	case WM2000_REG_ID1:
+	case WM2000_REG_ID2:
+	case WM2000_REG_REVISON:
+	case WM2000_REG_SYS_CTL1:
+	case WM2000_REG_SYS_CTL2:
+	case WM2000_REG_ANC_STAT:
+	case WM2000_REG_IF_CTL:
+		return true;
+	default:
+		return false;
+	}
+}
+
 static const struct regmap_config wm2000_regmap = {
 	.reg_bits = 8,
 	.val_bits = 8,
+
+	.max_register = WM2000_REG_IF_CTL,
+	.readable_reg = wm2000_readable_reg,
 };
 
 static int wm2000_probe(struct snd_soc_codec *codec)
diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c
index e167207..e239f4b 100644
--- a/sound/soc/codecs/wm5100-tables.c
+++ b/sound/soc/codecs/wm5100-tables.c
@@ -1,7 +1,7 @@
 /*
  * wm5100-tables.c  --  WM5100 ALSA SoC Audio driver data
  *
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c
index cb6d537..f481729 100644
--- a/sound/soc/codecs/wm5100.c
+++ b/sound/soc/codecs/wm5100.c
@@ -1,7 +1,7 @@
 /*
  * wm5100.c  --  WM5100 ALSA SoC Audio driver
  *
- * Copyright 2011 Wolfson Microelectronics plc
+ * Copyright 2011-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -2378,13 +2378,6 @@
 	return 0;
 }
 
-static int wm5100_soc_volatile(struct snd_soc_codec *codec,
-			       unsigned int reg)
-{
-	return true;
-}
-
-
 static struct snd_soc_codec_driver soc_codec_dev_wm5100 = {
 	.probe =	wm5100_probe,
 	.remove =	wm5100_remove,
@@ -2392,8 +2385,6 @@
 	.set_sysclk = wm5100_set_sysclk,
 	.set_pll = wm5100_set_fll,
 	.idle_bias_off = 1,
-	.reg_cache_size = WM5100_MAX_REGISTER,
-	.volatile_register = wm5100_soc_volatile,
 
 	.seq_notifier = wm5100_seq_notifier,
 	.controls = wm5100_snd_controls,
diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c
new file mode 100644
index 0000000..6537f16
--- /dev/null
+++ b/sound/soc/codecs/wm5102.c
@@ -0,0 +1,903 @@
+/*
+ * wm5102.c  --  WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5102.h"
+
+struct wm5102_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5102_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+	   ARIZONA_IN3_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+		       ARIZONA_IN1R_CONTROL,
+		       ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+		       ARIZONA_IN2R_CONTROL,
+		       ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+		       ARIZONA_IN3R_CONTROL,
+		       ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+	       ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	       ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
+		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
+		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+		     ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+		    0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+		    ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+		 ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+		 ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2L PGA" }, \
+	{ name, "IN2R", "IN2R PGA" }, \
+	{ name, "IN3L", "IN3L PGA" }, \
+	{ name, "IN3R", "IN3R PGA" }, \
+	{ name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ASRC1L", "ASRC1L" }, \
+	{ name, "ASRC1R", "ASRC1R" }, \
+	{ name, "ASRC2L", "ASRC2L" }, \
+	{ name, "ASRC2R", "ASRC2R" }
+
+static const struct snd_soc_dapm_route wm5102_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "AIF3 Capture", NULL, "DBVDD3" },
+	{ "AIF3 Playback", NULL, "DBVDD3" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT2L", NULL, "CPVDD" },
+	{ "OUT2R", NULL, "CPVDD" },
+	{ "OUT3L", NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDDL" },
+	{ "OUT4R", NULL, "SPKVDDR" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT4R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
+	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+	ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+	ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+
+	ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "HPOUT2L", NULL, "OUT2L" },
+	{ "HPOUT2R", NULL, "OUT2R" },
+
+	{ "EPOUTN", NULL, "OUT3L" },
+	{ "EPOUTP", NULL, "OUT3L" },
+
+	{ "SPKOUTLN", NULL, "OUT4L" },
+	{ "SPKOUTLP", NULL, "OUT4L" },
+
+	{ "SPKOUTRN", NULL, "OUT4R" },
+	{ "SPKOUTRP", NULL, "OUT4R" },
+
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+};
+
+static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM5102_FLL1:
+		return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout);
+	case WM5102_FLL2:
+		return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+#define WM5102_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm5102_dai[] = {
+	{
+		.name = "wm5102-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 8,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5102-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5102-aif3",
+		.id = 3,
+		.base = ARIZONA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5102_RATES,
+			.formats = WM5102_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF3 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5102_RATES,
+			 .formats = WM5102_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static int wm5102_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	codec->control_data = priv->core.arizona->regmap;
+	return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5102_DIG_VU 0x0200
+
+static unsigned int wm5102_digital_vu[] = {
+	ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	ARIZONA_ADC_DIGITAL_VOLUME_1R,
+	ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	ARIZONA_ADC_DIGITAL_VOLUME_2R,
+	ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	ARIZONA_DAC_DIGITAL_VOLUME_2R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_3R,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4R,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5102 = {
+	.probe = wm5102_codec_probe,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm5102_set_fll,
+
+	.controls = wm5102_snd_controls,
+	.num_controls = ARRAY_SIZE(wm5102_snd_controls),
+	.dapm_widgets = wm5102_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets),
+	.dapm_routes = wm5102_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes),
+};
+
+static int __devinit wm5102_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm5102_priv *wm5102;
+	int i;
+
+	wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv),
+			      GFP_KERNEL);
+	if (wm5102 == NULL)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm5102);
+
+	wm5102->core.arizona = arizona;
+
+	for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++)
+		wm5102->fll[i].vco_mult = 1;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm5102->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm5102->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++)
+		arizona_init_dai(&wm5102->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm5102_digital_vu[i],
+				   WM5102_DIG_VU, WM5102_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102,
+				      wm5102_dai, ARRAY_SIZE(wm5102_dai));
+}
+
+static int __devexit wm5102_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm5102_codec_driver = {
+	.driver = {
+		.name = "wm5102-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm5102_probe,
+	.remove = __devexit_p(wm5102_remove),
+};
+
+module_platform_driver(wm5102_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5102 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5102-codec");
diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h
new file mode 100644
index 0000000..d30477f
--- /dev/null
+++ b/sound/soc/codecs/wm5102.h
@@ -0,0 +1,21 @@
+/*
+ * wm5102.h  --  WM5102 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM5102_H
+#define _WM5102_H
+
+#include "arizona.h"
+
+#define WM5102_FLL1 1
+#define WM5102_FLL2 2
+
+#endif
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
new file mode 100644
index 0000000..8033f70
--- /dev/null
+++ b/sound/soc/codecs/wm5110.c
@@ -0,0 +1,950 @@
+/*
+ * wm5110.c  --  WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/pm_runtime.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include <linux/mfd/arizona/core.h>
+#include <linux/mfd/arizona/registers.h>
+
+#include "arizona.h"
+#include "wm5110.h"
+
+struct wm5110_priv {
+	struct arizona_priv core;
+	struct arizona_fll fll[2];
+};
+
+static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
+static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
+static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
+static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new wm5110_snd_controls[] = {
+SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL,
+	   ARIZONA_IN1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL,
+	   ARIZONA_IN2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL,
+	   ARIZONA_IN3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL,
+	   ARIZONA_IN4_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL,
+		       ARIZONA_IN1R_CONTROL,
+		       ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL,
+		       ARIZONA_IN2R_CONTROL,
+		       ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL,
+		       ARIZONA_IN3R_CONTROL,
+		       ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+
+SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+	     ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L,
+		 ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT,
+	       24, 0, eq_tlv),
+
+ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE),
+
+SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5,
+		   ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA),
+SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5,
+		   ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA),
+
+ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE),
+
+SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode),
+SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode),
+SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode),
+SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode),
+
+ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR,
+	       ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv),
+
+ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE),
+
+SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+	   ARIZONA_OUT1_OSR_SHIFT, 1, 0),
+SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+	   ARIZONA_OUT2_OSR_SHIFT, 1, 0),
+SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+	   ARIZONA_OUT3_OSR_SHIFT, 1, 0),
+SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L,
+	   ARIZONA_OUT4_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L,
+	   ARIZONA_OUT5_OSR_SHIFT, 1, 0),
+SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L,
+	   ARIZONA_OUT6_OSR_SHIFT, 1, 0),
+
+SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1),
+SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	   ARIZONA_OUT3L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+	     ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1),
+
+SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	       ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L,
+		 ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT,
+		 0xbf, 0, digital_tlv),
+
+SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_1R,
+		       ARIZONA_OUT1L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L,
+		       ARIZONA_OUTPUT_PATH_CONFIG_2R,
+		       ARIZONA_OUT2L_PGA_VOL_SHIFT,
+		       0x34, 0x40, 0, ana_tlv),
+SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L,
+		     ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv),
+
+SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT,
+	   ARIZONA_SPK1R_MUTE_SHIFT, 1, 1),
+SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT,
+	   ARIZONA_SPK2R_MUTE_SHIFT, 1, 1),
+
+ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE),
+
+ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE),
+ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE),
+};
+
+ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE);
+
+ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE);
+ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE);
+
+static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT,
+		    0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1,
+		    ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20),
+SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0),
+SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0),
+
+SND_SOC_DAPM_SIGGEN("TONE"),
+SND_SOC_DAPM_SIGGEN("NOISE"),
+
+SND_SOC_DAPM_INPUT("IN1L"),
+SND_SOC_DAPM_INPUT("IN1R"),
+SND_SOC_DAPM_INPUT("IN2L"),
+SND_SOC_DAPM_INPUT("IN2R"),
+SND_SOC_DAPM_INPUT("IN3L"),
+SND_SOC_DAPM_INPUT("IN3R"),
+SND_SOC_DAPM_INPUT("IN4L"),
+SND_SOC_DAPM_INPUT("IN4R"),
+
+SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT,
+		   0, NULL, 0, arizona_in_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3,
+		    ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR,
+		 ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1,
+		 ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1,
+		 ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0),
+SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT,
+		 0, NULL, 0),
+SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT,
+		 0, NULL, 0),
+
+SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0,
+		 NULL, 0),
+SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0,
+		 NULL, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0,
+		     ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0,
+		    ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0,
+		     ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0,
+		    ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0,
+		     ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0),
+SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0,
+		    ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1,
+		   ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev,
+		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+
+ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"),
+ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"),
+ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"),
+ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"),
+
+ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"),
+ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"),
+ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"),
+ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"),
+
+ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"),
+ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"),
+ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"),
+ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"),
+
+ARIZONA_MIXER_WIDGETS(Mic, "Mic"),
+ARIZONA_MIXER_WIDGETS(Noise, "Noise"),
+
+ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"),
+ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"),
+
+ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"),
+ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"),
+ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"),
+ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"),
+ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"),
+ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"),
+ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"),
+ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"),
+
+ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"),
+ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"),
+ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"),
+ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"),
+ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"),
+ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"),
+ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"),
+ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"),
+
+ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"),
+ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"),
+
+ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"),
+ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"),
+
+ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"),
+ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"),
+ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"),
+ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"),
+
+SND_SOC_DAPM_OUTPUT("HPOUT1L"),
+SND_SOC_DAPM_OUTPUT("HPOUT1R"),
+SND_SOC_DAPM_OUTPUT("HPOUT2L"),
+SND_SOC_DAPM_OUTPUT("HPOUT2R"),
+SND_SOC_DAPM_OUTPUT("EPOUTN"),
+SND_SOC_DAPM_OUTPUT("EPOUTP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
+SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT1R"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2L"),
+SND_SOC_DAPM_OUTPUT("SPKDAT2R"),
+};
+
+#define ARIZONA_MIXER_INPUT_ROUTES(name)	\
+	{ name, "Noise Generator", "Noise Generator" }, \
+	{ name, "Tone Generator 1", "Tone Generator 1" }, \
+	{ name, "Tone Generator 2", "Tone Generator 2" }, \
+	{ name, "IN1L", "IN1L PGA" }, \
+	{ name, "IN1R", "IN1R PGA" }, \
+	{ name, "IN2L", "IN2L PGA" }, \
+	{ name, "IN2R", "IN2R PGA" }, \
+	{ name, "IN3L", "IN3L PGA" }, \
+	{ name, "IN3R", "IN3R PGA" }, \
+	{ name, "IN4L", "IN4L PGA" }, \
+	{ name, "IN4R", "IN4R PGA" }, \
+	{ name, "Mic Mute Mixer", "Mic Mute Mixer" }, \
+	{ name, "AIF1RX1", "AIF1RX1" }, \
+	{ name, "AIF1RX2", "AIF1RX2" }, \
+	{ name, "AIF1RX3", "AIF1RX3" }, \
+	{ name, "AIF1RX4", "AIF1RX4" }, \
+	{ name, "AIF1RX5", "AIF1RX5" }, \
+	{ name, "AIF1RX6", "AIF1RX6" }, \
+	{ name, "AIF1RX7", "AIF1RX7" }, \
+	{ name, "AIF1RX8", "AIF1RX8" }, \
+	{ name, "AIF2RX1", "AIF2RX1" }, \
+	{ name, "AIF2RX2", "AIF2RX2" }, \
+	{ name, "AIF3RX1", "AIF3RX1" }, \
+	{ name, "AIF3RX2", "AIF3RX2" }, \
+	{ name, "EQ1", "EQ1" }, \
+	{ name, "EQ2", "EQ2" }, \
+	{ name, "EQ3", "EQ3" }, \
+	{ name, "EQ4", "EQ4" }, \
+	{ name, "DRC1L", "DRC1L" }, \
+	{ name, "DRC1R", "DRC1R" }, \
+	{ name, "DRC2L", "DRC2L" }, \
+	{ name, "DRC2R", "DRC2R" }, \
+	{ name, "LHPF1", "LHPF1" }, \
+	{ name, "LHPF2", "LHPF2" }, \
+	{ name, "LHPF3", "LHPF3" }, \
+	{ name, "LHPF4", "LHPF4" }, \
+	{ name, "ASRC1L", "ASRC1L" }, \
+	{ name, "ASRC1R", "ASRC1R" }, \
+	{ name, "ASRC2L", "ASRC2L" }, \
+	{ name, "ASRC2R", "ASRC2R" }
+
+static const struct snd_soc_dapm_route wm5110_dapm_routes[] = {
+	{ "AIF2 Capture", NULL, "DBVDD2" },
+	{ "AIF2 Playback", NULL, "DBVDD2" },
+
+	{ "AIF3 Capture", NULL, "DBVDD3" },
+	{ "AIF3 Playback", NULL, "DBVDD3" },
+
+	{ "OUT1L", NULL, "CPVDD" },
+	{ "OUT1R", NULL, "CPVDD" },
+	{ "OUT2L", NULL, "CPVDD" },
+	{ "OUT2R", NULL, "CPVDD" },
+	{ "OUT3L", NULL, "CPVDD" },
+
+	{ "OUT4L", NULL, "SPKVDDL" },
+	{ "OUT4R", NULL, "SPKVDDR" },
+
+	{ "OUT1L", NULL, "SYSCLK" },
+	{ "OUT1R", NULL, "SYSCLK" },
+	{ "OUT2L", NULL, "SYSCLK" },
+	{ "OUT2R", NULL, "SYSCLK" },
+	{ "OUT3L", NULL, "SYSCLK" },
+	{ "OUT4L", NULL, "SYSCLK" },
+	{ "OUT4R", NULL, "SYSCLK" },
+	{ "OUT5L", NULL, "SYSCLK" },
+	{ "OUT5R", NULL, "SYSCLK" },
+	{ "OUT6L", NULL, "SYSCLK" },
+	{ "OUT6R", NULL, "SYSCLK" },
+
+	{ "MICBIAS1", NULL, "MICVDD" },
+	{ "MICBIAS2", NULL, "MICVDD" },
+	{ "MICBIAS3", NULL, "MICVDD" },
+
+	{ "Noise Generator", NULL, "NOISE" },
+	{ "Tone Generator 1", NULL, "TONE" },
+	{ "Tone Generator 2", NULL, "TONE" },
+
+	{ "Mic Mute Mixer", NULL, "Noise Mixer" },
+	{ "Mic Mute Mixer", NULL, "Mic Mixer" },
+
+	{ "AIF1 Capture", NULL, "AIF1TX1" },
+	{ "AIF1 Capture", NULL, "AIF1TX2" },
+	{ "AIF1 Capture", NULL, "AIF1TX3" },
+	{ "AIF1 Capture", NULL, "AIF1TX4" },
+	{ "AIF1 Capture", NULL, "AIF1TX5" },
+	{ "AIF1 Capture", NULL, "AIF1TX6" },
+	{ "AIF1 Capture", NULL, "AIF1TX7" },
+	{ "AIF1 Capture", NULL, "AIF1TX8" },
+
+	{ "AIF1RX1", NULL, "AIF1 Playback" },
+	{ "AIF1RX2", NULL, "AIF1 Playback" },
+	{ "AIF1RX3", NULL, "AIF1 Playback" },
+	{ "AIF1RX4", NULL, "AIF1 Playback" },
+	{ "AIF1RX5", NULL, "AIF1 Playback" },
+	{ "AIF1RX6", NULL, "AIF1 Playback" },
+	{ "AIF1RX7", NULL, "AIF1 Playback" },
+	{ "AIF1RX8", NULL, "AIF1 Playback" },
+
+	{ "AIF2 Capture", NULL, "AIF2TX1" },
+	{ "AIF2 Capture", NULL, "AIF2TX2" },
+
+	{ "AIF2RX1", NULL, "AIF2 Playback" },
+	{ "AIF2RX2", NULL, "AIF2 Playback" },
+
+	{ "AIF3 Capture", NULL, "AIF3TX1" },
+	{ "AIF3 Capture", NULL, "AIF3TX2" },
+
+	{ "AIF3RX1", NULL, "AIF3 Playback" },
+	{ "AIF3RX2", NULL, "AIF3 Playback" },
+
+	{ "AIF1 Playback", NULL, "SYSCLK" },
+	{ "AIF2 Playback", NULL, "SYSCLK" },
+	{ "AIF3 Playback", NULL, "SYSCLK" },
+
+	{ "AIF1 Capture", NULL, "SYSCLK" },
+	{ "AIF2 Capture", NULL, "SYSCLK" },
+	{ "AIF3 Capture", NULL, "SYSCLK" },
+
+	ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"),
+	ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"),
+	ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"),
+	ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"),
+	ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"),
+
+	ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"),
+	ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"),
+	ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"),
+	ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"),
+	ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"),
+	ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"),
+
+	ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"),
+	ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"),
+
+	ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"),
+	ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"),
+	ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"),
+	ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"),
+	ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"),
+	ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"),
+	ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"),
+	ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"),
+
+	ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"),
+	ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"),
+
+	ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"),
+	ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"),
+
+	ARIZONA_MIXER_ROUTES("EQ1", "EQ1"),
+	ARIZONA_MIXER_ROUTES("EQ2", "EQ2"),
+	ARIZONA_MIXER_ROUTES("EQ3", "EQ3"),
+	ARIZONA_MIXER_ROUTES("EQ4", "EQ4"),
+
+	ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"),
+	ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"),
+	ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"),
+	ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"),
+
+	ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"),
+	ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"),
+	ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"),
+	ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"),
+
+	ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"),
+	ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"),
+	ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"),
+	ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"),
+
+	{ "HPOUT1L", NULL, "OUT1L" },
+	{ "HPOUT1R", NULL, "OUT1R" },
+
+	{ "HPOUT2L", NULL, "OUT2L" },
+	{ "HPOUT2R", NULL, "OUT2R" },
+
+	{ "EPOUTN", NULL, "OUT3L" },
+	{ "EPOUTP", NULL, "OUT3L" },
+
+	{ "SPKOUTLN", NULL, "OUT4L" },
+	{ "SPKOUTLP", NULL, "OUT4L" },
+
+	{ "SPKOUTRN", NULL, "OUT4R" },
+	{ "SPKOUTRP", NULL, "OUT4R" },
+
+	{ "SPKDAT1L", NULL, "OUT5L" },
+	{ "SPKDAT1R", NULL, "OUT5R" },
+
+	{ "SPKDAT2L", NULL, "OUT6L" },
+	{ "SPKDAT2R", NULL, "OUT6R" },
+};
+
+static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source,
+			  unsigned int Fref, unsigned int Fout)
+{
+	struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+
+	switch (fll_id) {
+	case WM5110_FLL1:
+		return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout);
+	case WM5110_FLL2:
+		return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout);
+	default:
+		return -EINVAL;
+	}
+}
+
+#define WM5110_RATES SNDRV_PCM_RATE_8000_192000
+
+#define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver wm5110_dai[] = {
+	{
+		.name = "wm5110-aif1",
+		.id = 1,
+		.base = ARIZONA_AIF1_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF1 Playback",
+			.channels_min = 1,
+			.channels_max = 8,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF1 Capture",
+			 .channels_min = 1,
+			 .channels_max = 8,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5110-aif2",
+		.id = 2,
+		.base = ARIZONA_AIF2_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF2 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF2 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+	{
+		.name = "wm5110-aif3",
+		.id = 3,
+		.base = ARIZONA_AIF3_BCLK_CTRL,
+		.playback = {
+			.stream_name = "AIF3 Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM5110_RATES,
+			.formats = WM5110_FORMATS,
+		},
+		.capture = {
+			 .stream_name = "AIF3 Capture",
+			 .channels_min = 1,
+			 .channels_max = 2,
+			 .rates = WM5110_RATES,
+			 .formats = WM5110_FORMATS,
+		 },
+		.ops = &arizona_dai_ops,
+		.symmetric_rates = 1,
+	},
+};
+
+static int wm5110_codec_probe(struct snd_soc_codec *codec)
+{
+	struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	codec->control_data = priv->core.arizona->regmap;
+	return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP);
+}
+
+#define WM5110_DIG_VU 0x0200
+
+static unsigned int wm5110_digital_vu[] = {
+	ARIZONA_ADC_DIGITAL_VOLUME_1L,
+	ARIZONA_ADC_DIGITAL_VOLUME_1R,
+	ARIZONA_ADC_DIGITAL_VOLUME_2L,
+	ARIZONA_ADC_DIGITAL_VOLUME_2R,
+	ARIZONA_ADC_DIGITAL_VOLUME_3L,
+	ARIZONA_ADC_DIGITAL_VOLUME_3R,
+
+	ARIZONA_DAC_DIGITAL_VOLUME_1L,
+	ARIZONA_DAC_DIGITAL_VOLUME_1R,
+	ARIZONA_DAC_DIGITAL_VOLUME_2L,
+	ARIZONA_DAC_DIGITAL_VOLUME_2R,
+	ARIZONA_DAC_DIGITAL_VOLUME_3L,
+	ARIZONA_DAC_DIGITAL_VOLUME_3R,
+	ARIZONA_DAC_DIGITAL_VOLUME_4L,
+	ARIZONA_DAC_DIGITAL_VOLUME_4R,
+	ARIZONA_DAC_DIGITAL_VOLUME_5L,
+	ARIZONA_DAC_DIGITAL_VOLUME_5R,
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_wm5110 = {
+	.probe = wm5110_codec_probe,
+
+	.idle_bias_off = true,
+
+	.set_sysclk = arizona_set_sysclk,
+	.set_pll = wm5110_set_fll,
+
+	.controls = wm5110_snd_controls,
+	.num_controls = ARRAY_SIZE(wm5110_snd_controls),
+	.dapm_widgets = wm5110_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets),
+	.dapm_routes = wm5110_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes),
+};
+
+static int __devinit wm5110_probe(struct platform_device *pdev)
+{
+	struct arizona *arizona = dev_get_drvdata(pdev->dev.parent);
+	struct wm5110_priv *wm5110;
+	int i;
+
+	wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv),
+			      GFP_KERNEL);
+	if (wm5110 == NULL)
+		return -ENOMEM;
+	platform_set_drvdata(pdev, wm5110);
+
+	wm5110->core.arizona = arizona;
+
+	for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++)
+		wm5110->fll[i].vco_mult = 3;
+
+	arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK,
+			 &wm5110->fll[0]);
+	arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1,
+			 ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK,
+			 &wm5110->fll[1]);
+
+	for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++)
+		arizona_init_dai(&wm5110->core, i);
+
+	/* Latch volume update bits */
+	for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++)
+		regmap_update_bits(arizona->regmap, wm5110_digital_vu[i],
+				   WM5110_DIG_VU, WM5110_DIG_VU);
+
+	pm_runtime_enable(&pdev->dev);
+	pm_runtime_idle(&pdev->dev);
+
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110,
+				      wm5110_dai, ARRAY_SIZE(wm5110_dai));
+}
+
+static int __devexit wm5110_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver wm5110_codec_driver = {
+	.driver = {
+		.name = "wm5110-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm5110_probe,
+	.remove = __devexit_p(wm5110_remove),
+};
+
+module_platform_driver(wm5110_codec_driver);
+
+MODULE_DESCRIPTION("ASoC WM5110 driver");
+MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm5110-codec");
diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h
new file mode 100644
index 0000000..75e9351
--- /dev/null
+++ b/sound/soc/codecs/wm5110.h
@@ -0,0 +1,21 @@
+/*
+ * wm5110.h  --  WM5110 ALSA SoC Audio driver
+ *
+ * Copyright 2012 Wolfson Microelectronics plc
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _WM5110_H
+#define _WM5110_H
+
+#include "arizona.h"
+
+#define WM5110_FLL1 1
+#define WM5110_FLL2 2
+
+#endif
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 555ee14..d26c8ae 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -1,7 +1,7 @@
 /*
  * wm8350.c -- WM8350 ALSA SoC audio driver
  *
- * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC.
+ * Copyright (C) 2007-12 Wolfson Microelectronics PLC.
  *
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
@@ -71,20 +71,6 @@
 	int fll_freq_in;
 };
 
-static unsigned int wm8350_codec_read(struct snd_soc_codec *codec,
-				      unsigned int reg)
-{
-	struct wm8350 *wm8350 = codec->control_data;
-	return wm8350_reg_read(wm8350, reg);
-}
-
-static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg,
-			      unsigned int value)
-{
-	struct wm8350 *wm8350 = codec->control_data;
-	return wm8350_reg_write(wm8350, reg, value);
-}
-
 /*
  * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown.
  */
@@ -1519,7 +1505,9 @@
 	if (ret != 0)
 		return ret;
 
-	codec->control_data = wm8350;
+	codec->control_data = wm8350->regmap;
+
+	snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP);
 
 	/* Put the codec into reset if it wasn't already */
 	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA);
@@ -1629,8 +1617,6 @@
 	.remove =	wm8350_codec_remove,
 	.suspend = 	wm8350_suspend,
 	.resume =	wm8350_resume,
-	.read = wm8350_codec_read,
-	.write = wm8350_codec_write,
 	.set_bias_level = wm8350_set_bias_level,
 
 	.controls = wm8350_snd_controls,
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 5dc31eb..5d277a9 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -1,7 +1,7 @@
 /*
  * wm8400.c  --  WM8400 ALSA Soc Audio driver
  *
- * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 Wolfson Microelectronics PLC.
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 2112851..7c68226 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1,7 +1,7 @@
 /*
  * wm8580.c  --  WM8580 ALSA Soc Audio driver
  *
- * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Copyright 2008-11 Wolfson Microelectronics PLC.
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 9d1b9b02..bb1d269 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -2,6 +2,7 @@
  * wm8731.c  --  WM8731 ALSA SoC Audio driver
  *
  * Copyright 2005 Openedhand Ltd.
+ * Copyright 2006-12 Wolfson Microelectronics, plc
  *
  * Author: Richard Purdie <richard@openedhand.com>
  *
diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c
index 6e849cb..35f3d23 100644
--- a/sound/soc/codecs/wm8741.c
+++ b/sound/soc/codecs/wm8741.c
@@ -1,7 +1,7 @@
 /*
  * wm8741.c  --  WM8741 ALSA SoC Audio driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-1 Wolfson Microelectronics plc
  *
  * Author: Ian Lartey <ian@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index a26482c..13bff87 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -1,7 +1,7 @@
 /*
  * wm8753.c  --  WM8753 ALSA Soc Audio driver
  *
- * Copyright 2003 Wolfson Microelectronics PLC.
+ * Copyright 2003-11 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index a19db5a..879c356 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -1,7 +1,7 @@
 /*
  * wm8776.c  --  WM8776 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c
index 6bd1b76..c088020 100644
--- a/sound/soc/codecs/wm8804.c
+++ b/sound/soc/codecs/wm8804.c
@@ -1,7 +1,7 @@
 /*
  * wm8804.c  --  WM8804 S/PDIF transceiver driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-11 Wolfson Microelectronics plc
  *
  * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 86b8a29..73f1c8d 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -1,8 +1,8 @@
 /*
  * wm8903.c  --  WM8903 ALSA SoC Audio driver
  *
- * Copyright 2008 Wolfson Microelectronics
- * Copyright 2011 NVIDIA, Inc.
+ * Copyright 2008-12 Wolfson Microelectronics
+ * Copyright 2011-2012 NVIDIA, Inc.
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -116,6 +116,7 @@
 
 struct wm8903_priv {
 	struct wm8903_platform_data *pdata;
+	struct device *dev;
 	struct snd_soc_codec *codec;
 	struct regmap *regmap;
 
@@ -1635,17 +1636,27 @@
 
 static irqreturn_t wm8903_irq(int irq, void *data)
 {
-	struct snd_soc_codec *codec = data;
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	int mic_report;
-	int int_pol;
-	int int_val = 0;
-	int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK);
+	struct wm8903_priv *wm8903 = data;
+	int mic_report, ret;
+	unsigned int int_val, mask, int_pol;
 
