diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index 9eab263..466b1c7 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -103,13 +103,21 @@
         unsigned int lineout1fb:1;
         unsigned int lineout2fb:1;
 
-        /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+	/* IRQ for microphone detection if brought out directly as a
+	 * signal.
+	 */
+	int micdet_irq;
+
+        /* WM8994 microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
         unsigned int micbias1_lvl:1;
         unsigned int micbias2_lvl:1;
 
-        /* Jack detect threashold levels, see datasheet for values */
+        /* WM8994 jack detect threashold levels, see datasheet for values */
         unsigned int jd_scthr:2;
         unsigned int jd_thr:2;
+
+	/* WM8958 microphone bias configuration */
+	int micbias[2];
 };
 
 #endif
diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h
index be072fa..f3ee842 100644
--- a/include/linux/mfd/wm8994/registers.h
+++ b/include/linux/mfd/wm8994/registers.h
@@ -63,6 +63,8 @@
 #define WM8994_MICBIAS                          0x3A
 #define WM8994_LDO_1                            0x3B
 #define WM8994_LDO_2                            0x3C
+#define WM8958_MICBIAS1				0x3D
+#define WM8958_MICBIAS2				0x3E
 #define WM8994_CHARGE_PUMP_1                    0x4C
 #define WM8958_CHARGE_PUMP_2                    0x4D
 #define WM8994_CLASS_W_1                        0x51
diff --git a/include/sound/control.h b/include/sound/control.h
index 7715e6f..e67db28 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -115,6 +115,8 @@
 int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol);
 int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id);
 int snd_ctl_rename_id(struct snd_card * card, struct snd_ctl_elem_id *src_id, struct snd_ctl_elem_id *dst_id);
+int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
+			int active);
 struct snd_kcontrol *snd_ctl_find_numid(struct snd_card * card, unsigned int numid);
 struct snd_kcontrol *snd_ctl_find_id(struct snd_card * card, struct snd_ctl_elem_id *id);
 
diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h
index 16f8d32..50a059e 100644
--- a/include/sound/cs4271.h
+++ b/include/sound/cs4271.h
@@ -19,7 +19,6 @@
 
 struct cs4271_platform_data {
 	int gpio_nreset;	/* GPIO driving Reset pin, if any */
-	int gpio_disable;	/* GPIO that disable serial bus, if any */
 };
 
 #endif /* __CS4271_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4b6c0a8..bfa4836 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -234,6 +234,7 @@
 struct snd_soc_codec_driver;
 struct soc_enum;
 struct snd_soc_jack;
+struct snd_soc_jack_zone;
 struct snd_soc_jack_pin;
 struct snd_soc_cache_ops;
 #include <sound/soc-dapm.h>
@@ -258,6 +259,11 @@
 	SND_SOC_RBTREE_COMPRESSION
 };
 
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+			     unsigned int freq, int dir);
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+			  unsigned int freq_in, unsigned int freq_out);
+
 int snd_soc_register_card(struct snd_soc_card *card);
 int snd_soc_unregister_card(struct snd_soc_card *card);
 int snd_soc_suspend(struct device *dev);
@@ -307,6 +313,9 @@
 				    struct notifier_block *nb);
 void snd_soc_jack_notifier_unregister(struct snd_soc_jack *jack,
 				      struct notifier_block *nb);
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+			  struct snd_soc_jack_zone *zones);
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage);
 #ifdef CONFIG_GPIOLIB
 int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
 			struct snd_soc_jack_gpio *gpios);
@@ -331,7 +340,8 @@
  *Controls
  */
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
-	void *data, char *long_name);
+				  void *data, char *long_name,
+				  const char *prefix);
 int snd_soc_add_controls(struct snd_soc_codec *codec,
 	const struct snd_kcontrol_new *controls, int num_controls);
 int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol,
@@ -407,6 +417,24 @@
 };
 
 /**
+ * struct snd_soc_jack_zone - Describes voltage zones of jack detection
+ *
+ * @min_mv: start voltage in mv
+ * @max_mv: end voltage in mv
+ * @jack_type: type of jack that is expected for this voltage
+ * @debounce_time: debounce_time for jack, codec driver should wait for this
+ *		duration before reading the adc for voltages
+ * @:list: list container
+ */
+struct snd_soc_jack_zone {
+	unsigned int min_mv;
+	unsigned int max_mv;
+	unsigned int jack_type;
+	unsigned int debounce_time;
+	struct list_head list;
+};
+
+/**
  * struct snd_soc_jack_gpio - Describes a gpio pin for jack detection
  *
  * @gpio:         gpio number
@@ -414,6 +442,10 @@
  * @report:       value to report when jack detected
  * @invert:       report presence in low state
  * @debouce_time: debouce time in ms
+ * @wake:	  enable as wake source
+ * @jack_status_check: callback function which overrides the detection
+ *		       to provide more complex checks (eg, reading an
+ *		       ADC).
  */
 #ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio {
@@ -422,6 +454,8 @@
 	int report;
 	int invert;
 	int debounce_time;
+	bool wake;
+
 	struct snd_soc_jack *jack;
 	struct delayed_work work;
 
@@ -435,6 +469,7 @@
 	struct list_head pins;
 	int status;
 	struct blocking_notifier_head notifier;
+	struct list_head jack_zones;
 };
 
 /* SoC PCM stream information */
@@ -533,6 +568,18 @@
 			pm_message_t state);
 	int (*resume)(struct snd_soc_codec *);
 
+	/* Default DAPM setup, added after probe() is run */
+	const struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	const struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
+	/* codec wide operations */
+	int (*set_sysclk)(struct snd_soc_codec *codec,
+			  int clk_id, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source,
+		unsigned int freq_in, unsigned int freq_out);
+
 	/* codec IO */
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
@@ -653,6 +700,7 @@
 	bool instantiated;
 
 	int (*probe)(struct snd_soc_card *card);
+	int (*late_probe)(struct snd_soc_card *card);
 	int (*remove)(struct snd_soc_card *card);
 
 	/* the pre and post PM functions are used to do any PM work before and
@@ -689,6 +737,14 @@
 	struct snd_soc_pcm_runtime *rtd_aux;
 	int num_aux_rtd;
 
+	/*
+	 * Card-specific routes and widgets.
+	 */
+	struct snd_soc_dapm_widget *dapm_widgets;
+	int num_dapm_widgets;
+	struct snd_soc_dapm_route *dapm_routes;
+	int num_dapm_routes;
+
 	struct work_struct deferred_resume_work;
 
 	/* lists of probed devices belonging to this card */
@@ -700,6 +756,9 @@
 	struct list_head paths;
 	struct list_head dapm_list;
 
+	/* Generic DAPM context for the card */
+	struct snd_soc_dapm_context dapm;
+
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_card_root;
 	struct dentry *debugfs_pop_time;
diff --git a/include/sound/tlv320aic32x4.h b/include/sound/tlv320aic32x4.h
new file mode 100644
index 0000000..c009f70
--- /dev/null
+++ b/include/sound/tlv320aic32x4.h
@@ -0,0 +1,31 @@
+/*
+ * tlv320aic32x4.h  --  TLV320AIC32X4 Soc Audio driver platform data
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.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 _AIC32X4_PDATA_H
+#define _AIC32X4_PDATA_H
+
+#define AIC32X4_PWR_MICBIAS_2075_LDOIN		0x00000001
+#define AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE	0x00000002
+#define AIC32X4_PWR_AIC32X4_LDO_ENABLE		0x00000004
+#define AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36	0x00000008
+#define AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED	0x00000010
+
+#define AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K	0x00000001
+#define AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K	0x00000002
+
+struct aic32x4_pdata {
+	u32 power_cfg;
+	u32 micpga_routing;
+	bool swapdacs;
+};
+
+#endif
diff --git a/include/sound/wm8903.h b/include/sound/wm8903.h
index 86172cf..cf7ccb7 100644
--- a/include/sound/wm8903.h
+++ b/include/sound/wm8903.h
@@ -17,13 +17,9 @@
 /*
  * R6 (0x06) - Mic Bias Control 0
  */
-#define WM8903_MICDET_HYST_ENA                  0x0080  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_MASK             0x0080  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_SHIFT                 7  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_HYST_ENA_WIDTH                 1  /* MICDET_HYST_ENA */
-#define WM8903_MICDET_THR_MASK                  0x0070  /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_SHIFT                      4  /* MICDET_THR - [6:4] */
-#define WM8903_MICDET_THR_WIDTH                      3  /* MICDET_THR - [6:4] */
+#define WM8903_MICDET_THR_MASK                  0x0030  /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_SHIFT                      4  /* MICDET_THR - [5:4] */
+#define WM8903_MICDET_THR_WIDTH                      2  /* MICDET_THR - [5:4] */
 #define WM8903_MICSHORT_THR_MASK                0x000C  /* MICSHORT_THR - [3:2] */
 #define WM8903_MICSHORT_THR_SHIFT                    2  /* MICSHORT_THR - [3:2] */
 #define WM8903_MICSHORT_THR_WIDTH                    2  /* MICSHORT_THR - [3:2] */
diff --git a/include/sound/wm9081.h b/include/sound/wm9081.h
index e173ddb..f34b0b1 100644
--- a/include/sound/wm9081.h
+++ b/include/sound/wm9081.h
@@ -17,9 +17,12 @@
 	u16 config[20];
 };
 
-struct wm9081_retune_mobile_config {
-	struct wm9081_retune_mobile_setting *configs;
-	int num_configs;
+struct wm9081_pdata {
+	bool irq_high;   /* IRQ is active high */
+	bool irq_cmos;   /* IRQ is in CMOS mode */
+
+	struct wm9081_retune_mobile_setting *retune_configs;
+	int num_retune_configs;
 };
 
 #endif
diff --git a/sound/core/control.c b/sound/core/control.c
index 9ce00ed..db51e4e 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -466,6 +466,52 @@
 }
 
 /**
+ * snd_ctl_activate_id - activate/inactivate the control of the given id
+ * @card: the card instance
+ * @id: the control id to activate/inactivate
+ * @active: non-zero to activate
+ *
+ * Finds the control instance with the given id, and activate or
+ * inactivate the control together with notification, if changed.
+ *
+ * Returns 0 if unchanged, 1 if changed, or a negative error code on failure.
+ */
+int snd_ctl_activate_id(struct snd_card *card, struct snd_ctl_elem_id *id,
+			int active)
+{
+	struct snd_kcontrol *kctl;
+	struct snd_kcontrol_volatile *vd;
+	unsigned int index_offset;
+	int ret;
+
+	down_write(&card->controls_rwsem);
+	kctl = snd_ctl_find_id(card, id);
+	if (kctl == NULL) {
+		ret = -ENOENT;
+		goto unlock;
+	}
+	index_offset = snd_ctl_get_ioff(kctl, &kctl->id);
+	vd = &kctl->vd[index_offset];
+	ret = 0;
+	if (active) {
+		if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE))
+			goto unlock;
+		vd->access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+	} else {
+		if (vd->access & SNDRV_CTL_ELEM_ACCESS_INACTIVE)
+			goto unlock;
+		vd->access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE;
+	}
+	ret = 1;
+ unlock:
+	up_write(&card->controls_rwsem);
+	if (ret > 0)
+		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_INFO, id);
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_ctl_activate_id);
+
+/**
  * snd_ctl_rename_id - replace the id of a control on the card
  * @card: the card instance
  * @src_id: the old id
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e239345..d63c175 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -29,16 +29,21 @@
 	select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_CX20442
 	select SND_SOC_DA7210 if I2C
+	select SND_SOC_DFBMCS320
 	select SND_SOC_JZ4740_CODEC if SOC_JZ4740
+	select SND_SOC_LM4857 if I2C
 	select SND_SOC_MAX98088 if I2C
+	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
+	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
+	select SND_SOC_TVL320AIC32X4 if I2C
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
@@ -173,15 +178,25 @@
 config SND_SOC_DA7210
         tristate
 
+config SND_SOC_DFBMCS320
+	tristate
+
 config SND_SOC_DMIC
 	tristate
 
 config SND_SOC_MAX98088
        tristate
 
+config SND_SOC_MAX9850
+	tristate
+
 config SND_SOC_PCM3008
        tristate
 
+#Freescale sgtl5000 codec
+config SND_SOC_SGTL5000
+	tristate
+
 config SND_SOC_SN95031
 	tristate
 
@@ -201,6 +216,9 @@
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
 	depends on SPI
 
+config SND_SOC_TVL320AIC32X4
+	tristate
+
 config SND_SOC_TLV320AIC3X
 	tristate
 
@@ -338,6 +356,9 @@
 	tristate
 
 # Amp
+config SND_SOC_LM4857
+	tristate
+
 config SND_SOC_MAX9877
 	tristate
 
@@ -349,4 +370,3 @@
 
 config SND_SOC_WM9090
 	tristate
-
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 83b7acc..379bc55 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -15,10 +15,13 @@
 snd-soc-cs4271-objs := cs4271.o
 snd-soc-cx20442-objs := cx20442.o
 snd-soc-da7210-objs := da7210.o
+snd-soc-dfbmcs320-objs := dfbmcs320.o
 snd-soc-dmic-objs := dmic.o
 snd-soc-l3-objs := l3.o
 snd-soc-max98088-objs := max98088.o
+snd-soc-max9850-objs := max9850.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
 snd-soc-sn95031-objs := sn95031.o
 snd-soc-spdif-objs := spdif_transciever.o
@@ -27,6 +30,7 @@
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
+snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o
 snd-soc-tlv320dac33-objs := tlv320dac33.o
 snd-soc-twl4030-objs := twl4030.o
 snd-soc-twl6040-objs := twl6040.o
@@ -75,6 +79,7 @@
 snd-soc-jz4740-codec-objs := jz4740.o
 
 # Amp
+snd-soc-lm4857-objs := lm4857.o
 snd-soc-max9877-objs := max9877.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
@@ -91,18 +96,21 @@
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o
 obj-$(CONFIG_SND_SOC_CS42L51)	+= snd-soc-cs42l51.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(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-$(CONFIG_SND_SOC_DFBMCS320)	+= snd-soc-dfbmcs320.o
 obj-$(CONFIG_SND_SOC_DMIC)	+= snd-soc-dmic.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_JZ4740_CODEC)	+= snd-soc-jz4740-codec.o
 obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
+obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
-obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
+obj-$(CONFIG_SND_SOC_SGTL5000)  += snd-soc-sgtl5000.o
 obj-$(CONFIG_SND_SOC_SN95031)	+=snd-soc-sn95031.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
@@ -110,6 +118,7 @@
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
+obj-$(CONFIG_SND_SOC_TVL320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
@@ -157,6 +166,7 @@
 obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 
 # Amp
+obj-$(CONFIG_SND_SOC_LM4857)	+= snd-soc-lm4857.o
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
 obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index c27f8f5..cbf0b6d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -294,7 +294,6 @@
 
 static int __init ak4104_init(void)
 {
-	pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
 	return spi_register_driver(&ak4104_spi_driver);
 }
 module_init(ak4104_init);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index c0fccad..0206a17 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -719,7 +719,7 @@
 /*
  * cs4270_id - I2C device IDs supported by this driver
  */
-static struct i2c_device_id cs4270_id[] = {
+static const struct i2c_device_id cs4270_id[] = {
 	{"cs4270", 0},
 	{}
 };
@@ -743,8 +743,6 @@
 
 static int __init cs4270_init(void)
 {
-	pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
-
 	return i2c_add_driver(&cs4270_i2c_driver);
 }
 module_init(cs4270_init);
diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c
index 9c5b7db..083aab9 100644
--- a/sound/soc/codecs/cs4271.c
+++ b/sound/soc/codecs/cs4271.c
@@ -33,6 +33,7 @@
 #define CS4271_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
 			    SNDRV_PCM_FMTBIT_S24_LE | \
 			    SNDRV_PCM_FMTBIT_S32_LE)
+#define CS4271_PCM_RATES SNDRV_PCM_RATE_8000_192000
 
 /*
  * CS4271 registers
@@ -167,27 +168,6 @@
 	int				gpio_disable;
 };
 
-struct cs4271_clk_cfg {
-	unsigned int	ratio;		/* MCLK / sample rate */
-	u8		speed_mode;	/* codec speed mode: 1x, 2x, 4x */
-	u8		mclk_master;	/* ratio bit mask for Master mode */
-	u8		mclk_slave;	/* ratio bit mask for Slave mode */
-};
-
-static struct cs4271_clk_cfg cs4271_clk_tab[] = {
-	{64,   CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_1,  CS4271_MODE1_DIV_1},
-	{96,   CS4271_MODE1_MODE_4X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
-	{128,  CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_1,  CS4271_MODE1_DIV_1},
-	{192,  CS4271_MODE1_MODE_2X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
-	{256,  CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_1,  CS4271_MODE1_DIV_1},
-	{384,  CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_15, CS4271_MODE1_DIV_1},
-	{512,  CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_2,  CS4271_MODE1_DIV_1},
-	{768,  CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3,  CS4271_MODE1_DIV_3},
-	{1024, CS4271_MODE1_MODE_1X, CS4271_MODE1_DIV_3,  CS4271_MODE1_DIV_3}
-};
-
-#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
-
 /*
  * @freq is the desired MCLK rate
  * MCLK rate should (c) be the sample rate, multiplied by one of the
@@ -296,6 +276,45 @@
 	return cs4271_set_deemph(codec);
 }
 
+struct cs4271_clk_cfg {
+	bool		master;		/* codec mode */
+	u8		speed_mode;	/* codec speed mode: 1x, 2x, 4x */
+	unsigned short	ratio;		/* MCLK / sample rate */
+	u8		ratio_mask;	/* ratio bit mask for Master mode */
+};
+
+static struct cs4271_clk_cfg cs4271_clk_tab[] = {
+	{1, CS4271_MODE1_MODE_1X, 256,  CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_1X, 384,  CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_1X, 512,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_1X, 768,  CS4271_MODE1_DIV_3},
+	{1, CS4271_MODE1_MODE_2X, 128,  CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_2X, 192,  CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_2X, 256,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_2X, 384,  CS4271_MODE1_DIV_3},
+	{1, CS4271_MODE1_MODE_4X, 64,   CS4271_MODE1_DIV_1},
+	{1, CS4271_MODE1_MODE_4X, 96,   CS4271_MODE1_DIV_15},
+	{1, CS4271_MODE1_MODE_4X, 128,  CS4271_MODE1_DIV_2},
+	{1, CS4271_MODE1_MODE_4X, 192,  CS4271_MODE1_DIV_3},
+	{0, CS4271_MODE1_MODE_1X, 256,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 384,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 512,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_1X, 768,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_1X, 1024, CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_2X, 128,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 192,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 256,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_2X, 384,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_2X, 512,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_4X, 64,   CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 96,   CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 128,  CS4271_MODE1_DIV_1},
+	{0, CS4271_MODE1_MODE_4X, 192,  CS4271_MODE1_DIV_2},
+	{0, CS4271_MODE1_MODE_4X, 256,  CS4271_MODE1_DIV_2},
+};
+
+#define CS4171_NR_RATIOS ARRAY_SIZE(cs4271_clk_tab)
+
 static int cs4271_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_pcm_hw_params *params,
 			    struct snd_soc_dai *dai)
@@ -307,23 +326,28 @@
 	unsigned int ratio, val;
 
 	cs4271->rate = params_rate(params);
+
+	/* Configure DAC */
+	if (cs4271->rate < 50000)
+		val = CS4271_MODE1_MODE_1X;
+	else if (cs4271->rate < 100000)
+		val = CS4271_MODE1_MODE_2X;
+	else
+		val = CS4271_MODE1_MODE_4X;
+
 	ratio = cs4271->mclk / cs4271->rate;
 	for (i = 0; i < CS4171_NR_RATIOS; i++)
-		if (cs4271_clk_tab[i].ratio == ratio)
+		if ((cs4271_clk_tab[i].master == cs4271->master) &&
+		    (cs4271_clk_tab[i].speed_mode == val) &&
+		    (cs4271_clk_tab[i].ratio == ratio))
 			break;
 
-	if ((i == CS4171_NR_RATIOS) || ((ratio == 1024) && cs4271->master)) {
+	if (i == CS4171_NR_RATIOS) {
 		dev_err(codec->dev, "Invalid sample rate\n");
 		return -EINVAL;
 	}
 
-	/* Configure DAC */
-	val = cs4271_clk_tab[i].speed_mode;
-
-	if (cs4271->master)
-		val |= cs4271_clk_tab[i].mclk_master;
-	else
-		val |= cs4271_clk_tab[i].mclk_slave;
+	val |= cs4271_clk_tab[i].ratio_mask;
 
 	ret = snd_soc_update_bits(codec, CS4271_MODE1,
 		CS4271_MODE1_MODE_MASK | CS4271_MODE1_DIV_MASK, val);
@@ -392,14 +416,14 @@
 		.stream_name	= "Playback",
 		.channels_min	= 2,
 		.channels_max	= 2,
-		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.rates		= CS4271_PCM_RATES,
 		.formats	= CS4271_PCM_FORMATS,
 	},
 	.capture = {
 		.stream_name	= "Capture",
 		.channels_min	= 2,
 		.channels_max	= 2,
-		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.rates		= CS4271_PCM_RATES,
 		.formats	= CS4271_PCM_FORMATS,
 	},
 	.ops = &cs4271_dai_ops,
@@ -441,22 +465,11 @@
 	struct cs4271_platform_data *cs4271plat = codec->dev->platform_data;
 	int ret;
 	int gpio_nreset = -EINVAL;
-	int gpio_disable = -EINVAL;
 
 	codec->control_data = cs4271->control_data;
 
-	if (cs4271plat) {
-		if (gpio_is_valid(cs4271plat->gpio_nreset))
-			gpio_nreset = cs4271plat->gpio_nreset;
-		if (gpio_is_valid(cs4271plat->gpio_disable))
-			gpio_disable = cs4271plat->gpio_disable;
-	}
-
-	if (gpio_disable >= 0)
-		if (gpio_request(gpio_disable, "CS4271 Disable"))
-			gpio_disable = -EINVAL;
-	if (gpio_disable >= 0)
-		gpio_direction_output(gpio_disable, 0);
+	if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset))
+		gpio_nreset = cs4271plat->gpio_nreset;
 
 	if (gpio_nreset >= 0)
 		if (gpio_request(gpio_nreset, "CS4271 Reset"))
@@ -471,7 +484,6 @@
 	}
 
 	cs4271->gpio_nreset = gpio_nreset;
-	cs4271->gpio_disable = gpio_disable;
 
 	/*
 	 * In case of I2C, chip address specified in board data.
@@ -509,10 +521,9 @@
 static int cs4271_remove(struct snd_soc_codec *codec)
 {
 	struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec);
-	int gpio_nreset, gpio_disable;
+	int gpio_nreset;
 
 	gpio_nreset = cs4271->gpio_nreset;
-	gpio_disable = cs4271->gpio_disable;
 
 	if (gpio_is_valid(gpio_nreset)) {
 		/* Set codec to the reset state */
@@ -520,9 +531,6 @@
 		gpio_free(gpio_nreset);
 	}
 
-	if (gpio_is_valid(gpio_disable))
-		gpio_free(gpio_disable);
-
 	return 0;
 };
 
@@ -571,7 +579,7 @@
 #endif /* defined(CONFIG_SPI_MASTER) */
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-static struct i2c_device_id cs4271_i2c_id[] = {
+static const struct i2c_device_id cs4271_i2c_id[] = {
 	{"cs4271", 0},
 	{}
 };
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index bb4bf65..0bb424a 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -367,7 +367,7 @@
 	return 0;
 }
 
-static const u8 cx20442_reg = CX20442_TELOUT | CX20442_MIC;
+static const u8 cx20442_reg;
 
