Merge branch 'topic/asoc' into for-linus

MAINTAINERS

@@ -530,6 +530,7 @@ S:	Maintained
 F:	drivers/infiniband/hw/amso1100/
 
 ANALOG DEVICES INC ASOC CODEC DRIVERS
+M:	Lars-Peter Clausen <lars@metafoo.de>
 L:	device-drivers-devel@blackfin.uclinux.org
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 W:	http://wiki.analog.com/
@@ -7199,6 +7200,7 @@ T:	git git://opensource.wolfsonmicro.com/linux-2.6-audioplus
 W:	http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices
 S:	Supported
 F:	Documentation/hwmon/wm83??
+F:	arch/arm/mach-s3c64xx/mach-crag6410*
 F:	drivers/leds/leds-wm83*.c
 F:	drivers/input/misc/wm831x-on.c
 F:	drivers/input/touchscreen/wm831x-ts.c

arch/arm/mach-omap2/board-rx51-peripherals.c

@@ -900,7 +900,6 @@ static struct twl4030_platform_data rx51_twldata __initdata = {
 };
 
 static struct tpa6130a2_platform_data rx51_tpa6130a2_data __initdata_or_module = {
-	.id			= TPA6130A2,
 	.power_gpio		= 98,
 };
 

arch/arm/mach-omap2/devices.c

@@ -330,6 +330,38 @@ static void omap_init_audio(void)
 static inline void omap_init_audio(void) {}
 #endif
 
+#if defined(CONFIG_SND_OMAP_SOC_MCPDM) || \
+		defined(CONFIG_SND_OMAP_SOC_MCPDM_MODULE)
+
+static struct omap_device_pm_latency omap_mcpdm_latency[] = {
+	{
+		.deactivate_func = omap_device_idle_hwmods,
+		.activate_func = omap_device_enable_hwmods,
+		.flags = OMAP_DEVICE_LATENCY_AUTO_ADJUST,
+	},
+};
+
+static void omap_init_mcpdm(void)
+{
+	struct omap_hwmod *oh;
+	struct omap_device *od;
+
+	oh = omap_hwmod_lookup("mcpdm");
+	if (!oh) {
+		printk(KERN_ERR "Could not look up mcpdm hw_mod\n");
+		return;
+	}
+
+	od = omap_device_build("omap-mcpdm", -1, oh, NULL, 0,
+				omap_mcpdm_latency,
+				ARRAY_SIZE(omap_mcpdm_latency), 0);
+	if (IS_ERR(od))
+		printk(KERN_ERR "Could not build omap_device for omap-mcpdm-dai\n");
+}
+#else
+static inline void omap_init_mcpdm(void) {}
+#endif
+
 #if defined(CONFIG_SPI_OMAP24XX) || defined(CONFIG_SPI_OMAP24XX_MODULE)
 
 #include <plat/mcspi.h>
@@ -683,6 +715,7 @@ static int __init omap2_init_devices(void)
 	 * in alphabetical order so they're easier to sort through.
 	 */
 	omap_init_audio();
+	omap_init_mcpdm();
 	omap_init_camera();
 	omap_init_mbox();
 	omap_init_mcspi();

arch/arm/mach-omap2/omap_hwmod_44xx_data.c

@@ -5430,7 +5430,7 @@ static __initdata struct omap_hwmod *omap44xx_hwmods[] = {
 	&omap44xx_mcbsp4_hwmod,
 
 	/* mcpdm class */
-/*	&omap44xx_mcpdm_hwmod, */
+	&omap44xx_mcpdm_hwmod,
 
 	/* mcspi class */
 	&omap44xx_mcspi1_hwmod,

arch/arm/plat-omap/devices.c

@@ -74,41 +74,6 @@ void omap_mcbsp_register_board_cfg(struct resource *res, int res_count,
 