-	int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK,
+			  &mask);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret);
+		return IRQ_NONE;
+	}
+
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret);
+		return IRQ_NONE;
+	}
+
+	int_val &= ~mask;
 
 	if (int_val & WM8903_WSEQ_BUSY_EINT) {
-		dev_warn(codec->dev, "Write sequencer done\n");
+		dev_warn(wm8903->dev, "Write sequencer done\n");
 	}
 
 	/*
@@ -1656,22 +1667,28 @@
 	 * the polarity register.
 	 */
 	mic_report = wm8903->mic_last_report;
-	int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1);
+	ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+			  &int_pol);
+	if (ret != 0) {
+		dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n",
+			ret);
+		return IRQ_HANDLED;
+	}
 
 #ifndef CONFIG_SND_SOC_WM8903_MODULE
 	if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT))
-		trace_snd_soc_jack_irq(dev_name(codec->dev));
+		trace_snd_soc_jack_irq(dev_name(wm8903->dev));
 #endif
 
 	if (int_val & WM8903_MICSHRT_EINT) {
-		dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol);
+		dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol);
 
 		mic_report ^= wm8903->mic_short;
 		int_pol ^= WM8903_MICSHRT_INV;
 	}
 
 	if (int_val & WM8903_MICDET_EINT) {
-		dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol);
+		dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol);
 
 		mic_report ^= wm8903->mic_det;
 		int_pol ^= WM8903_MICDET_INV;
@@ -1679,8 +1696,8 @@
 		msleep(wm8903->mic_delay);
 	}
 
-	snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1,
-			    WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
+	regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1,
+			   WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol);
 
 	snd_soc_jack_report(wm8903->mic_jack, mic_report,
 			    wm8903->mic_short | wm8903->mic_det);
@@ -1774,7 +1791,6 @@
 static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 	unsigned int mask, val;
 	int ret;
 
@@ -1782,8 +1798,8 @@
 	val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) |
 		WM8903_GP1_DIR;
 
-	ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-				  mask, val);
+	ret = regmap_update_bits(wm8903->regmap,
+				 WM8903_GPIO_CONTROL_1 + offset, mask, val);
 	if (ret < 0)
 		return ret;
 
@@ -1793,10 +1809,9 @@
 static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
-	int reg;
+	unsigned int reg;
 
-	reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset);
+	regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, &reg);
 
 	return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT;
 }
@@ -1805,7 +1820,6 @@
 				     unsigned offset, int value)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 	unsigned int mask, val;
 	int ret;
 
@@ -1813,8 +1827,8 @@
 	val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) |
 		(value << WM8903_GP2_LVL_SHIFT);
 
-	ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-				  mask, val);
+	ret = regmap_update_bits(wm8903->regmap,
+				 WM8903_GPIO_CONTROL_1 + offset, mask, val);
 	if (ret < 0)
 		return ret;
 
@@ -1824,11 +1838,10 @@
 static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value)
 {
 	struct wm8903_priv *wm8903 = gpio_to_wm8903(chip);
-	struct snd_soc_codec *codec = wm8903->codec;
 
-	snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset,
-			    WM8903_GP1_LVL_MASK,
-			    !!value << WM8903_GP1_LVL_SHIFT);
+	regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset,
+			   WM8903_GP1_LVL_MASK,
+			   !!value << WM8903_GP1_LVL_SHIFT);
 }
 
 static struct gpio_chip wm8903_template_chip = {
@@ -1842,15 +1855,14 @@
 	.can_sleep		= 1,
 };
 
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	struct wm8903_platform_data *pdata = wm8903->pdata;
 	int ret;
 
 	wm8903->gpio_chip = wm8903_template_chip;
 	wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO;
-	wm8903->gpio_chip.dev = codec->dev;
+	wm8903->gpio_chip.dev = wm8903->dev;
 
 	if (pdata->gpio_base)
 		wm8903->gpio_chip.base = pdata->gpio_base;
@@ -1859,24 +1871,23 @@
 
 	ret = gpiochip_add(&wm8903->gpio_chip);
 	if (ret != 0)
-		dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret);
+		dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret);
 }
 
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
 	ret = gpiochip_remove(&wm8903->gpio_chip);
 	if (ret != 0)
-		dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret);
+		dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret);
 }
 #else
-static void wm8903_init_gpio(struct snd_soc_codec *codec)
+static void wm8903_init_gpio(struct wm8903_priv *wm8903)
 {
 }
 
-static void wm8903_free_gpio(struct snd_soc_codec *codec)
+static void wm8903_free_gpio(struct wm8903_priv *wm8903)
 {
 }
 #endif
@@ -1884,11 +1895,7 @@
 static int wm8903_probe(struct snd_soc_codec *codec)
 {
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-	struct wm8903_platform_data *pdata = wm8903->pdata;
-	int ret, i;
-	int trigger, irq_pol;
-	u16 val;
-	bool mic_gpio = false;
+	int ret;
 
 	wm8903->codec = codec;
 	codec->control_data = wm8903->regmap;
@@ -1899,121 +1906,16 @@
 		return ret;
 	}
 
-	/* Set up GPIOs, detect if any are MIC detect outputs */
-	for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
-		if ((!pdata->gpio_cfg[i]) ||
-		    (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
-			continue;
-
-		snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
-				pdata->gpio_cfg[i] & 0x7fff);
-
-		val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
-			>> WM8903_GP1_FN_SHIFT;
-
-		switch (val) {
-		case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
-		case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
-			mic_gpio = true;
-			break;
-		default:
-			break;
-		}
-	}
-
-	/* Set up microphone detection */
-	snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
-			pdata->micdet_cfg);
-
-	/* Microphone detection needs the WSEQ clock */
-	if (pdata->micdet_cfg)
-		snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
-				    WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
-
-	/* If microphone detection is enabled by pdata but
-	    * detected via IRQ then interrupts can be lost before
-	    * the machine driver has set up microphone detection
-	    * IRQs as the IRQs are clear on read.  The detection
-	    * will be enabled when the machine driver configures.
-	    */
-	WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
-
-	wm8903->mic_delay = pdata->micdet_delay;
-
-	if (wm8903->irq) {
-		if (pdata->irq_active_low) {
-			trigger = IRQF_TRIGGER_LOW;
-			irq_pol = WM8903_IRQ_POL;
-		} else {
-			trigger = IRQF_TRIGGER_HIGH;
-			irq_pol = 0;
-		}
-
-		snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL,
-				    WM8903_IRQ_POL, irq_pol);
-		
-		ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq,
-					   trigger | IRQF_ONESHOT,
-					   "wm8903", codec);
-		if (ret != 0) {
-			dev_err(codec->dev, "Failed to request IRQ: %d\n",
-				ret);
-			return ret;
-		}
-
-		/* Enable write sequencer interrupts */
-		snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK,
-				    WM8903_IM_WSEQ_BUSY_EINT, 0);
-	}
-
 	/* power on device */
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	/* Latch volume update bits */
-	val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT);
-	val |= WM8903_ADCVU;
-	snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val);
-	snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT);
-	val |= WM8903_DACVU;
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val);
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT);
-	val |= WM8903_HPOUTVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT);
-	val |= WM8903_LINEOUTVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val);
-
-	val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT);
-	val |= WM8903_SPKVU;
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val);
-	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
-
-	/* Enable DAC soft mute by default */
-	snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
-			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
-			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
-
-	wm8903_init_gpio(codec);
-
 	return ret;
 }
 
 /* power down chip */
 static int wm8903_remove(struct snd_soc_codec *codec)
 {
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
-	wm8903_free_gpio(codec);
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	if (wm8903->irq)
-		free_irq(wm8903->irq, codec);
 
 	return 0;
 }
@@ -2123,15 +2025,18 @@
 {
 	struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev);
 	struct wm8903_priv *wm8903;
-	unsigned int val;
-	int ret;
+	int trigger;
+	bool mic_gpio = false;
+	unsigned int val, irq_pol;
+	int ret, i;
 
 	wm8903 = devm_kzalloc(&i2c->dev,  sizeof(struct wm8903_priv),
 			      GFP_KERNEL);
 	if (wm8903 == NULL)
 		return -ENOMEM;
+	wm8903->dev = &i2c->dev;
 
-	wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap);
+	wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap);
 	if (IS_ERR(wm8903->regmap)) {
 		ret = PTR_ERR(wm8903->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2140,7 +2045,6 @@
 	}
 
 	i2c_set_clientdata(i2c, wm8903);
-	wm8903->irq = i2c->irq;
 
 	/* If no platform data was supplied, create storage for defaults */
 	if (pdata) {
@@ -2167,6 +2071,8 @@
 		}
 	}
 
+	pdata = wm8903->pdata;
+
 	ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val);
 	if (ret != 0) {
 		dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret);
@@ -2189,6 +2095,107 @@
 	/* Reset the device */
 	regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903);
 
+	wm8903_init_gpio(wm8903);
+
+	/* Set up GPIO pin state, detect if any are MIC detect outputs */
+	for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
+		if ((!pdata->gpio_cfg[i]) ||
+		    (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO))
+			continue;
+
+		regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i,
+				pdata->gpio_cfg[i] & 0x7fff);
+
+		val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK)
+			>> WM8903_GP1_FN_SHIFT;
+
+		switch (val) {
+		case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT:
+		case WM8903_GPn_FN_MICBIAS_SHORT_DETECT:
+			mic_gpio = true;
+			break;
+		default:
+			break;
+		}
+	}
+
+	/* Set up microphone detection */
+	regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0,
+		     pdata->micdet_cfg);
+
+	/* Microphone detection needs the WSEQ clock */
+	if (pdata->micdet_cfg)
+		regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0,
+				   WM8903_WSEQ_ENA, WM8903_WSEQ_ENA);
+
+	/* If microphone detection is enabled by pdata but
+	 * detected via IRQ then interrupts can be lost before
+	 * the machine driver has set up microphone detection
+	 * IRQs as the IRQs are clear on read.  The detection
+	 * will be enabled when the machine driver configures.
+	 */
+	WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA));
+
+	wm8903->mic_delay = pdata->micdet_delay;
+
+	if (i2c->irq) {
+		if (pdata->irq_active_low) {
+			trigger = IRQF_TRIGGER_LOW;
+			irq_pol = WM8903_IRQ_POL;
+		} else {
+			trigger = IRQF_TRIGGER_HIGH;
+			irq_pol = 0;
+		}
+
+		regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL,
+				   WM8903_IRQ_POL, irq_pol);
+
+		ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq,
+					   trigger | IRQF_ONESHOT,
+					   "wm8903", wm8903);
+		if (ret != 0) {
+			dev_err(wm8903->dev, "Failed to request IRQ: %d\n",
+				ret);
+			return ret;
+		}
+
+		/* Enable write sequencer interrupts */
+		regmap_update_bits(wm8903->regmap,
+				   WM8903_INTERRUPT_STATUS_1_MASK,
+				   WM8903_IM_WSEQ_BUSY_EINT, 0);
+	}
+
+	/* Latch volume update bits */
+	regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT,
+			   WM8903_ADCVU, WM8903_ADCVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT,
+			   WM8903_ADCVU, WM8903_ADCVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT,
+			   WM8903_DACVU, WM8903_DACVU);
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT,
+			   WM8903_DACVU, WM8903_DACVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT,
+			   WM8903_HPOUTVU, WM8903_HPOUTVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT,
+			   WM8903_HPOUTVU, WM8903_HPOUTVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT,
+			   WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT,
+			   WM8903_LINEOUTVU, WM8903_LINEOUTVU);
+
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT,
+			   WM8903_SPKVU, WM8903_SPKVU);
+	regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT,
+			   WM8903_SPKVU, WM8903_SPKVU);
+
+	/* Enable DAC soft mute by default */
+	regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1,
+			   WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+			   WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8903, &wm8903_dai, 1);
 	if (ret != 0)
@@ -2196,7 +2203,6 @@
 
 	return 0;
 err:
-	regmap_exit(wm8903->regmap);
 	return ret;
 }
 
@@ -2204,7 +2210,9 @@
 {
 	struct wm8903_priv *wm8903 = i2c_get_clientdata(client);
 
-	regmap_exit(wm8903->regmap);
+	if (client->irq)
+		free_irq(client->irq, wm8903);
+	wm8903_free_gpio(wm8903);
 	snd_soc_unregister_codec(&client->dev);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 812acd8..0013afe 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1,7 +1,7 @@
 /*
  * wm8904.c  --  WM8904 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -314,11 +314,6 @@
 	}
 }
 
-static int wm8904_reset(struct snd_soc_codec *codec)
-{
-	return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0);
-}
-
 static int wm8904_configure_clocking(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -1945,25 +1940,6 @@
 	.symmetric_rates = 1,
 };
 
-#ifdef CONFIG_PM
-static int wm8904_suspend(struct snd_soc_codec *codec)
-{
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
-
-	return 0;
-}
-
-static int wm8904_resume(struct snd_soc_codec *codec)
-{
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	return 0;
-}
-#else
-#define wm8904_suspend NULL
-#define wm8904_resume NULL
-#endif
-
 static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
@@ -2078,8 +2054,7 @@
 static int wm8904_probe(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
-	struct wm8904_pdata *pdata = wm8904->pdata;
-	int ret, i;
+	int ret;
 
 	codec->control_data = wm8904->regmap;
 
@@ -2101,127 +2076,17 @@
 		return ret;
 	}
 
-	for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
-		wm8904->supplies[i].supply = wm8904_supply_names[i];
-
-	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies),
-				 wm8904->supplies);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
-		return ret;
-	}
-
-	ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
-				    wm8904->supplies);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
-		goto err_get;
-	}
-
-	ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read ID register\n");
-		goto err_enable;
-	}
-	if (ret != 0x8904) {
-		dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret);
-		ret = -EINVAL;
-		goto err_enable;
-	}
-
-	ret = snd_soc_read(codec, WM8904_REVISION);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to read device revision: %d\n",
-			ret);
-		goto err_enable;
-	}
-	dev_info(codec->dev, "revision %c\n", ret + 'A');
-
-	ret = wm8904_reset(codec);
-	if (ret < 0) {
-		dev_err(codec->dev, "Failed to issue reset\n");
-		goto err_enable;
-	}
-
-	regcache_cache_only(wm8904->regmap, true);
-	/* Change some default settings - latch VU and enable ZC */
-	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT,
-			    WM8904_ADC_VU, WM8904_ADC_VU);
-	snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
-			    WM8904_ADC_VU, WM8904_ADC_VU);
-	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT,
-			    WM8904_DAC_VU, WM8904_DAC_VU);
-	snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
-			    WM8904_DAC_VU, WM8904_DAC_VU);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT,
-			    WM8904_HPOUT_VU | WM8904_HPOUTLZC,
-			    WM8904_HPOUT_VU | WM8904_HPOUTLZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT,
-			    WM8904_HPOUT_VU | WM8904_HPOUTRZC,
-			    WM8904_HPOUT_VU | WM8904_HPOUTRZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
-	snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
-			    WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
-	snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0,
-			    WM8904_SR_MODE, 0);
-
-	/* Apply configuration from the platform data. */
-	if (wm8904->pdata) {
-		for (i = 0; i < WM8904_GPIO_REGS; i++) {
-			if (!pdata->gpio_cfg[i])
-				continue;
-
-			regmap_update_bits(wm8904->regmap,
-					   WM8904_GPIO_CONTROL_1 + i,
-					   0xffff,
-					   pdata->gpio_cfg[i]);
-		}
-
-		/* Zero is the default value for these anyway */
-		for (i = 0; i < WM8904_MIC_REGS; i++)
-			regmap_update_bits(wm8904->regmap,
-					   WM8904_MIC_BIAS_CONTROL_0 + i,
-					   0xffff,
-					   pdata->mic_cfg[i]);
-	}
-
-	/* Set Class W by default - this will be managed by the Class
-	 * G widget at runtime where bypass paths are available.
-	 */
-	snd_soc_update_bits(codec, WM8904_CLASS_W_0,
-			    WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
-
-	/* Use normal bias source */
-	snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0,
-			    WM8904_POBCTRL, 0);
-
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	/* Bias level configuration will have done an extra enable */
-	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-
 	wm8904_handle_pdata(codec);
 
 	wm8904_add_widgets(codec);
 
 	return 0;
-
-err_enable:
-	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-err_get:
-	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
-	return ret;
 }
 
 static int wm8904_remove(struct snd_soc_codec *codec)
 {
 	struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec);
 
-	wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 	kfree(wm8904->retune_mobile_texts);
 	kfree(wm8904->drc_texts);
 
@@ -2231,8 +2096,6 @@
 static struct snd_soc_codec_driver soc_codec_dev_wm8904 = {
 	.probe =	wm8904_probe,
 	.remove =	wm8904_remove,
-	.suspend =	wm8904_suspend,
-	.resume =	wm8904_resume,
 	.set_bias_level = wm8904_set_bias_level,
 	.idle_bias_off = true,
 };
@@ -2254,14 +2117,15 @@
 				      const struct i2c_device_id *id)
 {
 	struct wm8904_priv *wm8904;
-	int ret;
+	unsigned int val;
+	int ret, i;
 
 	wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv),
 			      GFP_KERNEL);
 	if (wm8904 == NULL)
 		return -ENOMEM;
 
-	wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap);
+	wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap);
 	if (IS_ERR(wm8904->regmap)) {
 		ret = PTR_ERR(wm8904->regmap);
 		dev_err(&i2c->dev, "Failed to allocate register map: %d\n",
@@ -2273,23 +2137,121 @@
 	i2c_set_clientdata(i2c, wm8904);
 	wm8904->pdata = i2c->dev.platform_data;
 
+	for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++)
+		wm8904->supplies[i].supply = wm8904_supply_names[i];
+
+	ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies),
+				      wm8904->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies),
+				    wm8904->supplies);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret);
+		return ret;
+	}
+
+	ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret);
+		goto err_enable;
+	}
+	if (val != 0x8904) {
+		dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val);
+		ret = -EINVAL;
+		goto err_enable;
+	}
+
+	ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to read device revision: %d\n",
+			ret);
+		goto err_enable;
+	}
+	dev_info(&i2c->dev, "revision %c\n", val + 'A');
+
+	ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0);
+	if (ret < 0) {
+		dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+		goto err_enable;
+	}
+
+	/* Change some default settings - latch VU and enable ZC */
+	regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT,
+			   WM8904_ADC_VU, WM8904_ADC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT,
+			   WM8904_ADC_VU, WM8904_ADC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT,
+			   WM8904_DAC_VU, WM8904_DAC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT,
+			   WM8904_DAC_VU, WM8904_DAC_VU);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT,
+			   WM8904_HPOUT_VU | WM8904_HPOUTLZC,
+			   WM8904_HPOUT_VU | WM8904_HPOUTLZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT,
+			   WM8904_HPOUT_VU | WM8904_HPOUTRZC,
+			   WM8904_HPOUT_VU | WM8904_HPOUTRZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTLZC,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTLZC);
+	regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTRZC,
+			   WM8904_LINEOUT_VU | WM8904_LINEOUTRZC);
+	regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0,
+			   WM8904_SR_MODE, 0);
+
+	/* Apply configuration from the platform data. */
+	if (wm8904->pdata) {
+		for (i = 0; i < WM8904_GPIO_REGS; i++) {
+			if (!wm8904->pdata->gpio_cfg[i])
+				continue;
+
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_GPIO_CONTROL_1 + i,
+					   0xffff,
+					   wm8904->pdata->gpio_cfg[i]);
+		}
+
+		/* Zero is the default value for these anyway */
+		for (i = 0; i < WM8904_MIC_REGS; i++)
+			regmap_update_bits(wm8904->regmap,
+					   WM8904_MIC_BIAS_CONTROL_0 + i,
+					   0xffff,
+					   wm8904->pdata->mic_cfg[i]);
+	}
+
+	/* Set Class W by default - this will be managed by the Class
+	 * G widget at runtime where bypass paths are available.
+	 */
+	regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0,
+			    WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR);
+
+	/* Use normal bias source */
+	regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0,
+			    WM8904_POBCTRL, 0);
+
+	/* Can leave the device powered off until we need it */
+	regcache_cache_only(wm8904->regmap, true);
+	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8904, &wm8904_dai, 1);
 	if (ret != 0)
-		goto err;
+		return ret;
 
 	return 0;
 
-err:
-	regmap_exit(wm8904->regmap);
+err_enable:
+	regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies);
 	return ret;
 }
 
 static __devexit int wm8904_i2c_remove(struct i2c_client *client)
 {
-	struct wm8904_priv *wm8904 = i2c_get_clientdata(client);
 	snd_soc_unregister_codec(&client->dev);
-	regmap_exit(wm8904->regmap);
 	return 0;
 }
 
@@ -2311,23 +2273,7 @@
 	.id_table = wm8904_i2c_id,
 };
 
-static int __init wm8904_modinit(void)
-{
-	int ret = 0;
-	ret = i2c_add_driver(&wm8904_i2c_driver);
-	if (ret != 0) {
-		printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n",
-		       ret);
-	}
-	return ret;
-}
-module_init(wm8904_modinit);
-
-static void __exit wm8904_exit(void)
-{
-	i2c_del_driver(&wm8904_i2c_driver);
-}
-module_exit(wm8904_exit);
+module_i2c_driver(wm8904_i2c_driver);
 
 MODULE_DESCRIPTION("ASoC WM8904 driver");
 MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index 8bc659d..96518ac 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -1,6 +1,8 @@
 /*
  * wm8960.c  --  WM8960 ALSA SoC Audio driver
  *
+ * Copyright 2007-11 Wolfson Microelectronics, plc
+ *
  * Author: Liam Girdwood
  *
  * This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c
index 05ea7c2..01edbcc 100644
--- a/sound/soc/codecs/wm8961.c
+++ b/sound/soc/codecs/wm8961.c
@@ -1,6 +1,8 @@
 /*
  * wm8961.c  --  WM8961 ALSA SoC Audio driver
  *
+ * Copyright 2009-10 Wolfson Microelectronics, plc
+ *
  * Author: Mark Brown
  *
  * This program is free software; you can redistribute it and/or modify
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 0cfce99..eaf6586 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -1,7 +1,7 @@
 /*
  * wm8962.c  --  WM8962 ALSA SoC Audio driver
  *
- * Copyright 2010 Wolfson Microelectronics plc
+ * Copyright 2010-2 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -2580,6 +2580,9 @@
 			    WM8962_SAMPLE_RATE_INT_MODE |
 			    WM8962_SAMPLE_RATE_MASK, adctl3);
 
+	dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n",
+		wm8962->bclk, wm8962->lrclk);
+
 	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
 		wm8962_configure_bclk(codec);
 
@@ -3722,6 +3725,9 @@
 	}
 
 	regcache_cache_only(wm8962->regmap, false);
+
+	wm8962_reset(wm8962);
+
 	regcache_sync(wm8962->regmap);
 
 	regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 36acfcc..9fd80d6 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -1,7 +1,7 @@
 /*
  * wm8993.c -- WM8993 ALSA SoC audio driver
  *
- * Copyright 2009, 2010 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 1436b6c..bb62f4b 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1,7 +1,7 @@
 /*
  * wm8994.c  --  WM8994 ALSA SoC Audio driver
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
@@ -2967,23 +2967,8 @@
 static int wm8994_codec_suspend(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	struct wm8994 *control = wm8994->wm8994;
 	int i, ret;
 
-	switch (control->type) {
-	case WM8994:
-		snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
-		break;
-	case WM1811:
-		snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-				    WM1811_JACKDET_MODE_MASK, 0);
-		/* Fall through */
-	case WM8958:
-		snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
-				    WM8958_MICD_ENA, 0);
-		break;
-	}
-
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
 		memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
 		       sizeof(struct wm8994_fll_config));
@@ -3033,28 +3018,6 @@
 				 i + 1, ret);
 	}
 
-	switch (control->type) {
-	case WM8994:
-		if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
-			snd_soc_update_bits(codec, WM8994_MICBIAS,
-					    WM8994_MICD_ENA, WM8994_MICD_ENA);
-		break;
-	case WM1811:
-		if (wm8994->jackdet && wm8994->jack_cb) {
-			/* Restart from idle */
-			snd_soc_update_bits(codec, WM8994_ANTIPOP_2,
-					    WM1811_JACKDET_MODE_MASK,
-					    WM1811_JACKDET_MODE_JACK);
-			break;
-		}
-		break;
-	case WM8958:
-		if (wm8994->jack_cb)
-			snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
-					    WM8958_MICD_ENA, WM8958_MICD_ENA);
-		break;
-	}
-
 	return 0;
 }
 #else
@@ -3729,9 +3692,6 @@
 
 	if (wm8994->pdata && wm8994->pdata->micdet_irq)
 		wm8994->micdet_irq = wm8994->pdata->micdet_irq;
-	else if (wm8994->pdata && wm8994->pdata->irq_base)
-		wm8994->micdet_irq = wm8994->pdata->irq_base +
-				     WM8994_IRQ_MIC1_DET;
 
 	pm_runtime_enable(codec->dev);
 	pm_runtime_idle(codec->dev);
@@ -3870,6 +3830,10 @@
 				dev_warn(codec->dev,
 					 "Failed to request Mic detect IRQ: %d\n",
 					 ret);
+		} else {
+			wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET,
+					   wm8958_mic_irq, "Mic detect",
+					   wm8994);
 		}
 	}
 
diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c
index dc9b42b..00f183d 100644
--- a/sound/soc/codecs/wm8996.c
+++ b/sound/soc/codecs/wm8996.c
@@ -1,7 +1,7 @@
 /*
  * wm8996.c - WM8996 audio codec interface
  *
- * Copyright 2011 Wolfson Microelectronics PLC.
+ * Copyright 2011-2 Wolfson Microelectronics PLC.
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -296,184 +296,6 @@
 	{ WM8996_RIGHT_PDM_SPEAKER, 0x1 },
 	{ WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 },
 	{ WM8996_PDM_SPEAKER_VOLUME, 0x66 },
-	{ WM8996_WRITE_SEQUENCER_0, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_1, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_3, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_4, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_5, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_6, 0xf },
-	{ WM8996_WRITE_SEQUENCER_7, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_8, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_9, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_10, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_12, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_13, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_14, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_16, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_17, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_18, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_20, 0x51 },
-	{ WM8996_WRITE_SEQUENCER_21, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_22, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_23, 0xa },
-	{ WM8996_WRITE_SEQUENCER_24, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_25, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_26, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_27, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_28, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_32, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_36, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_40, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_44, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_48, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_52, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_56, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_60, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_64, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_65, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_67, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_68, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_69, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_70, 0xf },
-	{ WM8996_WRITE_SEQUENCER_71, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_72, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_73, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_74, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_76, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_77, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_78, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_80, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_81, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_82, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_84, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_85, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_86, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_87, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_88, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_92, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_96, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_100, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_104, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_108, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_112, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_116, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_120, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_124, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_128, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_129, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_131, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_132, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_133, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_134, 0xf },
-	{ WM8996_WRITE_SEQUENCER_135, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_136, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_137, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_138, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_140, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_141, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_142, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_144, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_145, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_146, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_148, 0x51 },
-	{ WM8996_WRITE_SEQUENCER_149, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_150, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_151, 0xa },
-	{ WM8996_WRITE_SEQUENCER_152, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_153, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_154, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_155, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_156, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_160, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_164, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_168, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_172, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_176, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_180, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_184, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_188, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_192, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_193, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_195, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_196, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_197, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_198, 0xf },
-	{ WM8996_WRITE_SEQUENCER_199, 0x6 },
-	{ WM8996_WRITE_SEQUENCER_200, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_201, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_202, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_204, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_205, 0x11 },
-	{ WM8996_WRITE_SEQUENCER_206, 0x401 },
-	{ WM8996_WRITE_SEQUENCER_208, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_209, 0x3 },
-	{ WM8996_WRITE_SEQUENCER_210, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_212, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_213, 0x3b },
-	{ WM8996_WRITE_SEQUENCER_214, 0x502 },
-	{ WM8996_WRITE_SEQUENCER_215, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_216, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_220, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_224, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_228, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_232, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_236, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_240, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_244, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_248, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_252, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_256, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_258, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_260, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_262, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_264, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_266, 0x104 },
-	{ WM8996_WRITE_SEQUENCER_267, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_268, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_272, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_276, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_280, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_284, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_288, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_292, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_296, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_300, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_304, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_308, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_312, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_316, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_320, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_322, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_324, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_326, 0x102 },
-	{ WM8996_WRITE_SEQUENCER_328, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_330, 0x106 },
-	{ WM8996_WRITE_SEQUENCER_331, 0x100 },
-	{ WM8996_WRITE_SEQUENCER_332, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_336, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_340, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_344, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_348, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_352, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_356, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_360, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_364, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_368, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_372, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_376, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_380, 0x2fff },
-	{ WM8996_WRITE_SEQUENCER_384, 0x60 },
-	{ WM8996_WRITE_SEQUENCER_386, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_388, 0x61 },
-	{ WM8996_WRITE_SEQUENCER_390, 0x601 },
-	{ WM8996_WRITE_SEQUENCER_392, 0x50 },
-	{ WM8996_WRITE_SEQUENCER_394, 0x300 },
-	{ WM8996_WRITE_SEQUENCER_396, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_398, 0x304 },
-	{ WM8996_WRITE_SEQUENCER_400, 0x40 },
-	{ WM8996_WRITE_SEQUENCER_402, 0xf },
-	{ WM8996_WRITE_SEQUENCER_404, 0x1 },
-	{ WM8996_WRITE_SEQUENCER_407, 0x100 },
 };
 