 static struct snd_soc_codec_driver cx20442_codec_dev = {
 	.probe = 	cx20442_codec_probe,
diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/dfbmcs320.c
new file mode 100644
index 0000000..704bbde
--- /dev/null
+++ b/sound/soc/codecs/dfbmcs320.c
@@ -0,0 +1,72 @@
+/*
+ * Driver for the DFBM-CS320 bluetooth module
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+
+#include <sound/soc.h>
+
+static struct snd_soc_dai_driver dfbmcs320_dai = {
+	.name = "dfbmcs320-pcm",
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+};
+
+static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320;
+
+static int __devinit dfbmcs320_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320,
+			&dfbmcs320_dai, 1);
+}
+
+static int __devexit dfbmcs320_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+
+	return 0;
+}
+
+static struct platform_driver dfmcs320_driver = {
+	.driver = {
+		.name = "dfbmcs320",
+		.owner = THIS_MODULE,
+	},
+	.probe = dfbmcs320_probe,
+	.remove = __devexit_p(dfbmcs320_remove),
+};
+
+static int __init dfbmcs320_init(void)
+{
+	return platform_driver_register(&dfmcs320_driver);
+}
+module_init(dfbmcs320_init);
+
+static void __exit dfbmcs320_exit(void)
+{
+	platform_driver_unregister(&dfmcs320_driver);
+}
+module_exit(dfbmcs320_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c
new file mode 100644
index 0000000..72de47e
--- /dev/null
+++ b/sound/soc/codecs/lm4857.c
@@ -0,0 +1,276 @@
+/*
+ * LM4857 AMP driver
+ *
+ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
+ *
+ *  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/init.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+struct lm4857 {
+	struct i2c_client *i2c;
+	uint8_t mode;
+};
+
+static const uint8_t lm4857_default_regs[] = {
+	0x00, 0x00, 0x00, 0x00,
+};
+
+/* The register offsets in the cache array */
+#define LM4857_MVOL 0
+#define LM4857_LVOL 1
+#define LM4857_RVOL 2
+#define LM4857_CTRL 3
+
+/* the shifts required to set these bits */
+#define LM4857_3D 5
+#define LM4857_WAKEUP 5
+#define LM4857_EPGAIN 4
+
+static int lm4857_write(struct snd_soc_codec *codec, unsigned int reg,
+		unsigned int value)
+{
+	uint8_t data;
+	int ret;
+
+	ret = snd_soc_cache_write(codec, reg, value);
+	if (ret < 0)
+		return ret;
+
+	data = (reg << 6) | value;
+	ret = i2c_master_send(codec->control_data, &data, 1);
+	if (ret != 1) {
+		dev_err(codec->dev, "Failed to write register: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static unsigned int lm4857_read(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	unsigned int val;
+	int ret;
+
+	ret = snd_soc_cache_read(codec, reg, &val);
+	if (ret)
+		return -1;
+
+	return val;
+}
+
+static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.integer.value[0] = lm4857->mode;
+
+	return 0;
+}
+
+static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+	uint8_t value = ucontrol->value.integer.value[0];
+
+	lm4857->mode = value;
+
+	if (codec->dapm.bias_level == SND_SOC_BIAS_ON)
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, value + 6);
+
+	return 1;
+}
+
+static int lm4857_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, lm4857->mode + 6);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, LM4857_CTRL, 0x0F, 0);
+		break;
+	default:
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
+
+static const char *lm4857_mode[] = {
+	"Earpiece",
+	"Loudspeaker",
+	"Loudspeaker + Headphone",
+	"Headphone",
+};
+
+static const struct soc_enum lm4857_mode_enum =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode);
+
+static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("IN"),
+
+	SND_SOC_DAPM_OUTPUT("LS"),
+	SND_SOC_DAPM_OUTPUT("HP"),
+	SND_SOC_DAPM_OUTPUT("EP"),
+};
+
+static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+
+static const struct snd_kcontrol_new lm4857_controls[] = {
+	SOC_SINGLE_TLV("Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+		stereo_tlv),
+	SOC_SINGLE_TLV("Right Playback Volume", LM4857_RVOL, 0, 31, 0,
+		stereo_tlv),
+	SOC_SINGLE_TLV("Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
+		mono_tlv),
+	SOC_SINGLE("Spk 3D Playback Switch", LM4857_LVOL, LM4857_3D, 1, 0),
+	SOC_SINGLE("HP 3D Playback Switch", LM4857_RVOL, LM4857_3D, 1, 0),
+	SOC_SINGLE("Fast Wakeup Playback Switch", LM4857_CTRL,
+		LM4857_WAKEUP, 1, 0),
+	SOC_SINGLE("Earpiece 6dB Playback Switch", LM4857_CTRL,
+		LM4857_EPGAIN, 1, 0),
+
+	SOC_ENUM_EXT("Mode", lm4857_mode_enum,
+		lm4857_get_mode, lm4857_set_mode),
+};
+
+/* There is a demux inbetween the the input signal and the output signals.
+ * Currently there is no easy way to model it in ASoC and since it does not make
+ * much of a difference in practice simply connect the input direclty to the
+ * outputs. */
+static const struct snd_soc_dapm_route lm4857_routes[] = {
+	{"LS", NULL, "IN"},
+	{"HP", NULL, "IN"},
+	{"EP", NULL, "IN"},
+};
+
+static int lm4857_probe(struct snd_soc_codec *codec)
+{
+	struct lm4857 *lm4857 = snd_soc_codec_get_drvdata(codec);
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	codec->control_data = lm4857->i2c;
+
+	ret = snd_soc_add_controls(codec, lm4857_controls,
+			ARRAY_SIZE(lm4857_controls));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, lm4857_dapm_widgets,
+			ARRAY_SIZE(lm4857_dapm_widgets));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, lm4857_routes,
+			ARRAY_SIZE(lm4857_routes));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_new_widgets(dapm);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_lm4857 = {
+	.write = lm4857_write,
+	.read = lm4857_read,
+	.probe = lm4857_probe,
+	.reg_cache_size = ARRAY_SIZE(lm4857_default_regs),
+	.reg_word_size = sizeof(uint8_t),
+	.reg_cache_default = lm4857_default_regs,
+	.set_bias_level = lm4857_set_bias_level,
+};
+
+static int __devinit lm4857_i2c_probe(struct i2c_client *i2c,
+	const struct i2c_device_id *id)
+{
+	struct lm4857 *lm4857;
+	int ret;
+
+	lm4857 = kzalloc(sizeof(*lm4857), GFP_KERNEL);
+	if (!lm4857)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, lm4857);
+
+	lm4857->i2c = i2c;
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_lm4857, NULL, 0);
+
+	if (ret) {
+		kfree(lm4857);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit lm4857_i2c_remove(struct i2c_client *i2c)
+{
+	struct lm4857 *lm4857 = i2c_get_clientdata(i2c);
+
+	snd_soc_unregister_codec(&i2c->dev);
+	kfree(lm4857);
+
+	return 0;
+}
+
+static const struct i2c_device_id lm4857_i2c_id[] = {
+	{ "lm4857", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, lm4857_i2c_id);
+
+static struct i2c_driver lm4857_i2c_driver = {
+	.driver = {
+		.name = "lm4857",
+		.owner = THIS_MODULE,
+	},
+	.probe = lm4857_i2c_probe,
+	.remove = __devexit_p(lm4857_i2c_remove),
+	.id_table = lm4857_i2c_id,
+};
+
+static int __init lm4857_init(void)
+{
+	return i2c_add_driver(&lm4857_i2c_driver);
+}
+module_init(lm4857_init);
+
+static void __exit lm4857_exit(void)
+{
+	i2c_del_driver(&lm4857_i2c_driver);
+}
+module_exit(lm4857_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("LM4857 amplifier driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c
new file mode 100644
index 0000000..208d2ee
--- /dev/null
+++ b/sound/soc/codecs/max9850.c
@@ -0,0 +1,389 @@
+/*
+ * max9850.c  --  codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ *
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * Initial development of this code was funded by
+ * MICRONIC Computer Systeme GmbH, http://www.mcsberlin.de/
+ *
+ * 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/i2c.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+
+#include "max9850.h"
+
+struct max9850_priv {
+	unsigned int sysclk;
+};
+
+/* max9850 register cache */
+static const u8 max9850_reg[MAX9850_CACHEREGNUM] = {
+	0x00, 0x00, 0x0c, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
+};
+
+/* these registers are not used at the moment but provided for the sake of
+ * completeness */
+static int max9850_volatile_register(struct snd_soc_codec *codec,
+		unsigned int reg)
+{
+	switch (reg) {
+	case MAX9850_STATUSA:
+	case MAX9850_STATUSB:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
+static const unsigned int max9850_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0x18, 0x1f, TLV_DB_SCALE_ITEM(-7450, 400, 0),
+	0x20, 0x33, TLV_DB_SCALE_ITEM(-4150, 200, 0),
+	0x34, 0x37, TLV_DB_SCALE_ITEM(-150, 100, 0),
+	0x38, 0x3f, TLV_DB_SCALE_ITEM(250, 50, 0),
+};
+
+static const struct snd_kcontrol_new max9850_controls[] = {
+SOC_SINGLE_TLV("Headphone Volume", MAX9850_VOLUME, 0, 0x3f, 1, max9850_tlv),
+SOC_SINGLE("Headphone Switch", MAX9850_VOLUME, 7, 1, 1),
+SOC_SINGLE("Mono Switch", MAX9850_GENERAL_PURPOSE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new max9850_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line In Switch", MAX9850_ENABLE, 1, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget max9850_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("Charge Pump 1", MAX9850_ENABLE, 4, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("Charge Pump 2", MAX9850_ENABLE, 5, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("MCLK", MAX9850_ENABLE, 6, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("SHDN", MAX9850_ENABLE, 7, 0, NULL, 0),
+SND_SOC_DAPM_MIXER_NAMED_CTL("Output Mixer", MAX9850_ENABLE, 2, 0,
+		&max9850_mixer_controls[0],
+		ARRAY_SIZE(max9850_mixer_controls)),
+SND_SOC_DAPM_PGA("Headphone Output", MAX9850_ENABLE, 3, 0, NULL, 0),
+SND_SOC_DAPM_DAC("DAC", "HiFi Playback", MAX9850_ENABLE, 0, 0),
+SND_SOC_DAPM_OUTPUT("OUTL"),
+SND_SOC_DAPM_OUTPUT("HPL"),
+SND_SOC_DAPM_OUTPUT("OUTR"),
+SND_SOC_DAPM_OUTPUT("HPR"),
+SND_SOC_DAPM_MIXER("Line Input", SND_SOC_NOPM, 0, 0, NULL, 0),
+SND_SOC_DAPM_INPUT("INL"),
+SND_SOC_DAPM_INPUT("INR"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* output mixer */
+	{"Output Mixer", NULL, "DAC"},
+	{"Output Mixer", "Line In Switch", "Line Input"},
+
+	/* outputs */
+	{"Headphone Output", NULL, "Output Mixer"},
+	{"HPL", NULL, "Headphone Output"},
+	{"HPR", NULL, "Headphone Output"},
+	{"OUTL", NULL, "Output Mixer"},
+	{"OUTR", NULL, "Output Mixer"},
+
+	/* inputs */
+	{"Line Input", NULL, "INL"},
+	{"Line Input", NULL, "INR"},
+
+	/* supplies */
+	{"Output Mixer", NULL, "Charge Pump 1"},
+	{"Output Mixer", NULL, "Charge Pump 2"},
+	{"Output Mixer", NULL, "SHDN"},
+	{"DAC", NULL, "MCLK"},
+};
+
+static int max9850_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 max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+	u64 lrclk_div;
+	u8 sf, da;
+
+	if (!max9850->sysclk)
+		return -EINVAL;
+
+	/* lrclk_div = 2^22 * rate / iclk with iclk = mclk / sf */
+	sf = (snd_soc_read(codec, MAX9850_CLOCK) >> 2) + 1;
+	lrclk_div = (1 << 22);
+	lrclk_div *= params_rate(params);
+	lrclk_div *= sf;
+	do_div(lrclk_div, max9850->sysclk);
+
+	snd_soc_write(codec, MAX9850_LRCLK_MSB, (lrclk_div >> 8) & 0x7f);
+	snd_soc_write(codec, MAX9850_LRCLK_LSB, lrclk_div & 0xff);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		da = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		da = 0x2;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		da = 0x3;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, MAX9850_DIGITAL_AUDIO, 0x3, da);
+
+	return 0;
+}
+
+static int max9850_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct max9850_priv *max9850 = snd_soc_codec_get_drvdata(codec);
+
+	/* calculate mclk -> iclk divider */
+	if (freq <= 13000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x0);
+	else if (freq <= 26000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x4);
+	else if (freq <= 40000000)
+		snd_soc_write(codec, MAX9850_CLOCK, 0x8);
+	else
+		return -EINVAL;
+
+	max9850->sysclk = freq;
+	return 0;
+}
+
+static int max9850_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 da = 0;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		da |= MAX9850_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		da |= MAX9850_DLY;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		da |= MAX9850_RTJ;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		da |= MAX9850_BCINV | MAX9850_INV;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		da |= MAX9850_BCINV;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		da |= MAX9850_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set da */
+	snd_soc_write(codec, MAX9850_DIGITAL_AUDIO, da);
+
+	return 0;
+}
+
+static int max9850_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = snd_soc_cache_sync(codec);
+			if (ret) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define MAX9850_RATES SNDRV_PCM_RATE_8000_48000
+
+#define MAX9850_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops max9850_dai_ops = {
+	.hw_params	= max9850_hw_params,
+	.set_sysclk	= max9850_set_dai_sysclk,
+	.set_fmt	= max9850_set_dai_fmt,
+};
+
+static struct snd_soc_dai_driver max9850_dai = {
+	.name = "max9850-hifi",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = MAX9850_RATES,
+		.formats = MAX9850_FORMATS
+	},
+	.ops = &max9850_dai_ops,
+};
+
+#ifdef CONFIG_PM
+static int max9850_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	max9850_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int max9850_resume(struct snd_soc_codec *codec)
+{
+	max9850_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+#else
+#define max9850_suspend NULL
+#define max9850_resume NULL
+#endif
+
+static int max9850_probe(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* enable zero-detect */
+	snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1);
+	/* enable slew-rate control */
+	snd_soc_update_bits(codec, MAX9850_VOLUME, 0x40, 0x40);
+	/* set slew-rate 125ms */
+	snd_soc_update_bits(codec, MAX9850_CHARGE_PUMP, 0xff, 0xc0);
+
+	snd_soc_dapm_new_controls(dapm, max9850_dapm_widgets,
+				  ARRAY_SIZE(max9850_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_add_controls(codec, max9850_controls,
+			ARRAY_SIZE(max9850_controls));
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_max9850 = {
+	.probe =	max9850_probe,
+	.suspend =	max9850_suspend,
+	.resume =	max9850_resume,
+	.set_bias_level = max9850_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(max9850_reg),
+	.reg_word_size = sizeof(u8),
+	.reg_cache_default = max9850_reg,
+	.volatile_register = max9850_volatile_register,
+};
+
+static int __devinit max9850_i2c_probe(struct i2c_client *i2c,
+		const struct i2c_device_id *id)
+{
+	struct max9850_priv *max9850;
+	int ret;
+
+	max9850 = kzalloc(sizeof(struct max9850_priv), GFP_KERNEL);
+	if (max9850 == NULL)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, max9850);
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_max9850, &max9850_dai, 1);
+	if (ret < 0)
+		kfree(max9850);
+	return ret;
+}
+
+static __devexit int max9850_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id max9850_i2c_id[] = {
+	{ "max9850", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, max9850_i2c_id);
+
+static struct i2c_driver max9850_i2c_driver = {
+	.driver = {
+		.name = "max9850",
+		.owner = THIS_MODULE,
+	},
+	.probe = max9850_i2c_probe,
+	.remove = __devexit_p(max9850_i2c_remove),
+	.id_table = max9850_i2c_id,
+};
+
+static int __init max9850_init(void)
+{
+	return i2c_add_driver(&max9850_i2c_driver);
+}
+module_init(max9850_init);
+
+static void __exit max9850_exit(void)
+{
+	i2c_del_driver(&max9850_i2c_driver);
+}
+module_exit(max9850_exit);
+
+MODULE_AUTHOR("Christian Glindkamp <christian.glindkamp@taskit.de>");
+MODULE_DESCRIPTION("ASoC MAX9850 codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/max9850.h b/sound/soc/codecs/max9850.h
new file mode 100644
index 0000000..72b1ddb
--- /dev/null
+++ b/sound/soc/codecs/max9850.h
@@ -0,0 +1,38 @@
+/*
+ * max9850.h  --  codec driver for max9850
+ *
+ * Copyright (C) 2011 taskit GmbH
+ * Author: Christian Glindkamp <christian.glindkamp@taskit.de>
+ *
+ * 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.
+ *
+ */
+
+#ifndef _MAX9850_H
+#define _MAX9850_H
+
+#define MAX9850_STATUSA			0x00
+#define MAX9850_STATUSB			0x01
+#define MAX9850_VOLUME			0x02
+#define MAX9850_GENERAL_PURPOSE		0x03
+#define MAX9850_INTERRUPT		0x04
+#define MAX9850_ENABLE			0x05
+#define MAX9850_CLOCK			0x06
+#define MAX9850_CHARGE_PUMP		0x07
+#define MAX9850_LRCLK_MSB		0x08
+#define MAX9850_LRCLK_LSB		0x09
+#define MAX9850_DIGITAL_AUDIO		0x0a
+
+#define MAX9850_CACHEREGNUM 11
+
+/* MAX9850_DIGITAL_AUDIO */
+#define MAX9850_MASTER			(1<<7)
+#define MAX9850_INV			(1<<6)
+#define MAX9850_BCINV			(1<<5)
+#define MAX9850_DLY			(1<<3)
+#define MAX9850_RTJ			(1<<2)
+
+#endif
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
new file mode 100644
index 0000000..1f7217f
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.c
@@ -0,0 +1,1513 @@
+/*
+ * sgtl5000.c  --  SGTL5000 ALSA SoC Audio driver
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, Inc. All Rights Reserved.
+ *
+ * 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/slab.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/clk.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/driver.h>
+#include <linux/regulator/machine.h>
+#include <linux/regulator/consumer.h>
+#include <sound/core.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "sgtl5000.h"
+
+#define SGTL5000_DAP_REG_OFFSET	0x0100
+#define SGTL5000_MAX_REG_OFFSET	0x013A
+
+/* default value of sgtl5000 registers except DAP */
+static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =  {
+	0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */
+	0x0000, /* 0x0002, CHIP_DIG_POWER. */
+	0x0008, /* 0x0004, CHIP_CKL_CTRL */
+	0x0010, /* 0x0006, CHIP_I2S_CTRL */
+	0x0000, /* 0x0008, reserved */
+	0x0008, /* 0x000A, CHIP_SSS_CTRL */
+	0x0000, /* 0x000C, reserved */
+	0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */
+	0x3c3c, /* 0x0010, CHIP_DAC_VOL */
+	0x0000, /* 0x0012, reserved */
+	0x015f, /* 0x0014, CHIP_PAD_STRENGTH */
+	0x0000, /* 0x0016, reserved */
+	0x0000, /* 0x0018, reserved */
+	0x0000, /* 0x001A, reserved */
+	0x0000, /* 0x001E, reserved */
+	0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */
+	0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */
+	0x0111, /* 0x0024, CHIP_ANN_CTRL */
+	0x0000, /* 0x0026, CHIP_LINREG_CTRL */
+	0x0000, /* 0x0028, CHIP_REF_CTRL */
+	0x0000, /* 0x002A, CHIP_MIC_CTRL */
+	0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */
+	0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */
+	0x7060, /* 0x0030, CHIP_ANA_POWER */
+	0x5000, /* 0x0032, CHIP_PLL_CTRL */
+	0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */
+	0x0000, /* 0x0036, CHIP_ANA_STATUS */
+	0x0000, /* 0x0038, reserved */
+	0x0000, /* 0x003A, CHIP_ANA_TEST2 */
+	0x0000, /* 0x003C, CHIP_SHORT_CTRL */
+	0x0000, /* reserved */
+};
+
+/* default value of dap registers */
+static const u16 sgtl5000_dap_regs[] = {
+	0x0000, /* 0x0100, DAP_CONTROL */
+	0x0000, /* 0x0102, DAP_PEQ */
+	0x0040, /* 0x0104, DAP_BASS_ENHANCE */
+	0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */
+	0x0000, /* 0x0108, DAP_AUDIO_EQ */
+	0x0040, /* 0x010A, DAP_SGTL_SURROUND */
+	0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */
+	0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */
+	0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */
+	0x0000, /* 0x0112, reserved */
+	0x0000, /* 0x0114, reserved */
+	0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */
+	0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */
+	0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */
+	0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */
+	0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */
+	0x8000, /* 0x0120, DAP_MAIN_CHAN */
+	0x0000, /* 0x0122, DAP_MIX_CHAN */
+	0x0510, /* 0x0124, DAP_AVC_CTRL */
+	0x1473, /* 0x0126, DAP_AVC_THRESHOLD */
+	0x0028, /* 0x0128, DAP_AVC_ATTACK */
+	0x0050, /* 0x012A, DAP_AVC_DECAY */
+	0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */
+	0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */
+	0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */
+	0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */
+	0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */
+	0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */
+	0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */
+	0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */
+};
+
+/* regulator supplies for sgtl5000, VDDD is an optional external supply */
+enum sgtl5000_regulator_supplies {
+	VDDA,
+	VDDIO,
+	VDDD,
+	SGTL5000_SUPPLY_NUM
+};
+
+/* vddd is optional supply */
+static const char *supply_names[SGTL5000_SUPPLY_NUM] = {
+	"VDDA",
+	"VDDIO",
+	"VDDD"
+};
+
+#define LDO_CONSUMER_NAME	"VDDD_LDO"
+#define LDO_VOLTAGE		1200000
+
+static struct regulator_consumer_supply ldo_consumer[] = {
+	REGULATOR_SUPPLY(LDO_CONSUMER_NAME, NULL),
+};
+
+static struct regulator_init_data ldo_init_data = {
+	.constraints = {
+		.min_uV                 = 850000,
+		.max_uV                 = 1600000,
+		.valid_modes_mask       = REGULATOR_MODE_NORMAL,
+		.valid_ops_mask         = REGULATOR_CHANGE_STATUS,
+	},
+	.num_consumer_supplies = 1,
+	.consumer_supplies = &ldo_consumer[0],
+};
+
+/*
+ * sgtl5000 internal ldo regulator,
+ * enabled when VDDD not provided
+ */
+struct ldo_regulator {
+	struct regulator_desc desc;
+	struct regulator_dev *dev;
+	int voltage;
+	void *codec_data;
+	bool enabled;
+};
+
+/* sgtl5000 private structure in codec */
+struct sgtl5000_priv {
+	int sysclk;	/* sysclk rate */
+	int master;	/* i2s master or not */
+	int fmt;	/* i2s data format */
+	struct regulator_bulk_data supplies[SGTL5000_SUPPLY_NUM];
+	struct ldo_regulator *ldo;
+};
+
+/*
+ * mic_bias power on/off share the same register bits with
+ * output impedance of mic bias, when power on mic bias, we
+ * need reclaim it to impedance value.
+ * 0x0 = Powered off
+ * 0x1 = 2Kohm
+ * 0x2 = 4Kohm
+ * 0x3 = 8Kohm
+ */
+static int mic_bias_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* change mic bias resistor to 4Kohm */
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+				SGTL5000_BIAS_R_4k, SGTL5000_BIAS_R_4k);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		/*
+		 * SGTL5000_BIAS_R_8k as mask to clean the two bits
+		 * of mic bias and output impedance
+		 */
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_MIC_CTRL,
+				SGTL5000_BIAS_R_8k, 0);
+		break;
+	}
+	return 0;
+}
+
+/*
+ * using codec assist to small pop, hp_powerup or lineout_powerup
+ * should stay setting until vag_powerup is fully ramped down,
+ * vag fully ramped down require 400ms.
+ */
+static int small_pop_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP);
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_VAG_POWERUP, 0);
+		msleep(400);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+/* input sources for ADC */
+static const char *adc_mux_text[] = {
+	"MIC_IN", "LINE_IN"
+};
+
+static const struct soc_enum adc_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text);
+
+static const struct snd_kcontrol_new adc_mux =
+SOC_DAPM_ENUM("Capture Mux", adc_enum);
+
+/* input sources for DAC */
+static const char *dac_mux_text[] = {
+	"DAC", "LINE_IN"
+};
+
+static const struct soc_enum dac_enum =
+SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text);
+
+static const struct snd_kcontrol_new dac_mux =
+SOC_DAPM_ENUM("Headphone Mux", dac_enum);
+
+static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = {
+	SND_SOC_DAPM_INPUT("LINE_IN"),
+	SND_SOC_DAPM_INPUT("MIC_IN"),
+
+	SND_SOC_DAPM_OUTPUT("HP_OUT"),
+	SND_SOC_DAPM_OUTPUT("LINE_OUT"),
+
+	SND_SOC_DAPM_MICBIAS_E("Mic Bias", SGTL5000_CHIP_MIC_CTRL, 8, 0,
+				mic_bias_event,
+				SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0,
+			small_pop_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0,
+			small_pop_event,
+			SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux),
+	SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux),
+
+	/* aif for i2s input */
+	SND_SOC_DAPM_AIF_IN("AIFIN", "Playback",
+				0, SGTL5000_CHIP_DIG_POWER,
+				0, 0),
+
+	/* aif for i2s output */
+	SND_SOC_DAPM_AIF_OUT("AIFOUT", "Capture",
+				0, SGTL5000_CHIP_DIG_POWER,
+				1, 0),
+
+	SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0),
+
+	SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0),
+};
+
+/* routes for sgtl5000 */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Capture Mux", "LINE_IN", "LINE_IN"},	/* line_in --> adc_mux */
+	{"Capture Mux", "MIC_IN", "MIC_IN"},	/* mic_in --> adc_mux */
+
+	{"ADC", NULL, "Capture Mux"},		/* adc_mux --> adc */
+	{"AIFOUT", NULL, "ADC"},		/* adc --> i2s_out */
+
+	{"DAC", NULL, "AIFIN"},			/* i2s-->dac,skip audio mux */
+	{"Headphone Mux", "DAC", "DAC"},	/* dac --> hp_mux */
+	{"LO", NULL, "DAC"},			/* dac --> line_out */
+
+	{"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */
+	{"HP", NULL, "Headphone Mux"},		/* hp_mux --> hp */
+
+	{"LINE_OUT", NULL, "LO"},
+	{"HP_OUT", NULL, "HP"},
+};
+
+/* custom function to fetch info of PCM playback volume */
+static int dac_info_volsw(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 0xfc - 0x3c;
+	return 0;
+}
+
+/*
+ * custom function to get of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ *  -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * register value map to userspace value
+ *
+ * register value	0x3c(0dB)	  0xf0(-90dB)0xfc
+ *			------------------------------
+ * userspace value	0xc0			     0
+ */
+static int dac_get_volsw(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg;
+	int l;
+	int r;
+
+	reg = snd_soc_read(codec, SGTL5000_CHIP_DAC_VOL);
+
+	/* get left channel volume */
+	l = (reg & SGTL5000_DAC_VOL_LEFT_MASK) >> SGTL5000_DAC_VOL_LEFT_SHIFT;
+
+	/* get right channel volume */
+	r = (reg & SGTL5000_DAC_VOL_RIGHT_MASK) >> SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+	/* make sure value fall in (0x3c,0xfc) */
+	l = clamp(l, 0x3c, 0xfc);
+	r = clamp(r, 0x3c, 0xfc);
+
+	/* invert it and map to userspace value */
+	l = 0xfc - l;
+	r = 0xfc - r;
+
+	ucontrol->value.integer.value[0] = l;
+	ucontrol->value.integer.value[1] = r;
+
+	return 0;
+}
+
+/*
+ * custom function to put of PCM playback volume
+ *
+ * dac volume register
+ * 15-------------8-7--------------0
+ * | R channel vol | L channel vol |
+ *  -------------------------------
+ *
+ * PCM volume with 0.5017 dB steps from 0 to -90 dB
+ *
+ * register values map to dB
+ * 0x3B and less = Reserved
+ * 0x3C = 0 dB
+ * 0x3D = -0.5 dB
+ * 0xF0 = -90 dB
+ * 0xFC and greater = Muted
+ *
+ * userspace value map to register value
+ *
+ * userspace value	0xc0			     0
+ *			------------------------------
+ * register value	0x3c(0dB)	0xf0(-90dB)0xfc
+ */
+static int dac_put_volsw(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	int reg;
+	int l;
+	int r;
+
+	l = ucontrol->value.integer.value[0];
+	r = ucontrol->value.integer.value[1];
+
+	/* make sure userspace volume fall in (0, 0xfc-0x3c) */
+	l = clamp(l, 0, 0xfc - 0x3c);
+	r = clamp(r, 0, 0xfc - 0x3c);
+
+	/* invert it, get the value can be set to register */
+	l = 0xfc - l;
+	r = 0xfc - r;
+
+	/* shift to get the register value */
+	reg = l << SGTL5000_DAC_VOL_LEFT_SHIFT |
+		r << SGTL5000_DAC_VOL_RIGHT_SHIFT;
+
+	snd_soc_write(codec, SGTL5000_CHIP_DAC_VOL, reg);
+
+	return 0;
+}
+
+static const DECLARE_TLV_DB_SCALE(capture_6db_attenuate, -600, 600, 0);
+
+/* tlv for mic gain, 0db 20db 30db 40db */
+static const unsigned int mic_gain_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
+	1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0),
+};
+
+/* tlv for hp volume, -51.5db to 12.0db, step .5db */
+static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0);
+
+static const struct snd_kcontrol_new sgtl5000_snd_controls[] = {
+	/* SOC_DOUBLE_S8_TLV with invert */
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "PCM Playback Volume",
+		.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+			SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.info = dac_info_volsw,
+		.get = dac_get_volsw,
+		.put = dac_put_volsw,
+	},
+
+	SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0),
+	SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)",
+			SGTL5000_CHIP_ANA_ADC_CTRL,
+			8, 2, 0, capture_6db_attenuate),
+	SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0),
+
+	SOC_DOUBLE_TLV("Headphone Playback Volume",
+			SGTL5000_CHIP_ANA_HP_CTRL,
+			0, 8,
+			0x7f, 1,
+			headphone_volume),
+	SOC_SINGLE("Headphone Playback ZC Switch", SGTL5000_CHIP_ANA_CTRL,
+			5, 1, 0),
+
+	SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL,
+			0, 4, 0, mic_gain_tlv),
+};
+
+/* mute the codec used by alsa core */
+static int sgtl5000_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 adcdac_ctrl = SGTL5000_DAC_MUTE_LEFT | SGTL5000_DAC_MUTE_RIGHT;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+			adcdac_ctrl, mute ? adcdac_ctrl : 0);
+
+	return 0;
+}
+
+/* set codec format */
+static int sgtl5000_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	u16 i2sctl = 0;
+
+	sgtl5000->master = 0;
+	/*
+	 * i2s clock and frame master setting.
+	 * ONLY support:
+	 *  - clock and frame slave,
+	 *  - clock and frame master
+	 */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		i2sctl |= SGTL5000_I2S_MASTER;
+		sgtl5000->master = 1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* setting i2s data format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		i2sctl |= SGTL5000_I2S_MODE_PCM;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		i2sctl |= SGTL5000_I2S_MODE_PCM;
+		i2sctl |= SGTL5000_I2S_LRALIGN;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		i2sctl |= SGTL5000_I2S_MODE_RJ;
+		i2sctl |= SGTL5000_I2S_LRPOL;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		i2sctl |= SGTL5000_I2S_MODE_I2S_LJ;
+		i2sctl |= SGTL5000_I2S_LRALIGN;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	sgtl5000->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	/* Clock inversion */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		i2sctl |= SGTL5000_I2S_SCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, SGTL5000_CHIP_I2S_CTRL, i2sctl);
+
+	return 0;
+}
+
+/* set codec sysclk */
+static int sgtl5000_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				   int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	switch (clk_id) {
+	case SGTL5000_SYSCLK:
+		sgtl5000->sysclk = freq;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * set clock according to i2s frame clock,
+ * sgtl5000 provide 2 clock sources.
+ * 1. sys_mclk. sample freq can only configure to
+ *	1/256, 1/384, 1/512 of sys_mclk.
+ * 2. pll. can derive any audio clocks.
+ *
+ * clock setting rules:
+ * 1. in slave mode, only sys_mclk can use.
+ * 2. as constraint by sys_mclk, sample freq should
+ *	set to 32k, 44.1k and above.
+ * 3. using sys_mclk prefer to pll to save power.
+ */
+static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	int clk_ctl = 0;
+	int sys_fs;	/* sample freq */
+
+	/*
+	 * sample freq should be divided by frame clock,
+	 * if frame clock lower than 44.1khz, sample feq should set to
+	 * 32khz or 44.1khz.
+	 */
+	switch (frame_rate) {
+	case 8000:
+	case 16000:
+		sys_fs = 32000;
+		break;
+	case 11025:
+	case 22050:
+		sys_fs = 44100;
+		break;
+	default:
+		sys_fs = frame_rate;
+		break;
+	}
+
+	/* set divided factor of frame clock */
+	switch (sys_fs / frame_rate) {
+	case 4:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_4 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	case 2:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_2 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	case 1:
+		clk_ctl |= SGTL5000_RATE_MODE_DIV_1 << SGTL5000_RATE_MODE_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* set the sys_fs according to frame rate */
+	switch (sys_fs) {
+	case 32000:
+		clk_ctl |= SGTL5000_SYS_FS_32k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 44100:
+		clk_ctl |= SGTL5000_SYS_FS_44_1k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 48000:
+		clk_ctl |= SGTL5000_SYS_FS_48k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	case 96000:
+		clk_ctl |= SGTL5000_SYS_FS_96k << SGTL5000_SYS_FS_SHIFT;
+		break;
+	default:
+		dev_err(codec->dev, "frame rate %d not supported\n",
+			frame_rate);
+		return -EINVAL;
+	}
+
+	/*
+	 * calculate the divider of mclk/sample_freq,
+	 * factor of freq =96k can only be 256, since mclk in range (12m,27m)
+	 */
+	switch (sgtl5000->sysclk / sys_fs) {
+	case 256:
+		clk_ctl |= SGTL5000_MCLK_FREQ_256FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	case 384:
+		clk_ctl |= SGTL5000_MCLK_FREQ_384FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	case 512:
+		clk_ctl |= SGTL5000_MCLK_FREQ_512FS <<
+			SGTL5000_MCLK_FREQ_SHIFT;
+		break;
+	default:
+		/* if mclk not satisify the divider, use pll */
+		if (sgtl5000->master) {
+			clk_ctl |= SGTL5000_MCLK_FREQ_PLL <<
+				SGTL5000_MCLK_FREQ_SHIFT;
+		} else {
+			dev_err(codec->dev,
+				"PLL not supported in slave mode\n");
+			return -EINVAL;
+		}
+	}
+
+	/* if using pll, please check manual 6.4.2 for detail */
+	if ((clk_ctl & SGTL5000_MCLK_FREQ_MASK) == SGTL5000_MCLK_FREQ_PLL) {
+		u64 out, t;
+		int div2;
+		int pll_ctl;
+		unsigned int in, int_div, frac_div;
+
+		if (sgtl5000->sysclk > 17000000) {
+			div2 = 1;
+			in = sgtl5000->sysclk / 2;
+		} else {
+			div2 = 0;
+			in = sgtl5000->sysclk;
+		}
+		if (sys_fs == 44100)
+			out = 180633600;
+		else
+			out = 196608000;
+		t = do_div(out, in);
+		int_div = out;
+		t *= 2048;
+		do_div(t, in);
+		frac_div = t;
+		pll_ctl = int_div << SGTL5000_PLL_INT_DIV_SHIFT |
+		    frac_div << SGTL5000_PLL_FRAC_DIV_SHIFT;
+
+		snd_soc_write(codec, SGTL5000_CHIP_PLL_CTRL, pll_ctl);
+		if (div2)
+			snd_soc_update_bits(codec,
+				SGTL5000_CHIP_CLK_TOP_CTRL,
+				SGTL5000_INPUT_FREQ_DIV2,
+				SGTL5000_INPUT_FREQ_DIV2);
+		else
+			snd_soc_update_bits(codec,
+				SGTL5000_CHIP_CLK_TOP_CTRL,
+				SGTL5000_INPUT_FREQ_DIV2,
+				0);
+
+		/* power up pll */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP);
+	} else {
+		/* power down pll */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+			SGTL5000_PLL_POWERUP | SGTL5000_VCOAMP_POWERUP,
+			0);
+	}
+
+	/* if using pll, clk_ctrl must be set after pll power up */
+	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, clk_ctl);
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ * input: params_rate, params_fmt
+ */
+static int sgtl5000_pcm_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;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	int channels = params_channels(params);
+	int i2s_ctl = 0;
+	int stereo;
+	int ret;
+
+	/* sysclk should already set */
+	if (!sgtl5000->sysclk) {
+		dev_err(codec->dev, "%s: set sysclk first!\n", __func__);
+		return -EFAULT;
+	}
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		stereo = SGTL5000_DAC_STEREO;
+	else
+		stereo = SGTL5000_ADC_STEREO;
+
+	/* set mono to save power */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, stereo,
+			channels == 1 ? 0 : stereo);
+
+	/* set codec clock base on lrclk */
+	ret = sgtl5000_set_clock(codec, params_rate(params));
+	if (ret)
+		return ret;
+
+	/* set i2s data format */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+			return -EINVAL;
+		i2s_ctl |= SGTL5000_I2S_DLEN_16 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_32FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		i2s_ctl |= SGTL5000_I2S_DLEN_20 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		i2s_ctl |= SGTL5000_I2S_DLEN_24 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		if (sgtl5000->fmt == SND_SOC_DAIFMT_RIGHT_J)
+			return -EINVAL;
+		i2s_ctl |= SGTL5000_I2S_DLEN_32 << SGTL5000_I2S_DLEN_SHIFT;
+		i2s_ctl |= SGTL5000_I2S_SCLKFREQ_64FS <<
+		    SGTL5000_I2S_SCLKFREQ_SHIFT;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl);
+
+	return 0;
+}
+
+static int ldo_regulator_is_enabled(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+	return ldo->enabled;
+}
+
+static int ldo_regulator_enable(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+	struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+	int reg;
+
+	if (ldo_regulator_is_enabled(dev))
+		return 0;
+
+	/* set regulator value firstly */
+	reg = (1600 - ldo->voltage / 1000) / 50;
+	reg = clamp(reg, 0x0, 0xf);
+
+	/* amend the voltage value, unit: uV */
+	ldo->voltage = (1600 - reg * 50) * 1000;
+
+	/* set voltage to register */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, reg);
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINEREG_D_POWERUP,
+				SGTL5000_LINEREG_D_POWERUP);
+
+	/* when internal ldo enabled, simple digital power can be disabled */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP,
+				0);
+
+	ldo->enabled = 1;
+	return 0;
+}
+
+static int ldo_regulator_disable(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+	struct snd_soc_codec *codec = (struct snd_soc_codec *)ldo->codec_data;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINEREG_D_POWERUP,
+				0);
+
+	/* clear voltage info */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, 0);
+
+	ldo->enabled = 0;
+
+	return 0;
+}
+
+static int ldo_regulator_get_voltage(struct regulator_dev *dev)
+{
+	struct ldo_regulator *ldo = rdev_get_drvdata(dev);
+
+	return ldo->voltage;
+}
+
+static struct regulator_ops ldo_regulator_ops = {
+	.is_enabled = ldo_regulator_is_enabled,
+	.enable = ldo_regulator_enable,
+	.disable = ldo_regulator_disable,
+	.get_voltage = ldo_regulator_get_voltage,
+};
+
+static int ldo_regulator_register(struct snd_soc_codec *codec,
+				struct regulator_init_data *init_data,
+				int voltage)
+{
+	struct ldo_regulator *ldo;
+
+	ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
+
+	if (!ldo) {
+		dev_err(codec->dev, "failed to allocate ldo_regulator\n");
+		return -ENOMEM;
+	}
+
+	ldo->desc.name = kstrdup(dev_name(codec->dev), GFP_KERNEL);
+	if (!ldo->desc.name) {
+		kfree(ldo);
+		dev_err(codec->dev, "failed to allocate decs name memory\n");
+		return -ENOMEM;
+	}
+
+	ldo->desc.type  = REGULATOR_VOLTAGE;
+	ldo->desc.owner = THIS_MODULE;
+	ldo->desc.ops   = &ldo_regulator_ops;
+	ldo->desc.n_voltages = 1;
+
+	ldo->codec_data = codec;
+	ldo->voltage = voltage;
+
+	ldo->dev = regulator_register(&ldo->desc, codec->dev,
+					  init_data, ldo);
+	if (IS_ERR(ldo->dev)) {
+		int ret = PTR_ERR(ldo->dev);
+
+		dev_err(codec->dev, "failed to register regulator\n");
+		kfree(ldo->desc.name);
+		kfree(ldo);
+
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ldo_regulator_remove(struct snd_soc_codec *codec)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+	struct ldo_regulator *ldo = sgtl5000->ldo;
+
+	if (!ldo)
+		return 0;
+
+	regulator_unregister(ldo->dev);
+	kfree(ldo->desc.name);
+	kfree(ldo);
+
+	return 0;
+}
+
+/*
+ * set dac bias
+ * common state changes:
+ * startup:
+ * off --> standby --> prepare --> on
+ * standby --> prepare --> on
+ *
+ * stop:
+ * on --> prepare --> standby
+ */
+static int sgtl5000_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	int ret;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(
+						ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+			if (ret)
+				return ret;
+			udelay(10);
+		}
+
+		break;
+	case SND_SOC_BIAS_OFF:
+		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define SGTL5000_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 sgtl5000_ops = {
+	.hw_params = sgtl5000_pcm_hw_params,
+	.digital_mute = sgtl5000_digital_mute,
+	.set_fmt = sgtl5000_set_dai_fmt,
+	.set_sysclk = sgtl5000_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver sgtl5000_dai = {
+	.name = "sgtl5000",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		/*
+		 * only support 8~48K + 96K,
+		 * TODO modify hw_param to support more
+		 */
+		.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+		.formats = SGTL5000_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_96000,
+		.formats = SGTL5000_FORMATS,
+	},
+	.ops = &sgtl5000_ops,
+	.symmetric_rates = 1,
+};
+
+static int sgtl5000_volatile_register(struct snd_soc_codec *codec,
+					unsigned int reg)
+{
+	switch (reg) {
+	case SGTL5000_CHIP_ID:
+	case SGTL5000_CHIP_ADCDAC_CTRL:
+	case SGTL5000_CHIP_ANA_STATUS:
+		return 1;
+	}
+
+	return 0;
+}
+
+#ifdef CONFIG_SUSPEND
+static int sgtl5000_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+/*
+ * restore all sgtl5000 registers,
+ * since a big hole between dap and regular registers,
+ * we will restore them respectively.
+ */
+static int sgtl5000_restore_regs(struct snd_soc_codec *codec)
+{
+	u16 *cache = codec->reg_cache;
+	int i;
+	int regular_regs = SGTL5000_CHIP_SHORT_CTRL >> 1;
+
+	/* restore regular registers */
+	for (i = 0; i < regular_regs; i++) {
+		int reg = i << 1;
+
+		/* this regs depends on the others */
+		if (reg == SGTL5000_CHIP_ANA_POWER ||
+			reg == SGTL5000_CHIP_CLK_CTRL ||
+			reg == SGTL5000_CHIP_LINREG_CTRL ||
+			reg == SGTL5000_CHIP_LINE_OUT_CTRL ||
+			reg == SGTL5000_CHIP_CLK_CTRL)
+			continue;
+
+		snd_soc_write(codec, reg, cache[i]);
+	}
+
+	/* restore dap registers */
+	for (i = SGTL5000_DAP_REG_OFFSET >> 1;
+			i < SGTL5000_MAX_REG_OFFSET >> 1; i++) {
+		int reg = i << 1;
+
+		snd_soc_write(codec, reg, cache[i]);
+	}
+
+	/*
+	 * restore power and other regs according
+	 * to set_power() and set_clock()
+	 */
+	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL,
+			cache[SGTL5000_CHIP_LINREG_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER,
+			cache[SGTL5000_CHIP_ANA_POWER >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL,
+			cache[SGTL5000_CHIP_CLK_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL,
+			cache[SGTL5000_CHIP_REF_CTRL >> 1]);
+
+	snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+			cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]);
+	return 0;
+}
+
+static int sgtl5000_resume(struct snd_soc_codec *codec)
+{
+	/* Bring the codec back up to standby to enable regulators */
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* Restore registers by cached in memory */
+	sgtl5000_restore_regs(codec);
+	return 0;
+}
+#else
+#define sgtl5000_suspend NULL
+#define sgtl5000_resume  NULL
+#endif	/* CONFIG_SUSPEND */
+
+/*
+ * sgtl5000 has 3 internal power supplies:
+ * 1. VAG, normally set to vdda/2
+ * 2. chargepump, set to different value
+ *	according to voltage of vdda and vddio
+ * 3. line out VAG, normally set to vddio/2
+ *
+ * and should be set according to:
+ * 1. vddd provided by external or not
+ * 2. vdda and vddio voltage value. > 3.1v or not
+ * 3. chip revision >=0x11 or not. If >=0x11, not use external vddd.
+ */
+static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
+{
+	int vddd;
+	int vdda;
+	int vddio;
+	u16 ana_pwr;
+	u16 lreg_ctrl;
+	int vag;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	vdda  = regulator_get_voltage(sgtl5000->supplies[VDDA].consumer);
+	vddio = regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer);
+	vddd  = regulator_get_voltage(sgtl5000->supplies[VDDD].consumer);
+
+	vdda  = vdda / 1000;
+	vddio = vddio / 1000;
+	vddd  = vddd / 1000;
+
+	if (vdda <= 0 || vddio <= 0 || vddd < 0) {
+		dev_err(codec->dev, "regulator voltage not set correctly\n");
+
+		return -EINVAL;
+	}
+
+	/* according to datasheet, maximum voltage of supplies */
+	if (vdda > 3600 || vddio > 3600 || vddd > 1980) {
+		dev_err(codec->dev,
+			"exceed max voltage vdda %dmv vddio %dma vddd %dma\n",
+			vdda, vddio, vddd);
+
+		return -EINVAL;
+	}
+
+	/* reset value */
+	ana_pwr = snd_soc_read(codec, SGTL5000_CHIP_ANA_POWER);
+	ana_pwr |= SGTL5000_DAC_STEREO |
+			SGTL5000_ADC_STEREO |
+			SGTL5000_REFTOP_POWERUP;
+	lreg_ctrl = snd_soc_read(codec, SGTL5000_CHIP_LINREG_CTRL);
+
+	if (vddio < 3100 && vdda < 3100) {
+		/* enable internal oscillator used for charge pump */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_CLK_TOP_CTRL,
+					SGTL5000_INT_OSC_EN,
+					SGTL5000_INT_OSC_EN);
+		/* Enable VDDC charge pump */
+		ana_pwr |= SGTL5000_VDDC_CHRGPMP_POWERUP;
+	} else if (vddio >= 3100 && vdda >= 3100) {
+		/*
+		 * if vddio and vddd > 3.1v,
+		 * charge pump should be clean before set ana_pwr
+		 */
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_VDDC_CHRGPMP_POWERUP, 0);
+
+		/* VDDC use VDDIO rail */
+		lreg_ctrl |= SGTL5000_VDDC_ASSN_OVRD;
+		lreg_ctrl |= SGTL5000_VDDC_MAN_ASSN_VDDIO <<
+			    SGTL5000_VDDC_MAN_ASSN_SHIFT;
+	}
+
+	snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, lreg_ctrl);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, ana_pwr);
+
+	/* set voltage to register */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
+				(0x1 << 4) - 1, 0x8);
+
+	/*
+	 * if vddd linear reg has been enabled,
+	 * simple digital supply should be clear to get
+	 * proper VDDD voltage.
+	 */
+	if (ana_pwr & SGTL5000_LINEREG_D_POWERUP)
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP,
+				0);
+	else
+		snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
+				SGTL5000_LINREG_SIMPLE_POWERUP |
+				SGTL5000_STARTUP_POWERUP,
+				0);
+
+	/*
+	 * set ADC/DAC VAG to vdda / 2,
+	 * should stay in range (0.8v, 1.575v)
+	 */
+	vag = vdda / 2;
+	if (vag <= SGTL5000_ANA_GND_BASE)
+		vag = 0;
+	else if (vag >= SGTL5000_ANA_GND_BASE + SGTL5000_ANA_GND_STP *
+		 (SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT))
+		vag = SGTL5000_ANA_GND_MASK >> SGTL5000_ANA_GND_SHIFT;
+	else
+		vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+			vag << SGTL5000_ANA_GND_SHIFT,
+			vag << SGTL5000_ANA_GND_SHIFT);
+
+	/* set line out VAG to vddio / 2, in range (0.8v, 1.675v) */
+	vag = vddio / 2;
+	if (vag <= SGTL5000_LINE_OUT_GND_BASE)
+		vag = 0;
+	else if (vag >= SGTL5000_LINE_OUT_GND_BASE +
+		SGTL5000_LINE_OUT_GND_STP * SGTL5000_LINE_OUT_GND_MAX)
+		vag = SGTL5000_LINE_OUT_GND_MAX;
+	else
+		vag = (vag - SGTL5000_LINE_OUT_GND_BASE) /
+		    SGTL5000_LINE_OUT_GND_STP;
+
+	snd_soc_update_bits(codec, SGTL5000_CHIP_LINE_OUT_CTRL,
+			vag << SGTL5000_LINE_OUT_GND_SHIFT |
+			SGTL5000_LINE_OUT_CURRENT_360u <<
+				SGTL5000_LINE_OUT_CURRENT_SHIFT,
+			vag << SGTL5000_LINE_OUT_GND_SHIFT |
+			SGTL5000_LINE_OUT_CURRENT_360u <<
+				SGTL5000_LINE_OUT_CURRENT_SHIFT);
+
+	return 0;
+}
+
+static int sgtl5000_enable_regulators(struct snd_soc_codec *codec)
+{
+	u16 reg;
+	int ret;
+	int rev;
+	int i;
+	int external_vddd = 0;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++)
+		sgtl5000->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	if (!ret)
+		external_vddd = 1;
+	else {
+		/* set internal ldo to 1.2v */
+		int voltage = LDO_VOLTAGE;
+
+		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+		if (ret) {
+			dev_err(codec->dev,
+			"Failed to register vddd internal supplies: %d\n",
+				ret);
+			return ret;
+		}
+
+		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+		ret = regulator_bulk_get(codec->dev,
+				ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+
+		if (ret) {
+			ldo_regulator_remove(codec);
+			dev_err(codec->dev,
+				"Failed to request supplies: %d\n", ret);
+
+			return ret;
+		}
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+	if (ret)
+		goto err_regulator_free;
+
+	/* wait for all power rails bring up */
+	udelay(10);
+
+	/* read chip information */
+	reg = snd_soc_read(codec, SGTL5000_CHIP_ID);
+	if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) !=
+	    SGTL5000_PARTID_PART_ID) {
+		dev_err(codec->dev,
+			"Device with ID register %x is not a sgtl5000\n", reg);
+		ret = -ENODEV;
+		goto err_regulator_disable;
+	}
+
+	rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT;
+	dev_info(codec->dev, "sgtl5000 revision %d\n", rev);
+
+	/*
+	 * workaround for revision 0x11 and later,
+	 * roll back to use internal LDO
+	 */
+	if (external_vddd && rev >= 0x11) {
+		int voltage = LDO_VOLTAGE;
+		/* disable all regulator first */
+		regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+		/* free VDDD regulator */
+		regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+					sgtl5000->supplies);
+
+		ret = ldo_regulator_register(codec, &ldo_init_data, voltage);
+		if (ret)
+			return ret;
+
+		sgtl5000->supplies[VDDD].supply = LDO_CONSUMER_NAME;
+
+		ret = regulator_bulk_get(codec->dev,
+				ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+		if (ret) {
+			ldo_regulator_remove(codec);
+			dev_err(codec->dev,
+				"Failed to request supplies: %d\n", ret);
+
+			return ret;
+		}
+
+		ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+		if (ret)
+			goto err_regulator_free;
+
+		/* wait for all power rails bring up */
+		udelay(10);
+	}
+
+	return 0;
+
+err_regulator_disable:
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+err_regulator_free:
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	if (external_vddd)
+		ldo_regulator_remove(codec);
+	return ret;
+
+}
+
+static int sgtl5000_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	/* setup i2c data ops */
+	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	ret = sgtl5000_enable_regulators(codec);
+	if (ret)
+		return ret;
+
+	/* power up sgtl5000 */
+	ret = sgtl5000_set_power_regs(codec);
+	if (ret)
+		goto err;
+
+	/* enable small pop, introduce 400ms delay in turning off */
+	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
+				SGTL5000_SMALL_POP,
+				SGTL5000_SMALL_POP);
+
+	/* disable short cut detector */
+	snd_soc_write(codec, SGTL5000_CHIP_SHORT_CTRL, 0);
+
+	/*
+	 * set i2s as default input of sound switch
+	 * TODO: add sound switch to control and dapm widge.
+	 */
+	snd_soc_write(codec, SGTL5000_CHIP_SSS_CTRL,
+			SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT);
+	snd_soc_write(codec, SGTL5000_CHIP_DIG_POWER,
+			SGTL5000_ADC_EN | SGTL5000_DAC_EN);
+
+	/* enable dac volume ramp by default */
+	snd_soc_write(codec, SGTL5000_CHIP_ADCDAC_CTRL,
+			SGTL5000_DAC_VOL_RAMP_EN |
+			SGTL5000_DAC_MUTE_RIGHT |
+			SGTL5000_DAC_MUTE_LEFT);
+
+	snd_soc_write(codec, SGTL5000_CHIP_PAD_STRENGTH, 0x015f);
+
+	snd_soc_write(codec, SGTL5000_CHIP_ANA_CTRL,
+			SGTL5000_HP_ZCD_EN |
+			SGTL5000_ADC_ZCD_EN);
+
+	snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0);
+
+	/*
+	 * disable DAP
+	 * TODO:
+	 * Enable DAP in kcontrol and dapm.
+	 */
+	snd_soc_write(codec, SGTL5000_DAP_CTRL, 0);
+
+	/* leading to standby state */
+	ret = sgtl5000_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	if (ret)
+		goto err;
+
+	snd_soc_add_controls(codec, sgtl5000_snd_controls,
+			     ARRAY_SIZE(sgtl5000_snd_controls));
+
+	snd_soc_dapm_new_controls(&codec->dapm, sgtl5000_dapm_widgets,
+				  ARRAY_SIZE(sgtl5000_dapm_widgets));
+
+	snd_soc_dapm_add_routes(&codec->dapm, audio_map,
+				ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(&codec->dapm);
+
+	return 0;
+
+err:
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	ldo_regulator_remove(codec);
+
+	return ret;
+}
+
+static int sgtl5000_remove(struct snd_soc_codec *codec)
+{
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
+
+	sgtl5000_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies),
+						sgtl5000->supplies);
+	regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies),
+				sgtl5000->supplies);
+	ldo_regulator_remove(codec);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver sgtl5000_driver = {
+	.probe = sgtl5000_probe,
+	.remove = sgtl5000_remove,
+	.suspend = sgtl5000_suspend,
+	.resume = sgtl5000_resume,
+	.set_bias_level = sgtl5000_set_bias_level,
+	.reg_cache_size = ARRAY_SIZE(sgtl5000_regs),
+	.reg_word_size = sizeof(u16),
+	.reg_cache_step = 2,
+	.reg_cache_default = sgtl5000_regs,
+	.volatile_register = sgtl5000_volatile_register,
+};
+
+static __devinit int sgtl5000_i2c_probe(struct i2c_client *client,
+					const struct i2c_device_id *id)
+{
+	struct sgtl5000_priv *sgtl5000;
+	int ret;
+
+	sgtl5000 = kzalloc(sizeof(struct sgtl5000_priv), GFP_KERNEL);
+	if (!sgtl5000)
+		return -ENOMEM;
+
+	/*
+	 * copy DAP default values to default value array.
+	 * sgtl5000 register space has a big hole, merge it
+	 * at init phase makes life easy.
+	 * FIXME: should we drop 'const' of sgtl5000_regs?
+	 */
+	memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)),
+			sgtl5000_dap_regs,
+			SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET);
+
+	i2c_set_clientdata(client, sgtl5000);
+
+	ret = snd_soc_register_codec(&client->dev,
+			&sgtl5000_driver, &sgtl5000_dai, 1);
+	if (ret) {
+		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
+		kfree(sgtl5000);
+		return ret;
+	}
+
+	return 0;
+}
+
+static __devexit int sgtl5000_i2c_remove(struct i2c_client *client)
+{
+	struct sgtl5000_priv *sgtl5000 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_codec(&client->dev);
+
+	kfree(sgtl5000);
+	return 0;
+}
+
+static const struct i2c_device_id sgtl5000_id[] = {
+	{"sgtl5000", 0},
+	{},
+};
+
+MODULE_DEVICE_TABLE(i2c, sgtl5000_id);
+
+static struct i2c_driver sgtl5000_i2c_driver = {
+	.driver = {
+		   .name = "sgtl5000",
+		   .owner = THIS_MODULE,
+		   },
+	.probe = sgtl5000_i2c_probe,
+	.remove = __devexit_p(sgtl5000_i2c_remove),
+	.id_table = sgtl5000_id,
+};
+
+static int __init sgtl5000_modinit(void)
+{
+	return i2c_add_driver(&sgtl5000_i2c_driver);
+}
+module_init(sgtl5000_modinit);
+
+static void __exit sgtl5000_exit(void)
+{
+	i2c_del_driver(&sgtl5000_i2c_driver);
+}
+module_exit(sgtl5000_exit);
+
+MODULE_DESCRIPTION("Freescale SGTL5000 ALSA SoC Codec Driver");
+MODULE_AUTHOR("Zeng Zhaoming <zhaoming.zeng@freescale.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
new file mode 100644
index 0000000..eec3ab3
--- /dev/null
+++ b/sound/soc/codecs/sgtl5000.h
@@ -0,0 +1,400 @@
+/*
+ * sgtl5000.h - SGTL5000 audio codec interface
+ *
+ * Copyright 2010-2011 Freescale Semiconductor, 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 _SGTL5000_H
+#define _SGTL5000_H
+
+/*
+ * Register values.
+ */
+#define SGTL5000_CHIP_ID			0x0000
+#define SGTL5000_CHIP_DIG_POWER			0x0002
+#define SGTL5000_CHIP_CLK_CTRL			0x0004
+#define SGTL5000_CHIP_I2S_CTRL			0x0006
+#define SGTL5000_CHIP_SSS_CTRL			0x000a
+#define SGTL5000_CHIP_ADCDAC_CTRL		0x000e
+#define SGTL5000_CHIP_DAC_VOL			0x0010
+#define SGTL5000_CHIP_PAD_STRENGTH		0x0014
+#define SGTL5000_CHIP_ANA_ADC_CTRL		0x0020
+#define SGTL5000_CHIP_ANA_HP_CTRL		0x0022
+#define SGTL5000_CHIP_ANA_CTRL			0x0024
+#define SGTL5000_CHIP_LINREG_CTRL		0x0026
+#define SGTL5000_CHIP_REF_CTRL			0x0028
+#define SGTL5000_CHIP_MIC_CTRL			0x002a
+#define SGTL5000_CHIP_LINE_OUT_CTRL		0x002c
+#define SGTL5000_CHIP_LINE_OUT_VOL		0x002e
+#define SGTL5000_CHIP_ANA_POWER			0x0030
+#define SGTL5000_CHIP_PLL_CTRL			0x0032
+#define SGTL5000_CHIP_CLK_TOP_CTRL		0x0034
+#define SGTL5000_CHIP_ANA_STATUS		0x0036
+#define SGTL5000_CHIP_SHORT_CTRL		0x003c
+#define SGTL5000_CHIP_ANA_TEST2			0x003a
+#define SGTL5000_DAP_CTRL			0x0100
+#define SGTL5000_DAP_PEQ			0x0102
+#define SGTL5000_DAP_BASS_ENHANCE		0x0104
+#define SGTL5000_DAP_BASS_ENHANCE_CTRL		0x0106
+#define SGTL5000_DAP_AUDIO_EQ			0x0108
+#define SGTL5000_DAP_SURROUND			0x010a
+#define SGTL5000_DAP_FLT_COEF_ACCESS		0x010c
+#define SGTL5000_DAP_COEF_WR_B0_MSB		0x010e
+#define SGTL5000_DAP_COEF_WR_B0_LSB		0x0110
+#define SGTL5000_DAP_EQ_BASS_BAND0		0x0116
+#define SGTL5000_DAP_EQ_BASS_BAND1		0x0118
+#define SGTL5000_DAP_EQ_BASS_BAND2		0x011a
+#define SGTL5000_DAP_EQ_BASS_BAND3		0x011c
+#define SGTL5000_DAP_EQ_BASS_BAND4		0x011e
+#define SGTL5000_DAP_MAIN_CHAN			0x0120
+#define SGTL5000_DAP_MIX_CHAN			0x0122
+#define SGTL5000_DAP_AVC_CTRL			0x0124
+#define SGTL5000_DAP_AVC_THRESHOLD		0x0126
+#define SGTL5000_DAP_AVC_ATTACK			0x0128
+#define SGTL5000_DAP_AVC_DECAY			0x012a
+#define SGTL5000_DAP_COEF_WR_B1_MSB		0x012c
+#define SGTL5000_DAP_COEF_WR_B1_LSB		0x012e
+#define SGTL5000_DAP_COEF_WR_B2_MSB		0x0130
+#define SGTL5000_DAP_COEF_WR_B2_LSB		0x0132
+#define SGTL5000_DAP_COEF_WR_A1_MSB		0x0134
+#define SGTL5000_DAP_COEF_WR_A1_LSB		0x0136
+#define SGTL5000_DAP_COEF_WR_A2_MSB		0x0138
+#define SGTL5000_DAP_COEF_WR_A2_LSB		0x013a
+
+/*
+ * Field Definitions.
+ */
+
+/*
+ * SGTL5000_CHIP_ID
+ */
+#define SGTL5000_PARTID_MASK			0xff00
+#define SGTL5000_PARTID_SHIFT			8
+#define SGTL5000_PARTID_WIDTH			8
+#define SGTL5000_PARTID_PART_ID			0xa0
+#define SGTL5000_REVID_MASK			0x00ff
+#define SGTL5000_REVID_SHIFT			0
+#define SGTL5000_REVID_WIDTH			8
+
+/*
+ * SGTL5000_CHIP_DIG_POWER
+ */
+#define SGTL5000_ADC_EN				0x0040
+#define SGTL5000_DAC_EN				0x0020
+#define SGTL5000_DAP_POWERUP			0x0010
+#define SGTL5000_I2S_OUT_POWERUP		0x0002
+#define SGTL5000_I2S_IN_POWERUP			0x0001
+
+/*
+ * SGTL5000_CHIP_CLK_CTRL
+ */
+#define SGTL5000_RATE_MODE_MASK			0x0030
+#define SGTL5000_RATE_MODE_SHIFT		4
+#define SGTL5000_RATE_MODE_WIDTH		2
+#define SGTL5000_RATE_MODE_DIV_1		0
+#define SGTL5000_RATE_MODE_DIV_2		1
+#define SGTL5000_RATE_MODE_DIV_4		2
+#define SGTL5000_RATE_MODE_DIV_6		3
+#define SGTL5000_SYS_FS_MASK			0x000c
+#define SGTL5000_SYS_FS_SHIFT			2
+#define SGTL5000_SYS_FS_WIDTH			2
+#define SGTL5000_SYS_FS_32k			0x0
+#define SGTL5000_SYS_FS_44_1k			0x1
+#define SGTL5000_SYS_FS_48k			0x2
+#define SGTL5000_SYS_FS_96k			0x3
+#define SGTL5000_MCLK_FREQ_MASK			0x0003
+#define SGTL5000_MCLK_FREQ_SHIFT		0
+#define SGTL5000_MCLK_FREQ_WIDTH		2
+#define SGTL5000_MCLK_FREQ_256FS		0x0
+#define SGTL5000_MCLK_FREQ_384FS		0x1
+#define SGTL5000_MCLK_FREQ_512FS		0x2
+#define SGTL5000_MCLK_FREQ_PLL			0x3
+
+/*
+ * SGTL5000_CHIP_I2S_CTRL
+ */
+#define SGTL5000_I2S_SCLKFREQ_MASK		0x0100
+#define SGTL5000_I2S_SCLKFREQ_SHIFT		8
+#define SGTL5000_I2S_SCLKFREQ_WIDTH		1
+#define SGTL5000_I2S_SCLKFREQ_64FS		0x0
+#define SGTL5000_I2S_SCLKFREQ_32FS		0x1	/* Not for RJ mode */
+#define SGTL5000_I2S_MASTER			0x0080
+#define SGTL5000_I2S_SCLK_INV			0x0040
+#define SGTL5000_I2S_DLEN_MASK			0x0030
+#define SGTL5000_I2S_DLEN_SHIFT			4
+#define SGTL5000_I2S_DLEN_WIDTH			2
+#define SGTL5000_I2S_DLEN_32			0x0
+#define SGTL5000_I2S_DLEN_24			0x1
+#define SGTL5000_I2S_DLEN_20			0x2
+#define SGTL5000_I2S_DLEN_16			0x3
+#define SGTL5000_I2S_MODE_MASK			0x000c
+#define SGTL5000_I2S_MODE_SHIFT			2
+#define SGTL5000_I2S_MODE_WIDTH			2
+#define SGTL5000_I2S_MODE_I2S_LJ		0x0
+#define SGTL5000_I2S_MODE_RJ			0x1
+#define SGTL5000_I2S_MODE_PCM			0x2
+#define SGTL5000_I2S_LRALIGN			0x0002
+#define SGTL5000_I2S_LRPOL			0x0001	/* set for which mode */
+
+/*
+ * SGTL5000_CHIP_SSS_CTRL
+ */
+#define SGTL5000_DAP_MIX_LRSWAP			0x4000
+#define SGTL5000_DAP_LRSWAP			0x2000
+#define SGTL5000_DAC_LRSWAP			0x1000
+#define SGTL5000_I2S_OUT_LRSWAP			0x0400
+#define SGTL5000_DAP_MIX_SEL_MASK		0x0300
+#define SGTL5000_DAP_MIX_SEL_SHIFT		8
+#define SGTL5000_DAP_MIX_SEL_WIDTH		2
+#define SGTL5000_DAP_MIX_SEL_ADC		0x0
+#define SGTL5000_DAP_MIX_SEL_I2S_IN		0x1
+#define SGTL5000_DAP_SEL_MASK			0x00c0
+#define SGTL5000_DAP_SEL_SHIFT			6
+#define SGTL5000_DAP_SEL_WIDTH			2
+#define SGTL5000_DAP_SEL_ADC			0x0
+#define SGTL5000_DAP_SEL_I2S_IN			0x1
+#define SGTL5000_DAC_SEL_MASK			0x0030
+#define SGTL5000_DAC_SEL_SHIFT			4
+#define SGTL5000_DAC_SEL_WIDTH			2
+#define SGTL5000_DAC_SEL_ADC			0x0
+#define SGTL5000_DAC_SEL_I2S_IN			0x1
+#define SGTL5000_DAC_SEL_DAP			0x3
+#define SGTL5000_I2S_OUT_SEL_MASK		0x0003
+#define SGTL5000_I2S_OUT_SEL_SHIFT		0
+#define SGTL5000_I2S_OUT_SEL_WIDTH		2
+#define SGTL5000_I2S_OUT_SEL_ADC		0x0
+#define SGTL5000_I2S_OUT_SEL_I2S_IN		0x1
+#define SGTL5000_I2S_OUT_SEL_DAP		0x3
+
+/*
+ * SGTL5000_CHIP_ADCDAC_CTRL
+ */
+#define SGTL5000_VOL_BUSY_DAC_RIGHT		0x2000
+#define SGTL5000_VOL_BUSY_DAC_LEFT		0x1000
+#define SGTL5000_DAC_VOL_RAMP_EN		0x0200
+#define SGTL5000_DAC_VOL_RAMP_EXPO		0x0100
+#define SGTL5000_DAC_MUTE_RIGHT			0x0008
+#define SGTL5000_DAC_MUTE_LEFT			0x0004
+#define SGTL5000_ADC_HPF_FREEZE			0x0002
+#define SGTL5000_ADC_HPF_BYPASS			0x0001
+
+/*
+ * SGTL5000_CHIP_DAC_VOL
+ */
+#define SGTL5000_DAC_VOL_RIGHT_MASK		0xff00
+#define SGTL5000_DAC_VOL_RIGHT_SHIFT		8
+#define SGTL5000_DAC_VOL_RIGHT_WIDTH		8
+#define SGTL5000_DAC_VOL_LEFT_MASK		0x00ff
+#define SGTL5000_DAC_VOL_LEFT_SHIFT		0
+#define SGTL5000_DAC_VOL_LEFT_WIDTH		8
+
+/*
+ * SGTL5000_CHIP_PAD_STRENGTH
+ */
+#define SGTL5000_PAD_I2S_LRCLK_MASK		0x0300
+#define SGTL5000_PAD_I2S_LRCLK_SHIFT		8
+#define SGTL5000_PAD_I2S_LRCLK_WIDTH		2
+#define SGTL5000_PAD_I2S_SCLK_MASK		0x00c0
+#define SGTL5000_PAD_I2S_SCLK_SHIFT		6
+#define SGTL5000_PAD_I2S_SCLK_WIDTH		2
+#define SGTL5000_PAD_I2S_DOUT_MASK		0x0030
+#define SGTL5000_PAD_I2S_DOUT_SHIFT		4
+#define SGTL5000_PAD_I2S_DOUT_WIDTH		2
+#define SGTL5000_PAD_I2C_SDA_MASK		0x000c
+#define SGTL5000_PAD_I2C_SDA_SHIFT		2
+#define SGTL5000_PAD_I2C_SDA_WIDTH		2
+#define SGTL5000_PAD_I2C_SCL_MASK		0x0003
+#define SGTL5000_PAD_I2C_SCL_SHIFT		0
+#define SGTL5000_PAD_I2C_SCL_WIDTH		2
+
+/*
+ * SGTL5000_CHIP_ANA_ADC_CTRL
+ */
+#define SGTL5000_ADC_VOL_M6DB			0x0100
+#define SGTL5000_ADC_VOL_RIGHT_MASK		0x00f0
+#define SGTL5000_ADC_VOL_RIGHT_SHIFT		4
+#define SGTL5000_ADC_VOL_RIGHT_WIDTH		4
+#define SGTL5000_ADC_VOL_LEFT_MASK		0x000f
+#define SGTL5000_ADC_VOL_LEFT_SHIFT		0
+#define SGTL5000_ADC_VOL_LEFT_WIDTH		4
+
+/*
+ * SGTL5000_CHIP_ANA_HP_CTRL
+ */
+#define SGTL5000_HP_VOL_RIGHT_MASK		0x7f00
+#define SGTL5000_HP_VOL_RIGHT_SHIFT		8
+#define SGTL5000_HP_VOL_RIGHT_WIDTH		7
+#define SGTL5000_HP_VOL_LEFT_MASK		0x007f
+#define SGTL5000_HP_VOL_LEFT_SHIFT		0
+#define SGTL5000_HP_VOL_LEFT_WIDTH		7
+
+/*
+ * SGTL5000_CHIP_ANA_CTRL
+ */
+#define SGTL5000_LINE_OUT_MUTE			0x0100
+#define SGTL5000_HP_SEL_MASK			0x0040
+#define SGTL5000_HP_SEL_SHIFT			6
+#define SGTL5000_HP_SEL_WIDTH			1
+#define SGTL5000_HP_SEL_DAC			0x0
+#define SGTL5000_HP_SEL_LINE_IN			0x1
+#define SGTL5000_HP_ZCD_EN			0x0020
+#define SGTL5000_HP_MUTE			0x0010
+#define SGTL5000_ADC_SEL_MASK			0x0004
+#define SGTL5000_ADC_SEL_SHIFT			2
+#define SGTL5000_ADC_SEL_WIDTH			1
+#define SGTL5000_ADC_SEL_MIC			0x0
+#define SGTL5000_ADC_SEL_LINE_IN		0x1
+#define SGTL5000_ADC_ZCD_EN			0x0002
+#define SGTL5000_ADC_MUTE			0x0001
+
+/*
+ * SGTL5000_CHIP_LINREG_CTRL
+ */
+#define SGTL5000_VDDC_MAN_ASSN_MASK		0x0040
+#define SGTL5000_VDDC_MAN_ASSN_SHIFT		6
+#define SGTL5000_VDDC_MAN_ASSN_WIDTH		1
+#define SGTL5000_VDDC_MAN_ASSN_VDDA		0x0
+#define SGTL5000_VDDC_MAN_ASSN_VDDIO		0x1
+#define SGTL5000_VDDC_ASSN_OVRD			0x0020
+#define SGTL5000_LINREG_VDDD_MASK		0x000f
+#define SGTL5000_LINREG_VDDD_SHIFT		0
+#define SGTL5000_LINREG_VDDD_WIDTH		4
+
+/*
+ * SGTL5000_CHIP_REF_CTRL
+ */
+#define SGTL5000_ANA_GND_MASK			0x01f0
+#define SGTL5000_ANA_GND_SHIFT			4
+#define SGTL5000_ANA_GND_WIDTH			5
+#define SGTL5000_ANA_GND_BASE			800	/* mv */
+#define SGTL5000_ANA_GND_STP			25	/*mv */
+#define SGTL5000_BIAS_CTRL_MASK			0x000e
+#define SGTL5000_BIAS_CTRL_SHIFT		1
+#define SGTL5000_BIAS_CTRL_WIDTH		3
+#define SGTL5000_SMALL_POP			0x0001
+
+/*
+ * SGTL5000_CHIP_MIC_CTRL
+ */
+#define SGTL5000_BIAS_R_MASK			0x0200
+#define SGTL5000_BIAS_R_SHIFT			8
+#define SGTL5000_BIAS_R_WIDTH			2
+#define SGTL5000_BIAS_R_off			0x0
+#define SGTL5000_BIAS_R_2K			0x1
+#define SGTL5000_BIAS_R_4k			0x2
+#define SGTL5000_BIAS_R_8k			0x3
+#define SGTL5000_BIAS_VOLT_MASK			0x0070
+#define SGTL5000_BIAS_VOLT_SHIFT		4
+#define SGTL5000_BIAS_VOLT_WIDTH		3
+#define SGTL5000_MIC_GAIN_MASK			0x0003
+#define SGTL5000_MIC_GAIN_SHIFT			0
+#define SGTL5000_MIC_GAIN_WIDTH			2
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_CTRL
+ */
+#define SGTL5000_LINE_OUT_CURRENT_MASK		0x0f00
+#define SGTL5000_LINE_OUT_CURRENT_SHIFT		8
+#define SGTL5000_LINE_OUT_CURRENT_WIDTH		4
+#define SGTL5000_LINE_OUT_CURRENT_180u		0x0
+#define SGTL5000_LINE_OUT_CURRENT_270u		0x1
+#define SGTL5000_LINE_OUT_CURRENT_360u		0x3
+#define SGTL5000_LINE_OUT_CURRENT_450u		0x7
+#define SGTL5000_LINE_OUT_CURRENT_540u		0xf
+#define SGTL5000_LINE_OUT_GND_MASK		0x003f
+#define SGTL5000_LINE_OUT_GND_SHIFT		0
+#define SGTL5000_LINE_OUT_GND_WIDTH		6
+#define SGTL5000_LINE_OUT_GND_BASE		800	/* mv */
+#define SGTL5000_LINE_OUT_GND_STP		25
+#define SGTL5000_LINE_OUT_GND_MAX		0x23
+
+/*
+ * SGTL5000_CHIP_LINE_OUT_VOL
+ */
+#define SGTL5000_LINE_OUT_VOL_RIGHT_MASK	0x1f00
+#define SGTL5000_LINE_OUT_VOL_RIGHT_SHIFT	8
+#define SGTL5000_LINE_OUT_VOL_RIGHT_WIDTH	5
+#define SGTL5000_LINE_OUT_VOL_LEFT_MASK		0x001f
+#define SGTL5000_LINE_OUT_VOL_LEFT_SHIFT	0
+#define SGTL5000_LINE_OUT_VOL_LEFT_WIDTH	5
+
+/*
+ * SGTL5000_CHIP_ANA_POWER
+ */
+#define SGTL5000_DAC_STEREO			0x4000
+#define SGTL5000_LINREG_SIMPLE_POWERUP		0x2000
+#define SGTL5000_STARTUP_POWERUP		0x1000
+#define SGTL5000_VDDC_CHRGPMP_POWERUP		0x0800
+#define SGTL5000_PLL_POWERUP			0x0400
+#define SGTL5000_LINEREG_D_POWERUP		0x0200
+#define SGTL5000_VCOAMP_POWERUP			0x0100
+#define SGTL5000_VAG_POWERUP			0x0080
+#define SGTL5000_ADC_STEREO			0x0040
+#define SGTL5000_REFTOP_POWERUP			0x0020
+#define SGTL5000_HP_POWERUP			0x0010
+#define SGTL5000_DAC_POWERUP			0x0008
+#define SGTL5000_CAPLESS_HP_POWERUP		0x0004
+#define SGTL5000_ADC_POWERUP			0x0002
+#define SGTL5000_LINE_OUT_POWERUP		0x0001
+
+/*
+ * SGTL5000_CHIP_PLL_CTRL
+ */
+#define SGTL5000_PLL_INT_DIV_MASK		0xf800
+#define SGTL5000_PLL_INT_DIV_SHIFT		11
+#define SGTL5000_PLL_INT_DIV_WIDTH		5
+#define SGTL5000_PLL_FRAC_DIV_MASK		0x0700
+#define SGTL5000_PLL_FRAC_DIV_SHIFT		0
+#define SGTL5000_PLL_FRAC_DIV_WIDTH		11
+
+/*
+ * SGTL5000_CHIP_CLK_TOP_CTRL
+ */
+#define SGTL5000_INT_OSC_EN			0x0800
+#define SGTL5000_INPUT_FREQ_DIV2		0x0008
+
+/*
+ * SGTL5000_CHIP_ANA_STATUS
+ */
+#define SGTL5000_HP_LRSHORT			0x0200
+#define SGTL5000_CAPLESS_SHORT			0x0100
+#define SGTL5000_PLL_LOCKED			0x0010
+
+/*
+ * SGTL5000_CHIP_SHORT_CTRL
+ */
+#define SGTL5000_LVLADJR_MASK			0x7000
+#define SGTL5000_LVLADJR_SHIFT			12
+#define SGTL5000_LVLADJR_WIDTH			3
+#define SGTL5000_LVLADJL_MASK			0x0700
+#define SGTL5000_LVLADJL_SHIFT			8
+#define SGTL5000_LVLADJL_WIDTH			3
+#define SGTL5000_LVLADJC_MASK			0x0070
+#define SGTL5000_LVLADJC_SHIFT			4
+#define SGTL5000_LVLADJC_WIDTH			3
+#define SGTL5000_LR_SHORT_MOD_MASK		0x000c
+#define SGTL5000_LR_SHORT_MOD_SHIFT		2
+#define SGTL5000_LR_SHORT_MOD_WIDTH		2
+#define SGTL5000_CM_SHORT_MOD_MASK		0x0003
+#define SGTL5000_CM_SHORT_MOD_SHIFT		0
+#define SGTL5000_CM_SHORT_MOD_WIDTH		2
+
+/*
+ *SGTL5000_CHIP_ANA_TEST2
+ */
+#define SGTL5000_MONO_DAC			0x1000
+
+/*
+ * SGTL5000_DAP_CTRL
+ */
+#define SGTL5000_DAP_MIX_EN			0x0010
+#define SGTL5000_DAP_EN				0x0001
+
+#define SGTL5000_SYSCLK				0x00
+#define SGTL5000_LRCLK				0x01
+
+#endif
diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c
index 40e285d..2a30eae 100644
--- a/sound/soc/codecs/sn95031.c
+++ b/sound/soc/codecs/sn95031.c
@@ -34,17 +34,135 @@
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
 #include <sound/tlv.h>
+#include <sound/jack.h>
 #include "sn95031.h"
 