 /*-------------------------------------------------------------------------*/
 
-#if defined(CONFIG_SND_OMAP_SOC_MCPDM) || \
-		defined(CONFIG_SND_OMAP_SOC_MCPDM_MODULE)
-
-static struct resource mcpdm_resources[] = {
-	{
-		.name		= "mcpdm_mem",
-		.start		= OMAP44XX_MCPDM_BASE,
-		.end		= OMAP44XX_MCPDM_BASE + SZ_4K,
-		.flags		= IORESOURCE_MEM,
-	},
-	{
-		.name		= "mcpdm_irq",
-		.start		= OMAP44XX_IRQ_MCPDM,
-		.end		= OMAP44XX_IRQ_MCPDM,
-		.flags		= IORESOURCE_IRQ,
-	},
-};
-
-static struct platform_device omap_mcpdm_device = {
-	.name		= "omap-mcpdm",
-	.id		= -1,
-	.num_resources	= ARRAY_SIZE(mcpdm_resources),
-	.resource	= mcpdm_resources,
-};
-
-static void omap_init_mcpdm(void)
-{
-	(void) platform_device_register(&omap_mcpdm_device);
-}
-#else
-static inline void omap_init_mcpdm(void) {}
-#endif
-
-/*-------------------------------------------------------------------------*/
-
 #if defined(CONFIG_MMC_OMAP) || defined(CONFIG_MMC_OMAP_MODULE) || \
 	defined(CONFIG_MMC_OMAP_HS) || defined(CONFIG_MMC_OMAP_HS_MODULE)
 
@@ -291,7 +256,6 @@ static int __init omap_init_devices(void)
 	 * in alphabetical order so they're easier to sort through.
 	 */
 	omap_init_rng();
-	omap_init_mcpdm();
 	omap_init_uwire();
 	return 0;
 }

drivers/input/misc/twl6040-vibra.c

@@ -74,12 +74,12 @@ static irqreturn_t twl6040_vib_irq_handler(int irq, void *data)
 	if (status & TWL6040_VIBLOCDET) {
 		dev_warn(info->dev, "Left Vibrator overcurrent detected\n");
 		twl6040_clear_bits(twl6040, TWL6040_REG_VIBCTLL,
-				   TWL6040_VIBENAL);
+				   TWL6040_VIBENA);
 	}
 	if (status & TWL6040_VIBROCDET) {
 		dev_warn(info->dev, "Right Vibrator overcurrent detected\n");
 		twl6040_clear_bits(twl6040, TWL6040_REG_VIBCTLR,
-				   TWL6040_VIBENAR);
+				   TWL6040_VIBENA);
 	}
 
 	return IRQ_HANDLED;
@@ -104,16 +104,16 @@ static void twl6040_vibra_enable(struct vibra_info *info)
 		 * overcurrent detection
 		 */
 		twl6040_reg_write(twl6040, TWL6040_REG_VIBCTLL,
-				  TWL6040_VIBENAL | TWL6040_VIBCTRLL);
+				  TWL6040_VIBENA | TWL6040_VIBCTRL);
 		twl6040_reg_write(twl6040, TWL6040_REG_VIBCTLR,
-				  TWL6040_VIBENAR | TWL6040_VIBCTRLR);
+				  TWL6040_VIBENA | TWL6040_VIBCTRL);
 		usleep_range(3000, 3500);
 	}
 
 	twl6040_reg_write(twl6040, TWL6040_REG_VIBCTLL,
-			  TWL6040_VIBENAL);
+			  TWL6040_VIBENA);
 	twl6040_reg_write(twl6040, TWL6040_REG_VIBCTLR,
-			  TWL6040_VIBENAR);
+			  TWL6040_VIBENA);
 
 	info->enabled = true;
 }
@@ -201,6 +201,13 @@ static int vibra_play(struct input_dev *input, void *data,
 	struct vibra_info *info = input_get_drvdata(input);
 	int ret;
 
+	/* Do not allow effect, while the routing is set to use audio */
+	ret = twl6040_get_vibralr_status(info->twl6040);
+	if (ret & TWL6040_VIBSEL) {
+		dev_info(&input->dev, "Vibra is configured for audio\n");
+		return -EBUSY;
+	}
+
 	info->weak_speed = effect->u.rumble.weak_magnitude;
 	info->strong_speed = effect->u.rumble.strong_magnitude;
 	info->direction = effect->direction < EFFECT_DIR_180_DEG ? 1 : -1;

drivers/mfd/twl6040-core.c

@@ -34,17 +34,25 @@
 #include <linux/mfd/core.h>
 #include <linux/mfd/twl6040.h>
 
+#define VIBRACTRL_MEMBER(reg) ((reg == TWL6040_REG_VIBCTLL) ? 0 : 1)
+
 int twl6040_reg_read(struct twl6040 *twl6040, unsigned int reg)
 {
 	int ret;
 	u8 val = 0;
 
 	mutex_lock(&twl6040->io_mutex);
+	/* Vibra control registers from cache */
+	if (unlikely(reg == TWL6040_REG_VIBCTLL ||
+		     reg == TWL6040_REG_VIBCTLR)) {
+		val = twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)];
+	} else {
 		ret = twl_i2c_read_u8(TWL_MODULE_AUDIO_VOICE, &val, reg);
 		if (ret < 0) {
 			mutex_unlock(&twl6040->io_mutex);
 			return ret;
 		}
+	}
 	mutex_unlock(&twl6040->io_mutex);
 