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0);
@@ -1706,18 +1528,6 @@
 	}
 }
 
-static int wm8996_reset(struct wm8996_priv *wm8996)
-{
-	if (wm8996->pdata.ldo_ena > 0) {
-		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
-		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1);
-		return 0;
-	} else {
-		return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
-				    0x8915);
-	}
-}
-
 static const int bclk_divs[] = {
 	1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96
 };
@@ -1809,8 +1619,10 @@
 
 	case SND_SOC_BIAS_OFF:
 		regcache_cache_only(codec->control_data, true);
-		if (wm8996->pdata.ldo_ena >= 0)
+		if (wm8996->pdata.ldo_ena >= 0) {
 			gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+			regcache_cache_only(codec->control_data, true);
+		}
 		regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies),
 				       wm8996->supplies);
 		break;
@@ -2807,7 +2619,7 @@
 	int ret;
 	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
-	int i, irq_flags;
+	int irq_flags;
 
 	wm8996->codec = codec;
 
@@ -2822,177 +2634,12 @@
 		goto err;
 	}
 
-	wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
-	wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
-	wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
-
-	/* This should really be moved into the regulator core */
-	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
-		ret = regulator_register_notifier(wm8996->supplies[i].consumer,
-						  &wm8996->disable_nb[i]);
-		if (ret != 0) {
-			dev_err(codec->dev,
-				"Failed to register regulator notifier: %d\n",
-				ret);
-		}
-	}
-
-	/* Apply platform data settings */
-	snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL,
-			    WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
-			    wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
-			    wm8996->pdata.inr_mode);
-
-	for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
-		if (!wm8996->pdata.gpio_default[i])
-			continue;
-
-		snd_soc_write(codec, WM8996_GPIO_1 + i,
-			      wm8996->pdata.gpio_default[i] & 0xffff);
-	}
-
-	if (wm8996->pdata.spkmute_seq)
-		snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
-				    WM8996_SPK_MUTE_ENDIAN |
-				    WM8996_SPK_MUTE_SEQ1_MASK,
-				    wm8996->pdata.spkmute_seq);
-
-	snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2,
-			    WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
-			    WM8996_MICD_SRC, wm8996->pdata.micdet_def);
-
-	/* Latch volume update bits */
-	snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME,
-			    WM8996_IN1_VU, WM8996_IN1_VU);
-	snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME,
-			    WM8996_IN1_VU, WM8996_IN1_VU);
-
-	snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-	snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-
-	snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME,
-			    WM8996_DAC1_VU, WM8996_DAC1_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-	snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME,
-			    WM8996_DAC2_VU, WM8996_DAC2_VU);
-
-	snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME,
-			    WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME,
-			    WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME,
-			    WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME,
-			    WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
-
-	snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME,
-			    WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME,
-			    WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME,
-			    WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
-	snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME,
-			    WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
-
-	/* No support currently for the underclocked TDM modes and
-	 * pick a default TDM layout with each channel pair working with
-	 * slots 0 and 1. */
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN2_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN3_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN4_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
-			    WM8996_AIF1RX_CHAN5_SLOTS_MASK |
-			    WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF2RX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF2RX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN2_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN3_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN4_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
-			    WM8996_AIF1TX_CHAN5_SLOTS_MASK |
-			    WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
-
-	snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
-			    WM8996_AIF2TX_CHAN0_SLOTS_MASK |
-			    WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
-			    1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
-	snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
-			    WM8996_AIF2TX_CHAN1_SLOTS_MASK |
-			    WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
-			    1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
-
 	if (wm8996->pdata.num_retune_mobile_cfgs)
 		wm8996_retune_mobile_pdata(codec);
 	else
 		snd_soc_add_codec_controls(codec, wm8996_eq_controls,
 				     ARRAY_SIZE(wm8996_eq_controls));
 
-	/* If the TX LRCLK pins are not in LRCLK mode configure the
-	 * AIFs to source their clocks from the RX LRCLKs.
-	 */
-	if ((snd_soc_read(codec, WM8996_GPIO_1)))
-		snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2,
-				    WM8996_AIF1TX_LRCLK_MODE,
-				    WM8996_AIF1TX_LRCLK_MODE);
-
-	if ((snd_soc_read(codec, WM8996_GPIO_2)))
-		snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2,
-				    WM8996_AIF2TX_LRCLK_MODE,
-				    WM8996_AIF2TX_LRCLK_MODE);
-
 	if (i2c->irq) {
 		if (wm8996->pdata.irq_flags)
 			irq_flags = wm8996->pdata.irq_flags;
@@ -3036,9 +2683,7 @@
 
 static int wm8996_remove(struct snd_soc_codec *codec)
 {
-	struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec);
 	struct i2c_client *i2c = to_i2c_client(codec->dev);
-	int i;
 
 	snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL,
 			    WM8996_IM_IRQ, WM8996_IM_IRQ);
@@ -3046,10 +2691,6 @@
 	if (i2c->irq)
 		free_irq(i2c->irq, codec);
 
-	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
-		regulator_unregister_notifier(wm8996->supplies[i].consumer,
-					      &wm8996->disable_nb[i]);
-
 	return 0;
 }
 
@@ -3163,6 +2804,21 @@
 		goto err_gpio;
 	}
 
+	wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0;
+	wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1;
+	wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2;
+
+	/* This should really be moved into the regulator core */
+	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) {
+		ret = regulator_register_notifier(wm8996->supplies[i].consumer,
+						  &wm8996->disable_nb[i]);
+		if (ret != 0) {
+			dev_err(&i2c->dev,
+				"Failed to register regulator notifier: %d\n",
+				ret);
+		}
+	}
+
 	ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies),
 				    wm8996->supplies);
 	if (ret != 0) {
@@ -3175,7 +2831,7 @@
 		msleep(5);
 	}
 
-	wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap);
+	wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap);
 	if (IS_ERR(wm8996->regmap)) {
 		ret = PTR_ERR(wm8996->regmap);
 		dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
@@ -3203,14 +2859,198 @@
 	dev_info(&i2c->dev, "revision %c\n",
 		 (reg & WM8996_CHIP_REV_MASK) + 'A');
 
-	ret = wm8996_reset(wm8996);
-	if (ret < 0) {
-		dev_err(&i2c->dev, "Failed to issue reset\n");
+	if (wm8996->pdata.ldo_ena > 0) {
+		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
+		regcache_cache_only(wm8996->regmap, true);
+	} else {
+		ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET,
+				   0x8915);
+		if (ret != 0) {
+			dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
+			goto err_regmap;
+		}
+	}
+
+	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+
+	/* Apply platform data settings */
+	regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL,
+			   WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK,
+			   wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT |
+			   wm8996->pdata.inr_mode);
+
+	for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) {
+		if (!wm8996->pdata.gpio_default[i])
+			continue;
+
+		regmap_write(wm8996->regmap, WM8996_GPIO_1 + i,
+			     wm8996->pdata.gpio_default[i] & 0xffff);
+	}
+
+	if (wm8996->pdata.spkmute_seq)
+		regmap_update_bits(wm8996->regmap,
+				   WM8996_PDM_SPEAKER_MUTE_SEQUENCE,
+				   WM8996_SPK_MUTE_ENDIAN |
+				   WM8996_SPK_MUTE_SEQ1_MASK,
+				   wm8996->pdata.spkmute_seq);
+
+	regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2,
+			   WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC |
+			   WM8996_MICD_SRC, wm8996->pdata.micdet_def);
+
+	/* Latch volume update bits */
+	regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME,
+			   WM8996_IN1_VU, WM8996_IN1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME,
+			   WM8996_IN1_VU, WM8996_IN1_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME,
+			   WM8996_DAC1_VU, WM8996_DAC1_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME,
+			   WM8996_DAC2_VU, WM8996_DAC2_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME,
+			   WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME,
+			   WM8996_DSP1TX_VU, WM8996_DSP1TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME,
+			   WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME,
+			   WM8996_DSP2TX_VU, WM8996_DSP2TX_VU);
+
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME,
+			   WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME,
+			   WM8996_DSP1RX_VU, WM8996_DSP1RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME,
+			   WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+	regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME,
+			   WM8996_DSP2RX_VU, WM8996_DSP2RX_VU);
+
+	/* No support currently for the underclocked TDM modes and
+	 * pick a default TDM layout with each channel pair working with
+	 * slots 0 and 1. */
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_2_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN2_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN2_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_3_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN3_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_4_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN4_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1RX_CHANNEL_5_CONFIGURATION,
+			   WM8996_AIF1RX_CHAN5_SLOTS_MASK |
+			   WM8996_AIF1RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2RX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF2RX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF2RX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2RX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF2RX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF2RX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_2_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN2_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_3_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN3_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_4_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN4_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_5_CONFIGURATION,
+			   WM8996_AIF1TX_CHAN5_SLOTS_MASK |
+			   WM8996_AIF1TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1);
+
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF2TX_CHANNEL_0_CONFIGURATION,
+			   WM8996_AIF2TX_CHAN0_SLOTS_MASK |
+			   WM8996_AIF2TX_CHAN0_START_SLOT_MASK,
+			   1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0);
+	regmap_update_bits(wm8996->regmap,
+			   WM8996_AIF1TX_CHANNEL_1_CONFIGURATION,
+			   WM8996_AIF2TX_CHAN1_SLOTS_MASK |
+			   WM8996_AIF2TX_CHAN1_START_SLOT_MASK,
+			   1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1);
+
+	/* If the TX LRCLK pins are not in LRCLK mode configure the
+	 * AIFs to source their clocks from the RX LRCLKs.
+	 */
+	ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret);
 		goto err_regmap;
 	}
 
-	regcache_cache_only(wm8996->regmap, true);
-	regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies);
+	if (reg & WM8996_GP1_FN_MASK)
+		regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2,
+				   WM8996_AIF1TX_LRCLK_MODE,
+				   WM8996_AIF1TX_LRCLK_MODE);
+
+	ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, &reg);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret);
+		goto err_regmap;
+	}
+
+	if (reg & WM8996_GP2_FN_MASK)
+		regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2,
+				   WM8996_AIF2TX_LRCLK_MODE,
+				   WM8996_AIF2TX_LRCLK_MODE);
 
 	wm8996_init_gpio(wm8996);
 
@@ -3225,7 +3065,6 @@
 err_gpiolib:
 	wm8996_free_gpio(wm8996);
 err_regmap:
-	regmap_exit(wm8996->regmap);
 err_enable:
 	if (wm8996->pdata.ldo_ena > 0)
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
@@ -3241,14 +3080,18 @@
 static __devexit int wm8996_i2c_remove(struct i2c_client *client)
 {
 	struct wm8996_priv *wm8996 = i2c_get_clientdata(client);
+	int i;
 
 	snd_soc_unregister_codec(&client->dev);
 	wm8996_free_gpio(wm8996);
-	regmap_exit(wm8996->regmap);
 	if (wm8996->pdata.ldo_ena > 0) {
 		gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0);
 		gpio_free(wm8996->pdata.ldo_ena);
 	}
+	for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++)
+		regulator_unregister_notifier(wm8996->supplies[i].consumer,
+					      &wm8996->disable_nb[i]);
+
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 9328270..2de74e1 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -3,7 +3,7 @@
  *
  * Author: Mark Brown
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c
index 4b263b6..2c2346f 100644
--- a/sound/soc/codecs/wm9090.c
+++ b/sound/soc/codecs/wm9090.c
@@ -1,7 +1,7 @@
 /*
  * ALSA SoC WM9090 driver
  *
- * Copyright 2009, 2010 Wolfson Microelectronics
+ * Copyright 2009-12 Wolfson Microelectronics
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index a154141..099e6ec 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -1,7 +1,7 @@
 /*
  * wm9712.c  --  ALSA Soc WM9712 codec support
  *
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-12 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 2d22cc7..3eb19fb 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -1,7 +1,7 @@
 /*
  * wm9713.c  --  ALSA Soc WM9713 codec support
  *
- * Copyright 2006 Wolfson Microelectronics PLC.
+ * Copyright 2006-10 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index dfe957a..61baa48 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -1,7 +1,7 @@
 /*
  * wm_hubs.c  --  WM8993/4 common code
  *
- * Copyright 2009 Wolfson Microelectronics plc
+ * Copyright 2009-12 Wolfson Microelectronics plc
  *
  * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig
new file mode 100644
index 0000000..e334900
--- /dev/null
+++ b/sound/soc/dwc/Kconfig
@@ -0,0 +1,9 @@
+config SND_DESIGNWARE_I2S
+	tristate "Synopsys I2S Device Driver"
+	depends on CLKDEV_LOOKUP
+	help
+	 Say Y or M if you want to add support for I2S driver for
+	 Synopsys desigwnware I2S device. The device supports upto
+	 maximum of 8 channels each for play and record.
+
+
diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile
new file mode 100644
index 0000000..319371f
--- /dev/null
+++ b/sound/soc/dwc/Makefile
@@ -0,0 +1,3 @@
+# SYNOPSYS Platform Support
+obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o
+
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
new file mode 100644
index 0000000..1aa5130
--- /dev/null
+++ b/sound/soc/dwc/designware_i2s.c
@@ -0,0 +1,455 @@
+/*
+ * ALSA SoC Synopsys I2S Audio Layer
+ *
+ * sound/soc/spear/designware_i2s.c
+ *
+ * Copyright (C) 2010 ST Microelectronics
+ * Rajeev Kumar <rajeev-dlh.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/device.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <sound/designware_i2s.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+/* common register for all channel */
+#define IER		0x000
+#define IRER		0x004
+#define ITER		0x008
+#define CER		0x00C
+#define CCR		0x010
+#define RXFFR		0x014
+#define TXFFR		0x018
+
+/* I2STxRxRegisters for all channels */
+#define LRBR_LTHR(x)	(0x40 * x + 0x020)
+#define RRBR_RTHR(x)	(0x40 * x + 0x024)
+#define RER(x)		(0x40 * x + 0x028)
+#define TER(x)		(0x40 * x + 0x02C)
+#define RCR(x)		(0x40 * x + 0x030)
+#define TCR(x)		(0x40 * x + 0x034)
+#define ISR(x)		(0x40 * x + 0x038)
+#define IMR(x)		(0x40 * x + 0x03C)
+#define ROR(x)		(0x40 * x + 0x040)
+#define TOR(x)		(0x40 * x + 0x044)
+#define RFCR(x)		(0x40 * x + 0x048)
+#define TFCR(x)		(0x40 * x + 0x04C)
+#define RFF(x)		(0x40 * x + 0x050)
+#define TFF(x)		(0x40 * x + 0x054)
+
+/* I2SCOMPRegisters */
+#define I2S_COMP_PARAM_2	0x01F0
+#define I2S_COMP_PARAM_1	0x01F4
+#define I2S_COMP_VERSION	0x01F8
+#define I2S_COMP_TYPE		0x01FC
+
+#define MAX_CHANNEL_NUM		8
+#define MIN_CHANNEL_NUM		2
+
+struct dw_i2s_dev {
+	void __iomem *i2s_base;
+	struct clk *clk;
+	int active;
+	unsigned int capability;
+	struct device *dev;
+
+	/* data related to DMA transfers b/w i2s and DMAC */
+	struct i2s_dma_data play_dma_data;
+	struct i2s_dma_data capture_dma_data;
+	struct i2s_clk_config_data config;
+	int (*i2s_clk_cfg)(struct i2s_clk_config_data *config);
+};
+
+static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val)
+{
+	writel(val, io_base + reg);
+}
+
+static inline u32 i2s_read_reg(void __iomem *io_base, int reg)
+{
+	return readl(io_base + reg);
+}
+
+static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream)
+{
+	u32 i = 0;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, TER(i), 0);
+	} else {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, RER(i), 0);
+	}
+}
+
+static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream)
+{
+	u32 i = 0;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, TOR(i), 0);
+	} else {
+		for (i = 0; i < 4; i++)
+			i2s_write_reg(dev->i2s_base, ROR(i), 0);
+	}
+}
+
+static void i2s_start(struct dw_i2s_dev *dev,
+		      struct snd_pcm_substream *substream)
+{
+
+	i2s_write_reg(dev->i2s_base, IER, 1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		i2s_write_reg(dev->i2s_base, ITER, 1);
+	else
+		i2s_write_reg(dev->i2s_base, IRER, 1);
+
+	i2s_write_reg(dev->i2s_base, CER, 1);
+}
+
+static void i2s_stop(struct dw_i2s_dev *dev,
+		struct snd_pcm_substream *substream)
+{
+	u32 i = 0, irq;
+
+	i2s_clear_irqs(dev, substream->stream);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		i2s_write_reg(dev->i2s_base, ITER, 0);
+
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30);
+		}
+	} else {
+		i2s_write_reg(dev->i2s_base, IRER, 0);
+
+		for (i = 0; i < 4; i++) {
+			irq = i2s_read_reg(dev->i2s_base, IMR(i));
+			i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03);
+		}
+	}
+
+	if (!dev->active) {
+		i2s_write_reg(dev->i2s_base, CER, 0);
+		i2s_write_reg(dev->i2s_base, IER, 0);
+	}
+}
+
+static int dw_i2s_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	struct i2s_dma_data *dma_data = NULL;
+
+	if (!(dev->capability & DWC_I2S_RECORD) &&
+			(substream->stream == SNDRV_PCM_STREAM_CAPTURE))
+		return -EINVAL;
+
+	if (!(dev->capability & DWC_I2S_PLAY) &&
+			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+		return -EINVAL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dma_data = &dev->play_dma_data;
+	else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+		dma_data = &dev->capture_dma_data;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data);
+
+	return 0;
+}
+
+static int dw_i2s_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct i2s_clk_config_data *config = &dev->config;
+	u32 ccr, xfer_resolution, ch_reg, irq;
+	int ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		config->data_width = 16;
+		ccr = 0x00;
+		xfer_resolution = 0x02;
+		break;
+
+	case SNDRV_PCM_FORMAT_S24_LE:
+		config->data_width = 24;
+		ccr = 0x08;
+		xfer_resolution = 0x04;
+		break;
+
+	case SNDRV_PCM_FORMAT_S32_LE:
+		config->data_width = 32;
+		ccr = 0x10;
+		xfer_resolution = 0x05;
+		break;
+
+	default:
+		dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt");
+		return -EINVAL;
+	}
+
+	config->chan_nr = params_channels(params);
+
+	switch (config->chan_nr) {
+	case EIGHT_CHANNEL_SUPPORT:
+		ch_reg = 3;
+	case SIX_CHANNEL_SUPPORT:
+		ch_reg = 2;
+	case FOUR_CHANNEL_SUPPORT:
+		ch_reg = 1;
+	case TWO_CHANNEL_SUPPORT:
+		ch_reg = 0;
+		break;
+	default:
+		dev_err(dev->dev, "channel not supported\n");
+	}
+
+	i2s_disable_channels(dev, substream->stream);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution);
+		i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02);
+		irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+		i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30);
+		i2s_write_reg(dev->i2s_base, TER(ch_reg), 1);
+	} else {
+		i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution);
+		i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07);
+		irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg));
+		i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03);
+		i2s_write_reg(dev->i2s_base, RER(ch_reg), 1);
+	}
+
+	i2s_write_reg(dev->i2s_base, CCR, ccr);
+
+	config->sample_rate = params_rate(params);
+
+	if (!dev->i2s_clk_cfg)
+		return -EINVAL;
+
+	ret = dev->i2s_clk_cfg(config);
+	if (ret < 0) {
+		dev_err(dev->dev, "runtime audio clk config fail\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static void dw_i2s_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static int dw_i2s_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		dev->active++;
+		i2s_start(dev, substream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		dev->active--;
+		i2s_stop(dev, substream);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static struct snd_soc_dai_ops dw_i2s_dai_ops = {
+	.startup	= dw_i2s_startup,
+	.shutdown	= dw_i2s_shutdown,
+	.hw_params	= dw_i2s_hw_params,
+	.trigger	= dw_i2s_trigger,
+};
+
+#ifdef CONFIG_PM
+
+static int dw_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable(dev->clk);
+	return 0;
+}
+
+static int dw_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	clk_enable(dev->clk);
+	return 0;
+}
+
+#else
+#define dw_i2s_suspend	NULL
+#define dw_i2s_resume	NULL
+#endif
+
+static int dw_i2s_probe(struct platform_device *pdev)
+{
+	const struct i2s_platform_data *pdata = pdev->dev.platform_data;
+	struct dw_i2s_dev *dev;
+	struct resource *res;
+	int ret;
+	unsigned int cap;
+	struct snd_soc_dai_driver *dw_i2s_dai;
+
+	if (!pdata) {
+		dev_err(&pdev->dev, "Invalid platform data\n");
+		return -EINVAL;
+	}
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res) {
+		dev_err(&pdev->dev, "no i2s resource defined\n");
+		return -ENODEV;
+	}
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_err(&pdev->dev, "i2s region already claimed\n");
+		return -EBUSY;
+	}
+
+	dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL);
+	if (!dev) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	dev->i2s_base = devm_ioremap(&pdev->dev, res->start,
+			resource_size(res));
+	if (!dev->i2s_base) {
+		dev_err(&pdev->dev, "ioremap fail for i2s_region\n");
+		return -ENOMEM;
+	}
+
+	cap = pdata->cap;
+	dev->capability = cap;
+	dev->i2s_clk_cfg = pdata->i2s_clk_cfg;
+
+	/* Set DMA slaves info */
+
+	dev->play_dma_data.data = pdata->play_dma_data;
+	dev->capture_dma_data.data = pdata->capture_dma_data;
+	dev->play_dma_data.addr = res->start + I2S_TXDMA;
+	dev->capture_dma_data.addr = res->start + I2S_RXDMA;
+	dev->play_dma_data.max_burst = 16;
+	dev->capture_dma_data.max_burst = 16;
+	dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
+	dev->play_dma_data.filter = pdata->filter;
+	dev->capture_dma_data.filter = pdata->filter;
+
+	dev->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(dev->clk))
+		return  PTR_ERR(dev->clk);
+
+	ret = clk_enable(dev->clk);
+	if (ret < 0)
+		goto err_clk_put;
+
+	dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL);
+	if (!dw_i2s_dai) {
+		dev_err(&pdev->dev, "mem allocation failed for dai driver\n");
+		ret = -ENOMEM;
+		goto err_clk_disable;
+	}
+
+	if (cap & DWC_I2S_PLAY) {
+		dev_dbg(&pdev->dev, " SPEAr: play supported\n");
+		dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM;
+		dw_i2s_dai->playback.channels_max = pdata->channel;
+		dw_i2s_dai->playback.formats = pdata->snd_fmts;
+		dw_i2s_dai->playback.rates = pdata->snd_rates;
+	}
+
+	if (cap & DWC_I2S_RECORD) {
+		dev_dbg(&pdev->dev, "SPEAr: record supported\n");
+		dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM;
+		dw_i2s_dai->capture.channels_max = pdata->channel;
+		dw_i2s_dai->capture.formats = pdata->snd_fmts;
+		dw_i2s_dai->capture.rates = pdata->snd_rates;
+	}
+
+	dw_i2s_dai->ops = &dw_i2s_dai_ops;
+	dw_i2s_dai->suspend = dw_i2s_suspend;
+	dw_i2s_dai->resume = dw_i2s_resume;
+
+	dev->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, dev);
+	ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "not able to register dai\n");
+		goto err_set_drvdata;
+	}
+
+	return 0;
+
+err_set_drvdata:
+	dev_set_drvdata(&pdev->dev, NULL);
+err_clk_disable:
+	clk_disable(dev->clk);
+err_clk_put:
+	clk_put(dev->clk);
+	return ret;
+}
+
+static int dw_i2s_remove(struct platform_device *pdev)
+{
+	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(dev->clk);
+
+	return 0;
+}
+
+static struct platform_driver dw_i2s_driver = {
+	.probe		= dw_i2s_probe,
+	.remove		= dw_i2s_remove,
+	.driver		= {
+		.name	= "designware-i2s",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(dw_i2s_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:designware_i2s");
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 162dbb7..4eea98b 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -136,7 +136,7 @@
 	.hw_params	= ep93xx_pcm_hw_params,
 	.hw_free	= ep93xx_pcm_hw_free,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= ep93xx_pcm_mmap,
 };
 
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index 0803274..e7c800e 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -156,7 +156,7 @@
 		return;
 	}
 
-	for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) {
+	for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) {
 		snprintf(buf, sizeof(buf), "ssi%d", i);
 		if (!debugfs_create_file(buf, 0444, audmux_debugfs_root,
 					 (void *)i, &audmux_debugfs_fops))
diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h
index 04ebbab..b8ff44b 100644
--- a/sound/soc/fsl/imx-audmux.h
+++ b/sound/soc/fsl/imx-audmux.h
@@ -14,6 +14,7 @@
 #define MX31_AUDMUX_PORT4_SSI_PINS_4	3
 #define MX31_AUDMUX_PORT5_SSI_PINS_5	4
 #define MX31_AUDMUX_PORT6_SSI_PINS_6	5
+#define MX31_AUDMUX_PORT7_SSI_PINS_7	6
 
 #define MX51_AUDMUX_PORT1_SSI0		0
 #define MX51_AUDMUX_PORT2_SSI1		1
diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c
index f59c349..549b31f 100644
--- a/sound/soc/fsl/imx-mc13783.c
+++ b/sound/soc/fsl/imx-mc13783.c
@@ -111,22 +111,39 @@
 		return ret;
 	}
 
-	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
-		IMX_AUDMUX_V2_PTCR_SYN,
-		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
-		IMX_AUDMUX_V2_PDCR_MODE(1) |
-		IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
-	imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
-		IMX_AUDMUX_V2_PTCR_SYN |
-		IMX_AUDMUX_V2_PTCR_TFSDIR |
-		IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_TCLKDIR |
-		IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_RFSDIR |
-		IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
-		IMX_AUDMUX_V2_PTCR_RCLKDIR |
-		IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
-		IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+	if (machine_is_mx31_3ds()) {
+		imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4,
+			IMX_AUDMUX_V2_PTCR_SYN,
+			IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) |
+			IMX_AUDMUX_V2_PDCR_MODE(1) |
+			IMX_AUDMUX_V2_PDCR_INMMASK(0xfc));
+		imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0,
+			IMX_AUDMUX_V2_PTCR_SYN |
+			IMX_AUDMUX_V2_PTCR_TFSDIR |
+			IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_TCLKDIR |
+			IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_RFSDIR |
+			IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) |
+			IMX_AUDMUX_V2_PTCR_RCLKDIR |
+			IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4),
+			IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4));
+	} else if (machine_is_mx27_3ds()) {
+		imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+			IMX_AUDMUX_V1_PCR_SYN |
+			IMX_AUDMUX_V1_PCR_TFSDIR |
+			IMX_AUDMUX_V1_PCR_TCLKDIR |
+			IMX_AUDMUX_V1_PCR_RFSDIR |
+			IMX_AUDMUX_V1_PCR_RCLKDIR |
+			IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+			IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) |
+			IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4)
+		);
+		imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4,
+			IMX_AUDMUX_V1_PCR_SYN |
+			IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+		);
+	}
 
 	return ret;
 }
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index f3c0a5e..48f9d88 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/sound/soc/fsl/imx-pcm-dma.c
@@ -141,7 +141,7 @@
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_imx_pcm_hw_params,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= snd_imx_pcm_mmap,
 };
 
diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c
index 3a729ca..fb21b17 100644
--- a/sound/soc/fsl/imx-sgtl5000.c
+++ b/sound/soc/fsl/imx-sgtl5000.c
@@ -95,8 +95,7 @@
 		return ret;
 	}
 	imx_audmux_v2_configure_port(ext_port,
-			IMX_AUDMUX_V2_PTCR_SYN |
-			IMX_AUDMUX_V2_PTCR_TCSEL(int_port),
+			IMX_AUDMUX_V2_PTCR_SYN,
 			IMX_AUDMUX_V2_PDCR_RXDSEL(int_port));
 	if (ret) {
 		dev_err(&pdev->dev, "audmux external port setup failed\n");
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index 373dec9..f82d766 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/sound/soc/mxs/mxs-pcm.c
@@ -141,7 +141,7 @@
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= snd_mxs_pcm_hw_params,
 	.trigger	= snd_dmaengine_pcm_trigger,
-	.pointer	= snd_dmaengine_pcm_pointer,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
 	.mmap		= snd_mxs_pcm_mmap,
 };
 
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index 3e6e876..215113b 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -133,7 +133,7 @@
 		mxs_sgtl5000_dai[i].codec_name = NULL;
 		mxs_sgtl5000_dai[i].codec_of_node = codec_np;
 		mxs_sgtl5000_dai[i].cpu_dai_name = NULL;
-		mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i];
+		mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i];
 		mxs_sgtl5000_dai[i].platform_name = NULL;
 		mxs_sgtl5000_dai[i].platform_of_node = saif_np[i];
 	}
diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c
index 59d47ab5..2c66e249 100644
--- a/sound/soc/omap/omap-mcpdm.c
+++ b/sound/soc/omap/omap-mcpdm.c
@@ -527,6 +527,7 @@
 
 module_platform_driver(asoc_mcpdm_driver);
 
+MODULE_ALIAS("platform:omap-mcpdm");
 MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
 MODULE_DESCRIPTION("OMAP PDM SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index a0f7d3c..4d2e46f 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -8,6 +8,15 @@
 	  the PXA2xx AC97, I2S or SSP interface. You will also need
 	  to select the audio interfaces to support below.
 