 #define SN95031_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_44100)
 #define SN95031_FORMATS (SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE)
 
+/* adc helper functions */
+
+/* enables mic bias voltage */
+static void sn95031_enable_mic_bias(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_VAUD, BIT(2)|BIT(1)|BIT(0));
+	snd_soc_update_bits(codec, SN95031_MICBIAS, BIT(2), BIT(2));
+}
+
+/* Enable/Disable the ADC depending on the argument */
+static void configure_adc(struct snd_soc_codec *sn95031_codec, int val)
+{
+	int value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+	if (val) {
+		/* Enable and start the ADC */
+		value |= (SN95031_ADC_ENBL | SN95031_ADC_START);
+		value &= (~SN95031_ADC_NO_LOOP);
+	} else {
+		/* Just stop the ADC */
+		value &= (~SN95031_ADC_START);
+	}
+	snd_soc_write(sn95031_codec, SN95031_ADC1CNTL1, value);
+}
+
 /*
- * todo:
- * capture paths
- * jack detection
- * PM functions
+ * finds an empty channel for conversion
+ * If the ADC is not enabled then start using 0th channel
+ * itself. Otherwise find an empty channel by looking for a
+ * channel in which the stopbit is set to 1. returns the index
+ * of the first free channel if succeeds or an error code.
+ *
+ * Context: can sleep
+ *
  */
+static int find_free_channel(struct snd_soc_codec *sn95031_codec)
+{
+	int ret = 0, i, value;
+
+	/* check whether ADC is enabled */
+	value = snd_soc_read(sn95031_codec, SN95031_ADC1CNTL1);
+
+	if ((value & SN95031_ADC_ENBL) == 0)
+		return 0;
+
+	/* ADC is already enabled; Looking for an empty channel */
+	for (i = 0; i <	SN95031_ADC_CHANLS_MAX; i++) {
+		value = snd_soc_read(sn95031_codec,
+				SN95031_ADC_CHNL_START_ADDR + i);
+		if (value & SN95031_STOPBIT_MASK) {
+			ret = i;
+			break;
+		}
+	}
+	return (ret > SN95031_ADC_LOOP_MAX) ? (-EINVAL) : ret;
+}
+
+/* Initialize the ADC for reading micbias values. Can sleep. */
+static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
+{
+	int base_addr, chnl_addr;
+	int value;
+	static int channel_index;
+
+	/* Index of the first channel in which the stop bit is set */
+	channel_index = find_free_channel(sn95031_codec);
+	if (channel_index < 0) {
+		pr_err("No free ADC channels");
+		return channel_index;
+	}
+
+	base_addr = SN95031_ADC_CHNL_START_ADDR + channel_index;
+
+	if (!(channel_index == 0 || channel_index ==  SN95031_ADC_LOOP_MAX)) {
+		/* Reset stop bit for channels other than 0 and 12 */
+		value = snd_soc_read(sn95031_codec, base_addr);
+		/* Set the stop bit to zero */
+		snd_soc_write(sn95031_codec, base_addr, value & 0xEF);
+		/* Index of the first free channel */
+		base_addr++;
+		channel_index++;
+	}
+
+	/* Since this is the last channel, set the stop bit
+	   to 1 by ORing the DIE_SENSOR_CODE with 0x10 */
+	snd_soc_write(sn95031_codec, base_addr,
+				SN95031_AUDIO_DETECT_CODE | 0x10);
+
+	chnl_addr = SN95031_ADC_DATA_START_ADDR + 2 * channel_index;
+	pr_debug("mid_initialize : %x", chnl_addr);
+	configure_adc(sn95031_codec, 1);
+	return chnl_addr;
+}
+
+
+/* reads the ADC registers and gets the mic bias value in mV. */
+static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
+{
+	u16 adc_adr = sn95031_initialize_adc(codec);
+	u16 adc_val1, adc_val2;
+	unsigned int mic_bias;
+
+	sn95031_enable_mic_bias(codec);
+
+	/* Enable the sound card for conversion before reading */
+	snd_soc_write(codec, SN95031_ADC1CNTL3, 0x05);
+	/* Re-toggle the RRDATARD bit */
+	snd_soc_write(codec, SN95031_ADC1CNTL3, 0x04);
+
+	/* Read the higher bits of data */
+	msleep(1000);
+	adc_val1 = snd_soc_read(codec, adc_adr);
+	adc_adr++;
+	adc_val2 = snd_soc_read(codec, adc_adr);
+
+	/* Adding lower two bits to the higher bits */
+	mic_bias = (adc_val1 << 2) + (adc_val2 & 3);
+	mic_bias = (mic_bias * SN95031_ADC_ONE_LSB_MULTIPLIER) / 1000;
+	pr_debug("mic bias = %dmV\n", mic_bias);
+	return mic_bias;
+}
+EXPORT_SYMBOL_GPL(sn95031_get_mic_bias);
+/*end - adc helper functions */
 
 static inline unsigned int sn95031_read(struct snd_soc_codec *codec,
 			unsigned int reg)
@@ -241,7 +359,7 @@
 static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"};
 
 /* 0dB to 30dB in 10dB steps */
-static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 30);
+static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0);
 
 static const struct soc_enum sn95031_micmode1_enum =
 	SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text);
@@ -401,6 +519,8 @@
 
 static const struct snd_soc_dapm_route sn95031_audio_map[] = {
 	/* headset and earpiece map */
+	{ "HPOUTL", NULL, "Headset Rail"},
+	{ "HPOUTR", NULL, "Headset Rail"},
 	{ "HPOUTL", NULL, "Headset Left Playback" },
 	{ "HPOUTR", NULL, "Headset Right Playback" },
 	{ "EPOUT", NULL, "Earpiece Playback" },
@@ -409,18 +529,16 @@
 	{ "Earpiece Playback", NULL, "Headset Left Filter"},
 	{ "Headset Left Filter", NULL, "HSDAC Left"},
 	{ "Headset Right Filter", NULL, "HSDAC Right"},
-	{ "HSDAC Left", NULL, "Headset Rail"},
-	{ "HSDAC Right", NULL, "Headset Rail"},
 
 	/* speaker map */
+	{ "IHFOUTL", NULL, "Speaker Rail"},
+	{ "IHFOUTR", NULL, "Speaker Rail"},
 	{ "IHFOUTL", "NULL", "Speaker Left Playback"},
 	{ "IHFOUTR", "NULL", "Speaker Right Playback"},
 	{ "Speaker Left Playback", NULL, "Speaker Left Filter"},
 	{ "Speaker Right Playback", NULL, "Speaker Right Filter"},
 	{ "Speaker Left Filter", NULL, "IHFDAC Left"},
 	{ "Speaker Right Filter", NULL, "IHFDAC Right"},
-	{ "IHFDAC Left", NULL, "Speaker Rail"},
-	{ "IHFDAC Right", NULL, "Speaker Rail"},
 
 	/* vibra map */
 	{ "VIB1OUT", NULL, "Vibra1 Playback"},
@@ -484,30 +602,30 @@
 	{ "Txpath2 Capture Route", "ADC Right", "ADC Right"},
 	{ "Txpath3 Capture Route", "ADC Right", "ADC Right"},
 	{ "Txpath4 Capture Route", "ADC Right", "ADC Right"},
-	{ "Txpath1 Capture Route", NULL, "DMIC1"},
-	{ "Txpath2 Capture Route", NULL, "DMIC1"},
-	{ "Txpath3 Capture Route", NULL, "DMIC1"},
-	{ "Txpath4 Capture Route", NULL, "DMIC1"},
-	{ "Txpath1 Capture Route", NULL, "DMIC2"},
-	{ "Txpath2 Capture Route", NULL, "DMIC2"},
-	{ "Txpath3 Capture Route", NULL, "DMIC2"},
-	{ "Txpath4 Capture Route", NULL, "DMIC2"},
-	{ "Txpath1 Capture Route", NULL, "DMIC3"},
-	{ "Txpath2 Capture Route", NULL, "DMIC3"},
-	{ "Txpath3 Capture Route", NULL, "DMIC3"},
-	{ "Txpath4 Capture Route", NULL, "DMIC3"},
-	{ "Txpath1 Capture Route", NULL, "DMIC4"},
-	{ "Txpath2 Capture Route", NULL, "DMIC4"},
-	{ "Txpath3 Capture Route", NULL, "DMIC4"},
-	{ "Txpath4 Capture Route", NULL, "DMIC4"},
-	{ "Txpath1 Capture Route", NULL, "DMIC5"},
-	{ "Txpath2 Capture Route", NULL, "DMIC5"},
-	{ "Txpath3 Capture Route", NULL, "DMIC5"},
-	{ "Txpath4 Capture Route", NULL, "DMIC5"},
-	{ "Txpath1 Capture Route", NULL, "DMIC6"},
-	{ "Txpath2 Capture Route", NULL, "DMIC6"},
-	{ "Txpath3 Capture Route", NULL, "DMIC6"},
-	{ "Txpath4 Capture Route", NULL, "DMIC6"},
+	{ "Txpath1 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath2 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath3 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath4 Capture Route", "DMIC1", "DMIC1"},
+	{ "Txpath1 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath2 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath3 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath4 Capture Route", "DMIC2", "DMIC2"},
+	{ "Txpath1 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath2 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath3 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath4 Capture Route", "DMIC3", "DMIC3"},
+	{ "Txpath1 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath2 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath3 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath4 Capture Route", "DMIC4", "DMIC4"},
+	{ "Txpath1 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath2 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath3 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath4 Capture Route", "DMIC5", "DMIC5"},
+	{ "Txpath1 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath2 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath3 Capture Route", "DMIC6", "DMIC6"},
+	{ "Txpath4 Capture Route", "DMIC6", "DMIC6"},
 
 	/* tx path */
 	{ "TX1 Enable", NULL, "Txpath1 Capture Route"},
@@ -649,6 +767,61 @@
 },
 };
 
+static inline void sn95031_disable_jack_btn(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_BTNCTRL2, 0x00);
+}
+
+static inline void sn95031_enable_jack_btn(struct snd_soc_codec *codec)
+{
+	snd_soc_write(codec, SN95031_BTNCTRL1, 0x77);
+	snd_soc_write(codec, SN95031_BTNCTRL2, 0x01);
+}
+
+static int sn95031_get_headset_state(struct snd_soc_jack *mfld_jack)
+{
+	int micbias = sn95031_get_mic_bias(mfld_jack->codec);
+
+	int jack_type = snd_soc_jack_get_type(mfld_jack, micbias);
+
+	pr_debug("jack type detected = %d\n", jack_type);
+	if (jack_type == SND_JACK_HEADSET)
+		sn95031_enable_jack_btn(mfld_jack->codec);
+	return jack_type;
+}
+
+void sn95031_jack_detection(struct mfld_jack_data *jack_data)
+{
+	unsigned int status;
+	unsigned int mask = SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_HEADSET;
+
+	pr_debug("interrupt id read in sram = 0x%x\n", jack_data->intr_id);
+	if (jack_data->intr_id & 0x1) {
+		pr_debug("short_push detected\n");
+		status = SND_JACK_HEADSET | SND_JACK_BTN_0;
+	} else if (jack_data->intr_id & 0x2) {
+		pr_debug("long_push detected\n");
+		status = SND_JACK_HEADSET | SND_JACK_BTN_1;
+	} else if (jack_data->intr_id & 0x4) {
+		pr_debug("headset or headphones inserted\n");
+		status = sn95031_get_headset_state(jack_data->mfld_jack);
+	} else if (jack_data->intr_id & 0x8) {
+		pr_debug("headset or headphones removed\n");
+		status = 0;
+		sn95031_disable_jack_btn(jack_data->mfld_jack->codec);
+	} else {
+		pr_err("unidentified interrupt\n");
+		return;
+	}
+
+	snd_soc_jack_report(jack_data->mfld_jack, status, mask);
+	/*button pressed and released so we send explicit button release */
+	if ((status & SND_JACK_BTN_0) | (status & SND_JACK_BTN_1))
+		snd_soc_jack_report(jack_data->mfld_jack,
+				SND_JACK_HEADSET, mask);
+}
+EXPORT_SYMBOL_GPL(sn95031_jack_detection);
+
 /* codec registration */
 static int sn95031_codec_probe(struct snd_soc_codec *codec)
 {
diff --git a/sound/soc/codecs/sn95031.h b/sound/soc/codecs/sn95031.h
index e2b17d9..20376d2 100644
--- a/sound/soc/codecs/sn95031.h
+++ b/sound/soc/codecs/sn95031.h
@@ -96,4 +96,37 @@
 #define SN95031_SSR5			0x384
 #define SN95031_SSR6			0x385
 