 	return val;
@@ -57,6 +65,9 @@ int twl6040_reg_write(struct twl6040 *twl6040, unsigned int reg, u8 val)
 
 	mutex_lock(&twl6040->io_mutex);
 	ret = twl_i2c_write_u8(TWL_MODULE_AUDIO_VOICE, val, reg);
+	/* Cache the vibra control registers */
+	if (reg == TWL6040_REG_VIBCTLL || reg == TWL6040_REG_VIBCTLR)
+		twl6040->vibra_ctrl_cache[VIBRACTRL_MEMBER(reg)] = val;
 	mutex_unlock(&twl6040->io_mutex);
 
 	return ret;
@@ -433,6 +444,18 @@ unsigned int twl6040_get_sysclk(struct twl6040 *twl6040)
 }
 EXPORT_SYMBOL(twl6040_get_sysclk);
 
+/* Get the combined status of the vibra control register */
+int twl6040_get_vibralr_status(struct twl6040 *twl6040)
+{
+	u8 status;
+
+	status = twl6040->vibra_ctrl_cache[0] | twl6040->vibra_ctrl_cache[1];
+	status &= (TWL6040_VIBENA | TWL6040_VIBSEL);
+
+	return status;
+}
+EXPORT_SYMBOL(twl6040_get_vibralr_status);
+
 static struct resource twl6040_vibra_rsrc[] = {
 	{
 		.flags = IORESOURCE_IRQ,

drivers/mfd/wm8994-core.c

@@ -243,6 +243,18 @@ static struct mfd_cell wm8994_devs[] = {
  * and should be handled via the standard regulator API supply
  * management.
  */
+static const char *wm1811_main_supplies[] = {
+	"DBVDD1",
+	"DBVDD2",
+	"DBVDD3",
+	"DCVDD",
+	"AVDD1",
+	"AVDD2",
+	"CPVDD",
+	"SPKVDD1",
+	"SPKVDD2",
+};
+
 static const char *wm8994_main_supplies[] = {
 	"DBVDD",
 	"DCVDD",
@@ -401,6 +413,9 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq)
 	}
 
 	switch (wm8994->type) {
+	case WM1811:
+		wm8994->num_supplies = ARRAY_SIZE(wm1811_main_supplies);
+		break;
 	case WM8994:
 		wm8994->num_supplies = ARRAY_SIZE(wm8994_main_supplies);
 		break;
@@ -421,6 +436,10 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq)
 	}
 
 	switch (wm8994->type) {
+	case WM1811:
+		for (i = 0; i < ARRAY_SIZE(wm1811_main_supplies); i++)
+			wm8994->supplies[i].supply = wm1811_main_supplies[i];
+		break;
 	case WM8994:
 		for (i = 0; i < ARRAY_SIZE(wm8994_main_supplies); i++)
 			wm8994->supplies[i].supply = wm8994_main_supplies[i];
@@ -454,6 +473,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq)
 		goto err_enable;
 	}
 	switch (ret) {
+	case 0x1811:
+		devname = "WM1811";
+		if (wm8994->type != WM1811)
+			dev_warn(wm8994->dev, "Device registered as type %d\n",
+				 wm8994->type);
+		wm8994->type = WM1811;
+		break;
 	case 0x8994:
 		devname = "WM8994";
 		if (wm8994->type != WM8994)
@@ -651,6 +677,7 @@ static int wm8994_i2c_remove(struct i2c_client *i2c)
 }
 
 static const struct i2c_device_id wm8994_i2c_id[] = {
+	{ "wm1811", WM1811 },
 	{ "wm8994", WM8994 },
 	{ "wm8958", WM8958 },
 	{ }

drivers/regulator/core.c

@@ -1599,13 +1599,18 @@ static void regulator_disable_work(struct work_struct *work)
 int regulator_disable_deferred(struct regulator *regulator, int ms)
 {
 	struct regulator_dev *rdev = regulator->rdev;
+	int ret;
 
 	mutex_lock(&rdev->mutex);
 	rdev->deferred_disables++;
 	mutex_unlock(&rdev->mutex);
 