+config SND_MMP_SOC
+	bool "Soc Audio for Marvell MMP chips"
+	depends on ARCH_MMP
+	select SND_SOC_DMAENGINE_PCM
+	select SND_ARM
+	help
+	  Say Y if you want to add support for codecs attached to
+	  the MMP SSPA interface.
+
 config SND_PXA2XX_AC97
 	tristate
 	select SND_AC97_CODEC
@@ -26,6 +35,9 @@
 	tristate
 	select PXA_SSP
 
+config SND_MMP_SOC_SSPA
+	tristate
+
 config SND_PXA2XX_SOC_CORGI
 	tristate "SoC Audio support for Sharp Zaurus SL-C7x0"
 	depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx
@@ -138,6 +150,26 @@
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Saarb reference platform.
 
+config SND_PXA910_SOC
+	tristate "SoC Audio for Marvell PXA910 chip"
+	depends on ARCH_MMP && SND
+	select SND_PCM
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell PXA910 reference platform.
+
+config SND_SOC_TTC_DKB
+	bool "SoC Audio support for TTC DKB"
+	depends on SND_PXA910_SOC && MACH_TTC_DKB
+	select PXA_SSP
+	select SND_PXA_SOC_SSP
+	select SND_MMP_SOC
+	select MFD_88PM860X
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on TTC DKB
+
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
@@ -194,3 +226,13 @@
        help
          Say Y if you want to add support for SoC audio on the
 	 IMote 2.
+
+config SND_MMP_SOC_BROWNSTONE
+	tristate "SoC Audio support for Marvell Brownstone"
+	depends on SND_MMP_SOC && MACH_BROWNSTONE
+	select SND_MMP_SOC_SSPA
+	select MFD_WM8994
+	select SND_SOC_WM8994
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Brownstone reference platform.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index af35762..d8a265d 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -3,11 +3,15 @@
 snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
 snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o
 snd-soc-pxa-ssp-objs := pxa-ssp.o
+snd-soc-mmp-objs := mmp-pcm.o
+snd-soc-mmp-sspa-objs := mmp-sspa.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o
 obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o
 obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o
 obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
+obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o
+obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o
 
 # PXA Machine Support
 snd-soc-corgi-objs := corgi.o
@@ -28,6 +32,8 @@
 snd-soc-z2-objs := z2.o
 snd-soc-imote2-objs := imote2.o
 snd-soc-raumfeld-objs := raumfeld.o
+snd-soc-brownstone-objs := brownstone.o
+snd-soc-ttc-dkb-objs := ttc-dkb.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -47,3 +53,5 @@
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
+obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o
+obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
new file mode 100644
index 0000000..5e666e0
--- /dev/null
+++ b/sound/soc/pxa/brownstone.c
@@ -0,0 +1,174 @@
+/*
+ * linux/sound/soc/pxa/brownstone.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "../codecs/wm8994.h"
+#include "mmp-sspa.h"
+
+static const struct snd_kcontrol_new brownstone_dapm_control[] = {
+	SOC_DAPM_PIN_SWITCH("Ext Spk"),
+};
+
+static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route brownstone_audio_map[] = {
+	{"Ext Spk", NULL, "SPKOUTLP"},
+	{"Ext Spk", NULL, "SPKOUTLN"},
+	{"Ext Spk", NULL, "SPKOUTRP"},
+	{"Ext Spk", NULL, "SPKOUTRN"},
+
+	{"Headset Stereophone", NULL, "HPOUT1L"},
+	{"Headset Stereophone", NULL, "HPOUT1R"},
+
+	{"IN1RN", NULL, "Headset Mic"},
+
+	{"DMIC1DAT", NULL, "MICBIAS1"},
+	{"MICBIAS1", NULL, "Main Mic"},
+};
+
+static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
+	snd_soc_dapm_enable_pin(dapm, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_enable_pin(dapm, "Main Mic");
+
+	/* set endpoints to not connected */
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
+	snd_soc_dapm_nc_pin(dapm, "IN1LN");
+	snd_soc_dapm_nc_pin(dapm, "IN1LP");
+	snd_soc_dapm_nc_pin(dapm, "IN1RP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RN");
+	snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
+	snd_soc_dapm_nc_pin(dapm, "IN2LN");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
+				       struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int freq_out, sspa_mclk, sysclk;
+	int sspa_div;
+
+	if (params_rate(params) > 11025) {
+		freq_out  = params_rate(params) * 512;
+		sysclk    = params_rate(params) * 256;
+		sspa_mclk = params_rate(params) * 64;
+	} else {
+		freq_out  = params_rate(params) * 1024;
+		sysclk    = params_rate(params) * 512;
+		sspa_mclk = params_rate(params) * 64;
+	}
+	sspa_div = freq_out;
+	do_div(sspa_div, sspa_mclk);
+
+	snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk);
+	snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk);
+
+	/* set wm8994 sysclk */
+	snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0);
+
+	return 0;
+}
+
+/* machine stream operations */
+static struct snd_soc_ops brownstone_ops = {
+	.hw_params = brownstone_wm8994_hw_params,
+};
+
+static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
+{
+	.name		= "WM8994",
+	.stream_name	= "WM8994 HiFi",
+	.cpu_dai_name	= "mmp-sspa-dai.0",
+	.codec_dai_name	= "wm8994-aif1",
+	.platform_name	= "mmp-pcm-audio",
+	.codec_name	= "wm8994-codec",
+	.dai_fmt	= SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+				SND_SOC_DAIFMT_CBS_CFS,
+	.ops		= &brownstone_ops,
+	.init		= brownstone_wm8994_init,
+},
+};
+
+/* audio machine driver */
+static struct snd_soc_card brownstone = {
+	.name         = "brownstone",
+	.dai_link     = brownstone_wm8994_dai,
+	.num_links    = ARRAY_SIZE(brownstone_wm8994_dai),
+
+	.controls = brownstone_dapm_control,
+	.num_controls = ARRAY_SIZE(brownstone_dapm_control),
+	.dapm_widgets = brownstone_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
+	.dapm_routes = brownstone_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+};
+
+static int __devinit brownstone_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	brownstone.dev = &pdev->dev;
+	ret = snd_soc_register_card(&brownstone);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+				ret);
+	return ret;
+}
+
+static int __devexit brownstone_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&brownstone);
+	return 0;
+}
+
+static struct platform_driver mmp_driver = {
+	.driver		= {
+		.name	= "brownstone-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= brownstone_probe,
+	.remove		= __devexit_p(brownstone_remove),
+};
+
+module_platform_driver(mmp_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC Brownstone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 9c585af..8687c1c 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -186,36 +186,27 @@
 	.num_links = ARRAY_SIZE(mioa701_dai),
 };
 
-static struct platform_device *mioa701_snd_device;
-
-static int mioa701_wm9713_probe(struct platform_device *pdev)
+static int __devinit mioa701_wm9713_probe(struct platform_device *pdev)
 {
-	int ret;
+	int rc;
 
 	if (!machine_is_mioa701())
 		return -ENODEV;
 
-	dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
-		 "lead to overheating and possible destruction of your device."
-		 "Do not use without a good knowledge of mio's board design!\n");
-
-	mioa701_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!mioa701_snd_device)
-		return -ENOMEM;
-
-	platform_set_drvdata(mioa701_snd_device, &mioa701);
-
-	ret = platform_device_add(mioa701_snd_device);
-	if (!ret)
-		return 0;
-
-	platform_device_put(mioa701_snd_device);
-	return ret;
+	mioa701.dev = &pdev->dev;
+	rc =  snd_soc_register_card(&mioa701);
+	if (!rc)
+		dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+			 "lead to overheating and possible destruction of your device."
+			 " Do not use without a good knowledge of mio's board design!\n");
+	return rc;
 }
 
 static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
 {
-	platform_device_unregister(mioa701_snd_device);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
 	return 0;
 }
 
diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c
new file mode 100644
index 0000000..73ac546
--- /dev/null
+++ b/sound/soc/pxa/mmp-pcm.c
@@ -0,0 +1,297 @@
+/*
+ * linux/sound/soc/pxa/mmp-pcm.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ */
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/platform_data/mmp_audio.h>
+#include <sound/pxa2xx-lib.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <mach/sram.h>
+#include <sound/dmaengine_pcm.h>
+
+struct mmp_dma_data {
+	int ssp_id;
+	struct resource *dma_res;
+};
+
+#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP |	\
+		SNDRV_PCM_INFO_MMAP_VALID |	\
+		SNDRV_PCM_INFO_INTERLEAVED |	\
+		SNDRV_PCM_INFO_PAUSE |		\
+		SNDRV_PCM_INFO_RESUME)
+
+#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+			 SNDRV_PCM_FMTBIT_S24_LE | \
+			 SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_pcm_hardware mmp_pcm_hardware[] = {
+	{
+		.info			= MMP_PCM_INFO,
+		.formats		= MMP_PCM_FORMATS,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+	{
+		.info			= MMP_PCM_INFO,
+		.formats		= MMP_PCM_FORMATS,
+		.period_bytes_min	= 1024,
+		.period_bytes_max	= 2048,
+		.periods_min		= 2,
+		.periods_max		= 32,
+		.buffer_bytes_max	= 4096,
+		.fifo_size		= 32,
+	},
+};
+
+static int mmp_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	struct dma_slave_config slave_config;
+	int ret;
+
+	dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	if (!dma_params)
+		return 0;
+
+	ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config);
+	if (ret)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.dst_addr     = dma_params->dev_addr;
+		slave_config.dst_maxburst = 4;
+	} else {
+		slave_config.src_addr	  = dma_params->dev_addr;
+		slave_config.src_maxburst = 4;
+	}
+
+	ret = dmaengine_slave_config(chan, &slave_config);
+	if (ret)
+		return ret;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static bool filter(struct dma_chan *chan, void *param)
+{
+	struct mmp_dma_data *dma_data = param;
+	bool found = false;
+	char *devname;
+
+	devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name,
+		dma_data->ssp_id);
+	if ((strcmp(dev_name(chan->device->dev), devname) == 0) &&
+		(chan->chan_id == dma_data->dma_res->start)) {
+		found = true;
+	}
+
+	kfree(devname);
+	return found;
+}
+
+static int mmp_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct mmp_dma_data *dma_data;
+	struct resource *r;
+	int ret;
+
+	r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream);
+	if (!r)
+		return -EBUSY;
+
+	snd_soc_set_runtime_hwparams(substream,
+				&mmp_pcm_hardware[substream->stream]);
+	dma_data = devm_kzalloc(&pdev->dev,
+			sizeof(struct mmp_dma_data), GFP_KERNEL);
+	if (dma_data == NULL)
+		return -ENOMEM;
+
+	dma_data->dma_res = r;
+	dma_data->ssp_id = cpu_dai->id;
+
+	ret = snd_dmaengine_pcm_open(substream, filter, dma_data);
+	if (ret) {
+		devm_kfree(&pdev->dev, dma_data);
+		return ret;
+	}
+
+	snd_dmaengine_pcm_set_data(substream, dma_data);
+	return 0;
+}
+
+static int mmp_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream);
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct platform_device *pdev = to_platform_device(rtd->platform->dev);
+
+	snd_dmaengine_pcm_close(substream);
+	devm_kfree(&pdev->dev, dma_data);
+	return 0;
+}
+
+static int mmp_pcm_mmap(struct snd_pcm_substream *substream,
+			 struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long off = vma->vm_pgoff;
+
+	vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+	return remap_pfn_range(vma, vma->vm_start,
+		__phys_to_pfn(runtime->dma_addr) + off,
+		vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+
+struct snd_pcm_ops mmp_pcm_ops = {
+	.open		= mmp_pcm_open,
+	.close		= mmp_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= mmp_pcm_hw_params,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= mmp_pcm_mmap,
+};
+
+static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+	struct gen_pool *gpool;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return;
+
+	for (stream = 0; stream < 2; stream++) {
+		size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+		gen_pool_free(gpool, (unsigned long)buf->area, size);
+		buf->area = NULL;
+	}
+
+	return;
+}
+
+static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream,
+								int stream)
+{
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = mmp_pcm_hardware[stream].buffer_bytes_max;
+	struct gen_pool *gpool;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = substream->pcm->card->dev;
+	buf->private_data = NULL;
+
+	gpool = sram_get_gpool("asram");
+	if (!gpool)
+		return -ENOMEM;
+
+	buf->area = (unsigned char *)gen_pool_alloc(gpool, size);
+	if (!buf->area)
+		return -ENOMEM;
+	buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area);
+	buf->bytes = size;
+	return 0;
+}
+
+int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret = 0, stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+
+		ret = mmp_pcm_preallocate_dma_buffer(substream,	stream);
+		if (ret)
+			goto err;
+	}
+
+	return 0;
+
+err:
+	mmp_pcm_free_dma_buffers(pcm);
+	return ret;
+}
+
+struct snd_soc_platform_driver mmp_soc_platform = {
+	.ops		= &mmp_pcm_ops,
+	.pcm_new	= mmp_pcm_new,
+	.pcm_free	= mmp_pcm_free_dma_buffers,
+};
+
+static __devinit int mmp_pcm_probe(struct platform_device *pdev)
+{
+	struct mmp_audio_platdata *pdata = pdev->dev.platform_data;
+
+	if (pdata) {
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max =
+						pdata->buffer_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max =
+						pdata->period_max_playback;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max =
+						pdata->buffer_max_capture;
+		mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max =
+						pdata->period_max_capture;
+	}
+	return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform);
+}
+
+static int __devexit mmp_pcm_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver mmp_pcm_driver = {
+	.driver = {
+		.name = "mmp-pcm-audio",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = mmp_pcm_probe,
+	.remove = __devexit_p(mmp_pcm_remove),
+};
+
+module_platform_driver(mmp_pcm_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP Soc Audio DMA module");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c
new file mode 100644
index 0000000..4d6cb8a
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.c
@@ -0,0 +1,480 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.c
+ * Base on pxa2xx-ssp.c
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/slab.h>
+#include <linux/pxa2xx_ssp.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/pxa2xx-lib.h>
+#include "mmp-sspa.h"
+
+/*
+ * SSPA audio private data
+ */
+struct sspa_priv {
+	struct ssp_device *sspa;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	struct clk *audio_clk;
+	struct clk *sysclk;
+	int dai_fmt;
+	int running_cnt;
+};
+
+static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val)
+{
+	__raw_writel(val, sspa->mmio_base + reg);
+}
+
+static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg)
+{
+	return __raw_readl(sspa->mmio_base + reg);
+}
+
+static void mmp_sspa_tx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_tx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_enable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp |= SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static void mmp_sspa_rx_disable(struct ssp_device *sspa)
+{
+	unsigned int sspa_sp;
+
+	sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP);
+	sspa_sp &= ~SSPA_SP_S_EN;
+	sspa_sp |= SSPA_SP_WEN;
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+}
+
+static int mmp_sspa_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_enable(priv->sysclk);
+	clk_enable(priv->sspa->clk);
+
+	return 0;
+}
+
+static void mmp_sspa_shutdown(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	clk_disable(priv->sspa->clk);
+	clk_disable(priv->sysclk);
+
+	return;
+}
+
+/*
+ * Set the SSP ports SYSCLK.
+ */
+static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+				    int clk_id, unsigned int freq, int dir)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (clk_id) {
+	case MMP_SSPA_CLK_AUDIO:
+		ret = clk_set_rate(priv->audio_clk, freq);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK_PLL:
+	case MMP_SSPA_CLK_VCXO:
+		/* not support yet */
+		return -EINVAL;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id,
+				 int source, unsigned int freq_in,
+				 unsigned int freq_out)
+{
+	struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (pll_id) {
+	case MMP_SYSCLK:
+		ret = clk_set_rate(priv->sysclk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	case MMP_SSPA_CLK:
+		ret = clk_set_rate(priv->sspa->clk, freq_out);
+		if (ret)
+			return ret;
+		break;
+	default:
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+/*
+ * Set up the sspa dai format. The sspa port must be inactive
+ * before calling this function as the physical
+ * interface format is changed.
+ */
+static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				 unsigned int fmt)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	u32 sspa_sp, sspa_ctrl;
+
+	/* check if we need to change anything at all */
+	if (sspa_priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) ||
+	    (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) {
+		dev_err(&sspa->pdev->dev,
+			"can't change hardware dai format: stream is in use\n");
+		return -EINVAL;
+	}
+
+	/* reset port settings */
+	sspa_sp   = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH;
+	sspa_ctrl = 0;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		sspa_sp |= SSPA_SP_MSL;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		sspa_sp |= SSPA_SP_FSP;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		sspa_sp |= SSPA_TXSP_FPER(63);
+		sspa_sp |= SSPA_SP_FWID(31);
+		sspa_ctrl |= SSPA_CTL_XDATDLY(1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH);
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+	mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp);
+
+	/*
+	 * FIXME: hw issue, for the tx serial port,
+	 * can not config the master/slave mode;
+	 * so must clean this bit.
+	 * The master/slave mode has been set in the
+	 * rx port.
+	 */
+	sspa_sp &= ~SSPA_SP_MSL;
+	mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp);
+
+	mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+	mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+
+	/* Since we are configuring the timings for the format by hand
+	 * we have to defer some things until hw_params() where we
+	 * know parameters like the sample size.
+	 */
+	sspa_priv->dai_fmt = fmt;
+	return 0;
+}
+
+/*
+ * Set the SSPA audio DMA parameters and sample size.
+ * Can be called multiple times by oss emulation.
+ */
+static int mmp_sspa_hw_params(struct snd_pcm_substream *substream,
+			       struct snd_pcm_hw_params *params,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	struct pxa2xx_pcm_dma_params *dma_params;
+	u32 sspa_ctrl;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL);
+	else
+		sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL);
+
+	sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1);
+	sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK;
+	sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS);
+	sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1);
+	} else {
+		mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl);
+		mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0);
+	}
+
+	dma_params = &sspa_priv->dma_params[substream->stream];
+	dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ?
+				(sspa->phys_base + SSPA_TXD) :
+				(sspa->phys_base + SSPA_RXD);
+	snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params);
+	return 0;
+}
+
+static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd,
+			     struct snd_soc_dai *dai)
+{
+	struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai);
+	struct ssp_device *sspa = sspa_priv->sspa;
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		/*
+		 * whatever playback or capture, must enable rx.
+		 * this is a hw issue, so need check if rx has been
+		 * enabled or not; if has been enabled by another
+		 * stream, do not enable again.
+		 */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_enable(sspa);
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_enable(sspa);
+
+		sspa_priv->running_cnt++;
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		sspa_priv->running_cnt--;
+
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			mmp_sspa_tx_disable(sspa);
+
+		/* have no capture stream, disable rx port */
+		if (!sspa_priv->running_cnt)
+			mmp_sspa_rx_disable(sspa);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int mmp_sspa_probe(struct snd_soc_dai *dai)
+{
+	struct sspa_priv *priv = dev_get_drvdata(dai->dev);
+
+	snd_soc_dai_set_drvdata(dai, priv);
+	return 0;
+
+}
+
+#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000
+#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
+		SNDRV_PCM_FMTBIT_S16_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S24_LE | \
+		SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops mmp_sspa_dai_ops = {
+	.startup	= mmp_sspa_startup,
+	.shutdown	= mmp_sspa_shutdown,
+	.trigger	= mmp_sspa_trigger,
+	.hw_params	= mmp_sspa_hw_params,
+	.set_sysclk	= mmp_sspa_set_dai_sysclk,
+	.set_pll	= mmp_sspa_set_dai_pll,
+	.set_fmt	= mmp_sspa_set_dai_fmt,
+};
+
+struct snd_soc_dai_driver mmp_sspa_dai = {
+	.probe = mmp_sspa_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 128,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = MMP_SSPA_RATES,
+		.formats = MMP_SSPA_FORMATS,
+	},
+	.ops = &mmp_sspa_dai_ops,
+};
+
+static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev)
+{
+	struct sspa_priv *priv;
+	struct resource *res;
+
+	priv = devm_kzalloc(&pdev->dev,
+				sizeof(struct sspa_priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->sspa = devm_kzalloc(&pdev->dev,
+				sizeof(struct ssp_device), GFP_KERNEL);
+	if (priv->sspa == NULL)
+		return -ENOMEM;
+
+	priv->dma_params = devm_kzalloc(&pdev->dev,
+			2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL);
+	if (priv->dma_params == NULL)
+		return -ENOMEM;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (res == NULL)
+		return -ENOMEM;
+
+	priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res);
+	if (priv->sspa->mmio_base == NULL)
+		return -ENODEV;
+
+	priv->sspa->clk = devm_clk_get(&pdev->dev, NULL);
+	if (IS_ERR(priv->sspa->clk))
+		return PTR_ERR(priv->sspa->clk);
+
+	priv->audio_clk = clk_get(NULL, "mmp-audio");
+	if (IS_ERR(priv->audio_clk))
+		return PTR_ERR(priv->audio_clk);
+
+	priv->sysclk = clk_get(NULL, "mmp-sysclk");
+	if (IS_ERR(priv->sysclk)) {
+		clk_put(priv->audio_clk);
+		return PTR_ERR(priv->sysclk);
+	}
+	clk_enable(priv->audio_clk);
+	priv->dai_fmt = (unsigned int) -1;
+	platform_set_drvdata(pdev, priv);
+
+	return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai);
+}
+
+static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev)
+{
+	struct sspa_priv *priv = platform_get_drvdata(pdev);
+
+	clk_disable(priv->audio_clk);
+	clk_put(priv->audio_clk);
+	clk_put(priv->sysclk);
+	snd_soc_unregister_dai(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver asoc_mmp_sspa_driver = {
+	.driver = {
+		.name = "mmp-sspa-dai",
+		.owner = THIS_MODULE,
+	},
+	.probe = asoc_mmp_sspa_probe,
+	.remove = __devexit_p(asoc_mmp_sspa_remove),
+};
+
+module_platform_driver(asoc_mmp_sspa_driver);
+
+MODULE_AUTHOR("Leo Yan <leoy@marvell.com>");
+MODULE_DESCRIPTION("MMP SSPA SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h
new file mode 100644
index 0000000..ea365cb
--- /dev/null
+++ b/sound/soc/pxa/mmp-sspa.h
@@ -0,0 +1,92 @@
+/*
+ * linux/sound/soc/pxa/mmp-sspa.h
+ *
+ * Copyright (C) 2011 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#ifndef _MMP_SSPA_H
+#define _MMP_SSPA_H
+
+/*
+ * SSPA Registers
+ */
+#define SSPA_RXD		(0x00)
+#define SSPA_RXID		(0x04)
+#define SSPA_RXCTL		(0x08)
+#define SSPA_RXSP		(0x0c)
+#define SSPA_RXFIFO_UL		(0x10)
+#define SSPA_RXINT_MASK		(0x14)
+#define SSPA_RXC		(0x18)
+#define SSPA_RXFIFO_NOFS	(0x1c)
+#define SSPA_RXFIFO_SIZE	(0x20)
+
+#define SSPA_TXD		(0x80)
+#define SSPA_TXID		(0x84)
+#define SSPA_TXCTL		(0x88)
+#define SSPA_TXSP		(0x8c)
+#define SSPA_TXFIFO_LL		(0x90)
+#define SSPA_TXINT_MASK		(0x94)
+#define SSPA_TXC		(0x98)
+#define SSPA_TXFIFO_NOFS	(0x9c)
+#define SSPA_TXFIFO_SIZE	(0xa0)
+
+/* SSPA Control Register */
+#define	SSPA_CTL_XPH		(1 << 31)	/* Read Phase */
+#define	SSPA_CTL_XFIG		(1 << 15)	/* Transmit Zeros when FIFO Empty */
+#define	SSPA_CTL_JST		(1 << 3)	/* Audio Sample Justification */
+#define	SSPA_CTL_XFRLEN2_MASK	(7 << 24)
+#define	SSPA_CTL_XFRLEN2(x)	((x) << 24)	/* Transmit Frame Length in Phase 2 */
+#define	SSPA_CTL_XWDLEN2_MASK	(7 << 21)
+#define	SSPA_CTL_XWDLEN2(x)	((x) << 21)	/* Transmit Word Length in Phase 2 */
+#define	SSPA_CTL_XDATDLY(x)	((x) << 19)	/* Tansmit Data Delay */
+#define	SSPA_CTL_XSSZ2_MASK	(7 << 16)
+#define	SSPA_CTL_XSSZ2(x)	((x) << 16)	/* Transmit Sample Audio Size */
+#define	SSPA_CTL_XFRLEN1_MASK	(7 << 8)
+#define	SSPA_CTL_XFRLEN1(x)	((x) << 8)	/* Transmit Frame Length in Phase 1 */
+#define	SSPA_CTL_XWDLEN1_MASK	(7 << 5)
+#define	SSPA_CTL_XWDLEN1(x)	((x) << 5)	/* Transmit Word Length in Phase 1 */
+#define	SSPA_CTL_XSSZ1_MASK	(7 << 0)
+#define	SSPA_CTL_XSSZ1(x)	((x) << 0)	/* XSSZ1 */
+
+#define SSPA_CTL_8_BITS		(0x0)		/* Sample Size */
+#define SSPA_CTL_12_BITS	(0x1)
+#define SSPA_CTL_16_BITS	(0x2)
+#define SSPA_CTL_20_BITS	(0x3)
+#define SSPA_CTL_24_BITS	(0x4)
+#define SSPA_CTL_32_BITS	(0x5)
+
+/* SSPA Serial Port Register */
+#define	SSPA_SP_WEN		(1 << 31)	/* Write Configuration Enable */
+#define	SSPA_SP_MSL		(1 << 18)	/* Master Slave Configuration */
+#define	SSPA_SP_CLKP		(1 << 17)	/* CLKP Polarity Clock Edge Select */
+#define	SSPA_SP_FSP		(1 << 16)	/* FSP Polarity Clock Edge Select */
+#define	SSPA_SP_FFLUSH		(1 << 2)	/* FIFO Flush */
+#define	SSPA_SP_S_RST		(1 << 1)	/* Active High Reset Signal */
+#define	SSPA_SP_S_EN		(1 << 0)	/* Serial Clock Domain Enable */
+#define	SSPA_SP_FWID(x)		((x) << 20)	/* Frame-Sync Width */
+#define	SSPA_TXSP_FPER(x)	((x) << 4)	/* Frame-Sync Active */
+
+/* sspa clock sources */
+#define MMP_SSPA_CLK_PLL	0
+#define MMP_SSPA_CLK_VCXO	1
+#define MMP_SSPA_CLK_AUDIO	3
+
+/* sspa pll id */
+#define MMP_SYSCLK		0
+#define MMP_SSPA_CLK		1
+
+#endif /* _MMP_SSPA_H */
diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c
new file mode 100644
index 0000000..935491a
--- /dev/null
+++ b/sound/soc/pxa/ttc-dkb.c
@@ -0,0 +1,173 @@
+/*
+ * linux/sound/soc/pxa/ttc_dkb.c
+ *
+ * Copyright (C) 2012 Marvell International Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ */
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <asm/mach-types.h>
+#include <sound/pcm_params.h>
+#include "../codecs/88pm860x-codec.h"
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* ttc machine dapm widgets */
+static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* ttc machine audio map */
+static const struct snd_soc_dapm_route ttc_audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(dapm, "Ext Speaker");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(dapm, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(dapm, "Headset Stereophone");
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+
+	return 0;
+}
+
+/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = {
+{
+	 .name = "88pm860x i2s",
+	 .stream_name = "audio playback",
+	 .codec_name = "88pm860x-codec",
+	 .platform_name = "mmp-pcm-audio",
+	 .cpu_dai_name = "pxa-ssp-dai.1",
+	 .codec_dai_name = "88pm860x-i2s",
+	 .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM,
+	 .init = ttc_pm860x_init,
+},
+};
+
+/* ttc/td audio machine driver */
+static struct snd_soc_card ttc_dkb_card = {
+	.name = "ttc-dkb-hifi",
+	.dai_link = ttc_pm860x_hifi_dai,
+	.num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai),
+
+	.dapm_widgets = ttc_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets),
+	.dapm_routes = ttc_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ttc_audio_map),
+};
+
+static int __devinit ttc_dkb_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ttc_dkb_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit ttc_dkb_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static struct platform_driver ttc_dkb_driver = {
+	.driver		= {
+		.name	= "ttc-dkb-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= ttc_dkb_probe,
+	.remove		= __devexit_p(ttc_dkb_remove),
+};
+
+module_platform_driver(ttc_dkb_driver);
+
+/* Module information */
+MODULE_AUTHOR("Qiao Zhou, <zhouqiao@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC TTC DKB");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:ttc-dkb-audio");
diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c
index c82c646..ee52c8a0 100644
--- a/sound/soc/samsung/littlemill.c
+++ b/sound/soc/samsung/littlemill.c
@@ -211,6 +211,11 @@
 	return 0;
 }
 