+/* ADC registers */
+
+#define SN95031_ADC1CNTL1 0x1C0
+#define SN95031_ADC_ENBL 0x10
+#define SN95031_ADC_START 0x08
+#define SN95031_ADC1CNTL3 0x1C2
+#define SN95031_ADCTHERM_ENBL 0x04
+#define SN95031_ADCRRDATA_ENBL 0x05
+#define SN95031_STOPBIT_MASK 16
+#define SN95031_ADCTHERM_MASK 4
+#define SN95031_ADC_CHANLS_MAX 15 /* Number of ADC channels */
+#define SN95031_ADC_LOOP_MAX (SN95031_ADC_CHANLS_MAX - 1)
+#define SN95031_ADC_NO_LOOP 0x07
+#define SN95031_AUDIO_GPIO_CTRL 0x070
+
+/* ADC channel code values */
+#define SN95031_AUDIO_DETECT_CODE 0x06
+
+/* ADC base addresses */
+#define SN95031_ADC_CHNL_START_ADDR 0x1C5 /* increments by 1 */
+#define SN95031_ADC_DATA_START_ADDR 0x1D4  /* increments by 2 */
+/* multipier to convert to mV */
+#define SN95031_ADC_ONE_LSB_MULTIPLIER 2346
+
+
+struct mfld_jack_data {
+	int intr_id;
+	int micbias_vol;
+	struct snd_soc_jack *mfld_jack;
+};
+
+extern void sn95031_jack_detection(struct mfld_jack_data *jack_data);
+
 #endif
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
new file mode 100644
index 0000000..e93b9d1
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -0,0 +1,794 @@
+/*
+ * linux/sound/soc/codecs/tlv320aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.com>
+ *
+ * Based on sound/soc/codecs/wm8974 and TI driver for kernel 2.6.27.
+ *
+ * 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., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#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/platform_device.h>
+#include <linux/cdev.h>
+#include <linux/slab.h>
+
+#include <sound/tlv320aic32x4.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 "tlv320aic32x4.h"
+
+struct aic32x4_rate_divs {
+	u32 mclk;
+	u32 rate;
+	u8 p_val;
+	u8 pll_j;
+	u16 pll_d;
+	u16 dosr;
+	u8 ndac;
+	u8 mdac;
+	u8 aosr;
+	u8 nadc;
+	u8 madc;
+	u8 blck_N;
+};
+
+struct aic32x4_priv {
+	u32 sysclk;
+	s32 master;
+	u8 page_no;
+	void *control_data;
+	u32 power_cfg;
+	u32 micpga_routing;
+	bool swapdacs;
+};
+
+/* 0dB min, 1dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0);
+/* 0dB min, 0.5dB steps */
+static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0);
+
+static const struct snd_kcontrol_new aic32x4_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL,
+			AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5),
+	SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN,
+			AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1),
+	SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN,
+			AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1),
+	SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN,
+			AIC32X4_HPRGAIN, 6, 0x01, 1),
+	SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN,
+			AIC32X4_LORGAIN, 6, 0x01, 1),
+	SOC_DOUBLE_R("Mic PGA Switch", AIC32X4_LMICPGAVOL,
+			AIC32X4_RMICPGAVOL, 7, 0x01, 1),
+
+	SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0),
+	SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0),
+
+	SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL,
+			AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5),
+	SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL,
+			AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5),
+
+	SOC_SINGLE("Auto-mute Switch", AIC32X4_DACMUTE, 4, 7, 0),
+
+	SOC_SINGLE("AGC Left Switch", AIC32X4_LAGC1, 7, 1, 0),
+	SOC_SINGLE("AGC Right Switch", AIC32X4_RAGC1, 7, 1, 0),
+	SOC_DOUBLE_R("AGC Target Level", AIC32X4_LAGC1, AIC32X4_RAGC1,
+			4, 0x07, 0),
+	SOC_DOUBLE_R("AGC Gain Hysteresis", AIC32X4_LAGC1, AIC32X4_RAGC1,
+			0, 0x03, 0),
+	SOC_DOUBLE_R("AGC Hysteresis", AIC32X4_LAGC2, AIC32X4_RAGC2,
+			6, 0x03, 0),
+	SOC_DOUBLE_R("AGC Noise Threshold", AIC32X4_LAGC2, AIC32X4_RAGC2,
+			1, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Max PGA", AIC32X4_LAGC3, AIC32X4_RAGC3,
+			0, 0x7F, 0),
+	SOC_DOUBLE_R("AGC Attack Time", AIC32X4_LAGC4, AIC32X4_RAGC4,
+			3, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Decay Time", AIC32X4_LAGC5, AIC32X4_RAGC5,
+			3, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Noise Debounce", AIC32X4_LAGC6, AIC32X4_RAGC6,
+			0, 0x1F, 0),
+	SOC_DOUBLE_R("AGC Signal Debounce", AIC32X4_LAGC7, AIC32X4_RAGC7,
+			0, 0x0F, 0),
+};
+
+static const struct aic32x4_rate_divs aic32x4_divs[] = {
+	/* 8k rate */
+	{AIC32X4_FREQ_12000000, 8000, 1, 7, 6800, 768, 5, 3, 128, 5, 18, 24},
+	{AIC32X4_FREQ_24000000, 8000, 2, 7, 6800, 768, 15, 1, 64, 45, 4, 24},
+	{AIC32X4_FREQ_25000000, 8000, 2, 7, 3728, 768, 15, 1, 64, 45, 4, 24},
+	/* 11.025k rate */
+	{AIC32X4_FREQ_12000000, 11025, 1, 7, 5264, 512, 8, 2, 128, 8, 8, 16},
+	{AIC32X4_FREQ_24000000, 11025, 2, 7, 5264, 512, 16, 1, 64, 32, 4, 16},
+	/* 16k rate */
+	{AIC32X4_FREQ_12000000, 16000, 1, 7, 6800, 384, 5, 3, 128, 5, 9, 12},
+	{AIC32X4_FREQ_24000000, 16000, 2, 7, 6800, 384, 15, 1, 64, 18, 5, 12},
+	{AIC32X4_FREQ_25000000, 16000, 2, 7, 3728, 384, 15, 1, 64, 18, 5, 12},
+	/* 22.05k rate */
+	{AIC32X4_FREQ_12000000, 22050, 1, 7, 5264, 256, 4, 4, 128, 4, 8, 8},
+	{AIC32X4_FREQ_24000000, 22050, 2, 7, 5264, 256, 16, 1, 64, 16, 4, 8},
+	{AIC32X4_FREQ_25000000, 22050, 2, 7, 2253, 256, 16, 1, 64, 16, 4, 8},
+	/* 32k rate */
+	{AIC32X4_FREQ_12000000, 32000, 1, 7, 1680, 192, 2, 7, 64, 2, 21, 6},
+	{AIC32X4_FREQ_24000000, 32000, 2, 7, 1680, 192, 7, 2, 64, 7, 6, 6},
+	/* 44.1k rate */
+	{AIC32X4_FREQ_12000000, 44100, 1, 7, 5264, 128, 2, 8, 128, 2, 8, 4},
+	{AIC32X4_FREQ_24000000, 44100, 2, 7, 5264, 128, 8, 2, 64, 8, 4, 4},
+	{AIC32X4_FREQ_25000000, 44100, 2, 7, 2253, 128, 8, 2, 64, 8, 4, 4},
+	/* 48k rate */
+	{AIC32X4_FREQ_12000000, 48000, 1, 8, 1920, 128, 2, 8, 128, 2, 8, 4},
+	{AIC32X4_FREQ_24000000, 48000, 2, 8, 1920, 128, 8, 2, 64, 8, 4, 4},
+	{AIC32X4_FREQ_25000000, 48000, 2, 7, 8643, 128, 8, 2, 64, 8, 4, 4}
+};
+
+static const struct snd_kcontrol_new hpl_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_HPLROUTE, 3, 1, 0),
+	SOC_DAPM_SINGLE("IN1_L Switch", AIC32X4_HPLROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new hpr_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_HPRROUTE, 3, 1, 0),
+	SOC_DAPM_SINGLE("IN1_R Switch", AIC32X4_HPRROUTE, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new lol_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("L_DAC Switch", AIC32X4_LOLROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new lor_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("R_DAC Switch", AIC32X4_LORROUTE, 3, 1, 0),
+};
+
+static const struct snd_kcontrol_new left_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("IN1_L P Switch", AIC32X4_LMICPGAPIN, 6, 1, 0),
+	SOC_DAPM_SINGLE("IN2_L P Switch", AIC32X4_LMICPGAPIN, 4, 1, 0),
+	SOC_DAPM_SINGLE("IN3_L P Switch", AIC32X4_LMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_kcontrol_new right_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("IN1_R P Switch", AIC32X4_RMICPGAPIN, 6, 1, 0),
+	SOC_DAPM_SINGLE("IN2_R P Switch", AIC32X4_RMICPGAPIN, 4, 1, 0),
+	SOC_DAPM_SINGLE("IN3_R P Switch", AIC32X4_RMICPGAPIN, 2, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("Left DAC", "Left Playback", AIC32X4_DACSETUP, 7, 0),
+	SND_SOC_DAPM_MIXER("HPL Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &hpl_output_mixer_controls[0],
+			   ARRAY_SIZE(hpl_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPL Power", AIC32X4_OUTPWRCTL, 5, 0, NULL, 0),
+
+	SND_SOC_DAPM_MIXER("LOL Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &lol_output_mixer_controls[0],
+			   ARRAY_SIZE(lol_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("LOL Power", AIC32X4_OUTPWRCTL, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("Right DAC", "Right Playback", AIC32X4_DACSETUP, 6, 0),
+	SND_SOC_DAPM_MIXER("HPR Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &hpr_output_mixer_controls[0],
+			   ARRAY_SIZE(hpr_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("HPR Power", AIC32X4_OUTPWRCTL, 4, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("LOR Output Mixer", SND_SOC_NOPM, 0, 0,
+			   &lor_output_mixer_controls[0],
+			   ARRAY_SIZE(lor_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0,
+			   &left_input_mixer_controls[0],
+			   ARRAY_SIZE(left_input_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0,
+			   &right_input_mixer_controls[0],
+			   ARRAY_SIZE(right_input_mixer_controls)),
+	SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0),
+
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("LOL"),
+	SND_SOC_DAPM_OUTPUT("LOR"),
+	SND_SOC_DAPM_INPUT("IN1_L"),
+	SND_SOC_DAPM_INPUT("IN1_R"),
+	SND_SOC_DAPM_INPUT("IN2_L"),
+	SND_SOC_DAPM_INPUT("IN2_R"),
+	SND_SOC_DAPM_INPUT("IN3_L"),
+	SND_SOC_DAPM_INPUT("IN3_R"),
+};
+
+static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = {
+	/* Left Output */
+	{"HPL Output Mixer", "L_DAC Switch", "Left DAC"},
+	{"HPL Output Mixer", "IN1_L Switch", "IN1_L"},
+
+	{"HPL Power", NULL, "HPL Output Mixer"},
+	{"HPL", NULL, "HPL Power"},
+
+	{"LOL Output Mixer", "L_DAC Switch", "Left DAC"},
+
+	{"LOL Power", NULL, "LOL Output Mixer"},
+	{"LOL", NULL, "LOL Power"},
+
+	/* Right Output */
+	{"HPR Output Mixer", "R_DAC Switch", "Right DAC"},
+	{"HPR Output Mixer", "IN1_R Switch", "IN1_R"},
+
+	{"HPR Power", NULL, "HPR Output Mixer"},
+	{"HPR", NULL, "HPR Power"},
+
+	{"LOR Output Mixer", "R_DAC Switch", "Right DAC"},
+
+	{"LOR Power", NULL, "LOR Output Mixer"},
+	{"LOR", NULL, "LOR Power"},
+
+	/* Left input */
+	{"Left Input Mixer", "IN1_L P Switch", "IN1_L"},
+	{"Left Input Mixer", "IN2_L P Switch", "IN2_L"},
+	{"Left Input Mixer", "IN3_L P Switch", "IN3_L"},
+
+	{"Left ADC", NULL, "Left Input Mixer"},
+
+	/* Right Input */
+	{"Right Input Mixer", "IN1_R P Switch", "IN1_R"},
+	{"Right Input Mixer", "IN2_R P Switch", "IN2_R"},
+	{"Right Input Mixer", "IN3_R P Switch", "IN3_R"},
+
+	{"Right ADC", NULL, "Right Input Mixer"},
+};
+
+static inline int aic32x4_change_page(struct snd_soc_codec *codec,
+					unsigned int new_page)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 data[2];
+	int ret;
+
+	data[0] = 0x00;
+	data[1] = new_page & 0xff;
+
+	ret = codec->hw_write(codec->control_data, data, 2);
+	if (ret == 2) {
+		aic32x4->page_no = new_page;
+		return 0;
+	} else {
+		return ret;
+	}
+}
+
+static int aic32x4_write(struct snd_soc_codec *codec, unsigned int reg,
+				unsigned int val)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	unsigned int page = reg / 128;
+	unsigned int fixed_reg = reg % 128;
+	u8 data[2];
+	int ret;
+
+	/* A write to AIC32X4_PSEL is really a non-explicit page change */
+	if (reg == AIC32X4_PSEL)
+		return aic32x4_change_page(codec, val);
+
+	if (aic32x4->page_no != page) {
+		ret = aic32x4_change_page(codec, page);
+		if (ret != 0)
+			return ret;
+	}
+
+	data[0] = fixed_reg & 0xff;
+	data[1] = val & 0xff;
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+	else
+		return -EIO;
+}
+
+static unsigned int aic32x4_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	unsigned int page = reg / 128;
+	unsigned int fixed_reg = reg % 128;
+	int ret;
+
+	if (aic32x4->page_no != page) {
+		ret = aic32x4_change_page(codec, page);
+		if (ret != 0)
+			return ret;
+	}
+	return i2c_smbus_read_byte_data(codec->control_data, fixed_reg & 0xff);
+}
+
+static inline int aic32x4_get_divs(int mclk, int rate)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(aic32x4_divs); i++) {
+		if ((aic32x4_divs[i].rate == rate)
+		    && (aic32x4_divs[i].mclk == mclk)) {
+			return i;
+		}
+	}
+	printk(KERN_ERR "aic32x4: master clock and sample rate is not supported\n");
+	return -EINVAL;
+}
+
+static int aic32x4_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
+				  ARRAY_SIZE(aic32x4_dapm_widgets));
+
+	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
+				ARRAY_SIZE(aic32x4_dapm_routes));
+
+	snd_soc_dapm_new_widgets(&codec->dapm);
+	return 0;
+}
+
+static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+
+	switch (freq) {
+	case AIC32X4_FREQ_12000000:
+	case AIC32X4_FREQ_24000000:
+	case AIC32X4_FREQ_25000000:
+		aic32x4->sysclk = freq;
+		return 0;
+	}
+	printk(KERN_ERR "aic32x4: invalid frequency to set DAI system clock\n");
+	return -EINVAL;
+}
+
+static int aic32x4_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 iface_reg_1;
+	u8 iface_reg_2;
+	u8 iface_reg_3;
+
+	iface_reg_1 = snd_soc_read(codec, AIC32X4_IFACE1);
+	iface_reg_1 = iface_reg_1 & ~(3 << 6 | 3 << 2);
+	iface_reg_2 = snd_soc_read(codec, AIC32X4_IFACE2);
+	iface_reg_2 = 0;
+	iface_reg_3 = snd_soc_read(codec, AIC32X4_IFACE3);
+	iface_reg_3 = iface_reg_3 & ~(1 << 3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		aic32x4->master = 1;
+		iface_reg_1 |= AIC32X4_BCLKMASTER | AIC32X4_WCLKMASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		aic32x4->master = 0;
+		break;
+	default:
+		printk(KERN_ERR "aic32x4: invalid DAI master/slave interface\n");
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+		iface_reg_3 |= (1 << 3); /* invert bit clock */
+		iface_reg_2 = 0x01; /* add offset 1 */
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		iface_reg_1 |= (AIC32X4_DSP_MODE << AIC32X4_PLLJ_SHIFT);
+		iface_reg_3 |= (1 << 3); /* invert bit clock */
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iface_reg_1 |=
+			(AIC32X4_RIGHT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg_1 |=
+			(AIC32X4_LEFT_JUSTIFIED_MODE << AIC32X4_PLLJ_SHIFT);
+		break;
+	default:
+		printk(KERN_ERR "aic32x4: invalid DAI interface format\n");
+		return -EINVAL;
+	}
+
+	snd_soc_write(codec, AIC32X4_IFACE1, iface_reg_1);
+	snd_soc_write(codec, AIC32X4_IFACE2, iface_reg_2);
+	snd_soc_write(codec, AIC32X4_IFACE3, iface_reg_3);
+	return 0;
+}
+
+static int aic32x4_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 aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 data;
+	int i;
+
+	i = aic32x4_get_divs(aic32x4->sysclk, params_rate(params));
+	if (i < 0) {
+		printk(KERN_ERR "aic32x4: sampling rate not supported\n");
+		return i;
+	}
+
+	/* Use PLL as CODEC_CLKIN and DAC_MOD_CLK as BDIV_CLKIN */
+	snd_soc_write(codec, AIC32X4_CLKMUX, AIC32X4_PLLCLKIN);
+	snd_soc_write(codec, AIC32X4_IFACE3, AIC32X4_DACMOD2BCLK);
+
+	/* We will fix R value to 1 and will make P & J=K.D as varialble */
+	data = snd_soc_read(codec, AIC32X4_PLLPR);
+	data &= ~(7 << 4);
+	snd_soc_write(codec, AIC32X4_PLLPR,
+		      (data | (aic32x4_divs[i].p_val << 4) | 0x01));
+
+	snd_soc_write(codec, AIC32X4_PLLJ, aic32x4_divs[i].pll_j);
+
+	snd_soc_write(codec, AIC32X4_PLLDMSB, (aic32x4_divs[i].pll_d >> 8));
+	snd_soc_write(codec, AIC32X4_PLLDLSB,
+		      (aic32x4_divs[i].pll_d & 0xff));
+
+	/* NDAC divider value */
+	data = snd_soc_read(codec, AIC32X4_NDAC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_NDAC, data | aic32x4_divs[i].ndac);
+
+	/* MDAC divider value */
+	data = snd_soc_read(codec, AIC32X4_MDAC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_MDAC, data | aic32x4_divs[i].mdac);
+
+	/* DOSR MSB & LSB values */
+	snd_soc_write(codec, AIC32X4_DOSRMSB, aic32x4_divs[i].dosr >> 8);
+	snd_soc_write(codec, AIC32X4_DOSRLSB,
+		      (aic32x4_divs[i].dosr & 0xff));
+
+	/* NADC divider value */
+	data = snd_soc_read(codec, AIC32X4_NADC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_NADC, data | aic32x4_divs[i].nadc);
+
+	/* MADC divider value */
+	data = snd_soc_read(codec, AIC32X4_MADC);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_MADC, data | aic32x4_divs[i].madc);
+
+	/* AOSR value */
+	snd_soc_write(codec, AIC32X4_AOSR, aic32x4_divs[i].aosr);
+
+	/* BCLK N divider */
+	data = snd_soc_read(codec, AIC32X4_BCLKN);
+	data &= ~(0x7f);
+	snd_soc_write(codec, AIC32X4_BCLKN, data | aic32x4_divs[i].blck_N);
+
+	data = snd_soc_read(codec, AIC32X4_IFACE1);
+	data = data & ~(3 << 4);
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		data |= (AIC32X4_WORD_LEN_20BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		data |= (AIC32X4_WORD_LEN_24BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		data |= (AIC32X4_WORD_LEN_32BITS << AIC32X4_DOSRMSB_SHIFT);
+		break;
+	}
+	snd_soc_write(codec, AIC32X4_IFACE1, data);
+
+	return 0;
+}
+
+static int aic32x4_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 dac_reg;
+
+	dac_reg = snd_soc_read(codec, AIC32X4_DACMUTE) & ~AIC32X4_MUTEON;
+	if (mute)
+		snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg | AIC32X4_MUTEON);
+	else
+		snd_soc_write(codec, AIC32X4_DACMUTE, dac_reg);
+	return 0;
+}
+
+static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
+				  enum snd_soc_bias_level level)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u8 value;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		if (aic32x4->master) {
+			/* Switch on PLL */
+			value = snd_soc_read(codec, AIC32X4_PLLPR);
+			snd_soc_write(codec, AIC32X4_PLLPR,
+				      (value | AIC32X4_PLLEN));
+
+			/* Switch on NDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_write(codec, AIC32X4_NDAC,
+				      value | AIC32X4_NDACEN);
+
+			/* Switch on MDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_write(codec, AIC32X4_MDAC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on NADC Divider */
+			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_write(codec, AIC32X4_NADC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on MADC Divider */
+			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_write(codec, AIC32X4_MADC,
+				      value | AIC32X4_MDACEN);
+
+			/* Switch on BCLK_N Divider */
+			value = snd_soc_read(codec, AIC32X4_BCLKN);
+			snd_soc_write(codec, AIC32X4_BCLKN,
+				      value | AIC32X4_BCLKEN);
+		}
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (aic32x4->master) {
+			/* Switch off PLL */
+			value = snd_soc_read(codec, AIC32X4_PLLPR);
+			snd_soc_write(codec, AIC32X4_PLLPR,
+				      (value & ~AIC32X4_PLLEN));
+
+			/* Switch off NDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_write(codec, AIC32X4_NDAC,
+				      value & ~AIC32X4_NDACEN);
+
+			/* Switch off MDAC Divider */
+			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_write(codec, AIC32X4_MDAC,
+				      value & ~AIC32X4_MDACEN);
+
+			/* Switch off NADC Divider */
+			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_write(codec, AIC32X4_NADC,
+				      value & ~AIC32X4_NDACEN);
+
+			/* Switch off MADC Divider */
+			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_write(codec, AIC32X4_MADC,
+				      value & ~AIC32X4_MDACEN);
+			value = snd_soc_read(codec, AIC32X4_BCLKN);
+
+			/* Switch off BCLK_N Divider */
+			snd_soc_write(codec, AIC32X4_BCLKN,
+				      value & ~AIC32X4_BCLKEN);
+		}
+		break;
+	case SND_SOC_BIAS_OFF:
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define AIC32X4_RATES	SNDRV_PCM_RATE_8000_48000
+#define AIC32X4_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
+			 | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_ops aic32x4_ops = {
+	.hw_params = aic32x4_hw_params,
+	.digital_mute = aic32x4_mute,
+	.set_fmt = aic32x4_set_dai_fmt,
+	.set_sysclk = aic32x4_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_driver aic32x4_dai = {
+	.name = "tlv320aic32x4-hifi",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 1,
+		     .channels_max = 2,
+		     .rates = AIC32X4_RATES,
+		     .formats = AIC32X4_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 1,
+		    .channels_max = 2,
+		    .rates = AIC32X4_RATES,
+		    .formats = AIC32X4_FORMATS,},
+	.ops = &aic32x4_ops,
+	.symmetric_rates = 1,
+};
+
+static int aic32x4_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int aic32x4_resume(struct snd_soc_codec *codec)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int aic32x4_probe(struct snd_soc_codec *codec)
+{
+	struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec);
+	u32 tmp_reg;
+
+	codec->hw_write = (hw_write_t) i2c_master_send;
+	codec->control_data = aic32x4->control_data;
+
+	snd_soc_write(codec, AIC32X4_RESET, 0x01);
+
+	/* Power platform configuration */
+	if (aic32x4->power_cfg & AIC32X4_PWR_MICBIAS_2075_LDOIN) {
+		snd_soc_write(codec, AIC32X4_MICBIAS, AIC32X4_MICBIAS_LDOIN |
+						      AIC32X4_MICBIAS_2075V);
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_AVDD_DVDD_WEAK_DISABLE) {
+		snd_soc_write(codec, AIC32X4_PWRCFG, AIC32X4_AVDDWEAKDISABLE);
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_AIC32X4_LDO_ENABLE) {
+		snd_soc_write(codec, AIC32X4_LDOCTL, AIC32X4_LDOCTLEN);
+	}
+	tmp_reg = snd_soc_read(codec, AIC32X4_CMMODE);
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_LDOIN_RANGE_18_36) {
+		tmp_reg |= AIC32X4_LDOIN_18_36;
+	}
+	if (aic32x4->power_cfg & AIC32X4_PWR_CMMODE_HP_LDOIN_POWERED) {
+		tmp_reg |= AIC32X4_LDOIN2HP;
+	}
+	snd_soc_write(codec, AIC32X4_CMMODE, tmp_reg);
+
+	/* Do DACs need to be swapped? */
+	if (aic32x4->swapdacs) {
+		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2RCHN | AIC32X4_RDAC2LCHN);
+	} else {
+		snd_soc_write(codec, AIC32X4_DACSETUP, AIC32X4_LDAC2LCHN | AIC32X4_RDAC2RCHN);
+	}
+
+	/* Mic PGA routing */
+	if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
+		snd_soc_write(codec, AIC32X4_LMICPGANIN, AIC32X4_LMICPGANIN_IN2R_10K);
+	}
+	if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) {
+		snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
+	}
+
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	snd_soc_add_controls(codec, aic32x4_snd_controls,
+			     ARRAY_SIZE(aic32x4_snd_controls));
+	aic32x4_add_widgets(codec);
+
+	return 0;
+}
+
+static int aic32x4_remove(struct snd_soc_codec *codec)
+{
+	aic32x4_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
+	.read = aic32x4_read,
+	.write = aic32x4_write,
+	.probe = aic32x4_probe,
+	.remove = aic32x4_remove,
+	.suspend = aic32x4_suspend,
+	.resume = aic32x4_resume,
+	.set_bias_level = aic32x4_set_bias_level,
+};
+
+static __devinit int aic32x4_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct aic32x4_pdata *pdata = i2c->dev.platform_data;
+	struct aic32x4_priv *aic32x4;
+	int ret;
+
+	aic32x4 = kzalloc(sizeof(struct aic32x4_priv), GFP_KERNEL);
+	if (aic32x4 == NULL)
+		return -ENOMEM;
+
+	aic32x4->control_data = i2c;
+	i2c_set_clientdata(i2c, aic32x4);
+
+	if (pdata) {
+		aic32x4->power_cfg = pdata->power_cfg;
+		aic32x4->swapdacs = pdata->swapdacs;
+		aic32x4->micpga_routing = pdata->micpga_routing;
+	} else {
+		aic32x4->power_cfg = 0;
+		aic32x4->swapdacs = false;
+		aic32x4->micpga_routing = 0;
+	}
+
+	ret = snd_soc_register_codec(&i2c->dev,
+			&soc_codec_dev_aic32x4, &aic32x4_dai, 1);
+	if (ret < 0)
+		kfree(aic32x4);
+	return ret;
+}
+
+static __devexit int aic32x4_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	kfree(i2c_get_clientdata(client));
+	return 0;
+}
+
+static const struct i2c_device_id aic32x4_i2c_id[] = {
+	{ "tlv320aic32x4", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id);
+
+static struct i2c_driver aic32x4_i2c_driver = {
+	.driver = {
+		.name = "tlv320aic32x4",
+		.owner = THIS_MODULE,
+	},
+	.probe =    aic32x4_i2c_probe,
+	.remove =   __devexit_p(aic32x4_i2c_remove),
+	.id_table = aic32x4_i2c_id,
+};
+
+static int __init aic32x4_modinit(void)
+{
+	int ret = 0;
+
+	ret = i2c_add_driver(&aic32x4_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register aic32x4 I2C driver: %d\n",
+		       ret);
+	}
+	return ret;
+}
+module_init(aic32x4_modinit);
+
+static void __exit aic32x4_exit(void)
+{
+	i2c_del_driver(&aic32x4_i2c_driver);
+}
+module_exit(aic32x4_exit);
+
+MODULE_DESCRIPTION("ASoC tlv320aic32x4 codec driver");
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic32x4.h b/sound/soc/codecs/tlv320aic32x4.h
new file mode 100644
index 0000000..aae2b24
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic32x4.h
@@ -0,0 +1,143 @@
+/*
+ * tlv320aic32x4.h
+ *
+ * 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 _TLV320AIC32X4_H
+#define _TLV320AIC32X4_H
+
+/* tlv320aic32x4 register space (in decimal to match datasheet) */
+
+#define AIC32X4_PAGE1		128
+
+#define	AIC32X4_PSEL		0
+#define	AIC32X4_RESET		1
+#define	AIC32X4_CLKMUX		4
+#define	AIC32X4_PLLPR		5
+#define	AIC32X4_PLLJ		6
+#define	AIC32X4_PLLDMSB		7
+#define	AIC32X4_PLLDLSB		8
+#define	AIC32X4_NDAC		11
+#define	AIC32X4_MDAC		12
+#define AIC32X4_DOSRMSB		13
+#define AIC32X4_DOSRLSB		14
+#define	AIC32X4_NADC		18
+#define	AIC32X4_MADC		19
+#define AIC32X4_AOSR		20
+#define AIC32X4_CLKMUX2		25
+#define AIC32X4_CLKOUTM		26
+#define AIC32X4_IFACE1		27
+#define AIC32X4_IFACE2		28
+#define AIC32X4_IFACE3		29
+#define AIC32X4_BCLKN		30
+#define AIC32X4_IFACE4		31
+#define AIC32X4_IFACE5		32
+#define AIC32X4_IFACE6		33
+#define AIC32X4_DOUTCTL		53
+#define AIC32X4_DINCTL		54
+#define AIC32X4_DACSPB		60
+#define AIC32X4_ADCSPB		61
+#define AIC32X4_DACSETUP	63
+#define AIC32X4_DACMUTE		64
+#define AIC32X4_LDACVOL		65
+#define AIC32X4_RDACVOL		66
+#define AIC32X4_ADCSETUP	81
+#define	AIC32X4_ADCFGA		82
+#define AIC32X4_LADCVOL		83
+#define AIC32X4_RADCVOL		84
+#define AIC32X4_LAGC1		86
+#define AIC32X4_LAGC2		87
+#define AIC32X4_LAGC3		88
+#define AIC32X4_LAGC4		89
+#define AIC32X4_LAGC5		90
+#define AIC32X4_LAGC6		91
+#define AIC32X4_LAGC7		92
+#define AIC32X4_RAGC1		94
+#define AIC32X4_RAGC2		95
+#define AIC32X4_RAGC3		96
+#define AIC32X4_RAGC4		97
+#define AIC32X4_RAGC5		98
+#define AIC32X4_RAGC6		99
+#define AIC32X4_RAGC7		100
+#define AIC32X4_PWRCFG		(AIC32X4_PAGE1 + 1)
+#define AIC32X4_LDOCTL		(AIC32X4_PAGE1 + 2)
+#define AIC32X4_OUTPWRCTL	(AIC32X4_PAGE1 + 9)
+#define AIC32X4_CMMODE		(AIC32X4_PAGE1 + 10)
+#define AIC32X4_HPLROUTE	(AIC32X4_PAGE1 + 12)
+#define AIC32X4_HPRROUTE	(AIC32X4_PAGE1 + 13)
+#define AIC32X4_LOLROUTE	(AIC32X4_PAGE1 + 14)
+#define AIC32X4_LORROUTE	(AIC32X4_PAGE1 + 15)
+#define	AIC32X4_HPLGAIN		(AIC32X4_PAGE1 + 16)
+#define	AIC32X4_HPRGAIN		(AIC32X4_PAGE1 + 17)
+#define	AIC32X4_LOLGAIN		(AIC32X4_PAGE1 + 18)
+#define	AIC32X4_LORGAIN		(AIC32X4_PAGE1 + 19)
+#define AIC32X4_HEADSTART	(AIC32X4_PAGE1 + 20)
+#define AIC32X4_MICBIAS		(AIC32X4_PAGE1 + 51)
+#define AIC32X4_LMICPGAPIN	(AIC32X4_PAGE1 + 52)
+#define AIC32X4_LMICPGANIN	(AIC32X4_PAGE1 + 54)
+#define AIC32X4_RMICPGAPIN	(AIC32X4_PAGE1 + 55)
+#define AIC32X4_RMICPGANIN	(AIC32X4_PAGE1 + 57)
+#define AIC32X4_FLOATINGINPUT	(AIC32X4_PAGE1 + 58)
+#define AIC32X4_LMICPGAVOL	(AIC32X4_PAGE1 + 59)
+#define AIC32X4_RMICPGAVOL	(AIC32X4_PAGE1 + 60)
+
+#define AIC32X4_FREQ_12000000 12000000
+#define AIC32X4_FREQ_24000000 24000000
+#define AIC32X4_FREQ_25000000 25000000
+
+#define AIC32X4_WORD_LEN_16BITS		0x00
+#define AIC32X4_WORD_LEN_20BITS		0x01
+#define AIC32X4_WORD_LEN_24BITS		0x02
+#define AIC32X4_WORD_LEN_32BITS		0x03
+
+#define AIC32X4_I2S_MODE		0x00
+#define AIC32X4_DSP_MODE		0x01
+#define AIC32X4_RIGHT_JUSTIFIED_MODE	0x02
+#define AIC32X4_LEFT_JUSTIFIED_MODE	0x03
+
+#define AIC32X4_AVDDWEAKDISABLE		0x08
+#define AIC32X4_LDOCTLEN		0x01
+
+#define AIC32X4_LDOIN_18_36		0x01
+#define AIC32X4_LDOIN2HP		0x02
+
+#define AIC32X4_DACSPBLOCK_MASK		0x1f
+#define AIC32X4_ADCSPBLOCK_MASK		0x1f
+
+#define AIC32X4_PLLJ_SHIFT		6
+#define AIC32X4_DOSRMSB_SHIFT		4
+
+#define AIC32X4_PLLCLKIN		0x03
+
+#define AIC32X4_MICBIAS_LDOIN		0x08
+#define AIC32X4_MICBIAS_2075V		0x60
+
+#define AIC32X4_LMICPGANIN_IN2R_10K	0x10
+#define AIC32X4_RMICPGANIN_IN1L_10K	0x10
+
+#define AIC32X4_LMICPGAVOL_NOGAIN	0x80
+#define AIC32X4_RMICPGAVOL_NOGAIN	0x80
+
+#define AIC32X4_BCLKMASTER		0x08
+#define AIC32X4_WCLKMASTER		0x04
+#define AIC32X4_PLLEN			(0x01 << 7)
+#define AIC32X4_NDACEN			(0x01 << 7)
+#define AIC32X4_MDACEN			(0x01 << 7)
+#define AIC32X4_NADCEN			(0x01 << 7)
+#define AIC32X4_MADCEN			(0x01 << 7)
+#define AIC32X4_BCLKEN			(0x01 << 7)
+#define AIC32X4_DACEN			(0x03 << 6)
+#define AIC32X4_RDAC2LCHN		(0x02 << 2)
+#define AIC32X4_LDAC2RCHN		(0x02 << 4)
+#define AIC32X4_LDAC2LCHN		(0x01 << 4)
+#define AIC32X4_RDAC2RCHN		(0x01 << 2)
+
+#define AIC32X4_SSTEP2WCLK		0x01
+#define AIC32X4_MUTEON			0x0C
+#define	AIC32X4_DACMOD2BCLK		0x01
+
+#endif				/* _TLV320AIC32X4_H */
diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c
index 71d7be8..00b6d87 100644
--- a/sound/soc/codecs/tlv320dac33.c
+++ b/sound/soc/codecs/tlv320dac33.c
@@ -1615,6 +1615,7 @@
 	},
 	{ },
 };
+MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id);
 
 static struct i2c_driver tlv320dac33_i2c_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c
index 80ddf4f..a3b9cbb 100644
--- a/sound/soc/codecs/wm2000.c
+++ b/sound/soc/codecs/wm2000.c
@@ -836,24 +836,25 @@
 }
 
 #ifdef CONFIG_PM
-static int wm2000_i2c_suspend(struct i2c_client *i2c, pm_message_t mesg)
+static int wm2000_i2c_suspend(struct device *dev)
 {
+	struct i2c_client *i2c = to_i2c_client(dev);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 
 	return wm2000_anc_transition(wm2000, ANC_OFF);
 }
 
-static int wm2000_i2c_resume(struct i2c_client *i2c)
+static int wm2000_i2c_resume(struct device *dev)
 {
+	struct i2c_client *i2c = to_i2c_client(dev);
 	struct wm2000_priv *wm2000 = dev_get_drvdata(&i2c->dev);
 
 	return wm2000_anc_set_mode(wm2000);
 }
-#else
-#define wm2000_i2c_suspend NULL
-#define wm2000_i2c_resume NULL
 #endif
 
+static SIMPLE_DEV_PM_OPS(wm2000_pm, wm2000_i2c_suspend, wm2000_i2c_resume);
+
 static const struct i2c_device_id wm2000_i2c_id[] = {
 	{ "wm2000", 0 },
 	{ }
@@ -864,11 +865,10 @@
 	.driver = {
 		.name = "wm2000",
 		.owner = THIS_MODULE,
+		.pm = &wm2000_pm,
 	},
 	.probe = wm2000_i2c_probe,
 	.remove = __devexit_p(wm2000_i2c_remove),
-	.suspend = wm2000_i2c_suspend,
-	.resume = wm2000_i2c_resume,
 	.shutdown = wm2000_i2c_shutdown,
 	.id_table = wm2000_i2c_id,
 };
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 79b02ae..3f09dee 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -55,8 +55,10 @@
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
 
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, unsigned int hifi);
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt);
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt);
 
 /*
  * wm8753 register cache
@@ -87,6 +89,10 @@
 	enum snd_soc_control_type control_type;
 	unsigned int sysclk;
 	unsigned int pcmclk;
+
+	unsigned int voice_fmt;
+	unsigned int hifi_fmt;
+
 	int dai_func;
 };
 
@@ -170,9 +176,9 @@
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-	int mode = snd_soc_read(codec, WM8753_IOCTL);
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
 
-	ucontrol->value.integer.value[0] = (mode & 0xc) >> 2;
+	ucontrol->value.integer.value[0] = wm8753->dai_func;
 	return 0;
 }
 
@@ -180,16 +186,26 @@
 	struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
-	int mode = snd_soc_read(codec, WM8753_IOCTL);
 	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	u16 ioctl;
 
-	if (((mode & 0xc) >> 2) == ucontrol->value.integer.value[0])
-		return 0;
+	if (codec->active)
+		return -EBUSY;
 
-	mode &= 0xfff3;
-	mode |= (ucontrol->value.integer.value[0] << 2);
+	ioctl = snd_soc_read(codec, WM8753_IOCTL);
 
-	wm8753->dai_func =  ucontrol->value.integer.value[0];
+	wm8753->dai_func = ucontrol->value.integer.value[0];
+
+	if (((ioctl >> 2) & 0x3) == wm8753->dai_func)
+		return 1;
+
+	ioctl = (ioctl & 0x1f3) | (wm8753->dai_func << 2);
+	snd_soc_write(codec, WM8753_IOCTL, ioctl);
+
+
+	wm8753_hifi_write_dai_fmt(codec, wm8753->hifi_fmt);
+	wm8753_voice_write_dai_fmt(codec, wm8753->voice_fmt);
+
 	return 1;
 }
 
@@ -828,10 +844,9 @@
 /*
  * Set's ADC and Voice DAC format.
  */
-static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_vdac_adc_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 voice = snd_soc_read(codec, WM8753_PCM) & 0x01ec;
 
 	/* interface format */
@@ -858,13 +873,6 @@
 	return 0;
 }
 
-static int wm8753_pcm_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	wm8753_set_dai_mode(dai->codec, dai, 0);
-	return 0;
-}
-
 /*
  * Set PCM DAI bit size and sample rate.
  */
@@ -905,10 +913,9 @@
 /*
  * Set's PCM dai fmt and BCLK.
  */
-static int wm8753_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_pcm_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 voice, ioctl;
 
 	voice = snd_soc_read(codec, WM8753_PCM) & 0x011f;
@@ -999,10 +1006,9 @@
 /*
  * Set's HiFi DAC format.
  */
-static int wm8753_hdac_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_hdac_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 hifi = snd_soc_read(codec, WM8753_HIFI) & 0x01e0;
 
 	/* interface format */
@@ -1032,10 +1038,9 @@
 /*
  * Set's I2S DAI format.
  */
-static int wm8753_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_i2s_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 ioctl, hifi;
 
 	hifi = snd_soc_read(codec, WM8753_HIFI) & 0x011f;
@@ -1098,13 +1103,6 @@
 	return 0;
 }
 
-static int wm8753_i2s_startup(struct snd_pcm_substream *substream,
-				struct snd_soc_dai *dai)
-{
-	wm8753_set_dai_mode(dai->codec, dai, 1);
-	return 0;
-}
-
 /*
  * Set PCM DAI bit size and sample rate.
  */
@@ -1147,61 +1145,117 @@
 	return 0;
 }
 
-static int wm8753_mode1v_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1v_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as pcmclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock);
 
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_pcm_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode1h_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode1h_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_hdac_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode2_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode2_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as pcmclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock);
 
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
-static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int wm8753_mode3_4_set_dai_fmt(struct snd_soc_codec *codec,
 		unsigned int fmt)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 clock;
 
 	/* set clk source as mclk */
 	clock = snd_soc_read(codec, WM8753_CLOCK) & 0xfffb;
 	snd_soc_write(codec, WM8753_CLOCK, clock | 0x4);
 
-	if (wm8753_hdac_set_dai_fmt(codec_dai, fmt) < 0)
+	if (wm8753_hdac_set_dai_fmt(codec, fmt) < 0)
 		return -EINVAL;
-	if (wm8753_vdac_adc_set_dai_fmt(codec_dai, fmt) < 0)
-		return -EINVAL;
-	return wm8753_i2s_set_dai_fmt(codec_dai, fmt);
+	return wm8753_vdac_adc_set_dai_fmt(codec, fmt);
 }
 
+static int wm8753_hifi_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt)
+{
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	switch (wm8753->dai_func) {
+	case 0:
+		ret = wm8753_mode1h_set_dai_fmt(codec, fmt);
+		break;
+	case 1:
+		ret = wm8753_mode2_set_dai_fmt(codec, fmt);
+		break;
+	case 2:
+	case 3:
+		ret = wm8753_mode3_4_set_dai_fmt(codec, fmt);
+		break;
+	default:
+		 break;
+	}
+	if (ret)
+		return ret;
+
+	return wm8753_i2s_set_dai_fmt(codec, fmt);
+}
+
+static int wm8753_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+	wm8753->hifi_fmt = fmt;
+
+	return wm8753_hifi_write_dai_fmt(codec, fmt);
+};
+
+static int wm8753_voice_write_dai_fmt(struct snd_soc_codec *codec,
+		unsigned int fmt)
+{
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+	int ret = 0;
+
+	if (wm8753->dai_func != 0)
+		return 0;
+
+	ret = wm8753_mode1v_set_dai_fmt(codec, fmt);
+	if (ret)
+		return ret;
+	ret = wm8753_pcm_set_dai_fmt(codec, fmt);
+	if (ret)
+		return ret;
+
+	return 0;
+};
+
+static int wm8753_voice_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
+
+	wm8753->voice_fmt = fmt;
+
+	return wm8753_voice_write_dai_fmt(codec, fmt);
+};
+
 static int wm8753_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
@@ -1268,57 +1322,25 @@
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
-	.startup = wm8753_i2s_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode = {
 	.hw_params	= wm8753_i2s_hw_params,
 	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode1h_set_dai_fmt,
+	.set_fmt	= wm8753_hifi_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
 	.set_sysclk	= wm8753_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
-	.startup = wm8753_pcm_startup,
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode = {
 	.hw_params	= wm8753_pcm_hw_params,
 	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_fmt	= wm8753_voice_set_dai_fmt,
 	.set_clkdiv	= wm8753_set_dai_clkdiv,
 	.set_pll	= wm8753_set_dai_pll,
 	.set_sysclk	= wm8753_set_dai_sysclk,
 };
 
-static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
-	.startup = wm8753_pcm_startup,
-	.hw_params	= wm8753_pcm_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode2_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
-	.startup = wm8753_i2s_startup,
-	.hw_params	= wm8753_i2s_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
-	.startup = wm8753_i2s_startup,
-	.hw_params	= wm8753_i2s_hw_params,
-	.digital_mute	= wm8753_mute,
-	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
-	.set_clkdiv	= wm8753_set_dai_clkdiv,
-	.set_pll	= wm8753_set_dai_pll,
-	.set_sysclk	= wm8753_set_dai_sysclk,
-};
-
-static struct snd_soc_dai_driver wm8753_all_dai[] = {
+static struct snd_soc_dai_driver wm8753_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "wm8753-hifi",
 	.playback = {
@@ -1326,14 +1348,16 @@
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS},
+		.formats = WM8753_FORMATS
+	},
 	.capture = { /* dummy for fast DAI switching */
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS},
-	.ops = &wm8753_dai_ops_hifi_mode1,
+		.formats = WM8753_FORMATS
+	},
+	.ops = &wm8753_dai_ops_hifi_mode,
 },
 /* DAI Voice mode 1 */
 {	.name = "wm8753-voice",
@@ -1342,97 +1366,19 @@
 		.channels_min = 1,
 		.channels_max = 1,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
+		.formats = WM8753_FORMATS,
+	},
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_voice_mode1,
-},
-/* DAI HiFi mode 2 - dummy */
-{	.name = "wm8753-hifi",
-},
-/* DAI Voice mode 2 */
-{	.name = "wm8753-voice",
-	.playback = {
-		.stream_name = "Voice Playback",
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_voice_mode2,
-},
-/* DAI HiFi mode 3 */
-{	.name = "wm8753-hifi",
-	.playback = {
-		.stream_name = "HiFi Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_hifi_mode3,
-},
-/* DAI Voice mode 3 - dummy */
-{	.name = "wm8753-voice",
-},
-/* DAI HiFi mode 4 */
-{	.name = "wm8753-hifi",
-	.playback = {
-		.stream_name = "HiFi Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = WM8753_RATES,
-		.formats = WM8753_FORMATS,},
-	.ops = &wm8753_dai_ops_hifi_mode4,
-},
-/* DAI Voice mode 4 - dummy */
-{	.name = "wm8753-voice",
+		.formats = WM8753_FORMATS,
+	},
+	.ops = &wm8753_dai_ops_voice_mode,
 },
 };
 
-static struct snd_soc_dai_driver wm8753_dai[] = {
-	{
-		.name = "wm8753-aif0",
-	},
-	{
-		.name = "wm8753-aif1",
-	},
-};
-
-static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
-		struct snd_soc_dai *dai, unsigned int hifi)
-{
-	struct wm8753_priv *wm8753 = snd_soc_codec_get_drvdata(codec);
-
-	if (wm8753->dai_func < 4) {
-		if (hifi)
-			dai->driver = &wm8753_all_dai[wm8753->dai_func << 1];
-		else
-			dai->driver = &wm8753_all_dai[(wm8753->dai_func << 1) + 1];
-	}
-	snd_soc_write(codec, WM8753_IOCTL, wm8753->dai_func);
-}
-
 static void wm8753_work(struct work_struct *work)
 {
 	struct snd_soc_dapm_context *dapm =
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index 3d4c55f..ae1cadf 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -223,11 +223,12 @@
 	int fs;
 	int deemph;
 
+	int dcs_pending;
+	int dcs_cache[4];
+
 	/* Reference count */
 	int class_w_users;
 