-	return schedule_delayed_work(&rdev->disable_work,
+	ret = schedule_delayed_work(&rdev->disable_work,
 				    msecs_to_jiffies(ms));
+	if (ret < 0)
+		return ret;
+	else
+		return 0;
 }
 EXPORT_SYMBOL_GPL(regulator_disable_deferred);
 

drivers/regulator/wm8994-regulator.c

@@ -140,6 +140,14 @@ static int wm8994_ldo2_list_voltage(struct regulator_dev *rdev,
 		return (selector * 100000) + 900000;
 	case WM8958:
 		return (selector * 100000) + 1000000;
+	case WM1811:
+		switch (selector) {
+		case 0:
+			return -EINVAL;
+		default:
+			return (selector * 100000) + 950000;
+		}
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -170,6 +178,11 @@ static int wm8994_ldo2_set_voltage(struct regulator_dev *rdev,
 	case WM8958:
 		selector = (min_uV - 1000000) / 100000;
 		break;
+	case WM1811:
+		selector = (min_uV - 950000) / 100000;
+		if (selector == 0)
+			selector = 1;
+		break;
 	default:
 		return -EINVAL;
 	}

include/linux/mfd/twl6040.h

@@ -68,8 +68,6 @@
 #define TWL6040_REG_ACCCTL		0x2D
 #define TWL6040_REG_STATUS		0x2E
 
-#define TWL6040_CACHEREGNUM		(TWL6040_REG_STATUS + 1)
-
 /* INTID (0x03) fields */
 
 #define TWL6040_THINT			0x01
@@ -122,34 +120,24 @@
 #define TWL6040_LPLLFIN			0x08
 #define TWL6040_HPLLSEL			0x10
 
-/* HSLCTL (0x10) fields */
+/* HSLCTL/R (0x10/0x11) fields */
 
-#define TWL6040_HSDACMODEL		0x02
+#define TWL6040_HSDACENA		(1 << 0)
-#define TWL6040_HSDRVMODEL		0x08
+#define TWL6040_HSDACMODE		(1 << 1)
+#define TWL6040_HSDRVMODE		(1 << 3)
 
-/* HSRCTL (0x11) fields */
+/* VIBCTLL/R (0x18/0x1A) fields */
 
-#define TWL6040_HSDACMODER		0x02
+#define TWL6040_VIBENA			(1 << 0)
-#define TWL6040_HSDRVMODER		0x08
+#define TWL6040_VIBSEL			(1 << 1)
+#define TWL6040_VIBCTRL			(1 << 2)
+#define TWL6040_VIBCTRL_P		(1 << 3)
+#define TWL6040_VIBCTRL_N		(1 << 4)
 
-/* VIBCTLL (0x18) fields */
+/* VIBDATL/R (0x19/0x1B) fields */
-
-#define TWL6040_VIBENAL			0x01
-#define TWL6040_VIBCTRLL		0x04
-#define TWL6040_VIBCTRLLP		0x08
-#define TWL6040_VIBCTRLLN		0x10
-
-/* VIBDATL (0x19) fields */
 
 #define TWL6040_VIBDAT_MAX		0x64
 
-/* VIBCTLR (0x1A) fields */
-
-#define TWL6040_VIBENAR			0x01
-#define TWL6040_VIBCTRLR		0x04
-#define TWL6040_VIBCTRLRP		0x08
-#define TWL6040_VIBCTRLRN		0x10
-
 /* GPOCTL (0x1E) fields */
 
 #define TWL6040_GPO1			0x01
@@ -197,6 +185,7 @@ struct twl6040 {
 	int audpwron;
 	int power_count;
 	int rev;
+	u8 vibra_ctrl_cache[2];
 
 	int pll;
 	unsigned int sysclk;
@@ -221,10 +210,13 @@ int twl6040_get_pll(struct twl6040 *twl6040);
 unsigned int twl6040_get_sysclk(struct twl6040 *twl6040);
 int twl6040_irq_init(struct twl6040 *twl6040);
 void twl6040_irq_exit(struct twl6040 *twl6040);
+/* Get the combined status of the vibra control register */
+int twl6040_get_vibralr_status(struct twl6040 *twl6040);
 
 static inline int twl6040_get_revid(struct twl6040 *twl6040)
 {
 	return twl6040->rev;
 }
 