+static const struct snd_kcontrol_new controls[] = {
+	SOC_DAPM_PIN_SWITCH("WM1250 Input"),
+	SOC_DAPM_PIN_SWITCH("WM1250 Output"),
+};
+
 static struct snd_soc_dapm_widget widgets[] = {
 	SND_SOC_DAPM_HP("Headphone", NULL),
 
@@ -282,6 +287,8 @@
 	.set_bias_level = littlemill_set_bias_level,
 	.set_bias_level_post = littlemill_set_bias_level_post,
 
+	.controls = controls,
+	.num_controls = ARRAY_SIZE(controls),
 	.dapm_widgets = widgets,
 	.num_dapm_widgets = ARRAY_SIZE(widgets),
 	.dapm_routes = audio_paths,
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 79fbeea..ac7701b 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -25,7 +25,6 @@
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
-#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "dma.h"
@@ -83,12 +82,9 @@
 
 	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
 
-	/* Configure the I2S pins in correct mode */
-	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
-	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
+	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+			      S3C_GPIO_PULL_NONE);
 
 	return 0;
 }
diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c
index c4aa4d4..0aae3a3 100644
--- a/sound/soc/samsung/s3c24xx-i2s.c
+++ b/sound/soc/samsung/s3c24xx-i2s.c
@@ -23,7 +23,6 @@
 #include <sound/soc.h>
 #include <sound/pcm_params.h>
 
-#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 #include <plat/regs-iis.h>
 
@@ -391,12 +390,9 @@
 	}
 	clk_enable(s3c24xx_i2s.iis_clk);
 
-	/* Configure the I2S pins in correct mode */
-	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK);
-	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
-	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
+	/* Configure the I2S pins (GPE0...GPE4) in correct mode */
+	s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2),
+			      S3C_GPIO_PULL_NONE);
 
 	writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON);
 
diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c
index 8eb309f..48dd4dd 100644
--- a/sound/soc/samsung/smdk_wm8994.c
+++ b/sound/soc/samsung/smdk_wm8994.c
@@ -149,31 +149,41 @@
 	.num_links = ARRAY_SIZE(smdk_dai),
 };
 
-static struct platform_device *smdk_snd_device;
 
-static int __init smdk_audio_init(void)
+static int __devinit smdk_audio_probe(struct platform_device *pdev)
 {
 	int ret;
+	struct snd_soc_card *card = &smdk;
 
-	smdk_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!smdk_snd_device)
-		return -ENOMEM;
+	card->dev = &pdev->dev;
+	ret = snd_soc_register_card(card);
 
-	platform_set_drvdata(smdk_snd_device, &smdk);
-
-	ret = platform_device_add(smdk_snd_device);
 	if (ret)
-		platform_device_put(smdk_snd_device);
+		dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret);
 
 	return ret;
 }
-module_init(smdk_audio_init);
 
-static void __exit smdk_audio_exit(void)
+static int __devexit smdk_audio_remove(struct platform_device *pdev)
 {
-	platform_device_unregister(smdk_snd_device);
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
 }
-module_exit(smdk_audio_exit);
+
+static struct platform_driver smdk_audio_driver = {
+	.driver		= {
+		.name	= "smdk-audio",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= smdk_audio_probe,
+	.remove		= __devexit_p(smdk_audio_remove),
+};
+
+module_platform_driver(smdk_audio_driver);
 
 MODULE_DESCRIPTION("ALSA SoC SMDK WM8994");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:smdk-audio");
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 2ef9853..53486ff 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -247,7 +247,7 @@
 struct fsi_stream_handler {
 	int (*init)(struct fsi_priv *fsi, struct fsi_stream *io);
 	int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io);
-	int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io);
+	int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev);
 	int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io);
 	int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io);
 	void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io,
@@ -571,16 +571,16 @@
 #define fsi_stream_stop(fsi, io)\
 	fsi_stream_handler_call(io, start_stop, fsi, io, 0)
 
-static int fsi_stream_probe(struct fsi_priv *fsi)
+static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev)
 {
 	struct fsi_stream *io;
 	int ret1, ret2;
 
 	io = &fsi->playback;
-	ret1 = fsi_stream_handler_call(io, probe, fsi, io);
+	ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev);
 
 	io = &fsi->capture;
-	ret2 = fsi_stream_handler_call(io, probe, fsi, io);
+	ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev);
 
 	if (ret1 < 0)
 		return ret1;
@@ -1089,13 +1089,10 @@
 {
 	struct fsi_stream *io = (struct fsi_stream *)data;
 	struct fsi_priv *fsi = fsi_stream_to_priv(io);
-	struct dma_chan *chan;
 	struct snd_soc_dai *dai;
 	struct dma_async_tx_descriptor *desc;
-	struct scatterlist sg;
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
-	dma_cookie_t cookie;
 	int is_play = fsi_stream_is_play(fsi, io);
 	int len;
 	dma_addr_t buf;
@@ -1104,7 +1101,6 @@
 		return;
 
 	dai	= fsi_get_dai(io->substream);
-	chan	= io->chan;
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
@@ -1112,14 +1108,8 @@
 
 	dma_sync_single_for_device(dai->dev, buf, len, dir);
 
-	sg_init_table(&sg, 1);
-	sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)),
-		    len , offset_in_page(buf));
-	sg_dma_address(&sg) = buf;
-	sg_dma_len(&sg) = len;
-
-	desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir,
-				       DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
 	if (!desc) {
 		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
 		return;
@@ -1128,13 +1118,12 @@
 	desc->callback		= fsi_dma_complete;
 	desc->callback_param	= io;
 
-	cookie = desc->tx_submit(desc);
-	if (cookie < 0) {
+	if (dmaengine_submit(desc) < 0) {
 		dev_err(dai->dev, "tx_submit() fail\n");
 		return;
 	}
 
-	dma_async_issue_pending(chan);
+	dma_async_issue_pending(io->chan);
 
 	/*
 	 * FIXME
@@ -1184,7 +1173,7 @@
 		fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0);
 }
 
-static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io)
+static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev)
 {
 	dma_cap_mask_t mask;
 
@@ -1192,8 +1181,19 @@
 	dma_cap_set(DMA_SLAVE, mask);
 
 	io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave);
-	if (!io->chan)
-		return -EIO;
+	if (!io->chan) {
+
+		/* switch to PIO handler */
+		if (fsi_stream_is_play(fsi, io))
+			fsi->playback.handler	= &fsi_pio_push_handler;
+		else
+			fsi->capture.handler	= &fsi_pio_pop_handler;
+
+		dev_info(dev, "switch handler (dma => pio)\n");
+
+		/* probe again */
+		return fsi_stream_probe(fsi, dev);
+	}
 
 	tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io);
 
@@ -1683,7 +1683,7 @@
 	master->fsia.master	= master;
 	master->fsia.info	= &info->port_a;
 	fsi_handler_init(&master->fsia);
-	ret = fsi_stream_probe(&master->fsia);
+	ret = fsi_stream_probe(&master->fsia, &pdev->dev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "FSIA stream probe failed\n");
 		goto exit_iounmap;
@@ -1694,7 +1694,7 @@
 	master->fsib.master	= master;
 	master->fsib.info	= &info->port_b;
 	fsi_handler_init(&master->fsib);
-	ret = fsi_stream_probe(&master->fsib);
+	ret = fsi_stream_probe(&master->fsib, &pdev->dev);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "FSIB stream probe failed\n");
 		goto exit_fsia;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index b37ee80..f219b2f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -812,13 +812,15 @@
 
 	/* Find CPU DAI from registered DAIs*/
 	list_for_each_entry(cpu_dai, &dai_list, list) {
-		if (dai_link->cpu_dai_of_node) {
-			if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node)
-				continue;
-		} else {
-			if (strcmp(cpu_dai->name, dai_link->cpu_dai_name))
-				continue;
-		}
+		if (dai_link->cpu_of_node &&
+		    (cpu_dai->dev->of_node != dai_link->cpu_of_node))
+			continue;
+		if (dai_link->cpu_name &&
+		    strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name))
+			continue;
+		if (dai_link->cpu_dai_name &&
+		    strcmp(cpu_dai->name, dai_link->cpu_dai_name))
+			continue;
 
 		rtd->cpu_dai = cpu_dai;
 	}
@@ -896,6 +898,28 @@
 	return 0;
 }
 
+static int soc_remove_platform(struct snd_soc_platform *platform)
+{
+	int ret;
+
+	if (platform->driver->remove) {
+		ret = platform->driver->remove(platform);
+		if (ret < 0)
+			pr_err("asoc: failed to remove %s: %d\n",
+				platform->name, ret);
+	}
+
+	/* Make sure all DAPM widgets are freed */
+	snd_soc_dapm_free(&platform->dapm);
+
+	soc_cleanup_platform_debugfs(platform);
+	platform->probed = 0;
+	list_del(&platform->card_list);
+	module_put(platform->dev->driver->owner);
+
+	return 0;
+}
+
 static void soc_remove_codec(struct snd_soc_codec *codec)
 {
 	int err;
@@ -917,11 +941,9 @@
 	module_put(codec->dev->driver->owner);
 }
 
-static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order)
+static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai;
 	int err;
 
@@ -946,30 +968,6 @@
 		list_del(&codec_dai->card_list);
 	}
 
-	/* remove the platform */
-	if (platform && platform->probed &&
-			platform->driver->remove_order == order) {
-		if (platform->driver->remove) {
-			err = platform->driver->remove(platform);
-			if (err < 0)
-				pr_err("asoc: failed to remove %s: %d\n",
-							platform->name, err);
-		}
-
-		/* Make sure all DAPM widgets are freed */
-		snd_soc_dapm_free(&platform->dapm);
-
-		soc_cleanup_platform_debugfs(platform);
-		platform->probed = 0;
-		list_del(&platform->card_list);
-		module_put(platform->dev->driver->owner);
-	}
-
-	/* remove the CODEC */
-	if (codec && codec->probed &&
-			codec->driver->remove_order == order)
-		soc_remove_codec(codec);
-
 	/* remove the cpu_dai */
 	if (cpu_dai && cpu_dai->probed &&
 			cpu_dai->driver->remove_order == order) {
@@ -981,7 +979,43 @@
 		}
 		cpu_dai->probed = 0;
 		list_del(&cpu_dai->card_list);
-		module_put(cpu_dai->dev->driver->owner);
+
+		if (!cpu_dai->codec) {
+			snd_soc_dapm_free(&cpu_dai->dapm);
+			module_put(cpu_dai->dev->driver->owner);
+		}
+	}
+}
+
+static void soc_remove_link_components(struct snd_soc_card *card, int num,
+				       int order)
+{
+	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct snd_soc_codec *codec;
+
+	/* remove the platform */
+	if (platform && platform->probed &&
+	    platform->driver->remove_order == order) {
+		soc_remove_platform(platform);
+	}
+
+	/* remove the CODEC-side CODEC */
+	if (codec_dai) {
+		codec = codec_dai->codec;
+		if (codec && codec->probed &&
+		    codec->driver->remove_order == order)
+			soc_remove_codec(codec);
+	}
+
+	/* remove any CPU-side CODEC */
+	if (cpu_dai) {
+		codec = cpu_dai->codec;
+		if (codec && codec->probed &&
+		    codec->driver->remove_order == order)
+			soc_remove_codec(codec);
 	}
 }
 
@@ -992,8 +1026,15 @@
 	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
 			order++) {
 		for (dai = 0; dai < card->num_rtd; dai++)
-			soc_remove_dai_link(card, dai, order);
+			soc_remove_link_dais(card, dai, order);
 	}
+
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (dai = 0; dai < card->num_rtd; dai++)
+			soc_remove_link_components(card, dai, order);
+	}
+
 	card->num_rtd = 0;
 }
 
@@ -1054,6 +1095,10 @@
 		}
 	}
 
+	/* If the driver didn't set I/O up try regmap */
+	if (!codec->control_data)
+		snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP);
+
 	if (driver->controls)
 		snd_soc_add_codec_controls(codec, driver->controls,
 				     driver->num_controls);
@@ -1230,7 +1275,44 @@
 	return 0;
 }
 
-static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
+static int soc_probe_link_components(struct snd_soc_card *card, int num,
+				     int order)
+{
+	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_platform *platform = rtd->platform;
+	int ret;
+
+	/* probe the CPU-side component, if it is a CODEC */
+	if (cpu_dai->codec &&
+	    !cpu_dai->codec->probed &&
+	    cpu_dai->codec->driver->probe_order == order) {
+		ret = soc_probe_codec(card, cpu_dai->codec);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* probe the CODEC-side component */
+	if (!codec_dai->codec->probed &&
+	    codec_dai->codec->driver->probe_order == order) {
+		ret = soc_probe_codec(card, codec_dai->codec);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* probe the platform */
+	if (!platform->probed &&
+	    platform->driver->probe_order == order) {
+		ret = soc_probe_platform(card, platform);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order)
 {
 	struct snd_soc_dai_link *dai_link = &card->dai_link[num];
 	struct snd_soc_pcm_runtime *rtd = &card->rtd[num];
@@ -1255,11 +1337,14 @@
 	/* probe the cpu_dai */
 	if (!cpu_dai->probed &&
 			cpu_dai->driver->probe_order == order) {
-		cpu_dai->dapm.card = card;
-		if (!try_module_get(cpu_dai->dev->driver->owner))
-			return -ENODEV;
+		if (!cpu_dai->codec) {
+			cpu_dai->dapm.card = card;
+			if (!try_module_get(cpu_dai->dev->driver->owner))
+				return -ENODEV;
 
-		snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+			list_add(&cpu_dai->dapm.list, &card->dapm_list);
+			snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai);
+		}
 
 		if (cpu_dai->driver->probe) {
 			ret = cpu_dai->driver->probe(cpu_dai);
@@ -1275,22 +1360,6 @@
 		list_add(&cpu_dai->card_list, &card->dai_dev_list);
 	}
 
-	/* probe the CODEC */
-	if (!codec->probed &&
-			codec->driver->probe_order == order) {
-		ret = soc_probe_codec(card, codec);
-		if (ret < 0)
-			return ret;
-	}
-
-	/* probe the platform */
-	if (!platform->probed &&
-			platform->driver->probe_order == order) {
-		ret = soc_probe_platform(card, platform);
-		if (ret < 0)
-			return ret;
-	}
-
 	/* probe the CODEC DAI */
 	if (!codec_dai->probed && codec_dai->driver->probe_order == order) {
 		if (codec_dai->driver->probe) {
@@ -1565,14 +1634,27 @@
 			goto card_probe_error;
 	}
 
-	/* early DAI link probe */
+	/* probe all components used by DAI links on this card */
 	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
 			order++) {
 		for (i = 0; i < card->num_links; i++) {
-			ret = soc_probe_dai_link(card, i, order);
+			ret = soc_probe_link_components(card, i, order);
 			if (ret < 0) {
 				pr_err("asoc: failed to instantiate card %s: %d\n",
-			       card->name, ret);
+				       card->name, ret);
+				goto probe_dai_err;
+			}
+		}
+	}
+
+	/* probe all DAI links on this card */
+	for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST;
+			order++) {
+		for (i = 0; i < card->num_links; i++) {
+			ret = soc_probe_link_dais(card, i, order);
+			if (ret < 0) {
+				pr_err("asoc: failed to instantiate card %s: %d\n",
+				       card->name, ret);
 				goto probe_dai_err;
 			}
 		}
@@ -2790,6 +2872,104 @@
 EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 
 /**
+ * snd_soc_info_volsw_range - single mixer info callback with range.
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information, within a range, about a single
+ * mixer control.
+ *
+ * returns 0 for success.
+ */
+int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int platform_max;
+	int min = mc->min;
+
+	if (!mc->platform_max)
+		mc->platform_max = mc->max;
+	platform_max = mc->platform_max;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = platform_max - min;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range);
+
+/**
+ * snd_soc_put_volsw_range - single mixer put value callback with range.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to set the value, within a range, for a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int min = mc->min;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+	unsigned int val, val_mask;
+
+	val = ((ucontrol->value.integer.value[0] + min) & mask);
+	if (invert)
+		val = max - val;
+	val_mask = mask << shift;
+	val = val << shift;
+
+	return snd_soc_update_bits_locked(codec, reg, val_mask, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range);
+
+/**
+ * snd_soc_get_volsw_range - single mixer get callback with range
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * Callback to get the value, within a range, of a single mixer control.
+ *
+ * Returns 0 for success.
+ */
+int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int shift = mc->shift;
+	int min = mc->min;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+
+	ucontrol->value.integer.value[0] =
+		(snd_soc_read(codec, reg) >> shift) & mask;
+	if (invert)
+		ucontrol->value.integer.value[0] =
+			max - ucontrol->value.integer.value[0];
+	ucontrol->value.integer.value[0] =
+		ucontrol->value.integer.value[0] - min;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range);
+
+/**
  * snd_soc_limit_volume - Set new limit to an existing volume control.
  *
  * @codec: where to look for the control
@@ -3346,6 +3526,12 @@
 				link->name);
 			return -EINVAL;
 		}
+		/* Codec DAI name must be specified */
+		if (!link->codec_dai_name) {
+			dev_err(card->dev, "codec_dai_name not set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
 
 		/*
 		 * Platform may be specified by either name or OF node, but
@@ -3358,12 +3544,24 @@
 		}
 
 		/*
-		 * CPU DAI must be specified by 1 of name or OF node,
-		 * not both or neither.
+		 * CPU device may be specified by either name or OF node, but
+		 * can be left unspecified, and will be matched based on DAI
+		 * name alone..
 		 */
-		if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) {
+		if (link->cpu_name && link->cpu_of_node) {
 			dev_err(card->dev,
-				"Neither/both cpu_dai name/of_node are set for %s\n",
+				"Neither/both cpu name/of_node are set for %s\n",
+				link->name);
+			return -EINVAL;
+		}
+		/*
+		 * At least one of CPU DAI name or CPU device name/node must be
+		 * specified
+		 */
+		if (!link->cpu_dai_name &&
+		    !(link->cpu_name || link->cpu_of_node)) {
+			dev_err(card->dev,
+				"Neither cpu_dai_name nor cpu_name/of_node are set for %s\n",
 				link->name);
 			return -EINVAL;
 		}
@@ -3938,6 +4136,7 @@
 			dev_err(card->dev,
 				"Property '%s' index %d could not be read: %d\n",
 				propname, 2 * i, ret);
+			kfree(routes);
 			return -EINVAL;
 		}
 		ret = of_property_read_string_index(np, propname,
@@ -3946,6 +4145,7 @@
 			dev_err(card->dev,
 				"Property '%s' index %d could not be read: %d\n",
 				propname, (2 * i) + 1, ret);
+			kfree(routes);
 			return -EINVAL;
 		}
 	}
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 89eae93..4d181df9 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -35,6 +35,7 @@
 #include <linux/debugfs.h>
 #include <linux/pm_runtime.h>
 #include <linux/regulator/consumer.h>
+#include <linux/clk.h>
 #include <linux/slab.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -51,6 +52,7 @@
 	[snd_soc_dapm_pre] = 0,
 	[snd_soc_dapm_supply] = 1,
 	[snd_soc_dapm_regulator_supply] = 1,
+	[snd_soc_dapm_clock_supply] = 1,
 	[snd_soc_dapm_micbias] = 2,
 	[snd_soc_dapm_dai_link] = 2,
 	[snd_soc_dapm_dai] = 3,
@@ -92,6 +94,7 @@
 	[snd_soc_dapm_aif_out] = 10,
 	[snd_soc_dapm_dai] = 10,
 	[snd_soc_dapm_dai_link] = 11,
+	[snd_soc_dapm_clock_supply] = 12,
 	[snd_soc_dapm_regulator_supply] = 12,
 	[snd_soc_dapm_supply] = 12,
 	[snd_soc_dapm_post] = 13,
@@ -288,9 +291,9 @@
 		if (dapm->codec->driver->set_bias_level)
 			ret = dapm->codec->driver->set_bias_level(dapm->codec,
 								  level);
-		else
-			dapm->bias_level = level;
-	}
+	} else
+		dapm->bias_level = level;
+
 	if (ret != 0)
 		goto out;
 
@@ -321,11 +324,10 @@
 
 		val = soc_widget_read(w, reg);
 		val = (val >> shift) & mask;
+		if (invert)
+			val = max - val;
 
-		if ((invert && !val) || (!invert && val))
-			p->connect = 1;
-		else
-			p->connect = 0;
+		p->connect = !!val;
 	}
 	break;
 	case snd_soc_dapm_mux: {
@@ -391,6 +393,7 @@
 	case snd_soc_dapm_vmid:
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai:
@@ -764,6 +767,7 @@
 	switch (widget->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		return 0;
 	default:
 		break;
@@ -850,6 +854,7 @@
 	switch (widget->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		return 0;
 	default:
 		break;
@@ -996,6 +1001,27 @@
 }
 EXPORT_SYMBOL_GPL(dapm_regulator_event);
 
+/*
+ * Handler for clock supply widget.
+ */
+int dapm_clock_event(struct snd_soc_dapm_widget *w,
+		   struct snd_kcontrol *kcontrol, int event)
+{
+	if (!w->clk)
+		return -EIO;
+
+#ifdef CONFIG_HAVE_CLK
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		return clk_enable(w->clk);
+	} else {
+		clk_disable(w->clk);
+		return 0;
+	}
+#endif
+	return 0;
+}
+EXPORT_SYMBOL_GPL(dapm_clock_event);
+
 static int dapm_widget_power_check(struct snd_soc_dapm_widget *w)
 {
 	if (w->power_checked)
@@ -1487,6 +1513,7 @@
 	switch (w->id) {
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		/* Supplies can't affect their outputs, only their inputs */
 		break;
 	default:
@@ -1570,7 +1597,15 @@
 	}
 
 	list_for_each_entry(w, &card->widgets, list) {
-		list_del_init(&w->dirty);
+		switch (w->id) {
+		case snd_soc_dapm_pre:
+		case snd_soc_dapm_post:
+			/* These widgets always need to be powered */
+			break;
+		default:
+			list_del_init(&w->dirty);
+			break;
+		}
 
 		if (w->power) {
 			d = w->dapm;
@@ -1587,6 +1622,7 @@
 				break;
 			case snd_soc_dapm_supply:
 			case snd_soc_dapm_regulator_supply:
+			case snd_soc_dapm_clock_supply:
 			case snd_soc_dapm_micbias:
 				if (d->target_bias_level < SND_SOC_BIAS_STANDBY)
 					d->target_bias_level = SND_SOC_BIAS_STANDBY;
@@ -1941,6 +1977,7 @@
 		case snd_soc_dapm_mixer_named_ctl:
 		case snd_soc_dapm_supply:
 		case snd_soc_dapm_regulator_supply:
+		case snd_soc_dapm_clock_supply:
 			if (w->name)
 				count += sprintf(buf + count, "%s: %s\n",
 					w->name, w->power ? "On":"Off");
@@ -2187,6 +2224,7 @@
 	case snd_soc_dapm_post:
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 	case snd_soc_dapm_aif_in:
 	case snd_soc_dapm_aif_out:
 	case snd_soc_dapm_dai:
@@ -2221,6 +2259,10 @@
 		path->connect = 0;
 		return 0;
 	}
+
+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	return 0;
 
 err:
@@ -2230,6 +2272,59 @@
 	return ret;
 }
 
+static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm,
+				  const struct snd_soc_dapm_route *route)
+{
+	struct snd_soc_dapm_path *path, *p;
+	const char *sink;
+	const char *source;
+	char prefixed_sink[80];
+	char prefixed_source[80];
+
+	if (route->control) {
+		dev_err(dapm->dev,
+			"Removal of routes with controls not supported\n");
+		return -EINVAL;
+	}
+
+	if (dapm->codec && dapm->codec->name_prefix) {
+		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
+			 dapm->codec->name_prefix, route->sink);
+		sink = prefixed_sink;
+		snprintf(prefixed_source, sizeof(prefixed_source), "%s %s",
+			 dapm->codec->name_prefix, route->source);
+		source = prefixed_source;
+	} else {
+		sink = route->sink;
+		source = route->source;
+	}
+
+	path = NULL;
+	list_for_each_entry(p, &dapm->card->paths, list) {
+		if (strcmp(p->source->name, source) != 0)
+			continue;
+		if (strcmp(p->sink->name, sink) != 0)
+			continue;
+		path = p;
+		break;
+	}
+
+	if (path) {
+		dapm_mark_dirty(path->source, "Route removed");
+		dapm_mark_dirty(path->sink, "Route removed");
+
+		list_del(&path->list);
+		list_del(&path->list_sink);
+		list_del(&path->list_source);
+		kfree(path);
+	} else {
+		dev_warn(dapm->dev, "Route %s->%s does not exist\n",
+			 source, sink);
+	}
+
+	return 0;
+}
+
 /**
  * snd_soc_dapm_add_routes - Add routes between DAPM widgets
  * @dapm: DAPM context
@@ -2246,15 +2341,15 @@
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num)
 {
-	int i, ret = 0;
+	int i, r, ret = 0;
 
 	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
 	for (i = 0; i < num; i++) {
-		ret = snd_soc_dapm_add_route(dapm, route);
-		if (ret < 0) {
+		r = snd_soc_dapm_add_route(dapm, route);
+		if (r < 0) {
 			dev_err(dapm->dev, "Failed to add route %s->%s\n",
 				route->source, route->sink);
-			break;
+			ret = r;
 		}
 		route++;
 	}
@@ -2264,6 +2359,30 @@
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes);
 
+/**
+ * snd_soc_dapm_del_routes - Remove routes between DAPM widgets
+ * @dapm: DAPM context
+ * @route: audio routes
+ * @num: number of routes
+ *
+ * Removes routes from the DAPM context.
+ */
+int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm,
+			    const struct snd_soc_dapm_route *route, int num)
+{
+	int i, ret = 0;
+
+	mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT);
+	for (i = 0; i < num; i++) {
+		snd_soc_dapm_del_route(dapm, route);
+		route++;
+	}
+	mutex_unlock(&dapm->card->dapm_mutex);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes);
+
 static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm,
 				   const struct snd_soc_dapm_route *route)
 {
@@ -2434,23 +2553,20 @@
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;
 	unsigned int shift = mc->shift;
-	unsigned int rshift = mc->rshift;
 	int max = mc->max;
-	unsigned int invert = mc->invert;
 	unsigned int mask = (1 << fls(max)) - 1;
+	unsigned int invert = mc->invert;
+
+	if (snd_soc_volsw_is_stereo(mc))
+		dev_warn(widget->dapm->dev,
+			 "Control '%s' is stereo, which is not supported\n",
+			 kcontrol->id.name);
 
 	ucontrol->value.integer.value[0] =
 		(snd_soc_read(widget->codec, reg) >> shift) & mask;
-	if (shift != rshift)
-		ucontrol->value.integer.value[1] =
-			(snd_soc_read(widget->codec, reg) >> rshift) & mask;
-	if (invert) {
+	if (invert)
 		ucontrol->value.integer.value[0] =
 			max - ucontrol->value.integer.value[0];
-		if (shift != rshift)
-			ucontrol->value.integer.value[1] =
-				max - ucontrol->value.integer.value[1];
-	}
 
 	return 0;
 }
@@ -2484,20 +2600,19 @@
 	struct snd_soc_dapm_update update;
 	int wi;
 
+	if (snd_soc_volsw_is_stereo(mc))
+		dev_warn(widget->dapm->dev,
+			 "Control '%s' is stereo, which is not supported\n",
+			 kcontrol->id.name);
+
 	val = (ucontrol->value.integer.value[0] & mask);
+	connect = !!val;
 
 	if (invert)
 		val = max - val;
 	mask = mask << shift;
 	val = val << shift;
 
-	if (val)
-		/* new connection */
-		connect = invert ? 0 : 1;
-	else
-		/* old connection must be powered down */
-		connect = invert ? 1 : 0;
-
 	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
 
 	change = snd_soc_test_bits(widget->codec, reg, mask, val);
@@ -2873,6 +2988,19 @@
 			return NULL;
 		}
 		break;
+	case snd_soc_dapm_clock_supply:
+#ifdef CONFIG_CLKDEV_LOOKUP
+		w->clk = devm_clk_get(dapm->dev, w->name);
+		if (IS_ERR(w->clk)) {
+			ret = PTR_ERR(w->clk);
+			dev_err(dapm->dev, "Failed to request %s: %d\n",
+				w->name, ret);
+			return NULL;
+		}
+#else
+		return NULL;
+#endif
+		break;
 	default:
 		break;
 	}
@@ -2924,6 +3052,7 @@
 		break;
 	case snd_soc_dapm_supply:
 	case snd_soc_dapm_regulator_supply:
+	case snd_soc_dapm_clock_supply:
 		w->power_check = dapm_supply_check_power;
 		break;
 	case snd_soc_dapm_dai:
@@ -3538,10 +3667,13 @@
 
 static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm)
 {
+	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	LIST_HEAD(down_list);
 	int powerdown = 0;
 