-	struct completion wseq;
-
 	struct snd_soc_jack *mic_jack;
 	int mic_det;
 	int mic_short;
@@ -246,6 +247,12 @@
 	case WM8903_REVISION_NUMBER:
 	case WM8903_INTERRUPT_STATUS_1:
 	case WM8903_WRITE_SEQUENCER_4:
+	case WM8903_POWER_MANAGEMENT_3:
+	case WM8903_POWER_MANAGEMENT_2:
+	case WM8903_DC_SERVO_READBACK_1:
+	case WM8903_DC_SERVO_READBACK_2:
+	case WM8903_DC_SERVO_READBACK_3:
+	case WM8903_DC_SERVO_READBACK_4:
 		return 1;
 
 	default:
@@ -253,50 +260,6 @@
 	}
 }
 
-static int wm8903_run_sequence(struct snd_soc_codec *codec, unsigned int start)
-{
-	u16 reg[5];
-	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
-
-	BUG_ON(start > 48);
-
-	/* Enable the sequencer if it's not already on */
-	reg[0] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_0);
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0,
-		      reg[0] | WM8903_WSEQ_ENA);
-
-	dev_dbg(codec->dev, "Starting sequence at %d\n", start);
-
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_3,
-		     start | WM8903_WSEQ_START);
-
-	/* Wait for it to complete.  If we have the interrupt wired up then
-	 * that will break us out of the poll early.
-	 */
-	do {
-		wait_for_completion_timeout(&wm8903->wseq,
-					    msecs_to_jiffies(10));
-
-		reg[4] = snd_soc_read(codec, WM8903_WRITE_SEQUENCER_4);
-	} while (reg[4] & WM8903_WSEQ_BUSY);
-
-	dev_dbg(codec->dev, "Sequence complete\n");
-
-	/* Disable the sequencer again if we enabled it */
-	snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, reg[0]);
-
-	return 0;
-}
-
-static void wm8903_sync_reg_cache(struct snd_soc_codec *codec, u16 *cache)
-{
-	int i;
-
-	/* There really ought to be something better we can do here :/ */
-	for (i = 0; i < ARRAY_SIZE(wm8903_reg_defaults); i++)
-		cache[i] = codec->hw_read(codec, i);
-}
-
 static void wm8903_reset(struct snd_soc_codec *codec)
 {
 	snd_soc_write(codec, WM8903_SW_RESET_AND_ID, 0);
@@ -304,11 +267,6 @@
 	       sizeof(wm8903_reg_defaults));
 }
 
-#define WM8903_OUTPUT_SHORT 0x8
-#define WM8903_OUTPUT_OUT   0x4
-#define WM8903_OUTPUT_INT   0x2
-#define WM8903_OUTPUT_IN    0x1
-
 static int wm8903_cp_event(struct snd_soc_dapm_widget *w,
 			   struct snd_kcontrol *kcontrol, int event)
 {
@@ -318,99 +276,103 @@
 	return 0;
 }
 
-/*
- * Event for headphone and line out amplifier power changes.  Special
- * power up/down sequences are required in order to maximise pop/click
- * performance.
- */
-static int wm8903_output_event(struct snd_soc_dapm_widget *w,
-			       struct snd_kcontrol *kcontrol, int event)
+static int wm8903_dcs_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	u16 val;
-	u16 reg;
-	u16 dcs_reg;
-	u16 dcs_bit;
-	int shift;
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 
-	switch (w->reg) {
-	case WM8903_POWER_MANAGEMENT_2:
-		reg = WM8903_ANALOGUE_HP_0;
-		dcs_bit = 0 + w->shift;
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		wm8903->dcs_pending |= 1 << w->shift;
 		break;
-	case WM8903_POWER_MANAGEMENT_3:
-		reg = WM8903_ANALOGUE_LINEOUT_0;
-		dcs_bit = 2 + w->shift;
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+				    1 << w->shift, 0);
 		break;
-	default:
-		BUG();
-		return -EINVAL;  /* Spurious warning from some compilers */
-	}
-
-	switch (w->shift) {
-	case 0:
-		shift = 0;
-		break;
-	case 1:
-		shift = 4;
-		break;
-	default:
-		BUG();
-		return -EINVAL;  /* Spurious warning from some compilers */
-	}
-
-	if (event & SND_SOC_DAPM_PRE_PMU) {
-		val = snd_soc_read(codec, reg);
-
-		/* Short the output */
-		val &= ~(WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
-	}
-
-	if (event & SND_SOC_DAPM_POST_PMU) {
-		val = snd_soc_read(codec, reg);
-
-		val |= (WM8903_OUTPUT_IN << shift);
-		snd_soc_write(codec, reg, val);
-
-		val |= (WM8903_OUTPUT_INT << shift);
-		snd_soc_write(codec, reg, val);
-
-		/* Turn on the output ENA_OUTP */
-		val |= (WM8903_OUTPUT_OUT << shift);
-		snd_soc_write(codec, reg, val);
-
-		/* Enable the DC servo */
-		dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
-		dcs_reg |= dcs_bit;
-		snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
-
-		/* Remove the short */
-		val |= (WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
-	}
-
-	if (event & SND_SOC_DAPM_PRE_PMD) {
-		val = snd_soc_read(codec, reg);
-
-		/* Short the output */
-		val &= ~(WM8903_OUTPUT_SHORT << shift);
-		snd_soc_write(codec, reg, val);
-
-		/* Disable the DC servo */
-		dcs_reg = snd_soc_read(codec, WM8903_DC_SERVO_0);
-		dcs_reg &= ~dcs_bit;
-		snd_soc_write(codec, WM8903_DC_SERVO_0, dcs_reg);
-
-		/* Then disable the intermediate and output stages */
-		val &= ~((WM8903_OUTPUT_OUT | WM8903_OUTPUT_INT |
-			  WM8903_OUTPUT_IN) << shift);
-		snd_soc_write(codec, reg, val);
 	}
 
 	return 0;
 }
 
+#define WM8903_DCS_MODE_WRITE_STOP 0
+#define WM8903_DCS_MODE_START_STOP 2
+
+static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
+				enum snd_soc_dapm_type event, int subseq)
+{
+	struct snd_soc_codec *codec = container_of(dapm,
+						   struct snd_soc_codec, dapm);
+	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
+	int dcs_mode = WM8903_DCS_MODE_WRITE_STOP;
+	int i, val;
+
+	/* Complete any pending DC servo starts */
+	if (wm8903->dcs_pending) {
+		dev_dbg(codec->dev, "Starting DC servo for %x\n",
+			wm8903->dcs_pending);
+
+		/* If we've no cached values then we need to do startup */
+		for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+			if (!(wm8903->dcs_pending & (1 << i)))
+				continue;
+
+			if (wm8903->dcs_cache[i]) {
+				dev_dbg(codec->dev,
+					"Restore DC servo %d value %x\n",
+					3 - i, wm8903->dcs_cache[i]);
+
+				snd_soc_write(codec, WM8903_DC_SERVO_4 + i,
+					      wm8903->dcs_cache[i] & 0xff);
+			} else {
+				dev_dbg(codec->dev,
+					"Calibrate DC servo %d\n", 3 - i);
+				dcs_mode = WM8903_DCS_MODE_START_STOP;
+			}
+		}
+
+		/* Don't trust the cache for analogue */
+		if (wm8903->class_w_users)
+			dcs_mode = WM8903_DCS_MODE_START_STOP;
+
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_2,
+				    WM8903_DCS_MODE_MASK, dcs_mode);
+
+		snd_soc_update_bits(codec, WM8903_DC_SERVO_0,
+				    WM8903_DCS_ENA_MASK, wm8903->dcs_pending);
+
+		switch (dcs_mode) {
+		case WM8903_DCS_MODE_WRITE_STOP:
+			break;
+
+		case WM8903_DCS_MODE_START_STOP:
+			msleep(270);
+
+			/* Cache the measured offsets for digital */
+			if (wm8903->class_w_users)
+				break;
+
+			for (i = 0; i < ARRAY_SIZE(wm8903->dcs_cache); i++) {
+				if (!(wm8903->dcs_pending & (1 << i)))
+					continue;
+
+				val = snd_soc_read(codec,
+						   WM8903_DC_SERVO_READBACK_1 + i);
+				dev_dbg(codec->dev, "DC servo %d: %x\n",
+					3 - i, val);
+				wm8903->dcs_cache[i] = val;
+			}
+			break;
+
+		default:
+			pr_warn("DCS mode %d delay not set\n", dcs_mode);
+			break;
+		}
+
+		wm8903->dcs_pending = 0;
+	}
+}
+
 /*
  * When used with DAC outputs only the WM8903 charge pump supports
  * operation in class W mode, providing very low power consumption
@@ -674,6 +636,22 @@
 static const struct soc_enum rsidetone_enum =
 	SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text);
 
+static const char *aif_text[] = {
+	"Left", "Right"
+};
+
+static const struct soc_enum lcapture_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text);
+
+static const struct soc_enum rcapture_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text);
+
+static const struct soc_enum lplay_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text);
+
+static const struct soc_enum rplay_enum =
+	SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text);
+
 static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
@@ -791,6 +769,18 @@
 static const struct snd_kcontrol_new rsidetone_mux =
 	SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum);
 
+static const struct snd_kcontrol_new lcapture_mux =
+	SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum);
+
+static const struct snd_kcontrol_new rcapture_mux =
+	SOC_DAPM_ENUM("Right Capture Mux", rcapture_enum);
+
+static const struct snd_kcontrol_new lplay_mux =
+	SOC_DAPM_ENUM("Left Playback Mux", lplay_enum);
+
+static const struct snd_kcontrol_new rplay_mux =
+	SOC_DAPM_ENUM("Right Playback Mux", rplay_enum);
+
 static const struct snd_kcontrol_new left_output_mixer[] = {
 SOC_DAPM_SINGLE("DACL Switch", WM8903_ANALOGUE_LEFT_MIX_0, 3, 1, 0),
 SOC_DAPM_SINGLE("DACR Switch", WM8903_ANALOGUE_LEFT_MIX_0, 2, 1, 0),
@@ -854,14 +844,26 @@
 SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0),
 
-SND_SOC_DAPM_ADC("ADCL", "Left HiFi Capture", WM8903_POWER_MANAGEMENT_6, 1, 0),
-SND_SOC_DAPM_ADC("ADCR", "Right HiFi Capture", WM8903_POWER_MANAGEMENT_6, 0, 0),
+SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0),
+SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Capture Mux", SND_SOC_NOPM, 0, 0, &lcapture_mux),
+SND_SOC_DAPM_MUX("Right Capture Mux", SND_SOC_NOPM, 0, 0, &rcapture_mux),
+
+SND_SOC_DAPM_AIF_OUT("AIFTXL", "Left HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIFTXR", "Right HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &lsidetone_mux),
 SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &rsidetone_mux),
 
-SND_SOC_DAPM_DAC("DACL", "Left Playback", WM8903_POWER_MANAGEMENT_6, 3, 0),
-SND_SOC_DAPM_DAC("DACR", "Right Playback", WM8903_POWER_MANAGEMENT_6, 2, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXL", "Left Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_IN("AIFRXR", "Right Playback", 0, SND_SOC_NOPM, 0, 0),
+
+SND_SOC_DAPM_MUX("Left Playback Mux", SND_SOC_NOPM, 0, 0, &lplay_mux),
+SND_SOC_DAPM_MUX("Right Playback Mux", SND_SOC_NOPM, 0, 0, &rplay_mux),
+
+SND_SOC_DAPM_DAC("DACL", NULL, WM8903_POWER_MANAGEMENT_6, 3, 0),
+SND_SOC_DAPM_DAC("DACR", NULL, WM8903_POWER_MANAGEMENT_6, 2, 0),
 
 SND_SOC_DAPM_MIXER("Left Output Mixer", WM8903_POWER_MANAGEMENT_1, 1, 0,
 		   left_output_mixer, ARRAY_SIZE(left_output_mixer)),
@@ -873,23 +875,45 @@
 SND_SOC_DAPM_MIXER("Right Speaker Mixer", WM8903_POWER_MANAGEMENT_4, 0, 0,
 		   right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)),
 
-SND_SOC_DAPM_PGA_E("Left Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
-		   1, 0, NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Headphone Output PGA", WM8903_POWER_MANAGEMENT_2,
-		   0, 0, NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("Left Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+		   4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Headphone Output PGA", 0, WM8903_ANALOGUE_HP_0,
+		   0, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_E("Left Line Output PGA", WM8903_POWER_MANAGEMENT_3, 1, 0,
-		   NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
-SND_SOC_DAPM_PGA_E("Right Line Output PGA", WM8903_POWER_MANAGEMENT_3, 0, 0,
-		   NULL, 0, wm8903_output_event,
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
-		   SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("Left Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 4, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("Right Line Output PGA", 0, WM8903_ANALOGUE_LINEOUT_0, 0, 0,
+		   NULL, 0),
+
+SND_SOC_DAPM_PGA_S("HPL_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 7, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 6, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 5, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_RMV_SHORT", 4, WM8903_ANALOGUE_HP_0, 3, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_OUTP", 3, WM8903_ANALOGUE_HP_0, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPR_ENA_DLY", 1, WM8903_ANALOGUE_HP_0, 1, 0, NULL, 0),
+
+SND_SOC_DAPM_PGA_S("LINEOUTL_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 7, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 6, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTL_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 5, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_RMV_SHORT", 4, WM8903_ANALOGUE_LINEOUT_0, 3, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_OUTP", 3, WM8903_ANALOGUE_LINEOUT_0, 2, 0,
+		   NULL, 0),
+SND_SOC_DAPM_PGA_S("LINEOUTR_ENA_DLY", 1, WM8903_ANALOGUE_LINEOUT_0, 1, 0,
+		   NULL, 0),
+
+SND_SOC_DAPM_SUPPLY("DCS Master", WM8903_DC_SERVO_0, 4, 0, NULL, 0),
+SND_SOC_DAPM_PGA_S("HPL_DCS", 3, SND_SOC_NOPM, 3, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("HPR_DCS", 3, SND_SOC_NOPM, 2, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTL_DCS", 3, SND_SOC_NOPM, 1, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA_S("LINEOUTR_DCS", 3, SND_SOC_NOPM, 0, 0, wm8903_dcs_event,
+		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
 SND_SOC_DAPM_PGA("Left Speaker PGA", WM8903_POWER_MANAGEMENT_5, 1, 0,
 		 NULL, 0),
@@ -899,10 +923,18 @@
 SND_SOC_DAPM_SUPPLY("Charge Pump", WM8903_CHARGE_PUMP_0, 0, 0,
 		    wm8903_cp_event, SND_SOC_DAPM_POST_PMU),
 SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8903_CLOCK_RATES_2, 1, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("CLK_SYS", WM8903_CLOCK_RATES_2, 2, 0, NULL, 0),
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
 
+	{ "CLK_DSP", NULL, "CLK_SYS" },
+	{ "Mic Bias", NULL, "CLK_SYS" },
+	{ "HPL_DCS", NULL, "CLK_SYS" },
+	{ "HPR_DCS", NULL, "CLK_SYS" },
+	{ "LINEOUTL_DCS", NULL, "CLK_SYS" },
+	{ "LINEOUTR_DCS", NULL, "CLK_SYS" },
+
 	{ "Left Input Mux", "IN1L", "IN1L" },
 	{ "Left Input Mux", "IN2L", "IN2L" },
 	{ "Left Input Mux", "IN3L", "IN3L" },
@@ -943,18 +975,36 @@
 	{ "Left Input PGA", NULL, "Left Input Mode Mux" },
 	{ "Right Input PGA", NULL, "Right Input Mode Mux" },
 
+	{ "Left Capture Mux", "Left", "ADCL" },
+	{ "Left Capture Mux", "Right", "ADCR" },
+
+	{ "Right Capture Mux", "Left", "ADCL" },
+	{ "Right Capture Mux", "Right", "ADCR" },
+
+	{ "AIFTXL", NULL, "Left Capture Mux" },
+	{ "AIFTXR", NULL, "Right Capture Mux" },
+
 	{ "ADCL", NULL, "Left Input PGA" },
 	{ "ADCL", NULL, "CLK_DSP" },
 	{ "ADCR", NULL, "Right Input PGA" },
 	{ "ADCR", NULL, "CLK_DSP" },
 
+	{ "Left Playback Mux", "Left", "AIFRXL" },
+	{ "Left Playback Mux", "Right", "AIFRXR" },
+
+	{ "Right Playback Mux", "Left", "AIFRXL" },
+	{ "Right Playback Mux", "Right", "AIFRXR" },
+
 	{ "DACL Sidetone", "Left", "ADCL" },
 	{ "DACL Sidetone", "Right", "ADCR" },
 	{ "DACR Sidetone", "Left", "ADCL" },
 	{ "DACR Sidetone", "Right", "ADCR" },
 
+	{ "DACL", NULL, "Left Playback Mux" },
 	{ "DACL", NULL, "DACL Sidetone" },
 	{ "DACL", NULL, "CLK_DSP" },
+
+	{ "DACR", NULL, "Right Playback Mux" },
 	{ "DACR", NULL, "DACR Sidetone" },
 	{ "DACR", NULL, "CLK_DSP" },
 
@@ -987,11 +1037,35 @@
 	{ "Left Speaker PGA", NULL, "Left Speaker Mixer" },
 	{ "Right Speaker PGA", NULL, "Right Speaker Mixer" },
 
-	{ "HPOUTL", NULL, "Left Headphone Output PGA" },
-	{ "HPOUTR", NULL, "Right Headphone Output PGA" },
+	{ "HPL_ENA_DLY", NULL, "Left Headphone Output PGA" },
+	{ "HPR_ENA_DLY", NULL, "Right Headphone Output PGA" },
+	{ "LINEOUTL_ENA_DLY", NULL, "Left Line Output PGA" },
+	{ "LINEOUTR_ENA_DLY", NULL, "Right Line Output PGA" },
 
-	{ "LINEOUTL", NULL, "Left Line Output PGA" },
-	{ "LINEOUTR", NULL, "Right Line Output PGA" },
+	{ "HPL_DCS", NULL, "DCS Master" },
+	{ "HPR_DCS", NULL, "DCS Master" },
+	{ "LINEOUTL_DCS", NULL, "DCS Master" },
+	{ "LINEOUTR_DCS", NULL, "DCS Master" },
+
+	{ "HPL_DCS", NULL, "HPL_ENA_DLY" },
+	{ "HPR_DCS", NULL, "HPR_ENA_DLY" },
+	{ "LINEOUTL_DCS", NULL, "LINEOUTL_ENA_DLY" },
+	{ "LINEOUTR_DCS", NULL, "LINEOUTR_ENA_DLY" },
+
+	{ "HPL_ENA_OUTP", NULL, "HPL_DCS" },
+	{ "HPR_ENA_OUTP", NULL, "HPR_DCS" },
+	{ "LINEOUTL_ENA_OUTP", NULL, "LINEOUTL_DCS" },
+	{ "LINEOUTR_ENA_OUTP", NULL, "LINEOUTR_DCS" },
+
+	{ "HPL_RMV_SHORT", NULL, "HPL_ENA_OUTP" },
+	{ "HPR_RMV_SHORT", NULL, "HPR_ENA_OUTP" },
+	{ "LINEOUTL_RMV_SHORT", NULL, "LINEOUTL_ENA_OUTP" },
+	{ "LINEOUTR_RMV_SHORT", NULL, "LINEOUTR_ENA_OUTP" },
+
+	{ "HPOUTL", NULL, "HPL_RMV_SHORT" },
+	{ "HPOUTR", NULL, "HPR_RMV_SHORT" },
+	{ "LINEOUTL", NULL, "LINEOUTL_RMV_SHORT" },
+	{ "LINEOUTR", NULL, "LINEOUTR_RMV_SHORT" },
 
 	{ "LOP", NULL, "Left Speaker PGA" },
 	{ "LON", NULL, "Left Speaker PGA" },
@@ -1019,29 +1093,71 @@
 static int wm8903_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
+		break;
+
 	case SND_SOC_BIAS_PREPARE:
-		reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
-		reg &= ~(WM8903_VMID_RES_MASK);
-		reg |= WM8903_VMID_RES_50K;
-		snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_RES_MASK,
+				    WM8903_VMID_RES_50K);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
-			snd_soc_write(codec, WM8903_CLOCK_RATES_2,
-				     WM8903_CLK_SYS_ENA);
+			snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+					    WM8903_POBCTRL | WM8903_ISEL_MASK |
+					    WM8903_STARTUP_BIAS_ENA |
+					    WM8903_BIAS_ENA,
+					    WM8903_POBCTRL |
+					    (2 << WM8903_ISEL_SHIFT) |
+					    WM8903_STARTUP_BIAS_ENA);
 
-			/* Change DC servo dither level in startup sequence */
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_0, 0x11);
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_1, 0x1257);
-			snd_soc_write(codec, WM8903_WRITE_SEQUENCER_2, 0x2);
+			snd_soc_update_bits(codec,
+					    WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+					    WM8903_SPK_DISCHARGE,
+					    WM8903_SPK_DISCHARGE);
 
-			wm8903_run_sequence(codec, 0);
-			wm8903_sync_reg_cache(codec, codec->reg_cache);
+			msleep(33);
+
+			snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA);
+
+			snd_soc_update_bits(codec,
+					    WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0,
+					    WM8903_SPK_DISCHARGE, 0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_TIE_ENA |
+					    WM8903_BUFIO_ENA |
+					    WM8903_VMID_IO_ENA |
+					    WM8903_VMID_SOFT_MASK |
+					    WM8903_VMID_RES_MASK |
+					    WM8903_VMID_BUF_ENA,
+					    WM8903_VMID_TIE_ENA |
+					    WM8903_BUFIO_ENA |
+					    WM8903_VMID_IO_ENA |
+					    (2 << WM8903_VMID_SOFT_SHIFT) |
+					    WM8903_VMID_RES_250K |
+					    WM8903_VMID_BUF_ENA);
+
+			msleep(129);
+
+			snd_soc_update_bits(codec, WM8903_POWER_MANAGEMENT_5,
+					    WM8903_SPKL_ENA | WM8903_SPKR_ENA,
+					    0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_SOFT_MASK, 0);
+
+			snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+					    WM8903_VMID_RES_MASK,
+					    WM8903_VMID_RES_50K);
+
+			snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+					    WM8903_BIAS_ENA | WM8903_POBCTRL,
+					    WM8903_BIAS_ENA);
 
 			/* By default no bypass paths are enabled so
 			 * enable Class W support.
@@ -1054,17 +1170,32 @@
 					    WM8903_CP_DYN_V);
 		}
 
-		reg = snd_soc_read(codec, WM8903_VMID_CONTROL_0);
-		reg &= ~(WM8903_VMID_RES_MASK);
-		reg |= WM8903_VMID_RES_250K;
-		snd_soc_write(codec, WM8903_VMID_CONTROL_0, reg);
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_RES_MASK,
+				    WM8903_VMID_RES_250K);
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		wm8903_run_sequence(codec, 32);
-		reg = snd_soc_read(codec, WM8903_CLOCK_RATES_2);
-		reg &= ~WM8903_CLK_SYS_ENA;
-		snd_soc_write(codec, WM8903_CLOCK_RATES_2, reg);
+		snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+				    WM8903_BIAS_ENA, 0);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_SOFT_MASK,
+				    2 << WM8903_VMID_SOFT_SHIFT);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_BUF_ENA, 0);
+
+		msleep(290);
+
+		snd_soc_update_bits(codec, WM8903_VMID_CONTROL_0,
+				    WM8903_VMID_TIE_ENA | WM8903_BUFIO_ENA |
+				    WM8903_VMID_IO_ENA | WM8903_VMID_RES_MASK |
+				    WM8903_VMID_SOFT_MASK |
+				    WM8903_VMID_BUF_ENA, 0);
+
+		snd_soc_update_bits(codec, WM8903_BIAS_CONTROL_0,
+				    WM8903_STARTUP_BIAS_ENA, 0);
 		break;
 	}
 
@@ -1489,7 +1620,7 @@
 			    WM8903_MICDET_EINT | WM8903_MICSHRT_EINT,
 			    irq_mask);
 
-	if (det && shrt) {
+	if (det || shrt) {
 		/* Enable mic detection, this may not have been set through
 		 * platform data (eg, if the defaults are OK). */
 		snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0,
@@ -1517,8 +1648,7 @@
 	int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask;
 
 	if (int_val & WM8903_WSEQ_BUSY_EINT) {
-		dev_dbg(codec->dev, "Write sequencer done\n");
-		complete(&wm8903->wseq);
+		dev_warn(codec->dev, "Write sequencer done\n");
 	}
 
 	/*
@@ -1765,7 +1895,6 @@
 	u16 val;
 
 	wm8903->codec = codec;
-	init_completion(&wm8903->wseq);
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
 	if (ret != 0) {
@@ -1781,19 +1910,33 @@
 	}
 
 	val = snd_soc_read(codec, WM8903_REVISION_NUMBER);
-	dev_info(codec->dev, "WM8903 revision %d\n",
-		 val & WM8903_CHIP_REV_MASK);
+	dev_info(codec->dev, "WM8903 revision %c\n",
+		 (val & WM8903_CHIP_REV_MASK) + 'A');
 
 	wm8903_reset(codec);
 
 	/* Set up GPIOs and microphone detection */
 	if (pdata) {
+		bool mic_gpio = false;
+
 		for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) {
 			if (pdata->gpio_cfg[i] == WM8903_GPIO_NO_CONFIG)
 				continue;
 
 			snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i,
 				      pdata->gpio_cfg[i] & 0xffff);
+
+			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;
+			}
 		}
 
 		snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0,
@@ -1804,6 +1947,14 @@
 			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;
 	}
 	
@@ -1863,9 +2014,9 @@
 	snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val);
 
 	/* Enable DAC soft mute by default */
-	val = snd_soc_read(codec, WM8903_DAC_DIGITAL_1);
-	val |= WM8903_DAC_MUTEMODE;
-	snd_soc_write(codec, WM8903_DAC_DIGITAL_1, val);
+	snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1,
+			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE,
+			    WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE);
 
 	snd_soc_add_controls(codec, wm8903_snd_controls,
 				ARRAY_SIZE(wm8903_snd_controls));
@@ -1894,6 +2045,7 @@
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8903_reg_defaults,
 	.volatile_register = wm8903_volatile_register,
+	.seq_notifier = wm8903_seq_notifier,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1932,7 +2084,7 @@
 
 static struct i2c_driver wm8903_i2c_driver = {
 	.driver = {
-		.name = "wm8903-codec",
+		.name = "wm8903",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8903_i2c_probe,
diff --git a/sound/soc/codecs/wm8903.h b/sound/soc/codecs/wm8903.h
index e8490f3..db94931 100644
--- a/sound/soc/codecs/wm8903.h
+++ b/sound/soc/codecs/wm8903.h
@@ -75,6 +75,14 @@
 #define WM8903_ANALOGUE_SPK_OUTPUT_CONTROL_0    0x41
 #define WM8903_DC_SERVO_0                       0x43
 #define WM8903_DC_SERVO_2                       0x45
+#define WM8903_DC_SERVO_4			0x47
+#define WM8903_DC_SERVO_5			0x48
+#define WM8903_DC_SERVO_6			0x49
+#define WM8903_DC_SERVO_7			0x4A
+#define WM8903_DC_SERVO_READBACK_1		0x51
+#define WM8903_DC_SERVO_READBACK_2		0x52
+#define WM8903_DC_SERVO_READBACK_3		0x53
+#define WM8903_DC_SERVO_READBACK_4		0x54
 #define WM8903_ANALOGUE_HP_0                    0x5A
 #define WM8903_ANALOGUE_LINEOUT_0               0x5E
 #define WM8903_CHARGE_PUMP_0                    0x62
@@ -165,7 +173,7 @@
 
 #define WM8903_VMID_RES_50K                          2
 #define WM8903_VMID_RES_250K                         3
-#define WM8903_VMID_RES_5K                           4
+#define WM8903_VMID_RES_5K                           6
 
 /*
  * R8 (0x08) - Analogue DAC 0
diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c
index 30fb48e..85e3e63 100644
--- a/sound/soc/codecs/wm8978.c
+++ b/sound/soc/codecs/wm8978.c
@@ -93,6 +93,7 @@
 static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1200, 75, 0);
 static const DECLARE_TLV_DB_SCALE(spk_tlv, -5700, 100, 0);
 static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1);
+static const DECLARE_TLV_DB_SCALE(limiter_tlv, 0, 100, 0);
 
 static const struct snd_kcontrol_new wm8978_snd_controls[] = {
 
@@ -144,19 +145,19 @@
 
 	SOC_SINGLE("DAC Playback Limiter Threshold",
 		WM8978_DAC_LIMITER_2, 4, 7, 0),
-	SOC_SINGLE("DAC Playback Limiter Boost",
-		WM8978_DAC_LIMITER_2, 0, 15, 0),
+	SOC_SINGLE_TLV("DAC Playback Limiter Volume",
+		WM8978_DAC_LIMITER_2, 0, 12, 0, limiter_tlv),
 
 	SOC_ENUM("ALC Enable Switch", alc1),
 	SOC_SINGLE("ALC Capture Min Gain", WM8978_ALC_CONTROL_1, 0, 7, 0),
 	SOC_SINGLE("ALC Capture Max Gain", WM8978_ALC_CONTROL_1, 3, 7, 0),
 
-	SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 7, 0),
+	SOC_SINGLE("ALC Capture Hold", WM8978_ALC_CONTROL_2, 4, 10, 0),
 	SOC_SINGLE("ALC Capture Target", WM8978_ALC_CONTROL_2, 0, 15, 0),
 
 	SOC_ENUM("ALC Capture Mode", alc3),
-	SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 15, 0),
-	SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 15, 0),
+	SOC_SINGLE("ALC Capture Decay", WM8978_ALC_CONTROL_3, 4, 10, 0),
+	SOC_SINGLE("ALC Capture Attack", WM8978_ALC_CONTROL_3, 0, 10, 0),
 
 	SOC_SINGLE("ALC Capture Noise Gate Switch", WM8978_NOISE_GATE, 3, 1, 0),
 	SOC_SINGLE("ALC Capture Noise Gate Threshold",
@@ -211,8 +212,10 @@
 		WM8978_LOUT2_SPK_CONTROL, WM8978_ROUT2_SPK_CONTROL, 6, 1, 1),
 
 	/* DAC / ADC oversampling */
-	SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL, 8, 1, 0),
-	SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL, 8, 1, 0),
+	SOC_SINGLE("DAC 128x Oversampling Switch", WM8978_DAC_CONTROL,
+		   5, 1, 0),
+	SOC_SINGLE("ADC 128x Oversampling Switch", WM8978_ADC_CONTROL,
+		   5, 1, 0),
 };
 
 /* Mixer #1: Output (OUT1, OUT2) Mixer: mix AUX, Input mixer output and DAC */
diff --git a/sound/soc/codecs/wm8994-tables.c b/sound/soc/codecs/wm8994-tables.c
index 68e9b02..a87adbd 100644
--- a/sound/soc/codecs/wm8994-tables.c
+++ b/sound/soc/codecs/wm8994-tables.c
@@ -62,8 +62,8 @@
 	{ 0x00FF, 0x00FF }, /* R58    - MICBIAS */
 	{ 0x000F, 0x000F }, /* R59    - LDO 1 */
 	{ 0x0007, 0x0007 }, /* R60    - LDO 2 */
-	{ 0x0000, 0x0000 }, /* R61 */
-	{ 0x0000, 0x0000 }, /* R62 */
+	{ 0xFFFF, 0xFFFF }, /* R61 */
+	{ 0xFFFF, 0xFFFF }, /* R62 */
 	{ 0x0000, 0x0000 }, /* R63 */
 	{ 0x0000, 0x0000 }, /* R64 */
 	{ 0x0000, 0x0000 }, /* R65 */
@@ -209,9 +209,9 @@
 	{ 0x0000, 0x0000 }, /* R205 */
 	{ 0x0000, 0x0000 }, /* R206 */
 	{ 0x0000, 0x0000 }, /* R207 */
-	{ 0x0000, 0x0000 }, /* R208 */
-	{ 0x0000, 0x0000 }, /* R209 */
-	{ 0x0000, 0x0000 }, /* R210 */
+	{ 0xFFFF, 0xFFFF }, /* R208 */
+	{ 0xFFFF, 0xFFFF }, /* R209 */
+	{ 0xFFFF, 0xFFFF }, /* R210 */
 	{ 0x0000, 0x0000 }, /* R211 */
 	{ 0x0000, 0x0000 }, /* R212 */
 	{ 0x0000, 0x0000 }, /* R213 */
@@ -1573,7 +1573,7 @@
 	{ 0x03C3, 0x03C3 }, /* R1569  - Sidetone */
 };
 
-const __devinitdata u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
+const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE] = {
 	0x8994,     /* R0     - Software Reset */
 	0x0000,     /* R1     - Power Management (1) */
 	0x6000,     /* R2     - Power Management (2) */
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index 0ca81d3..3dc64c8 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -102,11 +102,16 @@
 
 	wm8958_micdet_cb jack_cb;
 	void *jack_cb_data;
-	bool jack_is_mic;
-	bool jack_is_video;
+	int micdet_irq;
 
 	int revision;
 	struct wm8994_pdata *pdata;
+
+	unsigned int aif1clk_enable:1;
+	unsigned int aif2clk_enable:1;
+
+	unsigned int aif1clk_disable:1;
+	unsigned int aif2clk_disable:1;
 };
 
 static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg)
@@ -523,7 +528,7 @@
 					 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct wm8994_priv *wm8994 =snd_soc_codec_get_drvdata(codec);
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int block = wm8994_get_retune_mobile_block(kcontrol->id.name);
 
 	ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block];
@@ -1004,6 +1009,117 @@
 	}
 }
 
+static int late_enable_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (wm8994->aif1clk_enable) {
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK,
+					    WM8994_AIF1CLK_ENA);
+			wm8994->aif1clk_enable = 0;
+		}
+		if (wm8994->aif2clk_enable) {
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK,
+					    WM8994_AIF2CLK_ENA);
+			wm8994->aif2clk_enable = 0;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static int late_disable_ev(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMD:
+		if (wm8994->aif1clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1,
+					    WM8994_AIF1CLK_ENA_MASK, 0);
+			wm8994->aif1clk_disable = 0;
+		}
+		if (wm8994->aif2clk_disable) {
+			snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1,
+					    WM8994_AIF2CLK_ENA_MASK, 0);
+			wm8994->aif2clk_disable = 0;
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static int aif1clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		wm8994->aif1clk_enable = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		wm8994->aif1clk_disable = 1;
+		break;
+	}
+
+	return 0;
+}
+
+static int aif2clk_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		wm8994->aif2clk_enable = 1;
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		wm8994->aif2clk_disable = 1;
+		break;
+	}
+
+	return 0;
+}
+
+static int adc_mux_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	late_enable_ev(w, kcontrol, event);
+	return 0;
+}
+
+static int micbias_ev(struct snd_soc_dapm_widget *w,
+		      struct snd_kcontrol *kcontrol, int event)
+{
+	late_enable_ev(w, kcontrol, event);
+	return 0;
+}
+
+static int dac_ev(struct snd_soc_dapm_widget *w,
+		  struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int mask = 1 << w->shift;
+
+	snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+			    mask, mask);
+	return 0;
+}
+
 static const char *hp_mux_text[] = {
 	"Mixer",
 	"DAC",
@@ -1272,11 +1388,68 @@
 static const struct snd_kcontrol_new aif2dacr_src_mux =
 	SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum);
 
+static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev,
+	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev,
+	SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC1R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0,
+	late_enable_ev, SND_SOC_DAPM_PRE_PMU),
+
+SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev)
+};
+
+static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = {
+SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
+SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0)
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = {
+SND_SOC_DAPM_DAC_E("DAC2L", NULL, SND_SOC_NOPM, 3, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC2R", NULL, SND_SOC_NOPM, 2, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1L", NULL, SND_SOC_NOPM, 1, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_DAC_E("DAC1R", NULL, SND_SOC_NOPM, 0, 0,
+	dac_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_dac_widgets[] = {
+SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
+SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
+SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
+SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_revd_widgets[] = {
+SND_SOC_DAPM_MUX_E("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux,
+		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+SND_SOC_DAPM_MUX_E("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux,
+		   adc_mux_ev, SND_SOC_DAPM_PRE_PMU),
+};
+
+static const struct snd_soc_dapm_widget wm8994_adc_widgets[] = {
+SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
+SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
+};
+
 static const struct snd_soc_dapm_widget wm8994_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("DMIC1DAT"),
 SND_SOC_DAPM_INPUT("DMIC2DAT"),
 SND_SOC_DAPM_INPUT("Clock"),
 
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0),
+SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev,
+		      SND_SOC_DAPM_PRE_PMU),
+
 SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event,
 		    SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
 
@@ -1284,12 +1457,9 @@
 SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0),
 SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0),
 
-SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0),
-SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0),
-
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 9, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 8, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev,
@@ -1298,9 +1468,9 @@
 		      WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev,
 		      SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
 
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 11, 0),
-SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", "AIF1 Capture",
+SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL,
 		     0, WM8994_POWER_MANAGEMENT_4, 10, 0),
 SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0,
 		      WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev,
@@ -1345,6 +1515,7 @@
 
 SND_SOC_DAPM_AIF_IN("AIF1DACDAT", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_IN("AIF2DACDAT", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF1ADCDAT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_AIF_OUT("AIF2ADCDAT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_MUX("AIF1DAC Mux", SND_SOC_NOPM, 0, 0, &aif1dac_mux),
@@ -1368,14 +1539,6 @@
 SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0),
 SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0),
 
-SND_SOC_DAPM_MUX("ADCL Mux", WM8994_POWER_MANAGEMENT_4, 1, 0, &adcl_mux),
-SND_SOC_DAPM_MUX("ADCR Mux", WM8994_POWER_MANAGEMENT_4, 0, 0, &adcr_mux),
-
-SND_SOC_DAPM_DAC("DAC2L", NULL, WM8994_POWER_MANAGEMENT_5, 3, 0),
-SND_SOC_DAPM_DAC("DAC2R", NULL, WM8994_POWER_MANAGEMENT_5, 2, 0),
-SND_SOC_DAPM_DAC("DAC1L", NULL, WM8994_POWER_MANAGEMENT_5, 1, 0),
-SND_SOC_DAPM_DAC("DAC1R", NULL, WM8994_POWER_MANAGEMENT_5, 0, 0),
-
 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux),
 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux),
 
@@ -1515,14 +1678,12 @@
 	{ "AIF2ADC Mux", "AIF3DACDAT", "AIF3ADCDAT" },
 
 	/* DAC1 inputs */
-	{ "DAC1L", NULL, "DAC1L Mixer" },
 	{ "DAC1L Mixer", "AIF2 Switch", "AIF2DACL" },
 	{ "DAC1L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
 	{ "DAC1L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
 	{ "DAC1L Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "DAC1L Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
-	{ "DAC1R", NULL, "DAC1R Mixer" },
 	{ "DAC1R Mixer", "AIF2 Switch", "AIF2DACR" },
 	{ "DAC1R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
 	{ "DAC1R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
@@ -1531,7 +1692,6 @@
 
 	/* DAC2/AIF2 outputs  */
 	{ "AIF2ADCL", NULL, "AIF2DAC2L Mixer" },
-	{ "DAC2L", NULL, "AIF2DAC2L Mixer" },
 	{ "AIF2DAC2L Mixer", "AIF2 Switch", "AIF2DACL" },
 	{ "AIF2DAC2L Mixer", "AIF1.2 Switch", "AIF1DAC2L" },
 	{ "AIF2DAC2L Mixer", "AIF1.1 Switch", "AIF1DAC1L" },
@@ -1539,13 +1699,17 @@
 	{ "AIF2DAC2L Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
 	{ "AIF2ADCR", NULL, "AIF2DAC2R Mixer" },
-	{ "DAC2R", NULL, "AIF2DAC2R Mixer" },
 	{ "AIF2DAC2R Mixer", "AIF2 Switch", "AIF2DACR" },
 	{ "AIF2DAC2R Mixer", "AIF1.2 Switch", "AIF1DAC2R" },
 	{ "AIF2DAC2R Mixer", "AIF1.1 Switch", "AIF1DAC1R" },
 	{ "AIF2DAC2R Mixer", "Left Sidetone Switch", "Left Sidetone" },
 	{ "AIF2DAC2R Mixer", "Right Sidetone Switch", "Right Sidetone" },
 
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC1R" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2L" },
+	{ "AIF1ADCDAT", NULL, "AIF1ADC2R" },
+
 	{ "AIF2ADCDAT", NULL, "AIF2ADC Mux" },
 
 	/* AIF3 output */
@@ -1578,6 +1742,33 @@
 	{ "Right Headphone Mux", "DAC", "DAC1R" },
 };
 