+
 #endif  /* End of __TWL6040_CODEC_H__ */

include/linux/mfd/wm8994/core.h

@@ -20,6 +20,7 @@
 enum wm8994_type {
 	WM8994 = 0,
 	WM8958 = 1,
+	WM1811 = 2,
 };
 
 struct regulator_dev;

include/linux/mfd/wm8994/registers.h

@@ -2069,6 +2069,10 @@
 /*
  * R96 (0x60) - Analogue HP (1)
  */
+#define WM1811_HPOUT1_ATTN                      0x0100  /* HPOUT1_ATTN */
+#define WM1811_HPOUT1_ATTN_MASK                 0x0100  /* HPOUT1_ATTN */
+#define WM1811_HPOUT1_ATTN_SHIFT                     8  /* HPOUT1_ATTN */
+#define WM1811_HPOUT1_ATTN_WIDTH                     1  /* HPOUT1_ATTN */
 #define WM8994_HPOUT1L_RMV_SHORT                0x0080  /* HPOUT1L_RMV_SHORT */
 #define WM8994_HPOUT1L_RMV_SHORT_MASK           0x0080  /* HPOUT1L_RMV_SHORT */
 #define WM8994_HPOUT1L_RMV_SHORT_SHIFT               7  /* HPOUT1L_RMV_SHORT */

include/sound/soc-dai.h

@@ -24,13 +24,13 @@ struct snd_pcm_substream;
  * Describes the physical PCM data formating and clocking. Add new formats
  * to the end.
  */
-#define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
+#define SND_SOC_DAIFMT_I2S		1 /* I2S mode */
-#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
+#define SND_SOC_DAIFMT_RIGHT_J		2 /* Right Justified mode */
-#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J		3 /* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A		3 /* L data MSB after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_A		4 /* L data MSB after FRM LRC */
-#define SND_SOC_DAIFMT_DSP_B		4 /* L data MSB during FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B		5 /* L data MSB during FRM LRC */
-#define SND_SOC_DAIFMT_AC97		5 /* AC97 */
+#define SND_SOC_DAIFMT_AC97		6 /* AC97 */
-#define SND_SOC_DAIFMT_PDM		6 /* Pulse density modulation */
+#define SND_SOC_DAIFMT_PDM		7 /* Pulse density modulation */
 
 /* left and right justified also known as MSB and LSB respectively */
 #define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
@@ -42,8 +42,8 @@ struct snd_pcm_substream;
  * DAI bit clocks can be be gated (disabled) when the DAI is not
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
-#define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
+#define SND_SOC_DAIFMT_GATED		(2 << 4) /* clock is gated */
 
 /*
  * DAI hardware signal inversions.
@@ -51,10 +51,10 @@ struct snd_pcm_substream;
  * Specifies whether the DAI can also support inverted clocks for the specified
  * format.
  */
-#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_NF		(1 << 8) /* normal bit clock + frame */
-#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
-#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
-#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert BCLK + FRM */
+#define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */
 
 /*
  * DAI hardware clock masters.
@@ -63,10 +63,10 @@ struct snd_pcm_substream;
  * i.e. if the codec is clk and FRM master then the interface is
  * clk and frame slave.
  */
-#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFM		(1 << 12) /* codec clk & FRM master */
-#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk slave & FRM master */
-#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & FRM slave */
+#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM slave */
 
 #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
@@ -242,6 +242,9 @@ struct snd_soc_dai {
 	void *playback_dma_data;
 	void *capture_dma_data;
 
+	/* Symmetry data - only valid if symmetry is being enforced */
+	unsigned int rate;
+
 	/* parent platform/codec */
 	union {
 		struct snd_soc_platform *platform;

include/sound/soc-dapm.h

@@ -381,6 +381,9 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm,
 				const char *pin);
 
+/* Mostly internal - should not normally be used */
+void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
@@ -473,6 +476,8 @@ struct snd_soc_dapm_widget {
 	unsigned char ext:1;			/* has external widgets */
 	unsigned char force:1;			/* force state */
 	unsigned char ignore_suspend:1;         /* kept enabled over suspend */
+	unsigned char new_power:1;		/* power from this run */
+	unsigned char power_checked:1;		/* power checked this run */
 	int subseq;				/* sort within widget type */
 
 	int (*power_check)(struct snd_soc_dapm_widget *w);
@@ -492,6 +497,9 @@ struct snd_soc_dapm_widget {
 
 	/* used during DAPM updates */
 	struct list_head power_list;
+	struct list_head dirty;
+	int inputs;
+	int outputs;
 };
 
 struct snd_soc_dapm_update {
@@ -537,4 +545,10 @@ struct snd_soc_dapm_widget_list {
 	struct snd_soc_dapm_widget *widgets[0];
 };
 