+	mutex_lock(&card->dapm_mutex);
+
 	list_for_each_entry(w, &dapm->card->widgets, list) {
 		if (w->dapm != dapm)
 			continue;
@@ -3564,6 +3696,8 @@
 			snd_soc_dapm_set_bias_level(dapm,
 						    SND_SOC_BIAS_STANDBY);
 	}
+
+	mutex_unlock(&card->dapm_mutex);
 }
 
 /*
diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c
index 4756952..5df529e 100644
--- a/sound/soc/soc-dmaengine-pcm.c
+++ b/sound/soc/soc-dmaengine-pcm.c
@@ -30,6 +30,7 @@
 
 struct dmaengine_pcm_runtime_data {
 	struct dma_chan *dma_chan;
+	dma_cookie_t cookie;
 
 	unsigned int pos;
 
@@ -153,7 +154,7 @@
 
 	desc->callback = dmaengine_pcm_dma_complete;
 	desc->callback_param = substream;
-	dmaengine_submit(desc);
+	prtd->cookie = dmaengine_submit(desc);
 
 	return 0;
 }
@@ -200,6 +201,20 @@
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger);
 
 /**
+ * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation
+ * @substream: PCM substream
+ *
+ * This function is deprecated and should not be used by new drivers, as its
+ * results may be unreliable.
+ */
+snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream)
+{
+	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+	return bytes_to_frames(substream->runtime, prtd->pos);
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue);
+
+/**
  * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation
  * @substream: PCM substream
  *
@@ -209,7 +224,19 @@
 snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
-	return bytes_to_frames(substream->runtime, prtd->pos);
+	struct dma_tx_state state;
+	enum dma_status status;
+	unsigned int buf_size;
+	unsigned int pos = 0;
+
+	status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state);
+	if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) {
+		buf_size = snd_pcm_lib_buffer_bytes(substream);
+		if (state.residue > 0 && state.residue <= buf_size)
+			pos = buf_size - state.residue;
+	}
+
+	return bytes_to_frames(substream->runtime, pos);
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer);
 
@@ -243,7 +270,7 @@
  * Note that this function will use private_data field of the substream's
  * runtime. So it is not availabe to your pcm driver implementation. If you need
  * to keep additional data attached to a substream use
- * snd_dmaeinge_pcm_{set,get}_data.
+ * snd_dmaengine_pcm_{set,get}_data.
  */
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
 	dma_filter_fn filter_fn, void *filter_data)
diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c
index 4d8dc6a..29183ef 100644
--- a/sound/soc/soc-io.c
+++ b/sound/soc/soc-io.c
@@ -142,11 +142,16 @@
 	case SND_SOC_REGMAP:
 		/* Device has made its own regmap arrangements */
 		codec->using_regmap = true;
+		if (!codec->control_data)
+			codec->control_data = dev_get_regmap(codec->dev, NULL);
 
-		ret = regmap_get_val_bytes(codec->control_data);
-		/* Errors are legitimate for non-integer byte multiples */
-		if (ret > 0)
-			codec->val_bytes = ret;
+		if (codec->control_data) {
+			ret = regmap_get_val_bytes(codec->control_data);
+			/* Errors are legitimate for non-integer byte
+			 * multiples */
+			if (ret > 0)
+				codec->val_bytes = ret;
+		}
 		break;
 
 	default:
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 48fd15b..ef22d0b 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1955,10 +1955,8 @@
 	fe->dpcm[stream].runtime = fe_substream->runtime;
 
 	if (dpcm_path_get(fe, stream, &list) <= 0) {
-		dev_warn(fe->dev, "asoc: %s no valid %s route\n",
+		dev_dbg(fe->dev, "asoc: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
-			mutex_unlock(&fe->card->mutex);
-			return -EINVAL;
 	}
 
 	/* calculate valid and active FE <-> BE dpcms */
@@ -2003,7 +2001,6 @@
 /* create a new pcm */
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
-	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
@@ -2042,7 +2039,8 @@
 			capture, &pcm);
 	}
 	if (ret < 0) {
-		printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name);
+		dev_err(rtd->card->dev, "can't create pcm for %s\n",
+			rtd->dai_link->name);
 		return ret;
 	}
 	dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name);
@@ -2099,14 +2097,14 @@
 	if (platform->driver->pcm_new) {
 		ret = platform->driver->pcm_new(rtd);
 		if (ret < 0) {
-			pr_err("asoc: platform pcm constructor failed\n");
+			dev_err(platform->dev, "pcm constructor failed\n");
 			return ret;
 		}
 	}
 
 	pcm->private_free = platform->driver->pcm_free;
 out:
-	printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name,
+	dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name,
 		cpu_dai->name);
 	return ret;
 }
diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c
new file mode 100644
index 0000000..c7c4b20
--- /dev/null
+++ b/sound/soc/spear/spdif_in.c
@@ -0,0 +1,297 @@
+/*
+ * ALSA SoC SPDIF In Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_in_regs.h"
+
+struct spdif_in_params {
+	u32 format;
+};
+
+struct spdif_in_dev {
+	struct clk *clk;
+	struct spear_dma_data dma_params;
+	struct spdif_in_params saved_params;
+	void *io_base;
+	struct device *dev;
+	void (*reset_perip)(void);
+	int irq;
+};
+
+static void spdif_in_configure(struct spdif_in_dev *host)
+{
+	u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN |
+		SPDIF_IN_VALEN | SPDIF_IN_BLKEN;
+	ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16;
+
+	writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+	writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+}
+
+static int spdif_in_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+	return 0;
+}
+
+static void spdif_in_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return;
+
+	writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_in_format(struct spdif_in_dev *host, u32 format)
+{
+	u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ctrl |= SPDIF_XTRACT_16BIT;
+		break;
+
+	case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE:
+		ctrl &= ~SPDIF_XTRACT_16BIT;
+		break;
+	}
+
+	writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+}
+
+static int spdif_in_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 format;
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	format = params_format(params);
+	host->saved_params.format = format;
+
+	return 0;
+}
+
+static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 ctrl;
+	int ret = 0;
+
+	if (substream->stream != SNDRV_PCM_STREAM_CAPTURE)
+		return -EINVAL;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		clk_enable(host->clk);
+		spdif_in_configure(host);
+		spdif_in_format(host, host->saved_params.format);
+
+		ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+		ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB;
+		writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+		writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ctrl = readl(host->io_base + SPDIF_IN_CTRL);
+		ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB);
+		writel(ctrl, host->io_base + SPDIF_IN_CTRL);
+		writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK);
+
+		if (host->reset_perip)
+			host->reset_perip();
+		clk_disable(host->clk);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static struct snd_soc_dai_ops spdif_in_dai_ops = {
+	.startup	= spdif_in_startup,
+	.shutdown	= spdif_in_shutdown,
+	.trigger	= spdif_in_trigger,
+	.hw_params	= spdif_in_hw_params,
+};
+
+struct snd_soc_dai_driver spdif_in_dai = {
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+				 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+				 SNDRV_PCM_RATE_192000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | \
+			   SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE,
+	},
+	.ops = &spdif_in_dai_ops,
+};
+
+static irqreturn_t spdif_in_irq(int irq, void *arg)
+{
+	struct spdif_in_dev *host = (struct spdif_in_dev *)arg;
+
+	u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ);
+
+	if (!irq_status)
+		return IRQ_NONE;
+
+	if (irq_status & SPDIF_IRQ_FIFOWRITE)
+		dev_err(host->dev, "spdif in: fifo write error");
+	if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD)
+		dev_err(host->dev, "spdif in: empty fifo read error");
+	if (irq_status & SPDIF_IRQ_FIFOFULL)
+		dev_err(host->dev, "spdif in: fifo full error");
+	if (irq_status & SPDIF_IRQ_OUTOFRANGE)
+		dev_err(host->dev, "spdif in: out of range error");
+
+	writel(0, host->io_base + SPDIF_IN_IRQ);
+
+	return IRQ_HANDLED;
+}
+
+static int spdif_in_probe(struct platform_device *pdev)
+{
+	struct spdif_in_dev *host;
+	struct spear_spdif_platform_data *pdata;
+	struct resource *res, *res_fifo;
+	int ret;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res)
+		return -EINVAL;
+
+	res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0);
+	if (!res_fifo)
+		return -EINVAL;
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+		return -ENOENT;
+	}
+
+	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+	if (!host) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	host->io_base = devm_ioremap(&pdev->dev, res->start,
+				resource_size(res));
+	if (!host->io_base) {
+		dev_warn(&pdev->dev, "ioremap failed\n");
+		return -ENOMEM;
+	}
+
+	host->irq = platform_get_irq(pdev, 0);
+	if (host->irq < 0)
+		return -EINVAL;
+
+	host->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(host->clk))
+		return PTR_ERR(host->clk);
+
+	pdata = dev_get_platdata(&pdev->dev);
+
+	if (!pdata)
+		return -EINVAL;
+
+	host->dma_params.data = pdata->dma_params;
+	host->dma_params.addr = res_fifo->start;
+	host->dma_params.max_burst = 16;
+	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	host->dma_params.filter = pdata->filter;
+	host->reset_perip = pdata->reset_perip;
+
+	host->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, host);
+
+	ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0,
+			"spdif-in", host);
+	if (ret) {
+		clk_put(host->clk);
+		dev_warn(&pdev->dev, "request_irq failed\n");
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai);
+	if (ret != 0) {
+		clk_put(host->clk);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int spdif_in_remove(struct platform_device *pdev)
+{
+	struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(host->clk);
+
+	return 0;
+}
+
+
+static struct platform_driver spdif_in_driver = {
+	.probe		= spdif_in_probe,
+	.remove		= spdif_in_remove,
+	.driver		= {
+		.name	= "spdif-in",
+		.owner	= THIS_MODULE,
+	},
+};
+
+module_platform_driver(spdif_in_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_in");
diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h
new file mode 100644
index 0000000..37af7bc
--- /dev/null
+++ b/sound/soc/spear/spdif_in_regs.h
@@ -0,0 +1,60 @@
+/*
+ * SPEAr SPDIF IN controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_IN_REGS_H
+#define SPDIF_IN_REGS_H
+
+#define SPDIF_IN_CTRL		0x00
+	#define SPDIF_IN_PRTYEN		(1 << 20)
+	#define SPDIF_IN_STATEN		(1 << 19)
+	#define SPDIF_IN_USREN		(1 << 18)
+	#define SPDIF_IN_VALEN		(1 << 17)
+	#define SPDIF_IN_BLKEN		(1 << 16)
+
+	#define SPDIF_MODE_24BIT	(8 << 12)
+	#define SPDIF_MODE_23BIT	(7 << 12)
+	#define SPDIF_MODE_22BIT	(6 << 12)
+	#define SPDIF_MODE_21BIT	(5 << 12)
+	#define SPDIF_MODE_20BIT	(4 << 12)
+	#define SPDIF_MODE_19BIT	(3 << 12)
+	#define SPDIF_MODE_18BIT	(2 << 12)
+	#define SPDIF_MODE_17BIT	(1 << 12)
+	#define SPDIF_MODE_16BIT	(0 << 12)
+	#define SPDIF_MODE_MASK		(0x0F << 12)
+
+	#define SPDIF_IN_VALID		(1 << 11)
+	#define SPDIF_IN_SAMPLE		(1 << 10)
+	#define SPDIF_DATA_SWAP		(1 << 9)
+	#define SPDIF_IN_ENB		(1 << 8)
+	#define SPDIF_DATA_REVERT	(1 << 7)
+	#define SPDIF_XTRACT_16BIT	(1 << 6)
+	#define SPDIF_FIFO_THRES_16	(16 << 0)
+
+#define SPDIF_IN_IRQ_MASK	0x04
+#define SPDIF_IN_IRQ		0x08
+	#define SPDIF_IRQ_FIFOWRITE	(1 << 0)
+	#define SPDIF_IRQ_EMPTYFIFOREAD	(1 << 1)
+	#define SPDIF_IRQ_FIFOFULL	(1 << 2)
+	#define SPDIF_IRQ_OUTOFRANGE	(1 << 3)
+
+#define SPDIF_IN_STA		0x0C
+	#define SPDIF_IN_LOCK		(0x1 << 0)
+
+#endif /* SPDIF_IN_REGS_H */
diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c
new file mode 100644
index 0000000..5eac4cd
--- /dev/null
+++ b/sound/soc/spear/spdif_out.c
@@ -0,0 +1,389 @@
+/*
+ * ALSA SoC SPDIF Out Audio Layer for spear processors
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Vipin Kumar <vipin.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/device.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/io.h>
+#include <linux/ioport.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+#include <sound/spear_spdif.h>
+#include "spdif_out_regs.h"
+
+struct spdif_out_params {
+	u32 rate;
+	u32 core_freq;
+	u32 mute;
+};
+
+struct spdif_out_dev {
+	struct clk *clk;
+	struct spear_dma_data dma_params;
+	struct spdif_out_params saved_params;
+	u32 running;
+	void __iomem *io_base;
+};
+
+static void spdif_out_configure(struct spdif_out_dev *host)
+{
+	writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST);
+	mdelay(1);
+	writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET,
+			host->io_base + SPDIF_OUT_SOFT_RST);
+
+	writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 |
+			SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW |
+			SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW,
+			host->io_base + SPDIF_OUT_CFG);
+
+	writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR);
+	writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR);
+}
+
+static int spdif_out_startup(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *cpu_dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params);
+
+	ret = clk_enable(host->clk);
+	if (ret)
+		return ret;
+
+	host->running = true;
+	spdif_out_configure(host);
+
+	return 0;
+}
+
+static void spdif_out_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return;
+
+	clk_disable(host->clk);
+	host->running = false;
+	snd_soc_dai_set_dma_data(dai, substream, NULL);
+}
+
+static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq,
+		u32 rate)
+{
+	u32 divider, ctrl;
+
+	clk_set_rate(host->clk, core_freq);
+	divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128));
+
+	ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+	ctrl &= ~SPDIF_DIVIDER_MASK;
+	ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK;
+	writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+}
+
+static int spdif_out_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 rate, core_freq;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+	case 16000:
+	case 32000:
+	case 64000:
+		/*
+		 * The clock is multiplied by 10 to bring it to feasible range
+		 * of frequencies for sscg
+		 */
+		core_freq = 64000 * 128 * 10;	/* 81.92 MHz */
+		break;
+	case 5512:
+	case 11025:
+	case 22050:
+	case 44100:
+	case 88200:
+	case 176400:
+		core_freq = 176400 * 128;	/* 22.5792 MHz */
+		break;
+	case 48000:
+	case 96000:
+	case 192000:
+	default:
+		core_freq = 192000 * 128;	/* 24.576 MHz */
+		break;
+	}
+
+	spdif_out_clock(host, core_freq, rate);
+	host->saved_params.core_freq = core_freq;
+	host->saved_params.rate = rate;
+
+	return 0;
+}
+
+static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 ctrl;
+	int ret = 0;
+
+	if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
+		return -EINVAL;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+			ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+			ctrl &= ~SPDIF_OPMODE_MASK;
+			if (!host->saved_params.mute)
+				ctrl |= SPDIF_OPMODE_AUD_DATA |
+					SPDIF_STATE_NORMAL;
+			else
+				ctrl |= SPDIF_OPMODE_MUTE_PCM;
+			writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ctrl = readl(host->io_base + SPDIF_OUT_CTRL);
+		ctrl &= ~SPDIF_OPMODE_MASK;
+		ctrl |= SPDIF_OPMODE_OFF;
+		writel(ctrl, host->io_base + SPDIF_OUT_CTRL);
+		break;
+
+	default:
+		ret = -EINVAL;
+		break;
+	}
+	return ret;
+}
+
+static int spdif_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai);
+	u32 val;
+
+	host->saved_params.mute = mute;
+	val = readl(host->io_base + SPDIF_OUT_CTRL);
+	val &= ~SPDIF_OPMODE_MASK;
+
+	if (mute)
+		val |= SPDIF_OPMODE_MUTE_PCM;
+	else {
+		if (host->running)
+			val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL;
+		else
+			val |= SPDIF_OPMODE_OFF;
+	}
+
+	writel(val, host->io_base + SPDIF_OUT_CTRL);
+	return 0;
+}
+
+static int spdif_mute_get(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_pcm_runtime *rtd = card->rtd;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ucontrol->value.integer.value[0] = host->saved_params.mute;
+	return 0;
+}
+
+static int spdif_mute_put(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_card *card = codec->card;
+	struct snd_soc_pcm_runtime *rtd = card->rtd;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (host->saved_params.mute == ucontrol->value.integer.value[0])
+		return 0;
+
+	spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]);
+
+	return 1;
+}
+static const struct snd_kcontrol_new spdif_out_controls[] = {
+	SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0,
+			spdif_mute_get, spdif_mute_put),
+};
+
+int spdif_soc_dai_probe(struct snd_soc_dai *dai)
+{
+	return snd_soc_add_dai_controls(dai, spdif_out_controls,
+				ARRAY_SIZE(spdif_out_controls));
+}
+
+static const struct snd_soc_dai_ops spdif_out_dai_ops = {
+	.digital_mute	= spdif_digital_mute,
+	.startup	= spdif_out_startup,
+	.shutdown	= spdif_out_shutdown,
+	.trigger	= spdif_out_trigger,
+	.hw_params	= spdif_out_hw_params,
+};
+
+static struct snd_soc_dai_driver spdif_out_dai = {
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+				 SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \
+				 SNDRV_PCM_RATE_192000),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.probe = spdif_soc_dai_probe,
+	.ops = &spdif_out_dai_ops,
+};
+
+static int spdif_out_probe(struct platform_device *pdev)
+{
+	struct spdif_out_dev *host;
+	struct spear_spdif_platform_data *pdata;
+	struct resource *res;
+	int ret;
+
+	res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!res)
+		return -EINVAL;
+
+	if (!devm_request_mem_region(&pdev->dev, res->start,
+				resource_size(res), pdev->name)) {
+		dev_warn(&pdev->dev, "Failed to get memory resourse\n");
+		return -ENOENT;
+	}
+
+	host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL);
+	if (!host) {
+		dev_warn(&pdev->dev, "kzalloc fail\n");
+		return -ENOMEM;
+	}
+
+	host->io_base = devm_ioremap(&pdev->dev, res->start,
+				resource_size(res));
+	if (!host->io_base) {
+		dev_warn(&pdev->dev, "ioremap failed\n");
+		return -ENOMEM;
+	}
+
+	host->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(host->clk))
+		return PTR_ERR(host->clk);
+
+	pdata = dev_get_platdata(&pdev->dev);
+
+	host->dma_params.data = pdata->dma_params;
+	host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA;
+	host->dma_params.max_burst = 16;
+	host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	host->dma_params.filter = pdata->filter;
+
+	dev_set_drvdata(&pdev->dev, host);
+
+	ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai);
+	if (ret != 0) {
+		clk_put(host->clk);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int spdif_out_remove(struct platform_device *pdev)
+{
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	clk_put(host->clk);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int spdif_out_suspend(struct device *dev)
+{
+	struct platform_device *pdev = to_platform_device(dev);
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	if (host->running)
+		clk_disable(host->clk);
+
+	return 0;
+}
+
+static int spdif_out_resume(struct device *dev)
+{
+	struct platform_device *pdev = to_platform_device(dev);
+	struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev);
+
+	if (host->running) {
+		clk_enable(host->clk);
+		spdif_out_configure(host);
+		spdif_out_clock(host, host->saved_params.core_freq,
+				host->saved_params.rate);
+	}
+	return 0;
+}
+
+static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \
+		spdif_out_resume);
+
+#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops)
+
+#else
+#define SPDIF_OUT_DEV_PM_OPS NULL
+
+#endif
+
+static struct platform_driver spdif_out_driver = {
+	.probe		= spdif_out_probe,
+	.remove		= spdif_out_remove,
+	.driver		= {
+		.name	= "spdif-out",
+		.owner	= THIS_MODULE,
+		.pm	= SPDIF_OUT_DEV_PM_OPS,
+	},
+};
+
+module_platform_driver(spdif_out_driver);
+
+MODULE_AUTHOR("Vipin Kumar <vipin.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spdif_out");
diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h
new file mode 100644
index 0000000..a5e5332
--- /dev/null
+++ b/sound/soc/spear/spdif_out_regs.h
@@ -0,0 +1,79 @@
+/*
+ * SPEAr SPDIF OUT controller header file
+ *
+ * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com)
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ */
+
+#ifndef SPDIF_OUT_REGS_H
+#define SPDIF_OUT_REGS_H
+
+#define SPDIF_OUT_SOFT_RST	0x00
+	#define SPDIF_OUT_RESET		(1 << 0)
+#define SPDIF_OUT_FIFO_DATA	0x04
+#define SPDIF_OUT_INT_STA	0x08
+#define SPDIF_OUT_INT_STA_CLR	0x0C
+	#define SPDIF_INT_UNDERFLOW	(1 << 0)
+	#define SPDIF_INT_EODATA	(1 << 1)
+	#define SPDIF_INT_EOBLOCK	(1 << 2)
+	#define SPDIF_INT_EOLATENCY	(1 << 3)
+	#define SPDIF_INT_EOPD_DATA	(1 << 4)
+	#define SPDIF_INT_MEMFULLREAD	(1 << 5)
+	#define SPDIF_INT_EOPD_PAUSE	(1 << 6)
+
+#define SPDIF_OUT_INT_EN	0x10
+#define SPDIF_OUT_INT_EN_SET	0x14
+#define SPDIF_OUT_INT_EN_CLR	0x18
+#define SPDIF_OUT_CTRL		0x1C
+	#define SPDIF_OPMODE_MASK	(7 << 0)
+	#define SPDIF_OPMODE_OFF	(0 << 0)
+	#define SPDIF_OPMODE_MUTE_PCM	(1 << 0)
+	#define SPDIF_OPMODE_MUTE_PAUSE	(2 << 0)
+	#define SPDIF_OPMODE_AUD_DATA	(3 << 0)
+	#define SPDIF_OPMODE_ENCODE	(4 << 0)
+	#define SPDIF_STATE_NORMAL	(1 << 3)
+	#define SPDIF_DIVIDER_MASK	(0xff << 5)
+	#define SPDIF_DIVIDER_SHIFT	(5)
+	#define SPDIF_SAMPLEREAD_MASK	(0x1ffff << 15)
+	#define SPDIF_SAMPLEREAD_SHIFT	(15)
+#define SPDIF_OUT_STA		0x20
+#define SPDIF_OUT_PA_PB		0x24
+#define SPDIF_OUT_PC_PD		0x28
+#define SPDIF_OUT_CL1		0x2C
+#define SPDIF_OUT_CR1		0x30
+#define SPDIF_OUT_CL2_CR2_UV	0x34
+#define SPDIF_OUT_PAUSE_LAT	0x38
+#define SPDIF_OUT_FRMLEN_BRST	0x3C
+#define SPDIF_OUT_CFG		0x40
+	#define SPDIF_OUT_MEMFMT_16_0	(0 << 5)
+	#define SPDIF_OUT_MEMFMT_16_16	(1 << 5)
+	#define SPDIF_OUT_VALID_DMA	(0 << 3)
+	#define SPDIF_OUT_VALID_HW	(1 << 3)
+	#define SPDIF_OUT_USER_DMA	(0 << 2)
+	#define SPDIF_OUT_USER_HW	(1 << 2)
+	#define SPDIF_OUT_CHNLSTA_DMA	(0 << 1)
+	#define SPDIF_OUT_CHNLSTA_HW	(1 << 1)
+	#define SPDIF_OUT_PARITY_HW	(0 << 0)
+	#define SPDIF_OUT_PARITY_DMA	(1 << 0)
+	#define SPDIF_OUT_FDMA_TRIG_2	(2 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_6	(6 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_8	(8 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_10	(10 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_12	(12 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_16	(16 << 8)
+	#define SPDIF_OUT_FDMA_TRIG_18	(18 << 8)
+
+#endif /* SPDIF_OUT_REGS_H */
diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c
new file mode 100644
index 0000000..97c2cac
--- /dev/null
+++ b/sound/soc/spear/spear_pcm.c
@@ -0,0 +1,214 @@
+/*
+ * ALSA PCM interface for ST SPEAr Processors
+ *
+ * sound/soc/spear/spear_pcm.c
+ *
+ * Copyright (C) 2012 ST Microelectronics
+ * Rajeev Kumar<rajeev-dlh.kumar@st.com>
+ *
+ * This file is licensed under the terms of the GNU General Public
+ * License version 2. This program is licensed "as is" without any
+ * warranty of any kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/dmaengine.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <linux/scatterlist.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/spear_dma.h>
+
+struct snd_pcm_hardware spear_pcm_hardware = {
+	.info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		 SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+	.buffer_bytes_max = 16 * 1024, /* max buffer size */
+	.period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */
+	.period_bytes_max = 2 * 1024, /* maximum period size */
+	.periods_min = 1, /* min # periods */
+	.periods_max = 8, /* max # of periods */
+	.fifo_size = 0, /* fifo size in bytes */
+};
+
+static int spear_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+
+	return 0;
+}
+
+static int spear_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+
+	return 0;
+}
+
+static int spear_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	struct spear_dma_data *dma_data = (struct spear_dma_data *)
+		snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+	int ret;
+
+	ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware);
+	if (ret)
+		return ret;
+
+	ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data);
+	if (ret)
+		return ret;
+
+	snd_dmaengine_pcm_set_data(substream, dma_data);
+
+	return 0;
+}
+
+static int spear_pcm_close(struct snd_pcm_substream *substream)
+{
+
+	snd_dmaengine_pcm_close(substream);
+
+	return 0;
+}
+
+static int spear_pcm_mmap(struct snd_pcm_substream *substream,
+		struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+			runtime->dma_area, runtime->dma_addr,
+			runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops spear_pcm_ops = {
+	.open		= spear_pcm_open,
+	.close		= spear_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= spear_pcm_hw_params,
+	.hw_free	= spear_pcm_hw_free,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= spear_pcm_mmap,
+};
+
+static int
+spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream,
+		size_t size)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+			&buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+
+	dev_info(buf->dev.dev,
+			" preallocate_dma_buffer: area=%p, addr=%p, size=%d\n",
+			(void *)buf->area, (void *)buf->addr, size);
+
+	buf->bytes = size;
+	return 0;
+}
+
+static void spear_pcm_free(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf && !buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static u64 spear_pcm_dmamask = DMA_BIT_MASK(32);
+
+static int spear_pcm_new(struct snd_card *card,
+		struct snd_soc_dai *dai, struct snd_pcm *pcm)
+{
+	int ret;
+
+	if (!card->dev->dma_mask)
+		card->dev->dma_mask = &spear_pcm_dmamask;
+	if (!card->dev->coherent_dma_mask)
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+	if (dai->driver->playback.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(pcm,
+				SNDRV_PCM_STREAM_PLAYBACK,
+				spear_pcm_hardware.buffer_bytes_max);
+		if (ret)
+			return ret;
+	}
+
+	if (dai->driver->capture.channels_min) {
+		ret = spear_pcm_preallocate_dma_buffer(pcm,
+				SNDRV_PCM_STREAM_CAPTURE,
+				spear_pcm_hardware.buffer_bytes_max);
+		if (ret)
+			return ret;
+	}
+
+	return 0;
+}
+
+struct snd_soc_platform_driver spear_soc_platform = {
+	.ops		=	&spear_pcm_ops,
+	.pcm_new	=	spear_pcm_new,
+	.pcm_free	=	spear_pcm_free,
+};
+
+static int __devinit spear_soc_platform_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_platform(&pdev->dev, &spear_soc_platform);
+}
+
+static int __devexit spear_soc_platform_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver spear_pcm_driver = {
+	.driver = {
+		.name = "spear-pcm-audio",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = spear_soc_platform_probe,
+	.remove = __devexit_p(spear_soc_platform_remove),
+};
+
+module_platform_driver(spear_pcm_driver);
+
+MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar@st.com>");
+MODULE_DESCRIPTION("SPEAr PCM DMA module");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:spear-pcm-audio");
diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig
index 76dc230..02bcd30 100644
--- a/sound/soc/tegra/Kconfig
+++ b/sound/soc/tegra/Kconfig
@@ -1,7 +1,8 @@
 config SND_SOC_TEGRA
 	tristate "SoC Audio for the Tegra System-on-Chip"
-	depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA
+	depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA)
 	select REGMAP_MMIO
+	select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA
 	help
 	  Say Y or M here if you want support for SoC audio on Tegra.
 
diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c
index 1647dbf..0832e8a 100644
--- a/sound/soc/tegra/tegra20_i2s.c
+++ b/sound/soc/tegra/tegra20_i2s.c
@@ -46,18 +46,6 @@
 
 #define DRV_NAME "tegra20-i2s"
 
-static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val)
-{
-	regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg)
-{
-	u32 val;
-	regmap_read(i2s->regmap, reg, &val);
-	return val;
-}
-
 static int tegra20_i2s_runtime_suspend(struct device *dev)
 {
 	struct tegra20_i2s *i2s = dev_get_drvdata(dev);
@@ -85,6 +73,7 @@
 				unsigned int fmt)
 {
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -93,10 +82,10 @@
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE;
+	mask = TEGRA20_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE;
+		val = TEGRA20_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
@@ -104,33 +93,35 @@
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
-			   TEGRA20_I2S_CTRL_LRCK_MASK);
+	mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK |
+		TEGRA20_I2S_CTRL_LRCK_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_A:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP;
+		val |= TEGRA20_I2S_CTRL_LRCK_R_LOW;
 		break;
 	case SND_SOC_DAIFMT_I2S:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM;
+		val |= TEGRA20_I2S_CTRL_LRCK_L_LOW;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
 	return 0;
 }
 