+static const struct snd_soc_dapm_route wm8994_lateclk_revd_intercon[] = {
+	{ "DAC1L", NULL, "Late DAC1L Enable PGA" },
+	{ "Late DAC1L Enable PGA", NULL, "DAC1L Mixer" },
+	{ "DAC1R", NULL, "Late DAC1R Enable PGA" },
+	{ "Late DAC1R Enable PGA", NULL, "DAC1R Mixer" },
+	{ "DAC2L", NULL, "Late DAC2L Enable PGA" },
+	{ "Late DAC2L Enable PGA", NULL, "AIF2DAC2L Mixer" },
+	{ "DAC2R", NULL, "Late DAC2R Enable PGA" },
+	{ "Late DAC2R Enable PGA", NULL, "AIF2DAC2R Mixer" }
+};
+
+static const struct snd_soc_dapm_route wm8994_lateclk_intercon[] = {
+	{ "DAC1L", NULL, "DAC1L Mixer" },
+	{ "DAC1R", NULL, "DAC1R Mixer" },
+	{ "DAC2L", NULL, "AIF2DAC2L Mixer" },
+	{ "DAC2R", NULL, "AIF2DAC2R Mixer" },
+};
+
+static const struct snd_soc_dapm_route wm8994_revd_intercon[] = {
+	{ "AIF1DACDAT", NULL, "AIF2DACDAT" },
+	{ "AIF2DACDAT", NULL, "AIF1DACDAT" },
+	{ "AIF1ADCDAT", NULL, "AIF2ADCDAT" },
+	{ "AIF2ADCDAT", NULL, "AIF1ADCDAT" },
+	{ "MICBIAS", NULL, "CLK_SYS" },
+	{ "MICBIAS", NULL, "MICBIAS Supply" },
+};
+
 static const struct snd_soc_dapm_route wm8994_intercon[] = {
 	{ "AIF2DACL", NULL, "AIF2DAC Mux" },
 	{ "AIF2DACR", NULL, "AIF2DAC Mux" },
@@ -2501,6 +2692,22 @@
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
+	unsigned int val, mask;
+
+	if (wm8994->revision < 4) {
+		/* force a HW read */
+		val = wm8994_reg_read(codec->control_data,
+				      WM8994_POWER_MANAGEMENT_5);
+
+		/* modify the cache only */
+		codec->cache_only = 1;
+		mask =  WM8994_DAC1R_ENA | WM8994_DAC1L_ENA |
+			WM8994_DAC2R_ENA | WM8994_DAC2L_ENA;
+		val &= mask;
+		snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5,
+				    mask, val);
+		codec->cache_only = 0;
+	}
 
 	/* Restore the registers */
 	ret = snd_soc_cache_sync(codec);
@@ -2688,6 +2895,13 @@
 	else
 		snd_soc_add_controls(wm8994->codec, wm8994_eq_controls,
 				     ARRAY_SIZE(wm8994_eq_controls));
+
+	for (i = 0; i < ARRAY_SIZE(pdata->micbias); i++) {
+		if (pdata->micbias[i]) {
+			snd_soc_write(codec, WM8958_MICBIAS1 + i,
+				pdata->micbias[i] & 0xffff);
+		}
+	}
 }
 
 /**
@@ -2798,47 +3012,18 @@
 	int report = 0;
 
 	/* If nothing present then clear our statuses */
-	if (!(status & WM8958_MICD_STS)) {
-		wm8994->jack_is_video = false;
-		wm8994->jack_is_mic = false;
+	if (!(status & WM8958_MICD_STS))
 		goto done;
-	}
 
-	/* Assume anything over 475 ohms is a microphone and remember
-	 * that we've seen one (since buttons override it) */
-	if (status & 0x600)
-		wm8994->jack_is_mic = true;
-	if (wm8994->jack_is_mic)
-		report |= SND_JACK_MICROPHONE;
-
-	/* Video has an impedence of approximately 75 ohms; assume
-	 * this isn't used as a button and remember it since buttons
-	 * override it. */
-	if (status & 0x40)
-		wm8994->jack_is_video = true;
-	if (wm8994->jack_is_video)
-		report |= SND_JACK_VIDEOOUT;
+	report = SND_JACK_MICROPHONE;
 
 	/* Everything else is buttons; just assign slots */
-	if (status & 0x4)
+	if (status & 0x1c0)
 		report |= SND_JACK_BTN_0;
-	if (status & 0x8)
-		report |= SND_JACK_BTN_1;
-	if (status & 0x10)
-		report |= SND_JACK_BTN_2;
-	if (status & 0x20)
-		report |= SND_JACK_BTN_3;
-	if (status & 0x80)
-		report |= SND_JACK_BTN_4;
-	if (status & 0x100)
-		report |= SND_JACK_BTN_5;
 
 done:
-	snd_soc_jack_report(wm8994->micdet[0].jack,
-			    SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 |
-			    SND_JACK_BTN_3 | SND_JACK_BTN_4 | SND_JACK_BTN_5 |
-			    SND_JACK_MICROPHONE | SND_JACK_VIDEOOUT,
-			    report);
+	snd_soc_jack_report(wm8994->micdet[0].jack, report,
+			    SND_JACK_BTN_0 | SND_JACK_MICROPHONE);
 }
 
 /**
@@ -2937,6 +3122,12 @@
 	wm8994->pdata = dev_get_platdata(codec->dev->parent);
 	wm8994->codec = codec;
 
+	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_resume(codec->dev);
 
@@ -2985,14 +3176,17 @@
 
 	switch (control->type) {
 	case WM8994:
-		ret = wm8994_request_irq(codec->control_data,
-					 WM8994_IRQ_MIC1_DET,
-					 wm8994_mic_irq, "Mic 1 detect",
-					 wm8994);
-		if (ret != 0)
-			dev_warn(codec->dev,
-				 "Failed to request Mic1 detect IRQ: %d\n",
-				 ret);
+		if (wm8994->micdet_irq) {
+			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+						   wm8994_mic_irq,
+						   IRQF_TRIGGER_RISING,
+						   "Mic1 detect",
+						   wm8994);
+			if (ret != 0)
+				dev_warn(codec->dev,
+					 "Failed to request Mic1 detect IRQ: %d\n",
+					 ret);
+		}
 
 		ret = wm8994_request_irq(codec->control_data,
 					 WM8994_IRQ_MIC1_SHRT,
@@ -3023,15 +3217,17 @@
 		break;
 
 	case WM8958:
-		ret = wm8994_request_irq(codec->control_data,
-					 WM8994_IRQ_MIC1_DET,
-					 wm8958_mic_irq, "Mic detect",
-					 wm8994);
-		if (ret != 0)
-			dev_warn(codec->dev,
-				 "Failed to request Mic detect IRQ: %d\n",
-				 ret);
-		break;
+		if (wm8994->micdet_irq) {
+			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
+						   wm8958_mic_irq,
+						   IRQF_TRIGGER_RISING,
+						   "Mic detect",
+						   wm8994);
+			if (ret != 0)
+				dev_warn(codec->dev,
+					 "Failed to request Mic detect IRQ: %d\n",
+					 ret);
+		}
 	}
 
 	/* Remember if AIFnLRCLK is configured as a GPIO.  This should be
@@ -3112,10 +3308,31 @@
 	case WM8994:
 		snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets,
 					  ARRAY_SIZE(wm8994_specific_dapm_widgets));
+		if (wm8994->revision < 4) {
+			snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets,
+						  ARRAY_SIZE(wm8994_lateclk_revd_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets,
+						  ARRAY_SIZE(wm8994_adc_revd_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_dac_revd_widgets,
+						  ARRAY_SIZE(wm8994_dac_revd_widgets));
+		} else {
+			snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+						  ARRAY_SIZE(wm8994_lateclk_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+						  ARRAY_SIZE(wm8994_adc_widgets));
+			snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+						  ARRAY_SIZE(wm8994_dac_widgets));
+		}
 		break;
 	case WM8958:
 		snd_soc_add_controls(codec, wm8958_snd_controls,
 				     ARRAY_SIZE(wm8958_snd_controls));
+		snd_soc_dapm_new_controls(dapm, wm8994_lateclk_widgets,
+					  ARRAY_SIZE(wm8994_lateclk_widgets));
+		snd_soc_dapm_new_controls(dapm, wm8994_adc_widgets,
+					  ARRAY_SIZE(wm8994_adc_widgets));
+		snd_soc_dapm_new_controls(dapm, wm8994_dac_widgets,
+					  ARRAY_SIZE(wm8994_dac_widgets));
 		snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
 					  ARRAY_SIZE(wm8958_dapm_widgets));
 		break;
@@ -3129,8 +3346,20 @@
 	case WM8994:
 		snd_soc_dapm_add_routes(dapm, wm8994_intercon,
 					ARRAY_SIZE(wm8994_intercon));
+
+		if (wm8994->revision < 4) {
+			snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon,
+						ARRAY_SIZE(wm8994_revd_intercon));
+			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon,
+						ARRAY_SIZE(wm8994_lateclk_revd_intercon));
+		} else {
+			snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+						ARRAY_SIZE(wm8994_lateclk_intercon));
+		}
 		break;
 	case WM8958:
+		snd_soc_dapm_add_routes(dapm, wm8994_lateclk_intercon,
+					ARRAY_SIZE(wm8994_lateclk_intercon));
 		snd_soc_dapm_add_routes(dapm, wm8958_intercon,
 					ARRAY_SIZE(wm8958_intercon));
 		break;
@@ -3142,7 +3371,8 @@
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET, wm8994);
 	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994);
-	wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET, wm8994);
+	if (wm8994->micdet_irq)
+		free_irq(wm8994->micdet_irq, wm8994);
 err:
 	kfree(wm8994);
 	return ret;
@@ -3159,8 +3389,8 @@
 
 	switch (control->type) {
 	case WM8994:
-		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_SHRT,
-				wm8994);
+		if (wm8994->micdet_irq)
+			free_irq(wm8994->micdet_irq, wm8994);
 		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC2_DET,
 				wm8994);
 		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT,
@@ -3170,8 +3400,8 @@
 		break;
 
 	case WM8958:
-		wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_DET,
-				wm8994);
+		if (wm8994->micdet_irq)
+			free_irq(wm8994->micdet_irq, wm8994);
 		break;
 	}
 	kfree(wm8994->retune_mobile_texts);
diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h
index 0c355bf..999b885 100644
--- a/sound/soc/codecs/wm8994.h
+++ b/sound/soc/codecs/wm8994.h
@@ -43,6 +43,6 @@
 };
 
 extern const struct wm8994_access_mask wm8994_access_masks[WM8994_CACHE_SIZE];
-extern const __devinitdata  u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
+extern const u16 wm8994_reg_defaults[WM8994_CACHE_SIZE];
 
 #endif
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 5c224dd..55cdf29 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -15,6 +15,7 @@
 #include <linux/moduleparam.h>
 #include <linux/init.h>
 #include <linux/delay.h>
+#include <linux/device.h>
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
@@ -166,7 +167,7 @@
 	int fll_fref;
 	int fll_fout;
 	int tdm_width;
-	struct wm9081_retune_mobile_config *retune;
+	struct wm9081_pdata pdata;
 };
 
 static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg)
@@ -388,27 +389,6 @@
 SOC_DAPM_SINGLE("Playback Switch", WM9081_ANALOGUE_MIXER, 4, 1, 0),
 };
 
-static int speaker_event(struct snd_soc_dapm_widget *w,
-			 struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	unsigned int reg = snd_soc_read(codec, WM9081_POWER_MANAGEMENT);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		reg |= WM9081_SPK_ENA;
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		reg &= ~WM9081_SPK_ENA;
-		break;
-	}
-
-	snd_soc_write(codec, WM9081_POWER_MANAGEMENT, reg);
-
-	return 0;
-}
-
 struct _fll_div {
 	u16 fll_fratio;
 	u16 fll_outdiv;
@@ -746,9 +726,8 @@
 
 SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA_E("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0,
-		   speaker_event,
-		   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
+SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LINEOUT"),
 SND_SOC_DAPM_OUTPUT("SPKN"),
@@ -761,7 +740,7 @@
 };
 
 
-static const struct snd_soc_dapm_route audio_paths[] = {
+static const struct snd_soc_dapm_route wm9081_audio_paths[] = {
 	{ "DAC", NULL, "CLK_SYS" },
 	{ "DAC", NULL, "CLK_DSP" },
 
@@ -779,8 +758,10 @@
 	{ "Speaker PGA", NULL, "TOCLK" },
 	{ "Speaker PGA", NULL, "CLK_SYS" },
 
-	{ "SPKN", NULL, "Speaker PGA" },
-	{ "SPKP", NULL, "Speaker PGA" },
+	{ "Speaker", NULL, "Speaker PGA" },
+
+	{ "SPKN", NULL, "Speaker" },
+	{ "SPKP", NULL, "Speaker" },
 };
 
 static int wm9081_set_bias_level(struct snd_soc_codec *codec,
@@ -1081,21 +1062,22 @@
 	aif4 |= wm9081->bclk / wm9081->fs;
 
 	/* Apply a ReTune Mobile configuration if it's in use */
-	if (wm9081->retune) {
-		struct wm9081_retune_mobile_config *retune = wm9081->retune;
+	if (wm9081->pdata.num_retune_configs) {
+		struct wm9081_pdata *pdata = &wm9081->pdata;
 		struct wm9081_retune_mobile_setting *s;
 		int eq1;
 
 		best = 0;
-		best_val = abs(retune->configs[0].rate - wm9081->fs);
-		for (i = 0; i < retune->num_configs; i++) {
-			cur_val = abs(retune->configs[i].rate - wm9081->fs);
+		best_val = abs(pdata->retune_configs[0].rate - wm9081->fs);
+		for (i = 0; i < pdata->num_retune_configs; i++) {
+			cur_val = abs(pdata->retune_configs[i].rate -
+				      wm9081->fs);
 			if (cur_val < best_val) {
 				best_val = cur_val;
 				best = i;
 			}
 		}
-		s = &retune->configs[best];
+		s = &pdata->retune_configs[best];
 
 		dev_dbg(codec->dev, "ReTune Mobile %s tuned for %dHz\n",
 			s->name, s->rate);
@@ -1138,10 +1120,9 @@
 	return 0;
 }
 
-static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai,
+static int wm9081_set_sysclk(struct snd_soc_codec *codec,
 			     int clk_id, unsigned int freq, int dir)
 {
-	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
 
 	switch (clk_id) {
@@ -1206,7 +1187,6 @@
 
 static struct snd_soc_dai_ops wm9081_dai_ops = {
 	.hw_params = wm9081_hw_params,
-	.set_sysclk = wm9081_set_sysclk,
 	.set_fmt = wm9081_set_dai_fmt,
 	.digital_mute = wm9081_digital_mute,
 	.set_tdm_slot = wm9081_set_tdm_slot,
@@ -1230,7 +1210,6 @@
 static int wm9081_probe(struct snd_soc_codec *codec)
 {
 	struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec);
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret;
 	u16 reg;
 
@@ -1254,6 +1233,14 @@
 		return ret;
 	}
 
+	reg = 0;
+	if (wm9081->pdata.irq_high)
+		reg |= WM9081_IRQ_POL;
+	if (!wm9081->pdata.irq_cmos)
+		reg |= WM9081_IRQ_OP_CTRL;
+	snd_soc_update_bits(codec, WM9081_INTERRUPT_CONTROL,
+			    WM9081_IRQ_POL | WM9081_IRQ_OP_CTRL, reg);
+
 	wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Enable zero cross by default */
@@ -1265,17 +1252,13 @@
 
 	snd_soc_add_controls(codec, wm9081_snd_controls,
 			     ARRAY_SIZE(wm9081_snd_controls));
-	if (!wm9081->retune) {
+	if (!wm9081->pdata.num_retune_configs) {
 		dev_dbg(codec->dev,
 			"No ReTune Mobile data, using normal EQ\n");
 		snd_soc_add_controls(codec, wm9081_eq_controls,
 				     ARRAY_SIZE(wm9081_eq_controls));
 	}
 
-	snd_soc_dapm_new_controls(dapm, wm9081_dapm_widgets,
-				  ARRAY_SIZE(wm9081_dapm_widgets));
-	snd_soc_dapm_add_routes(dapm, audio_paths, ARRAY_SIZE(audio_paths));
-
 	return ret;
 }
 
@@ -1319,11 +1302,19 @@
 	.remove = 	wm9081_remove,
 	.suspend =	wm9081_suspend,
 	.resume =	wm9081_resume,
+
+	.set_sysclk = wm9081_set_sysclk,
 	.set_bias_level = wm9081_set_bias_level,
+
 	.reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm9081_reg_defaults,
 	.volatile_register = wm9081_volatile_register,
+
+	.dapm_widgets	  = wm9081_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(wm9081_dapm_widgets),
+	.dapm_routes     = wm9081_audio_paths,
+	.num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths),
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -1341,6 +1332,10 @@
 	wm9081->control_type = SND_SOC_I2C;
 	wm9081->control_data = i2c;
 
+	if (dev_get_platdata(&i2c->dev))
+		memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev),
+		       sizeof(wm9081->pdata));
+
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm9081, &wm9081_dai, 1);
 	if (ret < 0)
@@ -1363,7 +1358,7 @@
 
 static struct i2c_driver wm9081_i2c_driver = {
 	.driver = {
-		.name = "wm9081-codec",
+		.name = "wm9081",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm9081_i2c_probe,
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 613df5d..7b6b3c1 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -82,7 +82,8 @@
 	} while (reg & op && count < 400);
 
 	if (reg & op)
-		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
+		dev_err(codec->dev, "Timed out waiting for DC Servo %x\n",
+			op);
 }
 
 /*
@@ -674,6 +675,9 @@
 };
 
 static const struct snd_soc_dapm_route analogue_routes[] = {
+	{ "MICBIAS1", NULL, "CLK_SYS" },
+	{ "MICBIAS2", NULL, "CLK_SYS" },
+
 	{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
 	{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
 
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index b36f0b3..fe79842 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -218,7 +218,19 @@
 		.ops = &evm_spdif_ops,
 	},
 };
-static struct snd_soc_dai_link da8xx_evm_dai = {
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.cpu_dai_name = "davinci-mcasp.1",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.codec_name = "tlv320aic3x-codec.1-0018",
+	.platform_name = "davinci-pcm-audio",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+};
+
+static struct snd_soc_dai_link da850_evm_dai = {
 	.name = "TLV320AIC3X",
 	.stream_name = "AIC3X",
 	.cpu_dai_name= "davinci-mcasp.0",
@@ -259,13 +271,13 @@
 
 static struct snd_soc_card da830_snd_soc_card = {
 	.name = "DA830/OMAP-L137 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da830_evm_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_card da850_snd_soc_card = {
 	.name = "DA850/OMAP-L138 EVM",
-	.dai_link = &da8xx_evm_dai,
+	.dai_link = &da850_evm_dai,
 	.num_links = 1,
 };
 
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 9e0e565..d0d60b8 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -658,7 +658,7 @@
 		return -ENODEV;
 	}
 
-	ioarea = request_mem_region(mem->start, (mem->end - mem->start) + 1,
+	ioarea = request_mem_region(mem->start, resource_size(mem),
 				    pdev->name);
 	if (!ioarea) {
 		dev_err(&pdev->dev, "McBSP region already claimed\n");
@@ -694,20 +694,25 @@
 	}
 	clk_enable(dev->clk);
 
-	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+	dev->base = ioremap(mem->start, resource_size(mem));
+	if (!dev->base) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release_clk;
+	}
 
 	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].dma_addr =
-	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DXR_REG);
+	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
 
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].dma_addr =
-	    (dma_addr_t)(io_v2p(dev->base) + DAVINCI_MCBSP_DRR_REG);
+	    (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENXIO;
-		goto err_free_mem;
+		goto err_iounmap;
 	}
 	dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK].channel = res->start;
 
@@ -715,7 +720,7 @@
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENXIO;
-		goto err_free_mem;
+		goto err_iounmap;
 	}
 	dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start;
 	dev->dev = &pdev->dev;
@@ -724,14 +729,19 @@
 
 	ret = snd_soc_register_dai(&pdev->dev, &davinci_i2s_dai);
 	if (ret != 0)
-		goto err_free_mem;
+		goto err_iounmap;
 
 	return 0;
 
+err_iounmap:
+	iounmap(dev->base);
+err_release_clk:
+	clk_disable(dev->clk);
+	clk_put(dev->clk);
 err_free_mem:
 	kfree(dev);
 err_release_region:
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	return ret;
 }
@@ -747,7 +757,7 @@
 	dev->clk = NULL;
 	kfree(dev);
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	return 0;
 }
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index fb55d2c..a5af834 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -868,7 +868,7 @@
 	}
 
 	ioarea = request_mem_region(mem->start,
-			(mem->end - mem->start) + 1, pdev->name);
+			resource_size(mem), pdev->name);
 	if (!ioarea) {
 		dev_err(&pdev->dev, "Audio region already claimed\n");
 		ret = -EBUSY;
@@ -885,7 +885,13 @@
 	clk_enable(dev->clk);
 	dev->clk_active = 1;
 
-	dev->base = (void __iomem *)IO_ADDRESS(mem->start);
+	dev->base = ioremap(mem->start, resource_size(mem));
+	if (!dev->base) {
+		dev_err(&pdev->dev, "ioremap failed\n");
+		ret = -ENOMEM;
+		goto err_release_clk;
+	}
+
 	dev->op_mode = pdata->op_mode;
 	dev->tdm_slots = pdata->tdm_slots;
 	dev->num_serializer = pdata->num_serializer;
@@ -899,14 +905,14 @@
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
-							io_v2p(dev->base));
+							mem->start);
 
 	/* first TX, then RX */
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENODEV;
-		goto err_release_region;
+		goto err_iounmap;
 	}
 
 	dma_data->channel = res->start;
@@ -915,13 +921,13 @@
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
 	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
-							io_v2p(dev->base));
+							mem->start);
 
 	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
 	if (!res) {
 		dev_err(&pdev->dev, "no DMA resource\n");
 		ret = -ENODEV;
-		goto err_release_region;
+		goto err_iounmap;
 	}
 
 	dma_data->channel = res->start;
@@ -929,11 +935,16 @@
 	ret = snd_soc_register_dai(&pdev->dev, &davinci_mcasp_dai[pdata->op_mode]);
 
 	if (ret != 0)
-		goto err_release_region;
+		goto err_iounmap;
 	return 0;
 
+err_iounmap:
+	iounmap(dev->base);
+err_release_clk:
+	clk_disable(dev->clk);
+	clk_put(dev->clk);
 err_release_region:
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 err_release_data:
 	kfree(dev);
 
@@ -951,7 +962,7 @@
 	dev->clk = NULL;
 
 	mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	release_mem_region(mem->start, (mem->end - mem->start) + 1);
+	release_mem_region(mem->start, resource_size(mem));
 
 	kfree(dev);
 
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
index b270085..d3aa151 100644
--- a/sound/soc/ep93xx/edb93xx.c
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -41,17 +41,17 @@
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	int err;
+	unsigned int mclk_rate;
 	unsigned int rate = params_rate(params);
+
 	/*
-	 * We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
-	 * because our sample size is 32 bit * 2 channels.
-	 * I2S standard permits us to transmit more bits than
-	 * the codec uses.
-	 * MCLK = SCLK * 4 is the best recommended value,
-	 * but we have to fall back to ratio 2 for higher
-	 * sample rates.
+	 * According to CS4271 datasheet we use MCLK/LRCK=256 for
+	 * rates below 50kHz and 128 for higher sample rates
 	 */
-	unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2);
+	if (rate < 50000)
+		mclk_rate = rate * 64 * 4;
+	else
+		mclk_rate = rate * 64 * 2;
 
 	err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
 				  SND_SOC_DAIFMT_NB_IF |
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 68a0bae..104e95c 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -253,7 +253,6 @@
 	struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned v = 0;
 
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index fff579a..042f4e9 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -242,7 +242,7 @@
 {
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned word_len, div, sdiv, lrdiv;
-	int found = 0, err;
+	int err;
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -275,15 +275,14 @@
 	 * the codec uses.
 	 */
 	div = clk_get_rate(info->mclk) / params_rate(params);
-	for (sdiv = 2; sdiv <= 4; sdiv += 2)
-		for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
-			if (sdiv * lrdiv == div) {
-				found = 1;
-				goto out;
-			}
-out:
-	if (!found)
-		return -EINVAL;
+	sdiv = 4;
+	if (div > (256 + 512) / 2) {
+		lrdiv = 128;
+	} else {
+		lrdiv = 64;
+		if (div < (128 + 256) / 2)
+			sdiv = 2;
+	}
 
 	err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
 	if (err)
@@ -314,10 +313,12 @@
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 
 static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
@@ -325,10 +326,12 @@
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 #else
 #define ep93xx_i2s_suspend	NULL
@@ -352,13 +355,13 @@
 	.playback	= {
 		.channels_min	= 2,
 		.channels_max	= 2,
-		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
 		.formats	= EP93XX_I2S_FORMATS,
 	},
 	.capture	= {
 		 .channels_min	= 2,
 		 .channels_max	= 2,
-		 .rates		= SNDRV_PCM_RATE_8000_96000,
+		 .rates		= SNDRV_PCM_RATE_8000_192000,
 		 .formats	= EP93XX_I2S_FORMATS,
 	},
 	.ops		= &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 0667077..a456e49 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@
 				   SNDRV_PCM_INFO_INTERLEAVED	|
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER),
 				   
-	.rates			= SNDRV_PCM_RATE_8000_96000,
+	.rates			= SNDRV_PCM_RATE_8000_192000,
 	.rate_min		= SNDRV_PCM_RATE_8000,
-	.rate_max		= SNDRV_PCM_RATE_96000,
+	.rate_max		= SNDRV_PCM_RATE_192000,
 	
 	.formats		= (SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE |
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index 642270a..d8f130d 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -30,6 +30,16 @@
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
 	  PMIC board with WM8835x fitted.
 
+config SND_SOC_MX27VIS_AIC32X4
+	tristate "SoC audio support for Visstrim M10 boards"
+	depends on MACH_IMX27_VISSTRIM_M10
+	select SND_SOC_TVL320AIC32X4
+	select SND_MXC_SOC_SSI
+	select SND_MXC_SOC_MX2
+	help
+	  Say Y if you want to add support for SoC audio on Visstrim SM10
+	  board with TLV320AIC32X4 codec.
+
 config SND_SOC_PHYCORE_AC97
 	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
 	depends on MACH_PCM043 || MACH_PCA100
@@ -44,7 +54,8 @@
 	tristate "Eukrea TLV320"
 	depends on MACH_EUKREA_MBIMX27_BASEBOARD \
 		|| MACH_EUKREA_MBIMXSD25_BASEBOARD \
-		|| MACH_EUKREA_MBIMXSD35_BASEBOARD
+		|| MACH_EUKREA_MBIMXSD35_BASEBOARD \
+		|| MACH_EUKREA_MBIMXSD51_BASEBOARD
 	select SND_SOC_TLV320AIC23
 	select SND_MXC_SOC_SSI
 	select SND_MXC_SOC_FIQ
diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile
index b67fc02..d6d609b 100644
--- a/sound/soc/imx/Makefile
+++ b/sound/soc/imx/Makefile
@@ -10,8 +10,10 @@
 # i.MX Machine Support
 snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o
 snd-soc-phycore-ac97-objs := phycore-ac97.o
+snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
 obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o
+obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
diff --git a/sound/soc/imx/eukrea-tlv320.c b/sound/soc/imx/eukrea-tlv320.c
index e20c9e1..75fb4b8 100644
--- a/sound/soc/imx/eukrea-tlv320.c
+++ b/sound/soc/imx/eukrea-tlv320.c
@@ -79,7 +79,7 @@
 	.name		= "tlv320aic23",
 	.stream_name	= "TLV320AIC23",
 	.codec_dai_name	= "tlv320aic23-hifi",
-	.platform_name	= "imx-pcm-audio.0",
+	.platform_name	= "imx-fiq-pcm-audio.0",
 	.codec_name	= "tlv320aic23-codec.0-001a",
 	.cpu_dai_name	= "imx-ssi.0",
 	.ops		= &eukrea_tlv320_snd_ops,
@@ -98,7 +98,8 @@
 	int ret;
 
 	if (!machine_is_eukrea_cpuimx27() && !machine_is_eukrea_cpuimx25sd()
-		&& !machine_is_eukrea_cpuimx35sd())
+		&& !machine_is_eukrea_cpuimx35sd()
+		&& !machine_is_eukrea_cpuimx51sd())
 		/* return happy. We might run on a totally different machine */
 		return 0;
 
diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c
index 30894ea..bc92ec6 100644
--- a/sound/soc/imx/imx-ssi.c
+++ b/sound/soc/imx/imx-ssi.c
@@ -108,7 +108,7 @@
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		/* data on rising edge of bclk, frame high with data */
-		strcr |= SSI_STCR_TFSL;
+		strcr |= SSI_STCR_TFSL | SSI_STCR_TXBIT0;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 		/* data on rising edge of bclk, frame high 1clk before data */
@@ -656,6 +656,9 @@
 	ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0;
 	ssi->dma_params_tx.dma_addr = res->start + SSI_STX0;
 
+	ssi->dma_params_tx.burstsize = 4;
+	ssi->dma_params_rx.burstsize = 4;
+
 	res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0");
 	if (res)
 		ssi->dma_params_tx.dma = res->start;
diff --git a/sound/soc/imx/mx27vis-aic32x4.c b/sound/soc/imx/mx27vis-aic32x4.c
new file mode 100644
index 0000000..054110b
--- /dev/null
+++ b/sound/soc/imx/mx27vis-aic32x4.c
@@ -0,0 +1,137 @@
+/*
+ * mx27vis-aic32x4.c
+ *
+ * Copyright 2011 Vista Silicon S.L.
+ *
+ * Author: Javier Martin <javier.martin@vista-silicon.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., 51 Franklin Street, Fifth Floor, Boston,
+ * MA 02110-1301, USA.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <asm/mach-types.h>
+#include <mach/audmux.h>
+
+#include "../codecs/tlv320aic32x4.h"
+#include "imx-ssi.h"
+
+static int mx27vis_aic32x4_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 ret;
+	u32 dai_format;
+
+	dai_format = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		SND_SOC_DAIFMT_CBM_CFM;
+
+	/* set codec DAI configuration */
+	snd_soc_dai_set_fmt(codec_dai, dai_format);
+
+	/* set cpu DAI configuration */
+	snd_soc_dai_set_fmt(cpu_dai, dai_format);
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+				     25000000, SND_SOC_CLOCK_OUT);
+	if (ret) {
+		pr_err("%s: failed setting codec sysclk\n", __func__);
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops mx27vis_aic32x4_snd_ops = {
+	.hw_params	= mx27vis_aic32x4_hw_params,
+};
+
+static struct snd_soc_dai_link mx27vis_aic32x4_dai = {
+	.name		= "tlv320aic32x4",
+	.stream_name	= "TLV320AIC32X4",
+	.codec_dai_name	= "tlv320aic32x4-hifi",
+	.platform_name	= "imx-pcm-audio.0",
+	.codec_name	= "tlv320aic32x4.0-0018",
+	.cpu_dai_name	= "imx-ssi.0",
+	.ops		= &mx27vis_aic32x4_snd_ops,
+};
+
+static struct snd_soc_card mx27vis_aic32x4 = {
+	.name		= "visstrim_m10-audio",
+	.dai_link	= &mx27vis_aic32x4_dai,
+	.num_links	= 1,
+};
+
+static struct platform_device *mx27vis_aic32x4_snd_device;
+
+static int __init mx27vis_aic32x4_init(void)
+{
+	int ret;
+
+	mx27vis_aic32x4_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!mx27vis_aic32x4_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mx27vis_aic32x4_snd_device, &mx27vis_aic32x4);
+	ret = platform_device_add(mx27vis_aic32x4_snd_device);
+
+	if (ret) {
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
+		platform_device_put(mx27vis_aic32x4_snd_device);
+	}
+
+	/* Connect SSI0 as clock slave to SSI1 external pins */
+	mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0,
+			MXC_AUDMUX_V1_PCR_SYN |
+			MXC_AUDMUX_V1_PCR_TFSDIR |
+			MXC_AUDMUX_V1_PCR_TCLKDIR |
+			MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1) |
+			MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR1_SSI_PINS_1)
+	);
+	mxc_audmux_v1_configure_port(MX27_AUDMUX_PPCR1_SSI_PINS_1,
+			MXC_AUDMUX_V1_PCR_SYN |
+			MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)
+	);
+
+	return ret;
+}
+
+static void __exit mx27vis_aic32x4_exit(void)
+{
+	platform_device_unregister(mx27vis_aic32x4_snd_device);
+}
+
+module_init(mx27vis_aic32x4_init);
+module_exit(mx27vis_aic32x4_exit);
+
+MODULE_AUTHOR("Javier Martin <javier.martin@vista-silicon.com>");
+MODULE_DESCRIPTION("ALSA SoC AIC32X4 mx27 visstrim");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/mid-x86/Kconfig
index 1ad7538..2935042 100644
--- a/sound/soc/mid-x86/Kconfig
+++ b/sound/soc/mid-x86/Kconfig
@@ -1,6 +1,7 @@
 config SND_MFLD_MACHINE
 	tristate "SOC Machine Audio driver for Intel Medfield MID platform"
 	depends on INTEL_SCU_IPC
+	depends on SND_INTEL_SST
 	select SND_SOC_SN95031
 	select SND_SST_PLATFORM
 	help
@@ -11,4 +12,3 @@
 
 config SND_SST_PLATFORM
 	tristate
-	depends on SND_INTEL_SST
diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c
index 7925851..429aa1b 100644
--- a/sound/soc/mid-x86/mfld_machine.c
+++ b/sound/soc/mid-x86/mfld_machine.c
@@ -27,18 +27,59 @@
 #include <linux/init.h>
 #include <linux/device.h>
 #include <linux/slab.h>
+#include <linux/io.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
+#include <sound/jack.h>
 #include "../codecs/sn95031.h"
 
 #define MID_MONO 1
 #define MID_STEREO 2
 #define MID_MAX_CAP 5
+#define MFLD_JACK_INSERT 0x04
+
+enum soc_mic_bias_zones {
+	MFLD_MV_START = 0,
+	/* mic bias volutage range for Headphones*/
+	MFLD_MV_HP = 400,
+	/* mic bias volutage range for American Headset*/
+	MFLD_MV_AM_HS = 650,
+	/* mic bias volutage range for Headset*/
+	MFLD_MV_HS = 2000,
+	MFLD_MV_UNDEFINED,
+};
 
 static unsigned int	hs_switch;
 static unsigned int	lo_dac;
 
+struct mfld_mc_private {
+	struct platform_device *socdev;
+	void __iomem *int_base;
+	struct snd_soc_codec *codec;
+	u8 interrupt_status;
+};
+
+struct snd_soc_jack mfld_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin mfld_jack_pins[] = {
+	{
+		.pin = "Headphones",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	{
+		.pin = "AMIC1",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
+/* jack detection voltage zones */
+static struct snd_soc_jack_zone mfld_zones[] = {
+	{MFLD_MV_START, MFLD_MV_AM_HS, SND_JACK_HEADPHONE},
+	{MFLD_MV_AM_HS, MFLD_MV_HS, SND_JACK_HEADSET},
+};
+
 /* sound card controls */
 static const char *headset_switch_text[] = {"Earpiece", "Headset"};
 
@@ -67,13 +108,11 @@
 
 	if (ucontrol->value.integer.value[0]) {
 		pr_debug("hs_set HS path\n");
-		snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL");
-		snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR");
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
 		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
 	} else {
 		pr_debug("hs_set EP path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
 		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
 	}
 	snd_soc_dapm_sync(&codec->dapm);
@@ -91,12 +130,10 @@
 	snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT");
 	snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT");
 	if (hs_switch) {
-		snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTL");
-		snd_soc_dapm_enable_pin(&codec->dapm, "HPOUTR");
+		snd_soc_dapm_enable_pin(&codec->dapm, "Headphones");
 		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
 	} else {
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
 		snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT");
 	}
 }
@@ -130,8 +167,7 @@
 
 	case 1:
 		pr_debug("set hs  path\n");
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTL");
-		snd_soc_dapm_disable_pin(&codec->dapm, "HPOUTR");
+		snd_soc_dapm_disable_pin(&codec->dapm, "Headphones");
 		snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT");
 		snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22);
 		break;
@@ -162,12 +198,45 @@
 			lo_get_switch, lo_set_switch),
 };
 
+static const struct snd_soc_dapm_widget mfld_widgets[] = {
+	SND_SOC_DAPM_HP("Headphones", NULL),
+	SND_SOC_DAPM_MIC("Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route mfld_map[] = {
+	{"Headphones", NULL, "HPOUTR"},
+	{"Headphones", NULL, "HPOUTL"},
+	{"Mic", NULL, "AMIC1"},
+};
+
+static void mfld_jack_check(unsigned int intr_status)
+{
+	struct mfld_jack_data jack_data;
+
+	jack_data.mfld_jack = &mfld_jack;
+	jack_data.intr_id = intr_status;
+
+	sn95031_jack_detection(&jack_data);
+	/* TODO: add american headset detection post gpiolib support */
+}
+
 static int mfld_init(struct snd_soc_pcm_runtime *runtime)
 {
 	struct snd_soc_codec *codec = runtime->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret_val;
 
+	/* Add jack sense widgets */
+	snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets));
+
+	/* Set up the map */
+	snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(dapm, "Headphones");
+	snd_soc_dapm_enable_pin(dapm, "Mic");
+	snd_soc_dapm_sync(dapm);
+
 	ret_val = snd_soc_add_controls(codec, mfld_snd_controls,
 				ARRAY_SIZE(mfld_snd_controls));
 	if (ret_val) {
@@ -175,8 +244,7 @@
 		return ret_val;
 	}
 	/* default is earpiece pin, userspace sets it explcitly */
-	snd_soc_dapm_disable_pin(dapm, "HPOUTL");
-	snd_soc_dapm_disable_pin(dapm, "HPOUTR");
+	snd_soc_dapm_disable_pin(dapm, "Headphones");
 	/* default is lineout NC, userspace sets it explcitly */
 	snd_soc_dapm_disable_pin(dapm, "LINEOUTL");
 	snd_soc_dapm_disable_pin(dapm, "LINEOUTR");
@@ -185,7 +253,35 @@
 	/* we dont use linein in this so set to NC */
 	snd_soc_dapm_disable_pin(dapm, "LINEINL");
 	snd_soc_dapm_disable_pin(dapm, "LINEINR");
-	return snd_soc_dapm_sync(dapm);
+	snd_soc_dapm_sync(dapm);
+
+	/* Headset and button jack detection */
+	ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack",
+			SND_JACK_HEADSET | SND_JACK_BTN_0 |
+			SND_JACK_BTN_1, &mfld_jack);
+	if (ret_val) {
+		pr_err("jack creation failed\n");
+		return ret_val;
+	}
+
+	ret_val = snd_soc_jack_add_pins(&mfld_jack,
+			ARRAY_SIZE(mfld_jack_pins), mfld_jack_pins);
+	if (ret_val) {
+		pr_err("adding jack pins failed\n");
+		return ret_val;
+	}
+	ret_val = snd_soc_jack_add_zones(&mfld_jack,
+			ARRAY_SIZE(mfld_zones), mfld_zones);
+	if (ret_val) {
+		pr_err("adding jack zones failed\n");
+		return ret_val;
+	}
+
+	/* we want to check if anything is inserted at boot,
+	 * so send a fake event to codec and it will read adc
+	 * to find if anything is there or not */
+	mfld_jack_check(MFLD_JACK_INSERT);
+	return ret_val;
 }
 
 struct snd_soc_dai_link mfld_msic_dailink[] = {
@@ -234,37 +330,94 @@
 	.num_links = ARRAY_SIZE(mfld_msic_dailink),
 };
 
+static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev)
+{
+	struct mfld_mc_private *mc_private = (struct mfld_mc_private *) dev;
+
+	memcpy_fromio(&mc_private->interrupt_status,
+			((void *)(mc_private->int_base)),
+			sizeof(u8));
+	return IRQ_WAKE_THREAD;
+}
+
+static irqreturn_t snd_mfld_jack_detection(int irq, void *data)
+{
+	struct mfld_mc_private *mc_drv_ctx = (struct mfld_mc_private *) data;
+
+	if (mfld_jack.codec == NULL)
+		return IRQ_HANDLED;
+	mfld_jack_check(mc_drv_ctx->interrupt_status);
+
+	return IRQ_HANDLED;
+}
+
 static int __devinit snd_mfld_mc_probe(struct platform_device *pdev)
 {
-	struct platform_device *socdev;
-	int ret_val = 0;
+	int ret_val = 0, irq;
+	struct mfld_mc_private *mc_drv_ctx;
+	struct resource *irq_mem;
 
 	pr_debug("snd_mfld_mc_probe called\n");
 