@@ -138,29 +129,34 @@
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 reg;
+	unsigned int mask, val;
 	int ret, sample_size, srate, i2sclock, bitcnt;
 
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
+	mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_16;
 		sample_size = 16;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_24;
 		sample_size = 24;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
-		i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32;
+		val = TEGRA20_I2S_CTRL_BIT_SIZE_32;
 		sample_size = 32;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK;
+	val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
+
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val);
+
 	srate = params_rate(params);
 
 	/* Final "* 2" required by Tegra hardware */
@@ -175,42 +171,44 @@
 	bitcnt = (i2sclock / (2 * srate)) - 1;
 	if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US)
 		return -EINVAL;
-	reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
+	val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT;
 
 	if (i2sclock % (2 * srate))
-		reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
+		val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE;
 
-	tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg);
+	regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val);
 
-	tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR,
-		TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
-		TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
+	regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR,
+		     TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
+		     TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
 	return 0;
 }
 
 static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE);
 }
 
 static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0);
 }
 
 static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE);
 }
 
 static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s)
 {
-	i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE;
-	tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL,
+			   TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0);
 }
 
 static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -261,12 +259,14 @@
 static const struct snd_soc_dai_driver tegra20_i2s_dai_template = {
 	.probe = tegra20_i2s_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
+		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
@@ -412,8 +412,6 @@
 	i2s->playback_dma_data.width = 32;
 	i2s->playback_dma_data.req_sel = dma_ch;
 
-	i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED;
-
 	pm_runtime_enable(&pdev->dev);
 	if (!pm_runtime_enabled(&pdev->dev)) {
 		ret = tegra20_i2s_runtime_resume(&pdev->dev);
diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h
index a57efc6..c27069d 100644
--- a/sound/soc/tegra/tegra20_i2s.h
+++ b/sound/soc/tegra/tegra20_i2s.h
@@ -158,7 +158,6 @@
 	struct tegra_pcm_dma_params capture_dma_data;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c
index 2262e4f..3ebc867 100644
--- a/sound/soc/tegra/tegra20_spdif.c
+++ b/sound/soc/tegra/tegra20_spdif.c
@@ -37,19 +37,6 @@
 
 #define DRV_NAME "tegra20-spdif"
 
-static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg,
-					u32 val)
-{
-	regmap_write(spdif->regmap, reg, val);
-}
-
-static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg)
-{
-	u32 val;
-	regmap_read(spdif->regmap, reg, &val);
-	return val;
-}
-
 static int tegra20_spdif_runtime_suspend(struct device *dev)
 {
 	struct tegra20_spdif *spdif = dev_get_drvdata(dev);
@@ -77,21 +64,24 @@
 				struct snd_pcm_hw_params *params,
 				struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 	int ret, spdifclock;
 
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK;
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
+	mask = TEGRA20_SPDIF_CTRL_PACK |
+	       TEGRA20_SPDIF_CTRL_BIT_MODE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK;
-		spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
+		val = TEGRA20_SPDIF_CTRL_PACK |
+		      TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val);
+
 	switch (params_rate(params)) {
 	case 32000:
 		spdifclock = 4096000;
@@ -129,14 +119,15 @@
 
 static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif)
 {
-	spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN;
-	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+			   TEGRA20_SPDIF_CTRL_TX_EN,
+			   TEGRA20_SPDIF_CTRL_TX_EN);
 }
 
 static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif)
 {
-	spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN;
-	tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl);
+	regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL,
+			   TEGRA20_SPDIF_CTRL_TX_EN, 0);
 }
 
 static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -181,6 +172,7 @@
 	.name = DRV_NAME,
 	.probe = tegra20_spdif_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |
diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h
index ed75652..b48d699 100644
--- a/sound/soc/tegra/tegra20_spdif.h
+++ b/sound/soc/tegra/tegra20_spdif.h
@@ -465,7 +465,6 @@
 	struct tegra_pcm_dma_params capture_dma_data;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c
index d308faa..4418422 100644
--- a/sound/soc/tegra/tegra30_i2s.c
+++ b/sound/soc/tegra/tegra30_i2s.c
@@ -44,18 +44,6 @@
 
 #define DRV_NAME "tegra30-i2s"
 
-static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val)
-{
-	regmap_write(i2s->regmap, reg, val);
-}
-
-static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg)
-{
-	u32 val;
-	regmap_read(i2s->regmap, reg, &val);
-	return val;
-}
-
 static int tegra30_i2s_runtime_suspend(struct device *dev)
 {
 	struct tegra30_i2s *i2s = dev_get_drvdata(dev);
@@ -128,6 +116,7 @@
 				unsigned int fmt)
 {
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
+	unsigned int mask, val;
 
 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
@@ -136,10 +125,10 @@
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE;
+	mask = TEGRA30_I2S_CTRL_MASTER_ENABLE;
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBS_CFS:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE;
+		val = TEGRA30_I2S_CTRL_MASTER_ENABLE;
 		break;
 	case SND_SOC_DAIFMT_CBM_CFM:
 		break;
@@ -147,33 +136,37 @@
 		return -EINVAL;
 	}
 
-	i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
-			   TEGRA30_I2S_CTRL_LRCK_MASK);
+	mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK |
+		TEGRA30_I2S_CTRL_LRCK_MASK;
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_DSP_A:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC;
+		val |= TEGRA30_I2S_CTRL_LRCK_R_LOW;
 		break;
 	case SND_SOC_DAIFMT_I2S:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_RIGHT_J:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	case SND_SOC_DAIFMT_LEFT_J:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
+		val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK;
+		val |= TEGRA30_I2S_CTRL_LRCK_L_LOW;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	pm_runtime_get_sync(dai->dev);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+	pm_runtime_put(dai->dev);
+
 	return 0;
 }
 
@@ -181,24 +174,26 @@
 				 struct snd_pcm_hw_params *params,
 				 struct snd_soc_dai *dai)
 {
-	struct device *dev = substream->pcm->card->dev;
+	struct device *dev = dai->dev;
 	struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai);
-	u32 val;
+	unsigned int mask, val, reg;
 	int ret, sample_size, srate, i2sclock, bitcnt;
 
 	if (params_channels(params) != 2)
 		return -EINVAL;
 
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
+	mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK;
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
-		i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16;
+		val = TEGRA30_I2S_CTRL_BIT_SIZE_16;
 		sample_size = 16;
 		break;
 	default:
 		return -EINVAL;
 	}
 
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val);
+
 	srate = params_rate(params);
 
 	/* Final "* 2" required by Tegra hardware */
@@ -219,7 +214,7 @@
 	if (i2sclock % (2 * srate))
 		val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE;
 
-	tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val);
+	regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val);
 
 	val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) |
 	      (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) |
@@ -229,15 +224,17 @@
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX;
-		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val);
+		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	} else {
 		val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX;
-		tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val);
+		reg = TEGRA30_I2S_CIF_RX_CTRL;
 	}
 
+	regmap_write(i2s->regmap, reg, val);
+
 	val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) |
 	      (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT);
-	tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val);
+	regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val);
 
 	return 0;
 }
@@ -245,29 +242,31 @@
 static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif);
-	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX);
 }
 
 static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif);
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_TX, 0);
 }
 
 static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif);
-	i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX);
 }
 
 static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s)
 {
 	tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif);
-	i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX;
-	tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl);
+	regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL,
+			   TEGRA30_I2S_CTRL_XFER_EN_RX, 0);
 }
 
 static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
@@ -320,12 +319,14 @@
 static const struct snd_soc_dai_driver tegra30_i2s_dai_template = {
 	.probe = tegra30_i2s_probe,
 	.playback = {
+		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
 	.capture = {
+		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_8000_96000,
diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h
index 91adf29..34dc47b 100644
--- a/sound/soc/tegra/tegra30_i2s.h
+++ b/sound/soc/tegra/tegra30_i2s.h
@@ -236,7 +236,6 @@
 	enum tegra30_ahub_txcif playback_fifo_cif;
 	struct tegra_pcm_dma_params playback_dma_data;
 	struct regmap *regmap;
-	u32 reg_ctrl;
 };
 
 #endif
diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c
index 32de700..d684df2 100644
--- a/sound/soc/tegra/tegra_alc5632.c
+++ b/sound/soc/tegra/tegra_alc5632.c
@@ -1,5 +1,5 @@
 /*
- * tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
+* tegra_alc5632.c  --  Toshiba AC100(PAZ00) machine ASoC driver
  *
  * Copyright (C) 2011 The AC100 Kernel Team <ac100@lists.lauchpad.net>
  * Copyright (C) 2012 - NVIDIA, Inc.
@@ -33,11 +33,8 @@
 
 #define DRV_NAME "tegra-alc5632"
 
-#define GPIO_HP_DET     BIT(0)
-
 struct tegra_alc5632 {
 	struct tegra_asoc_utils_data util_data;
-	int gpio_requested;
 	int gpio_hp_det;
 };
 
@@ -46,7 +43,7 @@
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -108,9 +105,9 @@
 
 static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	struct device_node *np = codec->card->dev->of_node;
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card);
 
 	snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
@@ -119,14 +116,11 @@
 			ARRAY_SIZE(tegra_alc5632_hs_jack_pins),
 			tegra_alc5632_hs_jack_pins);
 
-	machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
-
 	if (gpio_is_valid(machine->gpio_hp_det)) {
 		tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det;
 		snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack,
 						1,
 						&tegra_alc5632_hp_jack_gpio);
-		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1");
@@ -159,6 +153,7 @@
 
 static __devinit int tegra_alc5632_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_alc5632;
 	struct tegra_alc5632 *alc5632;
 	int ret;
@@ -181,6 +176,10 @@
 		goto err;
 	}
 
+	alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0);
+	if (alc5632->gpio_hp_det == -ENODEV)
+		return -EPROBE_DEFER;
+
 	ret = snd_soc_of_parse_card_name(card, "nvidia,model");
 	if (ret)
 		goto err;
@@ -199,16 +198,16 @@
 		goto err;
 	}
 
-	tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle(
+	tegra_alc5632_dai.cpu_of_node = of_parse_phandle(
 			pdev->dev.of_node, "nvidia,i2s-controller", 0);
-	if (!tegra_alc5632_dai.cpu_dai_of_node) {
+	if (!tegra_alc5632_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
 		"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
 		goto err;
 	}
 
-	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node;
+	tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev);
 	if (ret)
@@ -234,11 +233,8 @@
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card);
 
-	if (machine->gpio_requested & GPIO_HP_DET)
-		snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack,
-					1,
-					&tegra_alc5632_hp_jack_gpio);
-	machine->gpio_requested = 0;
+	snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1,
+				&tegra_alc5632_hp_jack_gpio);
 
 	snd_soc_unregister_card(card);
 
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 127348d..5658bce 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -36,6 +36,7 @@
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
 
 #include "tegra_pcm.h"
 
@@ -56,6 +57,7 @@
 	.fifo_size		= 4,
 };
 
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
 static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd)
 {
 	struct snd_pcm_substream *substream = prtd->substream;
@@ -285,6 +287,119 @@
 	.pointer	= tegra_pcm_pointer,
 	.mmap		= tegra_pcm_mmap,
 };
+#else
+static int tegra_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->platform->dev;
+	int ret;
+
+	/* Set HW params now that initialization is complete */
+	snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware);
+
+	ret = snd_dmaengine_pcm_open(substream, NULL, NULL);
+	if (ret) {
+		dev_err(dev, "dmaengine pcm open failed with err %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int tegra_pcm_close(struct snd_pcm_substream *substream)
+{
+	snd_dmaengine_pcm_close(substream);
+	return 0;
+}
+
+static int tegra_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->platform->dev;
+	struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+	struct tegra_pcm_dma_params *dmap;
+	struct dma_slave_config slave_config;
+	int ret;
+
+	dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	ret = snd_hwparams_to_dma_slave_config(substream, params,
+						&slave_config);
+	if (ret) {
+		dev_err(dev, "hw params config failed with err %d\n", ret);
+		return ret;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		slave_config.dst_addr = dmap->addr;
+		slave_config.src_maxburst = 0;
+	} else {
+		slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+		slave_config.src_addr = dmap->addr;
+		slave_config.dst_maxburst = 0;
+	}
+	slave_config.slave_id = dmap->req_sel;
+
+	ret = dmaengine_slave_config(chan, &slave_config);
+	if (ret < 0) {
+		dev_err(dev, "dma slave config failed with err %d\n", ret);
+		return ret;
+	}
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	return 0;
+}
+
+static int tegra_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		return snd_dmaengine_pcm_trigger(substream,
+					SNDRV_PCM_TRIGGER_START);
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		return snd_dmaengine_pcm_trigger(substream,
+					SNDRV_PCM_TRIGGER_STOP);
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static int tegra_pcm_mmap(struct snd_pcm_substream *substream,
+				struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+					runtime->dma_area,
+					runtime->dma_addr,
+					runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops tegra_pcm_ops = {
+	.open		= tegra_pcm_open,
+	.close		= tegra_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= tegra_pcm_hw_params,
+	.hw_free	= tegra_pcm_hw_free,
+	.trigger	= tegra_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer,
+	.mmap		= tegra_pcm_mmap,
+};
+#endif
 
 static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 {
diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h
index 985d418..a3a4503 100644
--- a/sound/soc/tegra/tegra_pcm.h
+++ b/sound/soc/tegra/tegra_pcm.h
@@ -40,6 +40,7 @@
 	unsigned long req_sel;
 };
 
+#if defined(CONFIG_TEGRA_SYSTEM_DMA)
 struct tegra_runtime_data {
 	struct snd_pcm_substream *substream;
 	spinlock_t lock;
@@ -51,6 +52,7 @@
 	struct tegra_dma_req dma_req[2];
 	struct tegra_dma_channel *dma_chan;
 };
+#endif
 
 int tegra_pcm_platform_register(struct device *dev);
 void tegra_pcm_platform_unregister(struct device *dev);
diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c
index 4e77026..ea9166d 100644
--- a/sound/soc/tegra/tegra_wm8753.c
+++ b/sound/soc/tegra/tegra_wm8753.c
@@ -57,7 +57,7 @@
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -157,9 +157,9 @@
 		goto err;
 	}
 
-	tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle(
+	tegra_wm8753_dai.cpu_of_node = of_parse_phandle(
 			pdev->dev.of_node, "nvidia,i2s-controller", 0);
-	if (!tegra_wm8753_dai.cpu_dai_of_node) {
+	if (!tegra_wm8753_dai.cpu_of_node) {
 		dev_err(&pdev->dev,
 			"Property 'nvidia,i2s-controller' missing or invalid\n");
 		ret = -EINVAL;
@@ -167,7 +167,7 @@
 	}
 
 	tegra_wm8753_dai.platform_of_node =
-				tegra_wm8753_dai.cpu_dai_of_node;
+				tegra_wm8753_dai.cpu_of_node;
 
 	ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev);
 	if (ret)
diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c
index 3b6da91..0c5bb33 100644
--- a/sound/soc/tegra/tegra_wm8903.c
+++ b/sound/soc/tegra/tegra_wm8903.c
@@ -28,8 +28,6 @@
  *
  */
 
-#include <asm/mach-types.h>
-
 #include <linux/module.h>
 #include <linux/platform_device.h>
 #include <linux/slab.h>
@@ -50,16 +48,9 @@
 
 #define DRV_NAME "tegra-snd-wm8903"
 
-#define GPIO_SPKR_EN    BIT(0)
-#define GPIO_HP_MUTE    BIT(1)
-#define GPIO_INT_MIC_EN BIT(2)
-#define GPIO_EXT_MIC_EN BIT(3)
-#define GPIO_HP_DET     BIT(4)
-
 struct tegra_wm8903 {
 	struct tegra_wm8903_platform_data pdata;
 	struct tegra_asoc_utils_data util_data;
-	int gpio_requested;
 };
 
 static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream,
@@ -67,8 +58,7 @@
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -95,24 +85,6 @@
 		return err;
 	}
 
-	err = snd_soc_dai_set_fmt(codec_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "codec_dai fmt not set\n");
-		return err;
-	}
-
-	err = snd_soc_dai_set_fmt(cpu_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "cpu_dai fmt not set\n");
-		return err;
-	}
-
 	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 					SND_SOC_CLOCK_IN);
 	if (err < 0) {
@@ -160,7 +132,7 @@
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!(machine->gpio_requested & GPIO_SPKR_EN))
+	if (!gpio_is_valid(pdata->gpio_spkr_en))
 		return 0;
 
 	gpio_set_value_cansleep(pdata->gpio_spkr_en,
@@ -177,7 +149,7 @@
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (!(machine->gpio_requested & GPIO_HP_MUTE))
+	if (!gpio_is_valid(pdata->gpio_hp_mute))
 		return 0;
 
 	gpio_set_value_cansleep(pdata->gpio_hp_mute,
@@ -203,122 +175,18 @@
 	{"IN1L", NULL, "Mic Jack"},
 };
 
-static const struct snd_soc_dapm_route seaboard_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "ROP"},
-	{"Int Spk", NULL, "RON"},
-	{"Int Spk", NULL, "LOP"},
-	{"Int Spk", NULL, "LON"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN1R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route kaen_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "ROP"},
-	{"Int Spk", NULL, "RON"},
-	{"Int Spk", NULL, "LOP"},
-	{"Int Spk", NULL, "LON"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN2R", NULL, "Mic Jack"},
-};
-
-static const struct snd_soc_dapm_route aebl_audio_map[] = {
-	{"Headphone Jack", NULL, "HPOUTR"},
-	{"Headphone Jack", NULL, "HPOUTL"},
-	{"Int Spk", NULL, "LINEOUTR"},
-	{"Int Spk", NULL, "LINEOUTL"},
-	{"Mic Jack", NULL, "MICBIAS"},
-	{"IN1R", NULL, "Mic Jack"},
-};
-
 static const struct snd_kcontrol_new tegra_wm8903_controls[] = {
 	SOC_DAPM_PIN_SWITCH("Int Spk"),
 };
 
 static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd)
 {
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
 	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
-	struct device_node *np = card->dev->of_node;
-	int ret;
-
-	if (card->dev->platform_data) {
-		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
-	} else if (np) {
-		/*
-		 * This part must be in init() rather than probe() in order to
-		 * guarantee that the WM8903 has been probed, and hence its
-		 * GPIO controller registered, which is a pre-condition for
-		 * of_get_named_gpio() to be able to map the phandles in the
-		 * properties to the controller node. Given this, all
-		 * pdata handling is in init() for consistency.
-		 */
-		pdata->gpio_spkr_en = of_get_named_gpio(np,
-						"nvidia,spkr-en-gpios", 0);
-		pdata->gpio_hp_mute = of_get_named_gpio(np,
-						"nvidia,hp-mute-gpios", 0);
-		pdata->gpio_hp_det = of_get_named_gpio(np,
-						"nvidia,hp-det-gpios", 0);
-		pdata->gpio_int_mic_en = of_get_named_gpio(np,
-						"nvidia,int-mic-en-gpios", 0);
-		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
-						"nvidia,ext-mic-en-gpios", 0);
-	} else {
-		dev_err(card->dev, "No platform data supplied\n");
-		return -EINVAL;
-	}
-
-	if (gpio_is_valid(pdata->gpio_spkr_en)) {
-		ret = gpio_request(pdata->gpio_spkr_en, "spkr_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get spkr_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_SPKR_EN;
-
-		gpio_direction_output(pdata->gpio_spkr_en, 0);
-	}
-
-	if (gpio_is_valid(pdata->gpio_hp_mute)) {
-		ret = gpio_request(pdata->gpio_hp_mute, "hp_mute");
-		if (ret) {
-			dev_err(card->dev, "cannot get hp_mute gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_HP_MUTE;
-
-		gpio_direction_output(pdata->gpio_hp_mute, 1);
-	}
-
-	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
-		ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get int_mic_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_INT_MIC_EN;
-
-		/* Disable int mic; enable signal is active-high */
-		gpio_direction_output(pdata->gpio_int_mic_en, 0);
-	}
-
-	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
-		ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en");
-		if (ret) {
-			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
-			return ret;
-		}
-		machine->gpio_requested |= GPIO_EXT_MIC_EN;
-
-		/* Enable ext mic; enable signal is active-low */
-		gpio_direction_output(pdata->gpio_ext_mic_en, 0);
-	}
 
 	if (gpio_is_valid(pdata->gpio_hp_det)) {
 		tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det;
@@ -330,7 +198,6 @@
 		snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack,
 					1,
 					&tegra_wm8903_hp_jack_gpio);
-		machine->gpio_requested |= GPIO_HP_DET;
 	}
 
 	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
@@ -366,6 +233,9 @@
 	.codec_dai_name = "wm8903-hifi",
 	.init = tegra_wm8903_init,
 	.ops = &tegra_wm8903_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S |
+		   SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
 };
 
 static struct snd_soc_card snd_soc_tegra_wm8903 = {
@@ -385,8 +255,10 @@
 
 static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev)
 {
+	struct device_node *np = pdev->dev.of_node;
 	struct snd_soc_card *card = &snd_soc_tegra_wm8903;
 	struct tegra_wm8903 *machine;
+	struct tegra_wm8903_platform_data *pdata;
 	int ret;
 
 	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
@@ -401,12 +273,42 @@
 		ret = -ENOMEM;
 		goto err;
 	}
+	pdata = &machine->pdata;
 
 	card->dev = &pdev->dev;
 	platform_set_drvdata(pdev, card);
 	snd_soc_card_set_drvdata(card, machine);
 
-	if (pdev->dev.of_node) {
+	if (pdev->dev.platform_data) {
+		memcpy(pdata, card->dev->platform_data, sizeof(*pdata));
+	} else if (np) {
+		pdata->gpio_spkr_en = of_get_named_gpio(np,
+						"nvidia,spkr-en-gpios", 0);
+		if (pdata->gpio_spkr_en == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_hp_mute = of_get_named_gpio(np,
+						"nvidia,hp-mute-gpios", 0);
+		if (pdata->gpio_hp_mute == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_hp_det = of_get_named_gpio(np,
+						"nvidia,hp-det-gpios", 0);
+		if (pdata->gpio_hp_det == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_int_mic_en = of_get_named_gpio(np,
+						"nvidia,int-mic-en-gpios", 0);
+		if (pdata->gpio_int_mic_en == -ENODEV)
+			return -EPROBE_DEFER;
+
+		pdata->gpio_ext_mic_en = of_get_named_gpio(np,
+						"nvidia,ext-mic-en-gpios", 0);
+		if (pdata->gpio_ext_mic_en == -ENODEV)
+			return -EPROBE_DEFER;
+	}
+
+	if (np) {
 		ret = snd_soc_of_parse_card_name(card, "nvidia,model");
 		if (ret)
 			goto err;
@@ -417,8 +319,8 @@
 			goto err;
 
 		tegra_wm8903_dai.codec_name = NULL;
-		tegra_wm8903_dai.codec_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,audio-codec", 0);
+		tegra_wm8903_dai.codec_of_node = of_parse_phandle(np,
+				"nvidia,audio-codec", 0);
 		if (!tegra_wm8903_dai.codec_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,audio-codec' missing or invalid\n");
@@ -427,9 +329,9 @@
 		}
 
 		tegra_wm8903_dai.cpu_dai_name = NULL;
-		tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(
-				pdev->dev.of_node, "nvidia,i2s-controller", 0);
-		if (!tegra_wm8903_dai.cpu_dai_of_node) {
+		tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np,
+				"nvidia,i2s-controller", 0);
+		if (!tegra_wm8903_dai.cpu_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,i2s-controller' missing or invalid\n");
 			ret = -EINVAL;
@@ -438,20 +340,47 @@
 
 		tegra_wm8903_dai.platform_name = NULL;
 		tegra_wm8903_dai.platform_of_node =
-					tegra_wm8903_dai.cpu_dai_of_node;
+					tegra_wm8903_dai.cpu_of_node;
 	} else {
-		if (machine_is_harmony()) {
-			card->dapm_routes = harmony_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
-		} else if (machine_is_seaboard()) {
-			card->dapm_routes = seaboard_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map);
-		} else if (machine_is_kaen()) {
-			card->dapm_routes = kaen_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map);
-		} else {
-			card->dapm_routes = aebl_audio_map;
-			card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map);
+		card->dapm_routes = harmony_audio_map;
+		card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map);
+	}
+
+	if (gpio_is_valid(pdata->gpio_spkr_en)) {
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en,
+					    GPIOF_OUT_INIT_LOW, "spkr_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get spkr_en gpio\n");
+			return ret;
+		}
+	}
+
+	if (gpio_is_valid(pdata->gpio_hp_mute)) {
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute,
+					    GPIOF_OUT_INIT_HIGH, "hp_mute");
+		if (ret) {
+			dev_err(card->dev, "cannot get hp_mute gpio\n");
+			return ret;
+		}
+	}
+
+	if (gpio_is_valid(pdata->gpio_int_mic_en)) {
+		/* Disable int mic; enable signal is active-high */
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en,
+					    GPIOF_OUT_INIT_LOW, "int_mic_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get int_mic_en gpio\n");
+			return ret;
+		}
+	}
+
+	if (gpio_is_valid(pdata->gpio_ext_mic_en)) {
+		/* Enable ext mic; enable signal is active-low */
+		ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en,
+					    GPIOF_OUT_INIT_LOW, "ext_mic_en");
+		if (ret) {
+			dev_err(card->dev, "cannot get ext_mic_en gpio\n");
+			return ret;
 		}
 	}
 
@@ -478,21 +407,9 @@
 {
 	struct snd_soc_card *card = platform_get_drvdata(pdev);
 	struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card);
-	struct tegra_wm8903_platform_data *pdata = &machine->pdata;
 
-	if (machine->gpio_requested & GPIO_HP_DET)
-		snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack,
-					1,
-					&tegra_wm8903_hp_jack_gpio);
-	if (machine->gpio_requested & GPIO_EXT_MIC_EN)
-		gpio_free(pdata->gpio_ext_mic_en);
-	if (machine->gpio_requested & GPIO_INT_MIC_EN)
-		gpio_free(pdata->gpio_int_mic_en);
-	if (machine->gpio_requested & GPIO_HP_MUTE)
-		gpio_free(pdata->gpio_hp_mute);
-	if (machine->gpio_requested & GPIO_SPKR_EN)
-		gpio_free(pdata->gpio_spkr_en);
-	machine->gpio_requested = 0;
+	snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1,
+				&tegra_wm8903_hp_jack_gpio);
 
 	snd_soc_unregister_card(card);
 
diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c
index 4a8d5b6..e69a4f7 100644
--- a/sound/soc/tegra/trimslice.c
+++ b/sound/soc/tegra/trimslice.c
@@ -52,8 +52,7 @@
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_codec *codec = codec_dai->codec;
 	struct snd_soc_card *card = codec->card;
 	struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card);
 	int srate, mclk;
@@ -68,24 +67,6 @@
 		return err;
 	}
 
-	err = snd_soc_dai_set_fmt(codec_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "codec_dai fmt not set\n");
-		return err;
-	}
-
-	err = snd_soc_dai_set_fmt(cpu_dai,
-					SND_SOC_DAIFMT_I2S |
-					SND_SOC_DAIFMT_NB_NF |
-					SND_SOC_DAIFMT_CBS_CFS);
-	if (err < 0) {
-		dev_err(card->dev, "cpu_dai fmt not set\n");
-		return err;
-	}
-
 	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk,
 					SND_SOC_CLOCK_IN);
 	if (err < 0) {
@@ -121,6 +102,9 @@
 	.cpu_dai_name = "tegra20-i2s.0",
 	.codec_dai_name = "tlv320aic23-hifi",
 	.ops = &trimslice_asoc_ops,
+	.dai_fmt = SND_SOC_DAIFMT_I2S |
+		   SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBS_CFS,
 };
 
 static struct snd_soc_card snd_soc_trimslice = {
@@ -162,9 +146,9 @@
 		}
 
 		trimslice_tlv320aic23_dai.cpu_dai_name = NULL;
-		trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle(
+		trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle(
 				pdev->dev.of_node, "nvidia,i2s-controller", 0);
-		if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) {
+		if (!trimslice_tlv320aic23_dai.cpu_of_node) {
 			dev_err(&pdev->dev,
 				"Property 'nvidia,i2s-controller' missing or invalid\n");
 			ret = -EINVAL;
@@ -173,7 +157,7 @@
 
 		trimslice_tlv320aic23_dai.platform_name = NULL;
 		trimslice_tlv320aic23_dai.platform_of_node =
-				trimslice_tlv320aic23_dai.cpu_dai_of_node;
+				trimslice_tlv320aic23_dai.cpu_of_node;
 	}
 
 	ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev);
diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig
index 44cf434..069330d 100644
--- a/sound/soc/ux500/Kconfig
+++ b/sound/soc/ux500/Kconfig
@@ -12,3 +12,21 @@
 config SND_SOC_UX500_PLAT_MSP_I2S
 	tristate
 	depends on SND_SOC_UX500
+
+config SND_SOC_UX500_PLAT_DMA
+	tristate "Platform - DB8500 (DMA)"
+	depends on SND_SOC_UX500
+	select SND_SOC_DMAENGINE_PCM
+	help
+		Say Y if you want to enable the Ux500 platform-driver.
+
++config SND_SOC_UX500_MACH_MOP500
++	tristate "Machine - MOP500 (Ux500 + AB8500)"
+	depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500
+	select SND_SOC_AB8500_CODEC
+	select SND_SOC_UX500_PLAT_MSP_I2S
+	select SND_SOC_UX500_PLAT_DMA
+	help
+		Select this to enable the MOP500 machine-driver.
+		This will enable platform-drivers for: Ux500
+		This will enable codec-drivers for: AB8500
diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile
index 19974c5..cce0c11 100644
--- a/sound/soc/ux500/Makefile
+++ b/sound/soc/ux500/Makefile
@@ -2,3 +2,9 @@
 
 snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o
 obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o
+
+snd-soc-ux500-plat-dma-objs := ux500_pcm.o
+obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o
+
+snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o
+obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o
diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c
new file mode 100644
index 0000000..31c4d26
--- /dev/null
+++ b/sound/soc/ux500/mop500.c
@@ -0,0 +1,113 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja (ola.o.lilja@stericsson.com)
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <asm/mach-types.h>
+
+#include <linux/module.h>
+#include <linux/io.h>
+#include <linux/spi/spi.h>
+
+#include <sound/soc.h>
+#include <sound/initval.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+
+#include <mop500_ab8500.h>
+
+/* Define the whole MOP500 soundcard, linking platform to the codec-drivers  */
+struct snd_soc_dai_link mop500_dai_links[] = {
+	{
+		.name = "ab8500_0",
+		.stream_name = "ab8500_0",
+		.cpu_dai_name = "ux500-msp-i2s.1",
+		.codec_dai_name = "ab8500-codec-dai.0",
+		.platform_name = "ux500-pcm.0",
+		.codec_name = "ab8500-codec.0",
+		.init = mop500_ab8500_machine_init,
+		.ops = mop500_ab8500_ops,
+	},
+	{
+		.name = "ab8500_1",
+		.stream_name = "ab8500_1",
+		.cpu_dai_name = "ux500-msp-i2s.3",
+		.codec_dai_name = "ab8500-codec-dai.1",
+		.platform_name = "ux500-pcm.0",
+		.codec_name = "ab8500-codec.0",
+		.init = NULL,
+		.ops = mop500_ab8500_ops,
+	},
+};
+
+static struct snd_soc_card mop500_card = {
+	.name = "MOP500-card",
+	.probe = NULL,
+	.dai_link = mop500_dai_links,
+	.num_links = ARRAY_SIZE(mop500_dai_links),
+};
+
+static int __devinit mop500_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	pr_debug("%s: Enter.\n", __func__);
+
+	dev_dbg(&pdev->dev, "%s: Enter.\n", __func__);
+
+	mop500_card.dev = &pdev->dev;
+
+	dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n",
+		__func__, mop500_card.name);
+	platform_set_drvdata(pdev, &mop500_card);
+
+	snd_soc_card_set_drvdata(&mop500_card, NULL);
+
+	dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n",
+		__func__, mop500_card.name, mop500_card.num_links);
+	dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n",
+		__func__, mop500_card.name, mop500_card.dai_link[0].name);
+	dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n",
+		__func__, mop500_card.name,
+		mop500_card.dai_link[0].stream_name);
+
+	ret = snd_soc_register_card(&mop500_card);
+	if (ret)
+		dev_err(&pdev->dev,
+			"Error: snd_soc_register_card failed (%d)!\n",
+			ret);
+
+	return ret;
+}
+
+static int __devexit mop500_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *mop500_card = platform_get_drvdata(pdev);
+
+	pr_debug("%s: Enter.\n", __func__);
+
+	snd_soc_unregister_card(mop500_card);
+	mop500_ab8500_remove(mop500_card);
+	
+	return 0;
+}
+
+static struct platform_driver snd_soc_mop500_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "snd-soc-mop500",
+	},
+	.probe = mop500_probe,
+	.remove = __devexit_p(mop500_remove),
+};
+
+module_platform_driver(snd_soc_mop500_driver);
diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c
new file mode 100644
index 0000000..78cce23
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.c
@@ -0,0 +1,431 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Kristoffer Karlsson <kristoffer.karlsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+
+#include <mach/hardware.h>
+
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "ux500_pcm.h"
+#include "ux500_msp_dai.h"
+#include "../codecs/ab8500-codec.h"
+
+#define TX_SLOT_MONO	0x0008
+#define TX_SLOT_STEREO	0x000a
+#define RX_SLOT_MONO	0x0001
+#define RX_SLOT_STEREO	0x0003
+#define TX_SLOT_8CH	0x00FF
+#define RX_SLOT_8CH	0x00FF
+
+#define DEF_TX_SLOTS	TX_SLOT_STEREO
+#define DEF_RX_SLOTS	RX_SLOT_MONO
+
+#define DRIVERMODE_NORMAL	0
+#define DRIVERMODE_CODEC_ONLY	1
+
+/* Slot configuration */
+static unsigned int tx_slots = DEF_TX_SLOTS;
+static unsigned int rx_slots = DEF_RX_SLOTS;
+
+/* Clocks */
+static const char * const enum_mclk[] = {
+	"SYSCLK",
+	"ULPCLK"
+};
+enum mclk {
+	MCLK_SYSCLK,
+	MCLK_ULPCLK,
+};
+
+static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk);
+
+/* Private data for machine-part MOP500<->AB8500 */
+struct mop500_ab8500_drvdata {
+	/* Clocks */
+	enum mclk mclk_sel;
+	struct clk *clk_ptr_intclk;
+	struct clk *clk_ptr_sysclk;
+	struct clk *clk_ptr_ulpclk;
+};
+
+static inline const char *get_mclk_str(enum mclk mclk_sel)
+{
+	switch (mclk_sel) {
+	case MCLK_SYSCLK:
+		return "SYSCLK";
+	case MCLK_ULPCLK:
+		return "ULPCLK";
+	default:
+		return "Unknown";
+	}
+}
+
+static int mop500_ab8500_set_mclk(struct device *dev,
+				struct mop500_ab8500_drvdata *drvdata)
+{
+	int status;
+	struct clk *clk_ptr;
+
+	if (IS_ERR(drvdata->clk_ptr_intclk)) {
+		dev_err(dev,
+			"%s: ERROR: intclk not initialized!\n", __func__);
+		return -EIO;
+	}
+
+	switch (drvdata->mclk_sel) {
+	case MCLK_SYSCLK:
+		clk_ptr = drvdata->clk_ptr_sysclk;
+		break;
+	case MCLK_ULPCLK:
+		clk_ptr = drvdata->clk_ptr_ulpclk;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (IS_ERR(clk_ptr)) {
+		dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__,
+			get_mclk_str(drvdata->mclk_sel));
+		return -EIO;
+	}
+
+	status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr);
+	if (status)
+		dev_err(dev,
+			"%s: ERROR: Setting intclk parent to %s failed (ret = %d)!",
+			__func__, get_mclk_str(drvdata->mclk_sel), status);
+	else
+		dev_dbg(dev,
+			"%s: intclk parent changed to %s.\n",
+			__func__, get_mclk_str(drvdata->mclk_sel));
+
+	return status;
+}
+
+/*
+ * Control-events
+ */
+
+static int mclk_input_control_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct mop500_ab8500_drvdata *drvdata =
+				snd_soc_card_get_drvdata(codec->card);
+
+	ucontrol->value.enumerated.item[0] = drvdata->mclk_sel;
+
+	return 0;
+}
+
+static int mclk_input_control_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct mop500_ab8500_drvdata *drvdata =
+				snd_soc_card_get_drvdata(codec->card);
+	unsigned int val = ucontrol->value.enumerated.item[0];
+
+	if (val > (unsigned int)MCLK_ULPCLK)
+		return -EINVAL;
+	if (drvdata->mclk_sel == val)
+		return 0;
+
+	drvdata->mclk_sel = val;
+
+	return 1;
+}
+
+/*
+ * Controls
+ */
+
+static struct snd_kcontrol_new mop500_ab8500_ctrls[] = {
+	SOC_ENUM_EXT("Master Clock Select",
+		soc_enum_mclk,
+		mclk_input_control_get, mclk_input_control_put),
+	/* Digital interface - Clocks */
+	SOC_SINGLE("Digital Interface Master Generator Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN,
+		1, 0),
+	SOC_SINGLE("Digital Interface 0 Bit-clock Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0,
+		1, 0),
+	SOC_SINGLE("Digital Interface 1 Bit-clock Switch",
+		AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1,
+		1, 0),
+	SOC_DAPM_PIN_SWITCH("Headset Left"),
+	SOC_DAPM_PIN_SWITCH("Headset Right"),
+	SOC_DAPM_PIN_SWITCH("Earpiece"),
+	SOC_DAPM_PIN_SWITCH("Speaker Left"),
+	SOC_DAPM_PIN_SWITCH("Speaker Right"),
+	SOC_DAPM_PIN_SWITCH("LineOut Left"),
+	SOC_DAPM_PIN_SWITCH("LineOut Right"),
+	SOC_DAPM_PIN_SWITCH("Vibra 1"),
+	SOC_DAPM_PIN_SWITCH("Vibra 2"),
+	SOC_DAPM_PIN_SWITCH("Mic 1"),
+	SOC_DAPM_PIN_SWITCH("Mic 2"),
+	SOC_DAPM_PIN_SWITCH("LineIn Left"),
+	SOC_DAPM_PIN_SWITCH("LineIn Right"),
+	SOC_DAPM_PIN_SWITCH("DMic 1"),
+	SOC_DAPM_PIN_SWITCH("DMic 2"),
+	SOC_DAPM_PIN_SWITCH("DMic 3"),
+	SOC_DAPM_PIN_SWITCH("DMic 4"),
+	SOC_DAPM_PIN_SWITCH("DMic 5"),
+	SOC_DAPM_PIN_SWITCH("DMic 6"),
+};
+
+/* ASoC */
+
+int mop500_ab8500_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* Set audio-clock source */
+	return mop500_ab8500_set_mclk(rtd->card->dev,
+				snd_soc_card_get_drvdata(rtd->card));
+}
+
+void mop500_ab8500_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct device *dev = rtd->card->dev;
+
+	dev_dbg(dev, "%s: Enter\n", __func__);
+
+	/* Reset slots configuration to default(s) */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		tx_slots = DEF_TX_SLOTS;
+	else
+		rx_slots = DEF_RX_SLOTS;
+}
+
+int mop500_ab8500_hw_params(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct device *dev = rtd->card->dev;
+	unsigned int fmt;
+	int channels, ret = 0, driver_mode, slots;
+	unsigned int sw_codec, sw_cpu;
+	bool is_playback;
+
+	dev_dbg(dev, "%s: Enter\n", __func__);
+
+	dev_dbg(dev, "%s: substream->pcm->name = %s\n"
+		"substream->pcm->id = %s.\n"
+		"substream->name = %s.\n"
+		"substream->number = %d.\n",
+		__func__,
+		substream->pcm->name,
+		substream->pcm->id,
+		substream->name,
+		substream->number);
+
+	channels = params_channels(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S32_LE:
+		sw_cpu = 32;
+		break;
+
+	case SNDRV_PCM_FORMAT_S16_LE:
+		sw_cpu = 16;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	/* Setup codec depending on driver-mode */
+	if (channels == 8)
+		driver_mode = DRIVERMODE_CODEC_ONLY;
+	else
+		driver_mode = DRIVERMODE_NORMAL;
+	dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__,
+		(driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY");
+
+	/* Setup format */
+
+	if (driver_mode == DRIVERMODE_NORMAL) {
+		fmt = SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_CBM_CFM |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CONT;
+	} else {
+		fmt = SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_CBM_CFM |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_GATED;
+	}
+
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		dev_err(dev,
+			"%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		dev_err(dev,
+			"%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	/* Setup TDM-slots */
+
+	is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	switch (channels) {
+	case 1:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_MONO : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_MONO;
+		break;
+	case 2:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_STEREO : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO;
+		break;
+	case 8:
+		slots = 16;
+		tx_slots = (is_playback) ? TX_SLOT_8CH : 0;
+		rx_slots = (is_playback) ? 0 : RX_SLOT_8CH;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (driver_mode == DRIVERMODE_NORMAL)
+		sw_codec = sw_cpu;
+	else
+		sw_codec = 20;
+
+	dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+		tx_slots, rx_slots);
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots,
+				sw_cpu);
+	if (ret)
+		return ret;
+
+	dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__,
+		tx_slots, rx_slots);
+	ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots,
+				sw_codec);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+struct snd_soc_ops mop500_ab8500_ops[] = {
+	{
+		.hw_params = mop500_ab8500_hw_params,
+		.startup = mop500_ab8500_startup,
+		.shutdown = mop500_ab8500_shutdown,
+	}
+};
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct device *dev = rtd->card->dev;
+	struct mop500_ab8500_drvdata *drvdata;
+	int ret;
+
+	dev_dbg(dev, "%s Enter.\n", __func__);
+
+	/* Create driver private-data struct */
+	drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata),
+			GFP_KERNEL);
+	snd_soc_card_set_drvdata(rtd->card, drvdata);
+
+	/* Setup clocks */
+
+	drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk");
+	if (IS_ERR(drvdata->clk_ptr_sysclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n",
+			__func__);
+	drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk");
+	if (IS_ERR(drvdata->clk_ptr_ulpclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n",
+			__func__);
+	drvdata->clk_ptr_intclk = clk_get(dev, "intclk");
+	if (IS_ERR(drvdata->clk_ptr_intclk))
+		dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n",
+			__func__);
+
+	/* Set intclk default parent to ulpclk */
+	drvdata->mclk_sel = MCLK_ULPCLK;
+	ret = mop500_ab8500_set_mclk(dev, drvdata);
+	if (ret < 0)
+		dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n",
+			__func__);
+
+	drvdata->mclk_sel = MCLK_ULPCLK;
+
+	/* Add controls */
+	ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls,
+			ARRAY_SIZE(mop500_ab8500_ctrls));
+	if (ret < 0) {
+		pr_err("%s: Failed to add machine-controls (%d)!\n",
+				__func__, ret);
+		return ret;
+	}
+
+	ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5");
+	ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6");
+
+	return ret;
+}
+
+void mop500_ab8500_remove(struct snd_soc_card *card)
+{
+	struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card);
+
+	if (drvdata->clk_ptr_sysclk != NULL)
+		clk_put(drvdata->clk_ptr_sysclk);
+	if (drvdata->clk_ptr_ulpclk != NULL)
+		clk_put(drvdata->clk_ptr_ulpclk);
+	if (drvdata->clk_ptr_intclk != NULL)
+		clk_put(drvdata->clk_ptr_intclk);
+
+	snd_soc_card_set_drvdata(card, drvdata);
+}
diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h
new file mode 100644
index 0000000..cca5b33
--- /dev/null
+++ b/sound/soc/ux500/mop500_ab8500.h
@@ -0,0 +1,22 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#ifndef MOP500_AB8500_H
+#define MOP500_AB8500_H
+
+extern struct snd_soc_ops mop500_ab8500_ops[];
+
+int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime);
+void mop500_ab8500_remove(struct snd_soc_card *card);
+
+#endif
diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c
index 93c6c40..62ac028 100644
--- a/sound/soc/ux500/ux500_msp_dai.c
+++ b/sound/soc/ux500/ux500_msp_dai.c
@@ -840,4 +840,4 @@
 };
 module_platform_driver(msp_i2s_driver);
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c
index 496dec1..ee14d2d 100644
--- a/sound/soc/ux500/ux500_msp_i2s.c
+++ b/sound/soc/ux500/ux500_msp_i2s.c
@@ -739,4 +739,4 @@
 	devm_kfree(&pdev->dev, msp);
 }
 
-MODULE_LICENSE("GPLv2");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c
new file mode 100644
index 0000000..1a04e24
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.c
@@ -0,0 +1,318 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+
+#include <asm/page.h>
+
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/dmaengine.h>
+#include <linux/slab.h>
+
+#include <plat/ste_dma40.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "ux500_msp_i2s.h"
+#include "ux500_pcm.h"
+
+static struct snd_pcm_hardware ux500_pcm_hw_playback = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_RESUME |
+		SNDRV_PCM_INFO_PAUSE,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE |
+		SNDRV_PCM_FMTBIT_U16_LE |
+		SNDRV_PCM_FMTBIT_S16_BE |
+		SNDRV_PCM_FMTBIT_U16_BE,
+	.rates = SNDRV_PCM_RATE_KNOT,
+	.rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK,
+	.rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK,
+	.channels_min = UX500_PLATFORM_MIN_CHANNELS,
+	.channels_max = UX500_PLATFORM_MAX_CHANNELS,
+	.buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+	.period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+	.period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+	.periods_min = UX500_PLATFORM_PERIODS_MIN,
+	.periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static struct snd_pcm_hardware ux500_pcm_hw_capture = {
+	.info = SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_RESUME |
+		SNDRV_PCM_INFO_PAUSE,
+	.formats = SNDRV_PCM_FMTBIT_S16_LE |
+		SNDRV_PCM_FMTBIT_U16_LE |
+		SNDRV_PCM_FMTBIT_S16_BE |
+		SNDRV_PCM_FMTBIT_U16_BE,
+	.rates = SNDRV_PCM_RATE_KNOT,
+	.rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE,
+	.rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE,
+	.channels_min = UX500_PLATFORM_MIN_CHANNELS,
+	.channels_max = UX500_PLATFORM_MAX_CHANNELS,
+	.buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX,
+	.period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN,
+	.period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX,
+	.periods_min = UX500_PLATFORM_PERIODS_MIN,
+	.periods_max = UX500_PLATFORM_PERIODS_MAX,
+};
+
+static void ux500_pcm_dma_hw_free(struct device *dev,
+				struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+
+	if (runtime->dma_area == NULL)
+		return;
+
+	if (buf != &substream->dma_buffer) {
+		dma_free_coherent(buf->dev.dev, buf->bytes, buf->area,
+				buf->addr);
+		kfree(runtime->dma_buffer_p);
+	}
+
+	snd_pcm_set_runtime_buffer(substream, NULL);
+}
+
+static int ux500_pcm_open(struct snd_pcm_substream *substream)
+{
+	int stream_id = substream->pstr->stream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	struct device *dev = dai->dev;
+	int ret;
+	struct ux500_msp_dma_params *dma_params;
+	u16 per_data_width, mem_data_width;
+	struct stedma40_chan_cfg *dma_cfg;
+
+	dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id,
+		snd_pcm_stream_str(substream));
+
+	dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__);
+	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
+		snd_soc_set_runtime_hwparams(substream,
+					&ux500_pcm_hw_playback);
+	else
+		snd_soc_set_runtime_hwparams(substream,
+					&ux500_pcm_hw_capture);
+
+	/* ensure that buffer size is a multiple of period size */
+	ret = snd_pcm_hw_constraint_integer(runtime,
+					SNDRV_PCM_HW_PARAM_PERIODS);
+	if (ret < 0) {
+		dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n",
+			__func__, ret);
+		return ret;
+	}
+
+	dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__,
+		snd_pcm_stream_str(substream));
+	runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ?
+		ux500_pcm_hw_playback : ux500_pcm_hw_capture;
+
+	mem_data_width = STEDMA40_HALFWORD_WIDTH;
+
+	dma_params = snd_soc_dai_get_dma_data(dai, substream);
+	switch (dma_params->data_size) {
+	case 32:
+		per_data_width = STEDMA40_WORD_WIDTH;
+		break;
+	case 16:
+		per_data_width = STEDMA40_HALFWORD_WIDTH;
+		break;
+	case 8:
+		per_data_width = STEDMA40_BYTE_WIDTH;
+		break;
+	default:
+		per_data_width = STEDMA40_WORD_WIDTH;
+		dev_warn(rtd->platform->dev,
+			"%s: Unknown data-size (%d)! Assuming 32 bits.\n",
+			__func__, dma_params->data_size);
+	}
+
+	dma_cfg = dma_params->dma_cfg;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		dma_cfg->src_info.data_width = mem_data_width;
+		dma_cfg->dst_info.data_width = per_data_width;
+	} else {
+		dma_cfg->src_info.data_width = per_data_width;
+		dma_cfg->dst_info.data_width = mem_data_width;
+	}
+
+
+	ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg);
+	if (ret) {
+		dev_dbg(dai->dev,
+			"%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n",
+			__func__, ret);
+		return ret;
+	}
+
+	snd_dmaengine_pcm_set_data(substream, dma_cfg);
+
+	return 0;
+}
+
+static int ux500_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+
+	dev_dbg(dai->dev, "%s: Enter\n", __func__);
+
+	snd_dmaengine_pcm_close(substream);
+
+	return 0;
+}
+
+static int ux500_pcm_hw_params(struct snd_pcm_substream *substream,
+			struct snd_pcm_hw_params *hw_params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_dma_buffer *buf = runtime->dma_buffer_p;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	int ret = 0;
+	int size;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+	size = params_buffer_bytes(hw_params);
+
+	if (buf) {
+		if (buf->bytes >= size)
+			goto out;
+		ux500_pcm_dma_hw_free(NULL, substream);
+	}
+
+	if (substream->dma_buffer.area != NULL &&
+		substream->dma_buffer.bytes >= size) {
+		buf = &substream->dma_buffer;
+	} else {
+		buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL);
+		if (!buf)
+			goto nomem;
+
+		buf->dev.type = SNDRV_DMA_TYPE_DEV;
+		buf->dev.dev = NULL;
+		buf->area = dma_alloc_coherent(NULL, size, &buf->addr,
+					GFP_KERNEL);
+		buf->bytes = size;
+		buf->private_data = NULL;
+
+		if (!buf->area)
+			goto free;
+	}
+	snd_pcm_set_runtime_buffer(substream, buf);
+	ret = 1;
+ out:
+	runtime->dma_bytes = size;
+	return ret;
+
+ free:
+	kfree(buf);
+ nomem:
+	return -ENOMEM;
+}
+
+static int ux500_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__);
+
+	ux500_pcm_dma_hw_free(NULL, substream);
+
+	return 0;
+}
+
+static int ux500_pcm_mmap(struct snd_pcm_substream *substream,
+			struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__);
+
+	return dma_mmap_coherent(NULL, vma, runtime->dma_area,
+				runtime->dma_addr, runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops ux500_pcm_ops = {
+	.open		= ux500_pcm_open,
+	.close		= ux500_pcm_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= ux500_pcm_hw_params,
+	.hw_free	= ux500_pcm_hw_free,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= snd_dmaengine_pcm_pointer_no_residue,
+	.mmap		= ux500_pcm_mmap
+};
+
+int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_pcm *pcm = rtd->pcm;
+
+	dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__,
+		pcm->id);
+
+	pcm->info_flags = 0;
+
+	return 0;
+}
+
+static struct snd_soc_platform_driver ux500_pcm_soc_drv = {
+	.ops		= &ux500_pcm_ops,
+	.pcm_new        = ux500_pcm_new,
+};
+
+static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv);
+	if (ret < 0) {
+		dev_err(&pdev->dev,
+			"%s: ERROR: Failed to register platform '%s' (%d)!\n",
+			__func__, pdev->name, ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_platform(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver ux500_pcm_driver = {
+	.driver = {
+		.name = "ux500-pcm",
+		.owner = THIS_MODULE,
+	},
+
+	.probe = ux500_pcm_drv_probe,
+	.remove = __devexit_p(ux500_pcm_drv_remove),
+};
+module_platform_driver(ux500_pcm_driver);
+
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h
new file mode 100644
index 0000000..77ed44d
--- /dev/null
+++ b/sound/soc/ux500/ux500_pcm.h
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) ST-Ericsson SA 2012
+ *
+ * Author: Ola Lilja <ola.o.lilja@stericsson.com>,
+ *         Roger Nilsson <roger.xr.nilsson@stericsson.com>
+ *         for ST-Ericsson.
+ *
+ * License terms:
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef UX500_PCM_H
+#define UX500_PCM_H
+
+#include <asm/page.h>
+
+#include <linux/workqueue.h>
+
+#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000
+#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000
+#define UX500_PLATFORM_MIN_RATE_CAPTURE	8000
+#define UX500_PLATFORM_MAX_RATE_CAPTURE	48000
+
+#define UX500_PLATFORM_MIN_CHANNELS 1
+#define UX500_PLATFORM_MAX_CHANNELS 8
+
+#define UX500_PLATFORM_PERIODS_BYTES_MIN	128
+#define UX500_PLATFORM_PERIODS_BYTES_MAX	(64 * PAGE_SIZE)
+#define UX500_PLATFORM_PERIODS_MIN		2
+#define UX500_PLATFORM_PERIODS_MAX		48
+#define UX500_PLATFORM_BUFFER_BYTES_MAX		(2048 * PAGE_SIZE)
+
+#endif
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 64aed43..7da0d0a 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -485,7 +485,7 @@
 		     const struct usb_device_id *id)
 {
 	int ret;
-	struct snd_card *card;
+	struct snd_card *card = NULL;
 	struct usb_device *device = interface_to_usbdev(intf);
 
 	ret = create_card(device, intf, &card);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 41f4b69..690000d 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -42,6 +42,13 @@
 
 extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
 
+struct std_mono_table {
+	unsigned int unitid, control, cmask;
+	int val_type;
+	const char *name;
+	snd_kcontrol_tlv_rw_t *tlv_callback;
+};
+
 /* private_free callback */
 static void usb_mixer_elem_free(struct snd_kcontrol *kctl)
 {
@@ -114,6 +121,25 @@
 }
 
 /*
+ * Create a set of standard UAC controls from a table
+ */
+static int snd_create_std_mono_table(struct usb_mixer_interface *mixer,
+				struct std_mono_table *t)
+{
+	int err;
+
+	while (t->name != NULL) {
+		err = snd_create_std_mono_ctl(mixer, t->unitid, t->control,
+				t->cmask, t->val_type, t->name, t->tlv_callback);
+		if (err < 0)
+			return err;
+		t++;
+	}
+
+	return 0;
+}
+
+/*
  * Sound Blaster remote control configuration
  *
  * format of remote control data:
@@ -916,61 +942,6 @@
 	return 0;
 }
 
-
-/*
- * Create mixer for Electrix Ebox-44
- *
- * The mixer units from this device are corrupt, and even where they
- * are valid they presents mono controls as L and R channels of
- * stereo. So we create a good mixer in code.
- */
-
-static int snd_ebox44_create_mixer(struct usb_mixer_interface *mixer)
-{
-	int err;
-
-	err = snd_create_std_mono_ctl(mixer, 4, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Headphone Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x1, USB_MIXER_S16,
-				"Headphone A Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 4, 2, 0x2, USB_MIXER_S16,
-				"Headphone B Mix Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 7, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Output Playback Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x1, USB_MIXER_S16,
-				"Output A Playback Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 7, 2, 0x2, USB_MIXER_S16,
-				"Output B Playback Volume", NULL);
-	if (err < 0)
-		return err;
-
-	err = snd_create_std_mono_ctl(mixer, 10, 1, 0x0, USB_MIXER_INV_BOOLEAN,
-				"Input Capture Switch", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x1, USB_MIXER_S16,
-				"Input A Capture Volume", NULL);
-	if (err < 0)
-		return err;
-	err = snd_create_std_mono_ctl(mixer, 10, 2, 0x2, USB_MIXER_S16,
-				"Input B Capture Volume", NULL);
-	if (err < 0)
-		return err;
-
-	return 0;
-}
-
 void snd_emuusb_set_samplerate(struct snd_usb_audio *chip,
 			       unsigned char samplerate_id)
 {
@@ -990,6 +961,81 @@
 	}
 }
 
+/*
+ * The mixer units for Ebox-44 are corrupt, and even where they
+ * are valid they presents mono controls as L and R channels of
+ * stereo. So we provide a good mixer here.
+ */
+struct std_mono_table ebox44_table[] = {
+	{
+		.unitid = 4,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Headphone Playback Switch"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone A Mix Playback Volume"
+	},
+	{
+		.unitid = 4,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Headphone B Mix Playback Volume"
+	},
+
+	{
+		.unitid = 7,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Output Playback Switch"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Output A Playback Volume"
+	},
+	{
+		.unitid = 7,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Output B Playback Volume"
+	},
+
+	{
+		.unitid = 10,
+		.control = 1,
+		.cmask = 0x0,
+		.val_type = USB_MIXER_INV_BOOLEAN,
+		.name = "Input Capture Switch"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x1,
+		.val_type = USB_MIXER_S16,
+		.name = "Input A Capture Volume"
+	},
+	{
+		.unitid = 10,
+		.control = 2,
+		.cmask = 0x2,
+		.val_type = USB_MIXER_S16,
+		.name = "Input B Capture Volume"
+	},
+
+	{}
+};
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -1035,7 +1081,8 @@
 		break;
 
 	case USB_ID(0x200c, 0x1018): /* Electrix Ebox-44 */
-		err = snd_ebox44_create_mixer(mixer);
+		/* detection is disabled in mixer_maps.c */
+		err = snd_create_std_mono_table(mixer, ebox44_table);
 		break;
 	}