-	socdev =  platform_device_alloc("soc-audio", -1);
-	if (!socdev) {
-		pr_err("soc-audio device allocation failed\n");
+	/* retrive the irq number */
+	irq = platform_get_irq(pdev, 0);
+
+	/* audio interrupt base of SRAM location where
+	 * interrupts are stored by System FW */
+	mc_drv_ctx = kzalloc(sizeof(*mc_drv_ctx), GFP_ATOMIC);
+	if (!mc_drv_ctx) {
+		pr_err("allocation failed\n");
 		return -ENOMEM;
 	}
-	platform_set_drvdata(socdev, &snd_soc_card_mfld);
-	ret_val = platform_device_add(socdev);
-	if (ret_val) {
-		pr_err("Unable to add soc-audio device, err %d\n", ret_val);
-		platform_device_put(socdev);
+
+	irq_mem = platform_get_resource_byname(
+				pdev, IORESOURCE_MEM, "IRQ_BASE");
+	if (!irq_mem) {
+		pr_err("no mem resource given\n");
+		ret_val = -ENODEV;
+		goto unalloc;
 	}
-
-	platform_set_drvdata(pdev, socdev);
-
+	mc_drv_ctx->int_base = ioremap_nocache(irq_mem->start,
+					resource_size(irq_mem));
+	if (!mc_drv_ctx->int_base) {
+		pr_err("Mapping of cache failed\n");
+		ret_val = -ENOMEM;
+		goto unalloc;
+	}
+	/* register for interrupt */
+	ret_val = request_threaded_irq(irq, snd_mfld_jack_intr_handler,
+			snd_mfld_jack_detection,
+			IRQF_SHARED, pdev->dev.driver->name, mc_drv_ctx);
+	if (ret_val) {
+		pr_err("cannot register IRQ\n");
+		goto unalloc;
+	}
+	/* register the soc card */
+	snd_soc_card_mfld.dev = &pdev->dev;
+	ret_val = snd_soc_register_card(&snd_soc_card_mfld);
+	if (ret_val) {
+		pr_debug("snd_soc_register_card failed %d\n", ret_val);
+		goto freeirq;
+	}
+	platform_set_drvdata(pdev, mc_drv_ctx);
 	pr_debug("successfully exited probe\n");
 	return ret_val;
+
+freeirq:
+	free_irq(irq, mc_drv_ctx);
+unalloc:
+	kfree(mc_drv_ctx);
+	return ret_val;
 }
 
 static int __devexit snd_mfld_mc_remove(struct platform_device *pdev)
 {
-	struct platform_device *socdev =  platform_get_drvdata(pdev);
-	pr_debug("snd_mfld_mc_remove called\n");
+	struct mfld_mc_private *mc_drv_ctx = platform_get_drvdata(pdev);
 
-	platform_device_unregister(socdev);
+	pr_debug("snd_mfld_mc_remove called\n");
+	free_irq(platform_get_irq(pdev, 0), mc_drv_ctx);
+	snd_soc_unregister_card(&snd_soc_card_mfld);
+	kfree(mc_drv_ctx);
 	platform_set_drvdata(pdev, NULL);
 	return 0;
 }
diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c
index 96e6e9c..ee2c224 100644
--- a/sound/soc/mid-x86/sst_platform.c
+++ b/sound/soc/mid-x86/sst_platform.c
@@ -365,6 +365,14 @@
 	return stream->stream_info.buffer_ptr;
 }
 
+static int sst_platform_pcm_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *params)
+{
+	snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	memset(substream->runtime->dma_area, 0, params_buffer_bytes(params));
+
+	return 0;
+}
 
 static struct snd_pcm_ops sst_platform_ops = {
 	.open = sst_platform_open,
@@ -373,6 +381,7 @@
 	.prepare = sst_platform_pcm_prepare,
 	.trigger = sst_platform_pcm_trigger,
 	.pointer = sst_platform_pcm_pointer,
+	.hw_params = sst_platform_pcm_hw_params,
 };
 
 static void sst_pcm_free(struct snd_pcm *pcm)
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index 28333e7..dc65650 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -117,7 +117,7 @@
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9705-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
@@ -126,7 +126,7 @@
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name = "wm9705-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
index 01bf316..51897fc 100644
--- a/sound/soc/pxa/e750_wm9705.c
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -99,7 +99,7 @@
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9705-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
@@ -109,7 +109,7 @@
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9705-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9705-codec",
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index c6a37c6..053ed20 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -89,7 +89,7 @@
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9712-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
@@ -98,7 +98,7 @@
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fc22e6e..b13a425 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -37,7 +37,7 @@
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9712-hifi",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
@@ -45,7 +45,7 @@
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9712-aux",
 		.platform_name = "pxa-pcm-audio",
 		.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
index 0d70fc8..38ca675 100644
--- a/sound/soc/pxa/mioa701_wm9713.c
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -162,7 +162,7 @@
 	{
 		.name = "AC97",
 		.stream_name = "AC97 HiFi",
-		.cpu_dai_name = "pxa-ac97.0",
+		.cpu_dai_name = "pxa2xx-ac97",
 		.codec_dai_name = "wm9713-hifi",
 		.codec_name = "wm9713-codec",
 		.init = mioa701_wm9713_init,
@@ -172,7 +172,7 @@
 	{
 		.name = "AC97 Aux",
 		.stream_name = "AC97 Aux",
-		.cpu_dai_name = "pxa-ac97.1",
+		.cpu_dai_name = "pxa2xx-ac97-aux",
 		.codec_dai_name ="wm9713-aux",
 		.codec_name = "wm9713-codec",
 		.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 857db96..504e400 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -132,7 +132,7 @@
 {
 	.name = "AC97 HiFi",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_dai_name =  "wm9712-hifi",
 	.codec_name = "wm9712-codec",
 	.platform_name = "pxa-pcm-audio",
@@ -141,7 +141,7 @@
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_dai_name = "wm9712-aux",
 	.codec_name = "wm9712-codec",
 	.platform_name = "pxa-pcm-audio",
diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c
index db1dd56..2afabaf 100644
--- a/sound/soc/pxa/raumfeld.c
+++ b/sound/soc/pxa/raumfeld.c
@@ -229,19 +229,19 @@
 {
 	.name		= "ak4104",
 	.stream_name	= "Playback",
-	.cpu_dai_name = "pxa-ssp-dai.1",
-	.codec_dai_name = "ak4104-hifi",
-	.platform_name = "pxa-pcm-audio",
+	.cpu_dai_name	= "pxa-ssp-dai.1",
+	.codec_dai_name	= "ak4104-hifi",
+	.platform_name	= "pxa-pcm-audio",
 	.ops		= &raumfeld_ak4104_ops,
-	.codec_name = "ak4104-codec.0",
+	.codec_name	= "ak4104-codec.0",
 },
 {
 	.name		= "CS4270",
 	.stream_name	= "CS4270",
-	.cpu_dai_name = "pxa-ssp-dai.0",
-	.platform_name = "pxa-pcm-audio",
-	.codec_dai_name = "cs4270-hifi",
-	.codec_name = "cs4270-codec.0-0048",
+	.cpu_dai_name	= "pxa-ssp-dai.0",
+	.platform_name	= "pxa-pcm-audio",
+	.codec_dai_name	= "cs4270-hifi",
+	.codec_name	= "cs4270-codec.0-0048",
 	.ops		= &raumfeld_cs4270_ops,
 },};
 
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index 489139a..9a23513 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -219,7 +219,7 @@
 {
 	.name = "AC97",
 	.stream_name = "AC97 HiFi",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_dai_name = "wm9712-hifi",
 	.platform_name = "pxa-pcm-audio",
 	.codec_name = "wm9712-codec",
@@ -229,7 +229,7 @@
 {
 	.name = "AC97 Aux",
 	.stream_name = "AC97 Aux",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_dai_name = "wm9712-aux",
 	.platform_name = "pxa-pcm-audio",
 	.codec_name = "wm9712-codec",
diff --git a/sound/soc/pxa/z2.c b/sound/soc/pxa/z2.c
index 3ceaef6..d69d9fc 100644
--- a/sound/soc/pxa/z2.c
+++ b/sound/soc/pxa/z2.c
@@ -95,6 +95,11 @@
 		.pin	= "Headphone Jack",
 		.mask	= SND_JACK_HEADPHONE,
 	},
+	{
+		.pin    = "Ext Spk",
+		.mask   = SND_JACK_HEADPHONE,
+		.invert = 1
+	},
 };
 
 /* Headset jack detection gpios */
@@ -147,7 +152,7 @@
 	snd_soc_dapm_disable_pin(dapm, "LINPUT3");
 	snd_soc_dapm_disable_pin(dapm, "RINPUT3");
 	snd_soc_dapm_disable_pin(dapm, "OUT3");
-	snd_soc_dapm_disable_pin(dapm, "MONO");
+	snd_soc_dapm_disable_pin(dapm, "MONO1");
 
 	/* Add z2 specific widgets */
 	snd_soc_dapm_new_controls(dapm, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index c585829..ac57726 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -166,7 +166,7 @@
 	.stream_name = "AC97 HiFi",
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
-	.cpu_dai_name = "pxa-ac97.0",
+	.cpu_dai_name = "pxa2xx-ac97",
 	.codec_name = "wm9713-hifi",
 	.init = zylonite_wm9713_init,
 },
@@ -175,7 +175,7 @@
 	.stream_name = "AC97 Aux",
 	.codec_name = "wm9713-codec",
 	.platform_name = "pxa-pcm-audio",
-	.cpu_dai_name = "pxa-ac97.1",
+	.cpu_dai_name = "pxa2xx-ac97-aux",
 	.codec_name = "wm9713-aux",
 },
 {
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index a6a6b5f..a08237a 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -35,23 +35,16 @@
 	tristate
 
 config SND_SOC_SAMSUNG_NEO1973_WM8753
-	tristate "SoC I2S Audio support for NEO1973 - WM8753"
-	depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA01
+	tristate "Audio support for Openmoko Neo1973 Smartphones (GTA01/GTA02)"
+	depends on SND_SOC_SAMSUNG && (MACH_NEO1973_GTA01 || MACH_NEO1973_GTA02)
 	select SND_S3C24XX_I2S
 	select SND_SOC_WM8753
+	select SND_SOC_LM4857 if MACH_NEO1973_GTA01
+	select SND_SOC_DFBMCS320
 	help
-	  Say Y if you want to add support for SoC audio on smdk2440
-	  with the WM8753.
+	  Say Y here to enable audio support for the Openmoko Neo1973
+	  Smartphones.
 
-config SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753
-	tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
-	depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02
-	select SND_S3C24XX_I2S
-	select SND_SOC_WM8753
-	help
-	  This driver provides audio support for the Openmoko Neo FreeRunner
-	  smartphone.
-	  
 config SND_SOC_SAMSUNG_JIVE_WM8750
 	tristate "SoC I2S Audio support for Jive"
 	depends on SND_SOC_SAMSUNG && MACH_JIVE
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index 705d4e8..294dec0 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -20,7 +20,6 @@
 # S3C24XX Machine Support
 snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
-snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
@@ -38,7 +37,6 @@
 
 obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
-obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index 9bce1df..5cb3b88 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -310,7 +310,7 @@
 	/* we seem to be getting the odd error from the pcm library due
 	 * to out-of-bounds pointers. this is maybe due to the dma engine
 	 * not having loaded the new values for the channel before being
-	 * callled... (todo - fix )
+	 * called... (todo - fix )
 	 */
 
 	if (res >= snd_pcm_lib_buffer_bytes(substream)) {
diff --git a/sound/soc/samsung/lm4857.h b/sound/soc/samsung/lm4857.h
deleted file mode 100644
index 0cf5b70..0000000
--- a/sound/soc/samsung/lm4857.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * lm4857.h  --  ALSA Soc Audio Layer
- *
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory
- *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.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.
- *
- *  Revision history
- *    18th Jun 2007   Initial version.
- */
-
-#ifndef LM4857_H_
-#define LM4857_H_
-
-/* The register offsets in the cache array */
-#define LM4857_MVOL 0
-#define LM4857_LVOL 1
-#define LM4857_RVOL 2
-#define LM4857_CTRL 3
-
-/* the shifts required to set these bits */
-#define LM4857_3D 5
-#define LM4857_WAKEUP 5
-#define LM4857_EPGAIN 4
-
-#endif /*LM4857_H_*/
-
diff --git a/sound/soc/samsung/neo1973_gta02_wm8753.c b/sound/soc/samsung/neo1973_gta02_wm8753.c
deleted file mode 100644
index 95ebf81..0000000
--- a/sound/soc/samsung/neo1973_gta02_wm8753.c
+++ /dev/null
@@ -1,494 +0,0 @@
-/*
- * neo1973_gta02_wm8753.c  --  SoC audio for Openmoko Freerunner(GTA02)
- *
- * Copyright 2007 Openmoko Inc
- * Author: Graeme Gregory <graeme@openmoko.org>
- * Copyright 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory <linux@wolfsonmicro.com>
- * Copyright 2009 Wolfson Microelectronics
- *
- *  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/gpio.h>
-
-#include <sound/soc.h>
-
-#include <asm/mach-types.h>
-#include <plat/regs-iis.h>
-#include <mach/gta02.h>
-
-#include "../codecs/wm8753.h"
-#include "s3c24xx-i2s.h"
-
-static struct snd_soc_card neo1973_gta02;
-
-static int neo1973_gta02_hifi_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;
-	unsigned int pll_out = 0, bclk = 0;
-	int ret = 0;
-	unsigned long iis_clkrate;
-
-	iis_clkrate = s3c24xx_i2s_get_clockrate();
-
-	switch (params_rate(params)) {
-	case 8000:
-	case 16000:
-		pll_out = 12288000;
-		break;
-	case 48000:
-		bclk = WM8753_BCLK_DIV_4;
-		pll_out = 12288000;
-		break;
-	case 96000:
-		bclk = WM8753_BCLK_DIV_2;
-		pll_out = 12288000;
-		break;
-	case 11025:
-		bclk = WM8753_BCLK_DIV_16;
-		pll_out = 11289600;
-		break;
-	case 22050:
-		bclk = WM8753_BCLK_DIV_8;
-		pll_out = 11289600;
-		break;
-	case 44100:
-		bclk = WM8753_BCLK_DIV_4;
-		pll_out = 11289600;
-		break;
-	case 88200:
-		bclk = WM8753_BCLK_DIV_2;
-		pll_out = 11289600;
-		break;
-	}
-
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai,
-		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set cpu DAI configuration */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-		SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
-		SND_SOC_DAIFMT_CBM_CFM);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
-		SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* set MCLK division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
-		S3C2410_IISMOD_32FS);
-	if (ret < 0)
-		return ret;
-
-	/* set codec BCLK division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(codec_dai,
-					WM8753_BCLKDIV, bclk);
-	if (ret < 0)
-		return ret;
-
-	/* set prescaler division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
-		S3C24XX_PRESCALE(4, 4));
-	if (ret < 0)
-		return ret;
-
-	/* codec PLL input is PCLK/4 */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
-		iis_clkrate / 4, pll_out);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
-	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
-}
-
-/*
- * Neo1973 WM8753 HiFi DAI opserations.
- */
-static struct snd_soc_ops neo1973_gta02_hifi_ops = {
-	.hw_params = neo1973_gta02_hifi_hw_params,
-	.hw_free = neo1973_gta02_hifi_hw_free,
-};
-
-static int neo1973_gta02_voice_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;
-	unsigned int pcmdiv = 0;
-	int ret = 0;
-	unsigned long iis_clkrate;
-
-	iis_clkrate = s3c24xx_i2s_get_clockrate();
-
-	if (params_rate(params) != 8000)
-		return -EINVAL;
-	if (params_channels(params) != 1)
-		return -EINVAL;
-
-	pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
-
-	/* todo: gg check mode (DSP_B) against CSR datasheet */
-	/* set codec DAI configuration */
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
-		12288000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
-		return ret;
-
-	/* set codec PCM division for sample rate */
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
-					pcmdiv);
-	if (ret < 0)
-		return ret;
-
-	/* configure and enable PLL for 12.288MHz output */
-	ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
-		iis_clkrate / 4, 12288000);
-	if (ret < 0)
-		return ret;
-
-	return 0;
-}
-
-static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-
-	/* disable the PLL */
-	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
-}
-
-static struct snd_soc_ops neo1973_gta02_voice_ops = {
-	.hw_params = neo1973_gta02_voice_hw_params,
-	.hw_free = neo1973_gta02_voice_hw_free,
-};
-
-#define LM4853_AMP 1
-#define LM4853_SPK 2
-
-static u8 lm4853_state;
-
-/* This has no effect, it exists only to maintain compatibility with
- * existing ALSA state files.
- */
-static int lm4853_set_state(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	int val = ucontrol->value.integer.value[0];
-
-	if (val)
-		lm4853_state |= LM4853_AMP;
-	else
-		lm4853_state &= ~LM4853_AMP;
-
-	return 0;
-}
-
-static int lm4853_get_state(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
-
-	return 0;
-}
-
-static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	int val = ucontrol->value.integer.value[0];
-
-	if (val) {
-		lm4853_state |= LM4853_SPK;
-		gpio_set_value(GTA02_GPIO_HP_IN, 0);
-	} else {
-		lm4853_state &= ~LM4853_SPK;
-		gpio_set_value(GTA02_GPIO_HP_IN, 1);
-	}
-
-	return 0;
-}
-
-static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
-
-	return 0;
-}
-
-static int lm4853_event(struct snd_soc_dapm_widget *w,
-			struct snd_kcontrol *k,
-			int event)
-{
-	gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
-
-	return 0;
-}
-
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
-	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-	SND_SOC_DAPM_LINE("GSM Line In", NULL),
-	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Handset Mic", NULL),
-	SND_SOC_DAPM_SPK("Handset Spk", NULL),
-};
-
-
-/* example machine audio_mapnections */
-static const struct snd_soc_dapm_route audio_map[] = {
-
-	/* Connections to the lm4853 amp */
-	{"Stereo Out", NULL, "LOUT1"},
-	{"Stereo Out", NULL, "ROUT1"},
-
-	/* Connections to the GSM Module */
-	{"GSM Line Out", NULL, "MONO1"},
-	{"GSM Line Out", NULL, "MONO2"},
-	{"RXP", NULL, "GSM Line In"},
-	{"RXN", NULL, "GSM Line In"},
-
-	/* Connections to Headset */
-	{"MIC1", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Headset Mic"},
-
-	/* Call Mic */
-	{"MIC2", NULL, "Mic Bias"},
-	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Handset Mic"},
-
-	/* Call Speaker */
-	{"Handset Spk", NULL, "LOUT2"},
-	{"Handset Spk", NULL, "ROUT2"},
-
-	/* Connect the ALC pins */
-	{"ACIN", NULL, "ACOP"},
-};
-
-static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
-	SOC_DAPM_PIN_SWITCH("Stereo Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
-	SOC_DAPM_PIN_SWITCH("GSM Line In"),
-	SOC_DAPM_PIN_SWITCH("Headset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Mic"),
-	SOC_DAPM_PIN_SWITCH("Handset Spk"),
-
-	/* This has no effect, it exists only to maintain compatibility with
-	 * existing ALSA state files.
-	 */
-	SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
-		lm4853_get_state,
-		lm4853_set_state),
-	SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
-		lm4853_get_spk,
-		lm4853_set_spk),
-};
-
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 GTA02.
- */
-static int neo1973_gta02_wm8753_init(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_codec *codec = rtd->codec;
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
-
-	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "OUT3");
-	snd_soc_dapm_nc_pin(dapm, "OUT4");
-	snd_soc_dapm_nc_pin(dapm, "LINE1");
-	snd_soc_dapm_nc_pin(dapm, "LINE2");
-
-	/* Add neo1973 gta02 specific widgets */
-	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
-
-	/* add neo1973 gta02 specific controls */
-	err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
-		ARRAY_SIZE(wm8753_neo1973_gta02_controls));
-
-	if (err < 0)
-		return err;
-
-	/* set up neo1973 gta02 specific audio path audio_map */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-	/* set endpoints to default off mode */
-	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
-	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
-
-	/* allow audio paths from the GSM modem to run during suspend */
-	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
-	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
-	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
-	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
-
-	snd_soc_dapm_sync(dapm);
-
-	return 0;
-}
-
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai_driver bt_dai = {
-	.name = "bluetooth-dai",
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-};
-
-static struct snd_soc_dai_link neo1973_gta02_dai[] = {
-{ /* Hifi Playback - for similatious use with voice below */
-	.name = "WM8753",
-	.stream_name = "WM8753 HiFi",
-	.cpu_dai_name = "s3c24xx-iis",
-	.codec_dai_name = "wm8753-hifi",
-	.init = neo1973_gta02_wm8753_init,
-	.platform_name = "samsung-audio",
-	.codec_name = "wm8753-codec.0-001a",
-	.ops = &neo1973_gta02_hifi_ops,
-},
-{ /* Voice via BT */
-	.name = "Bluetooth",
-	.stream_name = "Voice",
-	.cpu_dai_name = "bluetooth-dai",
-	.codec_dai_name = "wm8753-voice",
-	.ops = &neo1973_gta02_voice_ops,
-	.codec_name = "wm8753-codec.0-001a",
-	.platform_name = "samsung-audio",
-},
-};
-
-static struct snd_soc_card neo1973_gta02 = {
-	.name = "neo1973-gta02",
-	.dai_link = neo1973_gta02_dai,
-	.num_links = ARRAY_SIZE(neo1973_gta02_dai),
-};
-
-static struct platform_device *neo1973_gta02_snd_device;
-
-static int __init neo1973_gta02_init(void)
-{
-	int ret;
-
-	if (!machine_is_neo1973_gta02()) {
-		printk(KERN_INFO
-		       "Only GTA02 is supported by this ASoC driver\n");
-		return -ENODEV;
-	}
-
-	neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!neo1973_gta02_snd_device)
-		return -ENOMEM;
-
-	/* register bluetooth DAI here */
-	ret = snd_soc_register_dai(&neo1973_gta02_snd_device->dev, &bt_dai);
-	if (ret)
-		goto err_put_device;
-
-	platform_set_drvdata(neo1973_gta02_snd_device, &neo1973_gta02);
-	ret = platform_device_add(neo1973_gta02_snd_device);
-
-	if (ret)
-		goto err_unregister_dai;
-
-	/* Initialise GPIOs used by amp */
-	ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
-		goto err_del_device;
-	}
-
-	ret = gpio_direction_output(GTA02_GPIO_HP_IN, 1);
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
-		goto err_free_gpio_hp_in;
-	}
-
-	ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
-		goto err_free_gpio_hp_in;
-	}
-
-	ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
-	if (ret) {
-		pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
-		goto err_free_gpio_amp_shut;
-	}
-
-	return 0;
-
-err_free_gpio_amp_shut:
-	gpio_free(GTA02_GPIO_AMP_SHUT);
-err_free_gpio_hp_in:
-	gpio_free(GTA02_GPIO_HP_IN);
-err_del_device:
-	platform_device_del(neo1973_gta02_snd_device);
-err_unregister_dai:
-	snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
-err_put_device:
-	platform_device_put(neo1973_gta02_snd_device);
-	return ret;
-}
-module_init(neo1973_gta02_init);
-
-static void __exit neo1973_gta02_exit(void)
-{
-	snd_soc_unregister_dai(&neo1973_gta02_snd_device->dev);
-	platform_device_unregister(neo1973_gta02_snd_device);
-	gpio_free(GTA02_GPIO_HP_IN);
-	gpio_free(GTA02_GPIO_AMP_SHUT);
-}
-module_exit(neo1973_gta02_exit);
-
-/* Module information */
-MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
-MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c
index d3cd688..78bfdb3 100644
--- a/sound/soc/samsung/neo1973_wm8753.c
+++ b/sound/soc/samsung/neo1973_wm8753.c
@@ -1,56 +1,32 @@
 /*
- * neo1973_wm8753.c  --  SoC audio for Neo1973
+ * neo1973_wm8753.c  --  SoC audio for Openmoko Neo1973 and Freerunner devices
  *
+ * Copyright 2007 Openmoko Inc
+ * Author: Graeme Gregory <graeme@openmoko.org>
  * Copyright 2007 Wolfson Microelectronics PLC.
  * Author: Graeme Gregory
  *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2009 Wolfson Microelectronics
  *
  *  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/moduleparam.h>
-#include <linux/timer.h>
-#include <linux/interrupt.h>
 #include <linux/platform_device.h>
-#include <linux/i2c.h>
-#include <sound/core.h>
-#include <sound/pcm.h>
+#include <linux/gpio.h>
+
 #include <sound/soc.h>
-#include <sound/tlv.h>
 
 #include <asm/mach-types.h>
-#include <mach/regs-clock.h>
-#include <mach/regs-gpio.h>
-#include <mach/hardware.h>
-#include <linux/io.h>
-#include <mach/spi-gpio.h>
-
 #include <plat/regs-iis.h>
+#include <mach/gta02.h>
 
 #include "../codecs/wm8753.h"
-#include "lm4857.h"
-#include "dma.h"
 #include "s3c24xx-i2s.h"
 
-/* define the scenarios */
-#define NEO_AUDIO_OFF			0
-#define NEO_GSM_CALL_AUDIO_HANDSET	1
-#define NEO_GSM_CALL_AUDIO_HEADSET	2
-#define NEO_GSM_CALL_AUDIO_BLUETOOTH	3
-#define NEO_STEREO_TO_SPEAKERS		4
-#define NEO_STEREO_TO_HEADPHONES	5
-#define NEO_CAPTURE_HANDSET		6
-#define NEO_CAPTURE_HEADSET		7
-#define NEO_CAPTURE_BLUETOOTH		8
-
-static struct snd_soc_card neo1973;
-static struct i2c_client *i2c;
-
 static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
@@ -61,8 +37,6 @@
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	pr_debug("Entered %s\n", __func__);
-
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	switch (params_rate(params)) {
@@ -147,8 +121,6 @@
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	pr_debug("Entered %s\n", __func__);
-
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
 }
@@ -170,8 +142,6 @@
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	pr_debug("Entered %s\n", __func__);
-
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
 	if (params_rate(params) != 8000)
@@ -213,8 +183,6 @@
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 
-	pr_debug("Entered %s\n", __func__);
-
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
 }
@@ -224,205 +192,16 @@
 	.hw_free = neo1973_voice_hw_free,
 };
 
-static int neo1973_scenario;
+/* Shared routes and controls */
 
-static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	ucontrol->value.integer.value[0] = neo1973_scenario;
-	return 0;
-}
-
-static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	pr_debug("Entered %s\n", __func__);
-
-	switch (neo1973_scenario) {
-	case NEO_AUDIO_OFF:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_HANDSET:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_enable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_HEADSET:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_GSM_CALL_AUDIO_BLUETOOTH:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_enable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_STEREO_TO_SPEAKERS:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_STEREO_TO_HEADPHONES:
-		snd_soc_dapm_enable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_HANDSET:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_enable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_HEADSET:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_enable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	case NEO_CAPTURE_BLUETOOTH:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-		break;
-	default:
-		snd_soc_dapm_disable_pin(dapm, "Audio Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
-		snd_soc_dapm_disable_pin(dapm, "GSM Line In");
-		snd_soc_dapm_disable_pin(dapm, "Headset Mic");
-		snd_soc_dapm_disable_pin(dapm, "Call Mic");
-	}
-
-	snd_soc_dapm_sync(dapm);
-
-	return 0;
-}
-
-static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-
-	pr_debug("Entered %s\n", __func__);
-
-	if (neo1973_scenario == ucontrol->value.integer.value[0])
-		return 0;
-
-	neo1973_scenario = ucontrol->value.integer.value[0];
-	set_scenario_endpoints(codec, neo1973_scenario);
-	return 1;
-}
-
-static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
-
-static void lm4857_write_regs(void)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
-		printk(KERN_ERR "lm4857: i2c write failed\n");
-}
-
-static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int reg = mc->reg;
-	int shift = mc->shift;
-	int mask = mc->max;
-
-	pr_debug("Entered %s\n", __func__);
-
-	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
-	return 0;
-}
-
-static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	struct soc_mixer_control *mc =
-		(struct soc_mixer_control *)kcontrol->private_value;
-	int reg = mc->reg;
-	int shift = mc->shift;
-	int mask = mc->max;
-
-	if (((lm4857_regs[reg] >> shift) & mask) ==
-		ucontrol->value.integer.value[0])
-		return 0;
-
-	lm4857_regs[reg] &= ~(mask << shift);
-	lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
-	lm4857_write_regs();
-	return 1;
-}
-
-static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
-
-	pr_debug("Entered %s\n", __func__);
-
-	if (value)
-		value -= 5;
-
-	ucontrol->value.integer.value[0] = value;
-	return 0;
-}
-
-static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol)
-{
-	u8 value = ucontrol->value.integer.value[0];
-
-	pr_debug("Entered %s\n", __func__);
-
-	if (value)
-		value += 5;
-
-	if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
-		return 0;
-
-	lm4857_regs[LM4857_CTRL] &= 0xF0;
-	lm4857_regs[LM4857_CTRL] |= value;
-	lm4857_write_regs();
-	return 1;
-}
-
-static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
-	SND_SOC_DAPM_LINE("Audio Out", NULL),
+static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
 	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
 	SND_SOC_DAPM_LINE("GSM Line In", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
-	SND_SOC_DAPM_MIC("Call Mic", NULL),
+	SND_SOC_DAPM_MIC("Handset Mic", NULL),
 };
 
-
-static const struct snd_soc_dapm_route dapm_routes[] = {
-
-	/* Connections to the lm4857 amp */
-	{"Audio Out", NULL, "LOUT1"},
-	{"Audio Out", NULL, "ROUT1"},
-
+static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
 	/* Connections to the GSM Module */
 	{"GSM Line Out", NULL, "MONO1"},
 	{"GSM Line Out", NULL, "MONO2"},
@@ -436,123 +215,209 @@
 	/* Call Mic */
 	{"MIC2", NULL, "Mic Bias"},
 	{"MIC2N", NULL, "Mic Bias"},
-	{"Mic Bias", NULL, "Call Mic"},
+	{"Mic Bias", NULL, "Handset Mic"},
 
 	/* Connect the ALC pins */
 	{"ACIN", NULL, "ACOP"},
 };
 
-static const char *lm4857_mode[] = {
-	"Off",
-	"Call Speaker",
-	"Stereo Speakers",
-	"Stereo Speakers + Headphones",
-	"Headphones"
+static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("GSM Line Out"),
+	SOC_DAPM_PIN_SWITCH("GSM Line In"),
+	SOC_DAPM_PIN_SWITCH("Headset Mic"),
+	SOC_DAPM_PIN_SWITCH("Handset Mic"),
 };
 
-static const struct soc_enum lm4857_mode_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
+/* GTA02 specific routes and controlls */
+
+#ifdef CONFIG_MACH_NEO1973_GTA02
+
+static int gta02_speaker_enabled;
+
+static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	gta02_speaker_enabled = ucontrol->value.integer.value[0];
+
+	gpio_set_value(GTA02_GPIO_HP_IN, !gta02_speaker_enabled);
+
+	return 0;
+}
+
+static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = gta02_speaker_enabled;
+	return 0;
+}
+
+static int lm4853_event(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(event));
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_route neo1973_gta02_routes[] = {
+	/* Connections to the amp */
+	{"Stereo Out", NULL, "LOUT1"},
+	{"Stereo Out", NULL, "ROUT1"},
+
+	/* Call Speaker */
+	{"Handset Spk", NULL, "LOUT2"},
+	{"Handset Spk", NULL, "ROUT2"},
 };
 
-static const char *neo_scenarios[] = {
-	"Off",
-	"GSM Handset",
-	"GSM Headset",
-	"GSM Bluetooth",
-	"Speakers",
-	"Headphones",
-	"Capture Handset",
-	"Capture Headset",
-	"Capture Bluetooth"
+static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Handset Spk"),
+	SOC_DAPM_PIN_SWITCH("Stereo Out"),
+
+	SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0,
+		lm4853_get_spk,
+		lm4853_set_spk),
 };
 
-static const struct soc_enum neo_scenario_enum[] = {
-	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
+static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
 };
 
-static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
-static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
+static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int ret;
 
-static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
-	SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
-	SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, stereo_tlv),
-	SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
-		lm4857_get_reg, lm4857_set_reg, mono_tlv),
-	SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
-		lm4857_get_mode, lm4857_set_mode),
-	SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
-		neo1973_get_scenario, neo1973_set_scenario),
-	SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-	SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
-		lm4857_get_reg, lm4857_set_reg),
-};
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets,
+			ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets));
+	if (ret)
+		return ret;
 
-/*
- * This is an example machine initialisation for a wm8753 connected to a
- * neo1973 II. It is missing logic to detect hp/mic insertions and logic
- * to re-route the audio in such an event.
- */
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes,
+			ARRAY_SIZE(neo1973_gta02_routes));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_add_controls(codec, neo1973_gta02_wm8753_controls,
+			ARRAY_SIZE(neo1973_gta02_wm8753_controls));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_disable_pin(dapm, "Stereo Out");
+	snd_soc_dapm_disable_pin(dapm, "Handset Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+
+	return 0;
+}
+
+#else
+static int neo1973_gta02_wm8753_init(struct snd_soc_code *codec) { return 0; }
+#endif
+
 static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int err;
-
-	pr_debug("Entered %s\n", __func__);
+	int ret;
 
 	/* set up NC codec pins */
-	snd_soc_dapm_nc_pin(dapm, "LOUT2");
-	snd_soc_dapm_nc_pin(dapm, "ROUT2");
+	if (machine_is_neo1973_gta01()) {
+		snd_soc_dapm_nc_pin(dapm, "LOUT2");
+		snd_soc_dapm_nc_pin(dapm, "ROUT2");
+	}
 	snd_soc_dapm_nc_pin(dapm, "OUT3");
 	snd_soc_dapm_nc_pin(dapm, "OUT4");
 	snd_soc_dapm_nc_pin(dapm, "LINE1");
 	snd_soc_dapm_nc_pin(dapm, "LINE2");
 
 	/* Add neo1973 specific widgets */
-	snd_soc_dapm_new_controls(dapm, wm8753_dapm_widgets,
-				  ARRAY_SIZE(wm8753_dapm_widgets));
-
-	/* set endpoints to default mode */
-	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets,
+			ARRAY_SIZE(neo1973_wm8753_dapm_widgets));
+	if (ret)
+		return ret;
 
 	/* add neo1973 specific controls */
-	err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
-				ARRAY_SIZE(8753_neo1973_controls));
-	if (err < 0)
-		return err;
+	ret = snd_soc_add_controls(codec, neo1973_wm8753_controls,
+			ARRAY_SIZE(neo1973_wm8753_controls));
+	if (ret)
+		return ret;
 
 	/* set up neo1973 specific audio routes */
-	err = snd_soc_dapm_add_routes(dapm, dapm_routes,
-				      ARRAY_SIZE(dapm_routes));
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes,
+			ARRAY_SIZE(neo1973_wm8753_routes));
+	if (ret)
+		return ret;
+
+	/* set endpoints to default off mode */
+	snd_soc_dapm_disable_pin(dapm, "GSM Line Out");
+	snd_soc_dapm_disable_pin(dapm, "GSM Line In");
+	snd_soc_dapm_disable_pin(dapm, "Headset Mic");
+	snd_soc_dapm_disable_pin(dapm, "Handset Mic");
+
+	/* allow audio paths from the GSM modem to run during suspend */
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out");
+	snd_soc_dapm_ignore_suspend(dapm, "GSM Line In");
+	snd_soc_dapm_ignore_suspend(dapm, "Headset Mic");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Mic");
+
+	if (machine_is_neo1973_gta02()) {
+		ret = neo1973_gta02_wm8753_init(codec);
+		if (ret)
+			return ret;
+	}
 
 	snd_soc_dapm_sync(dapm);
+
 	return 0;
 }
 
-/*
- * BT Codec DAI
- */
-static struct snd_soc_dai bt_dai = {
-	.name = "bluetooth-dai",
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 1,
-		.rates = SNDRV_PCM_RATE_8000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+/* GTA01 specific controlls */
+
+#ifdef CONFIG_MACH_NEO1973_GTA01
+
+static const struct snd_soc_dapm_route neo1973_lm4857_routes[] = {
+	{"Amp IN", NULL, "ROUT1"},
+	{"Amp IN", NULL, "LOUT1"},
+
+	{"Handset Spk", NULL, "Amp EP"},
+	{"Stereo Out", NULL, "Amp LS"},
+	{"Headphone", NULL, "Amp HP"},
 };
 
+static const struct snd_soc_dapm_widget neo1973_lm4857_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Handset Spk", NULL),
+	SND_SOC_DAPM_SPK("Stereo Out", NULL),
+	SND_SOC_DAPM_HP("Headphone", NULL),
+};
+
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm)
+{
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(dapm, neo1973_lm4857_dapm_widgets,
+			ARRAY_SIZE(neo1973_lm4857_dapm_widgets));
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dapm_add_routes(dapm, neo1973_lm4857_routes,
+			ARRAY_SIZE(neo1973_lm4857_routes));
+	if (ret)
+		return ret;
+
+	snd_soc_dapm_ignore_suspend(dapm, "Stereo Out");
+	snd_soc_dapm_ignore_suspend(dapm, "Handset Spk");
+	snd_soc_dapm_ignore_suspend(dapm, "Headphone");
+
+	snd_soc_dapm_sync(dapm);
+
+	return 0;
+}
+
+#else
+static int neo1973_lm4857_init(struct snd_soc_dapm_context *dapm) { return 0; };
+#endif
+
 static struct snd_soc_dai_link neo1973_dai[] = {
 { /* Hifi Playback - for similatious use with voice below */
 	.name = "WM8753",
@@ -568,90 +433,49 @@
 	.name = "Bluetooth",
 	.stream_name = "Voice",
 	.platform_name = "samsung-audio",
-	.cpu_dai_name = "bluetooth-dai",
+	.cpu_dai_name = "dfbmcs320-pcm",
 	.codec_dai_name = "wm8753-voice",
 	.codec_name = "wm8753-codec.0-001a",
 	.ops = &neo1973_voice_ops,
 },
 };
 
+static struct snd_soc_aux_dev neo1973_aux_devs[] = {
+	{
+		.name = "dfbmcs320",
+		.codec_name = "dfbmcs320.0",
+	},
+	{
+		.name = "lm4857",
+		.codec_name = "lm4857.0-007c",
+		.init = neo1973_lm4857_init,
+	},
+};
+
+static struct snd_soc_codec_conf neo1973_codec_conf[] = {
+	{
+		.dev_name = "lm4857.0-007c",
+		.name_prefix = "Amp",
+	},
+};
+
+#ifdef CONFIG_MACH_NEO1973_GTA02
+static const struct gpio neo1973_gta02_gpios[] = {
+	{ GTA02_GPIO_HP_IN, GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
+	{ GTA02_GPIO_AMP_SHUT, GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
+};
+#else
+static const struct gpio neo1973_gta02_gpios[] = {};
+#endif
+
 static struct snd_soc_card neo1973 = {
 	.name = "neo1973",
 	.dai_link = neo1973_dai,
 	.num_links = ARRAY_SIZE(neo1973_dai),
-};
-
-static int lm4857_i2c_probe(struct i2c_client *client,
-			    const struct i2c_device_id *id)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	i2c = client;
-
-	lm4857_write_regs();
-	return 0;
-}
-
-static int lm4857_i2c_remove(struct i2c_client *client)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	i2c = NULL;
-
-	return 0;
-}
-
-static u8 lm4857_state;
-
-static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	dev_dbg(&dev->dev, "lm4857_suspend\n");
-	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
-	if (lm4857_state) {
-		lm4857_regs[LM4857_CTRL] &= 0xf0;
-		lm4857_write_regs();
-	}
-	return 0;
-}
-
-static int lm4857_resume(struct i2c_client *dev)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	if (lm4857_state) {
-		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
-		lm4857_write_regs();
-	}
-	return 0;
-}
-
-static void lm4857_shutdown(struct i2c_client *dev)
-{
-	pr_debug("Entered %s\n", __func__);
-
-	dev_dbg(&dev->dev, "lm4857_shutdown\n");
-	lm4857_regs[LM4857_CTRL] &= 0xf0;
-	lm4857_write_regs();
-}
-
-static const struct i2c_device_id lm4857_i2c_id[] = {
-	{ "neo1973_lm4857", 0 },
-	{ }
-};
-
-static struct i2c_driver lm4857_i2c_driver = {
-	.driver = {
-		.name = "LM4857 I2C Amp",
-		.owner = THIS_MODULE,
-	},
-	.suspend =        lm4857_suspend,
-	.resume	=         lm4857_resume,
-	.shutdown =       lm4857_shutdown,
-	.probe =          lm4857_i2c_probe,
-	.remove =         lm4857_i2c_remove,
-	.id_table =       lm4857_i2c_id,
+	.aux_dev = neo1973_aux_devs,
+	.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
+	.codec_conf = neo1973_codec_conf,
+	.num_configs = ARRAY_SIZE(neo1973_codec_conf),
 };
 
 static struct platform_device *neo1973_snd_device;
@@ -660,46 +484,56 @@
 {
 	int ret;
 
-	pr_debug("Entered %s\n", __func__);
-
-	if (!machine_is_neo1973_gta01()) {
-		printk(KERN_INFO
-			"Only GTA01 hardware supported by ASoC driver\n");
+	if (!machine_is_neo1973_gta01() && !machine_is_neo1973_gta02())
 		return -ENODEV;
+
+	if (machine_is_neo1973_gta02()) {
+		neo1973.name = "neo1973gta02";
+		neo1973.num_aux_devs = 1;
+
+		ret = gpio_request_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
+		if (ret)
+			return ret;
 	}
 
 	neo1973_snd_device = platform_device_alloc("soc-audio", -1);
-	if (!neo1973_snd_device)
-		return -ENOMEM;
+	if (!neo1973_snd_device) {
+		ret = -ENOMEM;
+		goto err_gpio_free;
+	}
 
 	platform_set_drvdata(neo1973_snd_device, &neo1973);
 	ret = platform_device_add(neo1973_snd_device);
 
-	if (ret) {
-		platform_device_put(neo1973_snd_device);
-		return ret;
+	if (ret)
+		goto err_put_device;
+
+	return 0;
+
+err_put_device:
+	platform_device_put(neo1973_snd_device);
+err_gpio_free:
+	if (machine_is_neo1973_gta02()) {
+		gpio_free_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
 	}
-
-	ret = i2c_add_driver(&lm4857_i2c_driver);
-
-	if (ret != 0)
-		platform_device_unregister(neo1973_snd_device);
-
 	return ret;
 }
+module_init(neo1973_init);
 
 static void __exit neo1973_exit(void)
 {
-	pr_debug("Entered %s\n", __func__);
-
-	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
-}
 
-module_init(neo1973_init);
+	if (machine_is_neo1973_gta02()) {
+		gpio_free_array(neo1973_gta02_gpios,
+				ARRAY_SIZE(neo1973_gta02_gpios));
+	}
+}
 module_exit(neo1973_exit);
 
 /* Module information */
 MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
-MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c
index db66dc4..5d76da4 100644
--- a/sound/soc/soc-cache.c
+++ b/sound/soc/soc-cache.c
@@ -1609,24 +1609,23 @@
 		return 0;
 	}
 
+	if (!codec->cache_ops || !codec->cache_ops->sync)
+		return -EINVAL;
+
 	if (codec->cache_ops->name)
 		name = codec->cache_ops->name;
 	else
 		name = "unknown";
 
-	if (codec->cache_ops && codec->cache_ops->sync) {
-		if (codec->cache_ops->name)
-			dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
-				codec->cache_ops->name, codec->name);
-		trace_snd_soc_cache_sync(codec, name, "start");
-		ret = codec->cache_ops->sync(codec);
-		if (!ret)
-			codec->cache_sync = 0;
-		trace_snd_soc_cache_sync(codec, name, "end");
-		return ret;
-	}
-
-	return -EINVAL;
+	if (codec->cache_ops->name)
+		dev_dbg(codec->dev, "Syncing %s cache for %s codec\n",
+			codec->cache_ops->name, codec->name);
+	trace_snd_soc_cache_sync(codec, name, "start");
+	ret = codec->cache_ops->sync(codec);
+	if (!ret)
+		codec->cache_sync = 0;
+	trace_snd_soc_cache_sync(codec, name, "end");
+	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_cache_sync);
 
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 205cbd7..17efacd 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -87,15 +87,56 @@
 	return c;
 }
 
-/* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
+/* fill buf which is 'len' bytes with a formatted
+ * string of the form 'reg: value\n' */
+static int format_register_str(struct snd_soc_codec *codec,
+			       unsigned int reg, char *buf, size_t len)
 {
-	int ret, i, step = 1, count = 0;
+	int wordsize = codec->driver->reg_word_size * 2;
+	int regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
+	int ret;
+	char tmpbuf[len + 1];
+	char regbuf[regsize + 1];
+
+	/* since tmpbuf is allocated on the stack, warn the callers if they
+	 * try to abuse this function */
+	WARN_ON(len > 63);
+
+	/* +2 for ': ' and + 1 for '\n' */
+	if (wordsize + regsize + 2 + 1 != len)
+		return -EINVAL;
+
+	ret = snd_soc_read(codec , reg);
+	if (ret < 0) {
+		memset(regbuf, 'X', regsize);
+		regbuf[regsize] = '\0';
+	} else {
+		snprintf(regbuf, regsize + 1, "%.*x", regsize, ret);
+	}
+
+	/* prepare the buffer */
+	snprintf(tmpbuf, len + 1, "%.*x: %s\n", wordsize, reg, regbuf);
+	/* copy it back to the caller without the '\0' */
+	memcpy(buf, tmpbuf, len);
+
+	return 0;
+}
+
+/* codec register dump */
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf,
+				  size_t count, loff_t pos)
+{
+	int i, step = 1;
 	int wordsize, regsize;
+	int len;
+	size_t total = 0;
+	loff_t p = 0;
 
 	wordsize = codec->driver->reg_word_size * 2;
 	regsize = min_bytes_needed(codec->driver->reg_cache_size) * 2;
 
+	len = wordsize + regsize + 2 + 1;
+
 	if (!codec->driver->reg_cache_size)
 		return 0;
 
@@ -105,51 +146,34 @@
 	for (i = 0; i < codec->driver->reg_cache_size; i += step) {
 		if (codec->readable_register && !codec->readable_register(codec, i))
 			continue;
-
-		count += sprintf(buf + count, "%.*x: ", regsize, i);
-		if (count >= PAGE_SIZE - 1)
-			break;
-
 		if (codec->driver->display_register) {
 			count += codec->driver->display_register(codec, buf + count,
 							 PAGE_SIZE - count, i);
 		} else {
-			/* If the read fails it's almost certainly due to
-			 * the register being volatile and the device being
-			 * powered off.
-			 */
-			ret = snd_soc_read(codec, i);
-			if (ret >= 0)
-				count += snprintf(buf + count,
-						  PAGE_SIZE - count,
-						  "%.*x", wordsize, ret);
-			else
-				count += snprintf(buf + count,
-						  PAGE_SIZE - count,
-						  "<no data: %d>", ret);
+			/* only support larger than PAGE_SIZE bytes debugfs
+			 * entries for the default case */
+			if (p >= pos) {
+				if (total + len >= count - 1)
+					break;
+				format_register_str(codec, i, buf + total, len);
+				total += len;
+			}
+			p += len;
 		}
-
-		if (count >= PAGE_SIZE - 1)
-			break;
-
-		count += snprintf(buf + count, PAGE_SIZE - count, "\n");
-		if (count >= PAGE_SIZE - 1)
-			break;
 	}
 
-	/* Truncate count; min() would cause a warning */
-	if (count >= PAGE_SIZE)
-		count = PAGE_SIZE - 1;
+	total = min(total, count - 1);
 
-	return count;
+	return total;
 }
+
 static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_pcm_runtime *rtd =
 			container_of(dev, struct snd_soc_pcm_runtime, dev);
 
-	return soc_codec_reg_show(rtd->codec, buf);
+	return soc_codec_reg_show(rtd->codec, buf, PAGE_SIZE, 0);
 }
 
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -188,16 +212,28 @@
 }
 
 static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
-			       size_t count, loff_t *ppos)
+				   size_t count, loff_t *ppos)
 {
 	ssize_t ret;
 	struct snd_soc_codec *codec = file->private_data;
-	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
+	char *buf;
+
+	if (*ppos < 0 || !count)
+		return -EINVAL;
+
+	buf = kmalloc(count, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
-	ret = soc_codec_reg_show(codec, buf);
-	if (ret >= 0)
-		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
+
+	ret = soc_codec_reg_show(codec, buf, count, *ppos);
+	if (ret >= 0) {
+		if (copy_to_user(user_buf, buf, ret)) {
+			kfree(buf);
+			return -EFAULT;
+		}
+		*ppos += ret;
+	}
+
 	kfree(buf);
 	return ret;
 }
@@ -464,20 +500,30 @@
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
 	int ret;
 
-	if (codec_dai->driver->symmetric_rates || cpu_dai->driver->symmetric_rates ||
-			rtd->dai_link->symmetric_rates) {
-		dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n",
-				rtd->rate);
+	if (!codec_dai->driver->symmetric_rates &&
+	    !cpu_dai->driver->symmetric_rates &&
+	    !rtd->dai_link->symmetric_rates)
+		return 0;
 
-		ret = snd_pcm_hw_constraint_minmax(substream->runtime,
-						   SNDRV_PCM_HW_PARAM_RATE,
-						   rtd->rate,
-						   rtd->rate);
-		if (ret < 0) {
-			dev_err(&rtd->dev,
-				"Unable to apply rate symmetry constraint: %d\n", ret);
-			return ret;
-		}
+	/* This can happen if multiple streams are starting simultaneously -
+	 * the second can need to get its constraints before the first has
+	 * picked a rate.  Complain and allow the application to carry on.
+	 */
+	if (!rtd->rate) {
+		dev_warn(&rtd->dev,
+			 "Not enforcing symmetric_rates due to race\n");
+		return 0;
+	}
+
+	dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate);
+
+	ret = snd_pcm_hw_constraint_minmax(substream->runtime,
+					   SNDRV_PCM_HW_PARAM_RATE,
+					   rtd->rate, rtd->rate);
+	if (ret < 0) {
+		dev_err(&rtd->dev,
+			"Unable to apply rate symmetry constraint: %d\n", ret);
+		return ret;
 	}
 
 	return 0;
@@ -1428,6 +1474,7 @@
 			   struct snd_soc_codec *codec)
 {
 	int ret = 0;
+	const struct snd_soc_codec_driver *driver = codec->driver;
 
 	codec->card = card;
 	codec->dapm.card = card;
@@ -1436,8 +1483,8 @@
 	if (!try_module_get(codec->dev->driver->owner))
 		return -ENODEV;
 
-	if (codec->driver->probe) {
-		ret = codec->driver->probe(codec);
+	if (driver->probe) {
+		ret = driver->probe(codec);
 		if (ret < 0) {
 			dev_err(codec->dev,
 				"asoc: failed to probe CODEC %s: %d\n",
@@ -1446,6 +1493,13 @@
 		}
 	}
 
+	if (driver->dapm_widgets)
+		snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets,
+					  driver->num_dapm_widgets);
+	if (driver->dapm_routes)
+		snd_soc_dapm_add_routes(&codec->dapm, driver->dapm_routes,
+					driver->num_dapm_routes);
+
 	soc_init_codec_debugfs(codec);
 
 	/* mark codec as probed and add to card codec list */
@@ -1482,6 +1536,7 @@
 		rtd = &card->rtd_aux[num];
 		name = aux_dev->name;
 	}
+	rtd->card = card;
 
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
@@ -1503,7 +1558,6 @@
 
 	/* register the rtd device */
 	rtd->codec = codec;
-	rtd->card = card;
 	rtd->dev.parent = card->dev;
 	rtd->dev.release = rtd_release;
 	rtd->dev.init_name = name;
@@ -1801,7 +1855,12 @@
 	}
 	card->snd_card->dev = card->dev;
 
-#ifdef CONFIG_PM
+	card->dapm.bias_level = SND_SOC_BIAS_OFF;
+	card->dapm.dev = card->dev;
+	card->dapm.card = card;
+	list_add(&card->dapm.list, &card->dapm_list);
+
+#ifdef CONFIG_PM_SLEEP
 	/* deferred resume work */
 	INIT_WORK(&card->deferred_resume_work, soc_resume_deferred);
 #endif
@@ -1831,11 +1890,37 @@
 		}
 	}
 
+	if (card->dapm_widgets)
+		snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets,
+					  card->num_dapm_widgets);
+	if (card->dapm_routes)
+		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
+					card->num_dapm_routes);
+
+#ifdef CONFIG_DEBUG_FS
+	card->dapm.debugfs_dapm = debugfs_create_dir("dapm",
+						     card->debugfs_card_root);
+	if (!card->dapm.debugfs_dapm)
+		printk(KERN_WARNING
+		       "Failed to create card DAPM debugfs directory\n");
+
+	snd_soc_dapm_debugfs_init(&card->dapm);
+#endif
+
 	snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
 		 "%s",  card->name);
 	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->name);
 
+	if (card->late_probe) {
+		ret = card->late_probe(card);
+		if (ret < 0) {
+			dev_err(card->dev, "%s late_probe() failed: %d\n",
+				card->name, ret);
+			goto probe_aux_dev_err;
+		}
+	}
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		printk(KERN_ERR "asoc: failed to register soundcard for %s\n", card->name);
@@ -2259,22 +2344,45 @@
  * @_template: control template
  * @data: control private data
  * @long_name: control long name
+ * @prefix: control name prefix
  *
  * Create a new mixer control from a template control.
  *
  * Returns 0 for success, else error.
  */
 struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
-	void *data, char *long_name)
+				  void *data, char *long_name,
+				  const char *prefix)
 {
 	struct snd_kcontrol_new template;
+	struct snd_kcontrol *kcontrol;
+	char *name = NULL;
+	int name_len;
 
 	memcpy(&template, _template, sizeof(template));
-	if (long_name)
-		template.name = long_name;
 	template.index = 0;
 
-	return snd_ctl_new1(&template, data);
+	if (!long_name)
+		long_name = template.name;
+
+	if (prefix) {
+		name_len = strlen(long_name) + strlen(prefix) + 2;
+		name = kmalloc(name_len, GFP_ATOMIC);
+		if (!name)
+			return NULL;
+
+		snprintf(name, name_len, "%s %s", prefix, long_name);
+
+		template.name = name;
+	} else {
+		template.name = long_name;
+	}
+
+	kcontrol = snd_ctl_new1(&template, data);
+
+	kfree(name);
+
+	return kcontrol;
 }
 EXPORT_SYMBOL_GPL(snd_soc_cnew);
 
@@ -2293,22 +2401,16 @@
 	const struct snd_kcontrol_new *controls, int num_controls)
 {
 	struct snd_card *card = codec->card->snd_card;
-	char prefixed_name[44], *name;
 	int err, i;
 
 	for (i = 0; i < num_controls; i++) {
 		const struct snd_kcontrol_new *control = &controls[i];
-		if (codec->name_prefix) {
-			snprintf(prefixed_name, sizeof(prefixed_name), "%s %s",
-				 codec->name_prefix, control->name);
-			name = prefixed_name;
-		} else {
-			name = control->name;
-		}
-		err = snd_ctl_add(card, snd_soc_cnew(control, codec, name));
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec,
+						     control->name,
+						     codec->name_prefix));
 		if (err < 0) {
 			dev_err(codec->dev, "%s: Failed to add %s: %d\n",
-				codec->name, name, err);
+				codec->name, control->name, err);
 			return err;
 		}
 	}
@@ -2989,12 +3091,34 @@
 {
 	if (dai->driver && dai->driver->ops->set_sysclk)
 		return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir);
+	else if (dai->codec && dai->codec->driver->set_sysclk)
+		return dai->codec->driver->set_sysclk(dai->codec, clk_id,
+						      freq, dir);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 
 /**
+ * snd_soc_codec_set_sysclk - configure CODEC system or master clock.
+ * @codec: CODEC
+ * @clk_id: DAI specific clock ID
+ * @freq: new clock frequency in Hz
+ * @dir: new clock direction - input/output.
+ *
+ * Configures the CODEC master (MCLK) or system (SYSCLK) clocking.
+ */
+int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
+	unsigned int freq, int dir)
+{
+	if (codec->driver->set_sysclk)
+		return codec->driver->set_sysclk(codec, clk_id, freq, dir);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_sysclk);
+
+/**
  * snd_soc_dai_set_clkdiv - configure DAI clock dividers.
  * @dai: DAI
  * @div_id: DAI specific clock divider ID
@@ -3030,11 +3154,35 @@
 	if (dai->driver && dai->driver->ops->set_pll)
 		return dai->driver->ops->set_pll(dai, pll_id, source,
 					 freq_in, freq_out);
+	else if (dai->codec && dai->codec->driver->set_pll)
+		return dai->codec->driver->set_pll(dai->codec, pll_id, source,
+						   freq_in, freq_out);
 	else
 		return -EINVAL;
 }
 EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
 
+/*
+ * snd_soc_codec_set_pll - configure codec PLL.
+ * @codec: CODEC
+ * @pll_id: DAI specific PLL ID
+ * @source: DAI specific source for the PLL
+ * @freq_in: PLL input clock frequency in Hz
+ * @freq_out: requested PLL output clock frequency in Hz
+ *
+ * Configures and enables PLL to generate output clock based on input clock.
+ */
+int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
+			  unsigned int freq_in, unsigned int freq_out)
+{
+	if (codec->driver->set_pll)
+		return codec->driver->set_pll(codec, pll_id, source,
+					      freq_in, freq_out);
+	else
+		return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll);
+
 /**
  * snd_soc_dai_set_fmt - configure DAI hardware audio format.
  * @dai: DAI
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d0342aa..81c4052 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -32,6 +32,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/init.h>
+#include <linux/async.h>
 #include <linux/delay.h>
 #include <linux/pm.h>
 #include <linux/bitops.h>
@@ -125,17 +126,17 @@
 
 /**
  * snd_soc_dapm_set_bias_level - set the bias level for the system
- * @card: audio device
+ * @dapm: DAPM context
  * @level: level to configure
  *
  * Configure the bias (power) levels for the SoC audio device.
  *
  * Returns 0 for success else error.
  */
-static int snd_soc_dapm_set_bias_level(struct snd_soc_card *card,
-				       struct snd_soc_dapm_context *dapm,
+static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm,
 				       enum snd_soc_bias_level level)
 {
+	struct snd_soc_card *card = dapm->card;
 	int ret = 0;
 
 	switch (level) {
@@ -365,9 +366,20 @@
 	struct snd_soc_dapm_widget *w)
 {
 	int i, ret = 0;
-	size_t name_len;
+	size_t name_len, prefix_len;
 	struct snd_soc_dapm_path *path;
-	struct snd_card *card = dapm->codec->card->snd_card;
+	struct snd_card *card = dapm->card->snd_card;
+	const char *prefix;
+
+	if (dapm->codec)
+		prefix = dapm->codec->name_prefix;
+	else
+		prefix = NULL;
+
+	if (prefix)
+		prefix_len = strlen(prefix) + 1;
+	else
+		prefix_len = 0;
 
 	/* add kcontrol */
 	for (i = 0; i < w->num_kcontrols; i++) {
@@ -396,8 +408,15 @@
 
 			switch (w->id) {
 			default:
+				/* The control will get a prefix from
+				 * the control creation process but
+				 * we're also using the same prefix
+				 * for widgets so cut the prefix off
+				 * the front of the widget name.
+				 */
 				snprintf(path->long_name, name_len, "%s %s",
-					 w->name, w->kcontrols[i].name);
+					 w->name + prefix_len,
+					 w->kcontrols[i].name);
 				break;
 			case snd_soc_dapm_mixer_named_ctl:
 				snprintf(path->long_name, name_len, "%s",
@@ -408,7 +427,7 @@
 			path->long_name[name_len - 1] = '\0';
 
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
-				path->long_name);
+						      path->long_name, prefix);
 			ret = snd_ctl_add(card, path->kcontrol);
 			if (ret < 0) {
 				dev_err(dapm->dev,
@@ -429,7 +448,9 @@
 {
 	struct snd_soc_dapm_path *path = NULL;
 	struct snd_kcontrol *kcontrol;
-	struct snd_card *card = dapm->codec->card->snd_card;
+	struct snd_card *card = dapm->card->snd_card;
+	const char *prefix;
+	size_t prefix_len;
 	int ret = 0;
 
 	if (!w->num_kcontrols) {
@@ -437,7 +458,22 @@
 		return -EINVAL;
 	}
 
-	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name);
+	if (dapm->codec)
+		prefix = dapm->codec->name_prefix;
+	else
+		prefix = NULL;
+
+	if (prefix)
+		prefix_len = strlen(prefix) + 1;
+	else
+		prefix_len = 0;
+
+	/* The control will get a prefix from the control creation
+	 * process but we're also using the same prefix for widgets so
+	 * cut the prefix off the front of the widget name.
+	 */
+	kcontrol = snd_soc_cnew(&w->kcontrols[0], w, w->name + prefix_len,
+				prefix);
 	ret = snd_ctl_add(card, kcontrol);
 
 	if (ret < 0)
@@ -479,7 +515,7 @@
  */
 static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget)
 {
-	int level = snd_power_get_state(widget->dapm->codec->card->snd_card);
+	int level = snd_power_get_state(widget->dapm->card->snd_card);
 
 	switch (level) {
 	case SNDRV_CTL_POWER_D3hot:
@@ -712,7 +748,15 @@
 		    !path->connected(path->source, path->sink))
 			continue;
 
-		if (path->sink && path->sink->power_check &&
+		if (!path->sink)
+			continue;
+
+		if (path->sink->force) {
+			power = 1;
+			break;
+		}
+
+		if (path->sink->power_check &&
 		    path->sink->power_check(path->sink)) {
 			power = 1;
 			break;
@@ -963,7 +1007,7 @@
 	}
 
 	if (!list_empty(&pending))
-		dapm_seq_run_coalesced(dapm, &pending);
+		dapm_seq_run_coalesced(cur_dapm, &pending);
 
 	if (cur_dapm && cur_dapm->seq_notifier) {
 		for (i = 0; i < ARRAY_SIZE(dapm_up_seq); i++)
@@ -1006,7 +1050,62 @@
 	}
 }
 
+/* Async callback run prior to DAPM sequences - brings to _PREPARE if
+ * they're changing state.
+ */
+static void dapm_pre_sequence_async(void *data, async_cookie_t cookie)
+{
+	struct snd_soc_dapm_context *d = data;
+	int ret;
 
+	if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+		if (ret != 0)
+			dev_err(d->dev,
+				"Failed to turn on bias: %d\n", ret);
+	}
+
+	/* If we're changing to all on or all off then prepare */
+	if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
+	    (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_PREPARE);
+		if (ret != 0)
+			dev_err(d->dev,
+				"Failed to prepare bias: %d\n", ret);
+	}
+}
+
+/* Async callback run prior to DAPM sequences - brings to their final
+ * state.
+ */
+static void dapm_post_sequence_async(void *data, async_cookie_t cookie)
+{
+	struct snd_soc_dapm_context *d = data;
+	int ret;
+
+	/* If we just powered the last thing off drop to standby bias */
+	if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_STANDBY);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to apply standby bias: %d\n",
+				ret);
+	}
+
+	/* If we're in standby and can support bias off then do that */
+	if (d->bias_level == SND_SOC_BIAS_STANDBY && d->idle_bias_off) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_OFF);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to turn off bias: %d\n", ret);
+	}
+
+	/* If we just powered up then move to active bias */
+	if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
+		ret = snd_soc_dapm_set_bias_level(d, SND_SOC_BIAS_ON);
+		if (ret != 0)
+			dev_err(d->dev, "Failed to apply active bias: %d\n",
+				ret);
+	}
+}
 
 /*
  * Scan each dapm widget for complete audio path.
@@ -1019,12 +1118,12 @@
  */
 static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 {
-	struct snd_soc_card *card = dapm->codec->card;
+	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	struct snd_soc_dapm_context *d;
 	LIST_HEAD(up_list);
 	LIST_HEAD(down_list);
-	int ret = 0;
+	LIST_HEAD(async_domain);
 	int power;
 
 	trace_snd_soc_dapm_start(card);
@@ -1102,25 +1201,11 @@
 		}
 	}
 
-	list_for_each_entry(d, &dapm->card->dapm_list, list) {
-		if (d->dev_power && d->bias_level == SND_SOC_BIAS_OFF) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_STANDBY);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to turn on bias: %d\n", ret);
-		}
-
-		/* If we're changing to all on or all off then prepare */
-		if ((d->dev_power && d->bias_level == SND_SOC_BIAS_STANDBY) ||
-		    (!d->dev_power && d->bias_level == SND_SOC_BIAS_ON)) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_PREPARE);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to prepare bias: %d\n", ret);
-		}
-	}
+	/* Run all the bias changes in parallel */
+	list_for_each_entry(d, &dapm->card->dapm_list, list)
+		async_schedule_domain(dapm_pre_sequence_async, d,
+					&async_domain);
+	async_synchronize_full_domain(&async_domain);
 
 	/* Power down widgets first; try to avoid amplifying pops. */
 	dapm_seq_run(dapm, &down_list, event, false);
@@ -1130,37 +1215,11 @@
 	/* Now power up. */
 	dapm_seq_run(dapm, &up_list, event, true);
 
-	list_for_each_entry(d, &dapm->card->dapm_list, list) {
-		/* If we just powered the last thing off drop to standby bias */
-		if (d->bias_level == SND_SOC_BIAS_PREPARE && !d->dev_power) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_STANDBY);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to apply standby bias: %d\n",
-					ret);
-		}
-
-		/* If we're in standby and can support bias off then do that */
-		if (d->bias_level == SND_SOC_BIAS_STANDBY &&
-		    d->idle_bias_off) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_OFF);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to turn off bias: %d\n", ret);
-		}
-
-		/* If we just powered up then move to active bias */
-		if (d->bias_level == SND_SOC_BIAS_PREPARE && d->dev_power) {
-			ret = snd_soc_dapm_set_bias_level(card, d,
-							  SND_SOC_BIAS_ON);
-			if (ret != 0)
-				dev_err(d->dev,
-					"Failed to apply active bias: %d\n",
-					ret);
-		}
-	}
+	/* Run all the bias changes in parallel */
+	list_for_each_entry(d, &dapm->card->dapm_list, list)
+		async_schedule_domain(dapm_post_sequence_async, d,
+					&async_domain);
+	async_synchronize_full_domain(&async_domain);
 
 	pop_dbg(dapm->dev, card->pop_time,
 		"DAPM sequencing finished, waiting %dms\n", card->pop_time);
@@ -1218,7 +1277,7 @@
 
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
-					" in  %s %s\n",
+					" in  \"%s\" \"%s\"\n",
 					p->name ? p->name : "static",
 					p->source->name);
 	}
@@ -1228,7 +1287,7 @@
 
 		if (p->connect)
 			ret += snprintf(buf + ret, PAGE_SIZE - ret,
-					" out %s %s\n",
+					" out \"%s\" \"%s\"\n",
 					p->name ? p->name : "static",
 					p->sink->name);
 	}
@@ -1493,7 +1552,7 @@
 	char prefixed_source[80];
 	int ret = 0;
 
-	if (dapm->codec->name_prefix) {
+	if (dapm->codec && dapm->codec->name_prefix) {
 		snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s",
 			 dapm->codec->name_prefix, route->sink);
 		sink = prefixed_sink;
@@ -1664,6 +1723,7 @@
 int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
 {
 	struct snd_soc_dapm_widget *w;
+	unsigned int val;
 
 	list_for_each_entry(w, &dapm->card->widgets, list)
 	{
@@ -1712,6 +1772,18 @@
 		case snd_soc_dapm_post:
 			break;
 		}
+
+		/* Read the initial power state from the device */
+		if (w->reg >= 0) {
+			val = snd_soc_read(w->codec, w->reg);
+			val &= 1 << w->shift;
+			if (w->invert)
+				val = !val;
+
+			if (val)
+				w->power = 1;
+		}
+
 		w->new = 1;
 	}
 
@@ -2130,14 +2202,14 @@
 		return -ENOMEM;
 
 	name_len = strlen(widget->name) + 1;
-	if (dapm->codec->name_prefix)
+	if (dapm->codec && dapm->codec->name_prefix)
 		name_len += 1 + strlen(dapm->codec->name_prefix);
 	w->name = kmalloc(name_len, GFP_KERNEL);
 	if (w->name == NULL) {
 		kfree(w);
 		return -ENOMEM;
 	}
-	if (dapm->codec->name_prefix)
+	if (dapm->codec && dapm->codec->name_prefix)
 		snprintf(w->name, name_len, "%s %s",
 			dapm->codec->name_prefix, widget->name);
 	else
@@ -2242,7 +2314,6 @@
 	mutex_unlock(&codec->mutex);
 	return 0;
 }
-EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 
 /**
  * snd_soc_dapm_enable_pin - enable pin.
@@ -2419,9 +2490,9 @@
 	 * standby.
 	 */
 	if (powerdown) {
-		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_PREPARE);
+		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_PREPARE);
 		dapm_seq_run(dapm, &down_list, 0, false);
-		snd_soc_dapm_set_bias_level(NULL, dapm, SND_SOC_BIAS_STANDBY);
+		snd_soc_dapm_set_bias_level(dapm, SND_SOC_BIAS_STANDBY);
 	}
 }
 
@@ -2434,7 +2505,7 @@
 
 	list_for_each_entry(codec, &card->codec_dev_list, list) {
 		soc_dapm_shutdown_codec(&codec->dapm);
-		snd_soc_dapm_set_bias_level(card, &codec->dapm, SND_SOC_BIAS_OFF);
+		snd_soc_dapm_set_bias_level(&codec->dapm, SND_SOC_BIAS_OFF);
 	}
 }
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index ac5a5bc..fcab80b 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -37,6 +37,7 @@
 {
 	jack->codec = codec;
 	INIT_LIST_HEAD(&jack->pins);
+	INIT_LIST_HEAD(&jack->jack_zones);
 	BLOCKING_INIT_NOTIFIER_HEAD(&jack->notifier);
 
 	return snd_jack_new(codec->card->snd_card, id, type, &jack->jack);
@@ -100,7 +101,7 @@
 	}
 
 	/* Report before the DAPM sync to help users updating micbias status */
-	blocking_notifier_call_chain(&jack->notifier, status, NULL);
+	blocking_notifier_call_chain(&jack->notifier, status, jack);
 
 	snd_soc_dapm_sync(dapm);
 
@@ -112,6 +113,51 @@
 EXPORT_SYMBOL_GPL(snd_soc_jack_report);
 
 /**
+ * snd_soc_jack_add_zones - Associate voltage zones with jack
+ *
+ * @jack:  ASoC jack
+ * @count: Number of zones
+ * @zone:  Array of zones
+ *
+ * After this function has been called the zones specified in the
+ * array will be associated with the jack.
+ */
+int snd_soc_jack_add_zones(struct snd_soc_jack *jack, int count,
+			  struct snd_soc_jack_zone *zones)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		INIT_LIST_HEAD(&zones[i].list);
+		list_add(&(zones[i].list), &jack->jack_zones);
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_zones);
+
+/**
+ * snd_soc_jack_get_type - Based on the mic bias value, this function returns
+ * the type of jack from the zones delcared in the jack type
+ *
+ * @micbias_voltage:  mic bias voltage at adc channel when jack is plugged in
+ *
+ * Based on the mic bias value passed, this function helps identify
+ * the type of jack from the already delcared jack zones
+ */
+int snd_soc_jack_get_type(struct snd_soc_jack *jack, int micbias_voltage)
+{
+	struct snd_soc_jack_zone *zone;
+
+	list_for_each_entry(zone, &jack->jack_zones, list) {
+		if (micbias_voltage >= zone->min_mv &&
+			micbias_voltage < zone->max_mv)
+				return zone->jack_type;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_get_type);
+
+/**
  * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
  *
  * @jack:  ASoC jack
@@ -194,7 +240,7 @@
 	int enable;
 	int report;
 
-	enable = gpio_get_value(gpio->gpio);
+	enable = gpio_get_value_cansleep(gpio->gpio);
 	if (gpio->invert)
 		enable = !enable;
 
@@ -284,6 +330,14 @@
 		if (ret)
 			goto err;
 
+		if (gpios[i].wake) {
+			ret = set_irq_wake(gpio_to_irq(gpios[i].gpio), 1);
+			if (ret != 0)
+				printk(KERN_ERR
+				  "Failed to mark GPIO %d as wake source: %d\n",
+					gpios[i].gpio, ret);
+		}
+
 #ifdef CONFIG_GPIO_SYSFS
 		/* Expose GPIO value over sysfs for diagnostic purposes */
 		gpio_export(gpios[i].gpio, false);
diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile
index dfd2ab9..fd183d3 100644
--- a/sound/soc/tegra/Makefile
+++ b/sound/soc/tegra/Makefile
@@ -2,13 +2,14 @@
 snd-soc-tegra-das-objs := tegra_das.o
 snd-soc-tegra-pcm-objs := tegra_pcm.o
 snd-soc-tegra-i2s-objs := tegra_i2s.o
+snd-soc-tegra-utils-objs += tegra_asoc_utils.o
 
+obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-utils.o
 obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-das.o
 obj-$(CONFIG_SND_TEGRA_SOC) += snd-soc-tegra-pcm.o
 obj-$(CONFIG_SND_TEGRA_SOC_I2S) += snd-soc-tegra-i2s.o
 
 # Tegra machine Support
 snd-soc-tegra-harmony-objs := harmony.o
-snd-soc-tegra-harmony-objs += tegra_asoc_utils.o
 
 obj-$(CONFIG_SND_TEGRA_SOC_HARMONY) += snd-soc-tegra-harmony.o
diff --git a/sound/soc/tegra/harmony.c b/sound/soc/tegra/harmony.c
index 11e2cb8..8585957 100644
--- a/sound/soc/tegra/harmony.c
+++ b/sound/soc/tegra/harmony.c
@@ -38,10 +38,13 @@
 #include <mach/harmony_audio.h>
 
 #include <sound/core.h>
+#include <sound/jack.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 
+#include "../codecs/wm8903.h"
+
 #include "tegra_das.h"
 #include "tegra_i2s.h"
 #include "tegra_pcm.h"
@@ -49,10 +52,14 @@
 
 #define DRV_NAME "tegra-snd-harmony"
 
+#define GPIO_SPKR_EN    BIT(0)
+#define GPIO_INT_MIC_EN BIT(1)
+#define GPIO_EXT_MIC_EN BIT(2)
+
 struct tegra_harmony {
 	struct tegra_asoc_utils_data util_data;
 	struct harmony_audio_platform_data *pdata;
-	int gpio_spkr_en_requested;
+	int gpio_requested;
 };
 
 static int harmony_asoc_hw_params(struct snd_pcm_substream *substream,
@@ -123,6 +130,33 @@
 	.hw_params = harmony_asoc_hw_params,
 };
 
+static struct snd_soc_jack harmony_hp_jack;
+
+static struct snd_soc_jack_pin harmony_hp_jack_pins[] = {
+	{
+		.pin = "Headphone Jack",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+static struct snd_soc_jack_gpio harmony_hp_jack_gpios[] = {
+	{
+		.name = "headphone detect",
+		.report = SND_JACK_HEADPHONE,
+		.debounce_time = 150,
+		.invert = 1,
+	}
+};
+
+static struct snd_soc_jack harmony_mic_jack;
+
+static struct snd_soc_jack_pin harmony_mic_jack_pins[] = {
+	{
+		.pin = "Mic Jack",
+		.mask = SND_JACK_MICROPHONE,
+	},
+};
+
 static int harmony_event_int_spk(struct snd_soc_dapm_widget *w,
 					struct snd_kcontrol *k, int event)
 {
@@ -154,6 +188,10 @@
 	{"IN1L", NULL, "Mic Bias"},
 };
 
+static const struct snd_kcontrol_new harmony_controls[] = {
+	SOC_DAPM_PIN_SWITCH("Int Spk"),
+};
+
 static int harmony_asoc_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_codec *codec = rtd->codec;
@@ -168,19 +206,65 @@
 		dev_err(card->dev, "cannot get spkr_en gpio\n");
 		return ret;
 	}
-	harmony->gpio_spkr_en_requested = 1;
+	harmony->gpio_requested |= GPIO_SPKR_EN;
 
 	gpio_direction_output(pdata->gpio_spkr_en, 0);
 
+	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;
+	}
+	harmony->gpio_requested |= GPIO_INT_MIC_EN;
+
+	/* Disable int mic; enable signal is active-high */
+	gpio_direction_output(pdata->gpio_int_mic_en, 0);
+
+	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;
+	}
+	harmony->gpio_requested |= GPIO_EXT_MIC_EN;
+
+	/* Enable ext mic; enable signal is active-low */
+	gpio_direction_output(pdata->gpio_ext_mic_en, 0);
+
+	ret = snd_soc_add_controls(codec, harmony_controls,
+				   ARRAY_SIZE(harmony_controls));
+	if (ret < 0)
+		return ret;
+
 	snd_soc_dapm_new_controls(dapm, harmony_dapm_widgets,
 					ARRAY_SIZE(harmony_dapm_widgets));
 
 	snd_soc_dapm_add_routes(dapm, harmony_audio_map,
 				ARRAY_SIZE(harmony_audio_map));
 
-	snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
-	snd_soc_dapm_enable_pin(dapm, "Int Spk");
-	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
+	harmony_hp_jack_gpios[0].gpio = pdata->gpio_hp_det;
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
+			 &harmony_hp_jack);
+	snd_soc_jack_add_pins(&harmony_hp_jack,
+			      ARRAY_SIZE(harmony_hp_jack_pins),
+			      harmony_hp_jack_pins);
+	snd_soc_jack_add_gpios(&harmony_hp_jack,
+			       ARRAY_SIZE(harmony_hp_jack_gpios),
+			       harmony_hp_jack_gpios);
+
+	snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE,
+			 &harmony_mic_jack);
+	snd_soc_jack_add_pins(&harmony_mic_jack,
+			      ARRAY_SIZE(harmony_mic_jack_pins),
+			      harmony_mic_jack_pins);
+	wm8903_mic_detect(codec, &harmony_mic_jack, SND_JACK_MICROPHONE, 0);
+
+	snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+
+	snd_soc_dapm_nc_pin(dapm, "IN3L");
+	snd_soc_dapm_nc_pin(dapm, "IN3R");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUTL");
+	snd_soc_dapm_nc_pin(dapm, "LINEOUTR");
+
 	snd_soc_dapm_sync(dapm);
 
 	return 0;
@@ -189,7 +273,7 @@
 static struct snd_soc_dai_link harmony_wm8903_dai = {
 	.name = "WM8903",
 	.stream_name = "WM8903 PCM",
-	.codec_name = "wm8903-codec.0-001a",
+	.codec_name = "wm8903.0-001a",
 	.platform_name = "tegra-pcm-audio",
 	.cpu_dai_name = "tegra-i2s.0",
 	.codec_dai_name = "wm8903-hifi",
@@ -270,7 +354,11 @@
 
 	tegra_asoc_utils_fini(&harmony->util_data);
 
-	if (harmony->gpio_spkr_en_requested)
+	if (harmony->gpio_requested & GPIO_EXT_MIC_EN)
+		gpio_free(pdata->gpio_ext_mic_en);
+	if (harmony->gpio_requested & GPIO_INT_MIC_EN)
+		gpio_free(pdata->gpio_int_mic_en);
+	if (harmony->gpio_requested & GPIO_SPKR_EN)
 		gpio_free(pdata->gpio_spkr_en);
 
 	kfree(harmony);
@@ -302,3 +390,4 @@
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Harmony machine ASoC driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_asoc_utils.c b/sound/soc/tegra/tegra_asoc_utils.c
index cb4fc13..52f0a3f 100644
--- a/sound/soc/tegra/tegra_asoc_utils.c
+++ b/sound/soc/tegra/tegra_asoc_utils.c
@@ -101,6 +101,7 @@
 
 	return 0;
 }
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_set_rate);
 
 int tegra_asoc_utils_init(struct tegra_asoc_utils_data *data,
 			  struct device *dev)
@@ -139,6 +140,7 @@
 err:
 	return ret;
 }
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_init);
 
 void tegra_asoc_utils_fini(struct tegra_asoc_utils_data *data)
 {
@@ -146,4 +148,8 @@
 	clk_put(data->clk_pll_a_out0);
 	clk_put(data->clk_pll_a);
 }
+EXPORT_SYMBOL_GPL(tegra_asoc_utils_fini);
 
+MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
+MODULE_DESCRIPTION("Tegra ASoC utility code");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/tegra/tegra_das.c b/sound/soc/tegra/tegra_das.c
index 01eb9c9..9f24ef7 100644
--- a/sound/soc/tegra/tegra_das.c
+++ b/sound/soc/tegra/tegra_das.c
@@ -262,3 +262,4 @@
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra DAS driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 870ee36..4f5e2c9 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -500,3 +500,4 @@
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra I2S ASoC driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c
index 663ea9f..3c271f9 100644
--- a/sound/soc/tegra/tegra_pcm.c
+++ b/sound/soc/tegra/tegra_pcm.c
@@ -39,6 +39,8 @@
 
 #include "tegra_pcm.h"
 
+#define DRV_NAME "tegra-pcm-audio"
+
 static const struct snd_pcm_hardware tegra_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
@@ -159,8 +161,8 @@
 	prtd->dma_req[1].dev = prtd;
 
 	prtd->dma_chan = tegra_dma_allocate_channel(TEGRA_DMA_MODE_ONESHOT);
-	if (IS_ERR(prtd->dma_chan)) {
-		ret = PTR_ERR(prtd->dma_chan);
+	if (prtd->dma_chan == NULL) {
+		ret = -ENOMEM;
 		goto err;
 	}
 
@@ -377,7 +379,7 @@
 
 static struct platform_driver tegra_pcm_driver = {
 	.driver = {
-		.name = "tegra-pcm-audio",
+		.name = DRV_NAME,
 		.owner = THIS_MODULE,
 	},
 	.probe = tegra_pcm_platform_probe,
@@ -399,3 +401,4 @@
 MODULE_AUTHOR("Stephen Warren <swarren@nvidia.com>");
 MODULE_DESCRIPTION("Tegra PCM ASoC driver");
 MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);