+struct snd_soc_dapm_stats {
+	int power_checks;
+	int path_checks;
+	int neighbour_checks;
+};
+
 #endif

include/sound/soc.h

@@ -28,13 +28,20 @@
 /*
  * Convenience kcontrol builders
  */
-#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
+#define SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
-	{.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
+	{.reg = xreg, .rreg = xreg, .shift = shift_left, \
-	.platform_max = xmax, .invert = xinvert})
+	.rshift = shift_right, .max = xmax, .platform_max = xmax, \
+	.invert = xinvert})
+#define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
+	SOC_DOUBLE_VALUE(xreg, xshift, xshift, xmax, xinvert)
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
 	((unsigned long)&(struct soc_mixer_control) \
 	{.reg = xreg, .max = xmax, .platform_max = xmax, .invert = xinvert})
+#define SOC_DOUBLE_R_VALUE(xlreg, xrreg, xshift, xmax, xinvert) \
+	((unsigned long)&(struct soc_mixer_control) \
+	{.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \
+	.max = xmax, .platform_max = xmax, .invert = xinvert})
 #define SOC_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
@@ -48,40 +55,36 @@
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
 	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
-#define SOC_DOUBLE(xname, xreg, shift_left, shift_right, xmax, xinvert) \
+#define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
 	.put = snd_soc_put_volsw, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
-		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+					  max, invert) }
-		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R(xname, reg_left, reg_right, xshift, xmax, xinvert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
-	.info = snd_soc_info_volsw_2r, \
+	.info = snd_soc_info_volsw, \
-	.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
+	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+					    xmax, xinvert) }
-		.max = xmax, .platform_max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \
-#define SOC_DOUBLE_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
 		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
 	.tlv.p = (tlv_array), \
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
 	.put = snd_soc_put_volsw, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
-		{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
+					  max, invert) }
-		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
 		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
 	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r, \
+	.info = snd_soc_info_volsw, \
-	.get = snd_soc_get_volsw_2r, .put = snd_soc_put_volsw_2r, \
+	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+					    xmax, xinvert) }
-		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
 {	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -121,14 +124,13 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
-#define SOC_DOUBLE_EXT(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+#define SOC_DOUBLE_EXT(xname, reg, shift_left, shift_right, max, invert,\
 	 xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = \
-		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+		SOC_DOUBLE_VALUE(reg, shift_left, shift_right, max, invert) }
-		 .max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_EXT_TLV(xname, xreg, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -146,20 +148,18 @@
 	.tlv.p = (tlv_array), \
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_VALUE(xreg, shift_left, shift_right, \
-		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+					  xmax, xinvert) }
-		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
 	 xhandler_get, xhandler_put, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
 		 SNDRV_CTL_ELEM_ACCESS_READWRITE, \
 	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r, \
+	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
+	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+					    xmax, xinvert) }
-		.max = xmax, .platform_max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_bool_ext, \
@@ -393,12 +393,8 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
-int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol,
+#define snd_soc_get_volsw_2r snd_soc_get_volsw
-	struct snd_ctl_elem_info *uinfo);
+#define snd_soc_put_volsw_2r snd_soc_put_volsw
-int snd_soc_get_volsw_2r(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
-int snd_soc_put_volsw_2r(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
@@ -582,6 +578,7 @@ struct snd_soc_codec {
 
 	/* dapm */
 	struct snd_soc_dapm_context dapm;
+	unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_codec_root;
@@ -713,6 +710,8 @@ struct snd_soc_dai_link {
 	const char *cpu_dai_name;
 	const char *codec_dai_name;
 
+	unsigned int dai_fmt;           /* format to set on init */
+
 	/* Keep DAI active over suspend */
 	unsigned int ignore_suspend:1;
 
@@ -825,9 +824,11 @@ struct snd_soc_card {
 	struct list_head widgets;
 	struct list_head paths;
 	struct list_head dapm_list;
+	struct list_head dapm_dirty;
 
 	/* Generic DAPM context for the card */
 	struct snd_soc_dapm_context dapm;
+	struct snd_soc_dapm_stats dapm_stats;
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_card_root;
@@ -850,8 +851,6 @@ struct snd_soc_pcm_runtime  {
 	unsigned int complete:1;
 	unsigned int dev_registered:1;
 
-	/* Symmetry data - only valid if symmetry is being enforced */
-	unsigned int rate;
 	long pmdown_time;
 
 	/* runtime devices */
@@ -946,6 +945,18 @@ static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
 	INIT_LIST_HEAD(&card->dapm_list);
 }
 
+static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
+{
+	if (mc->reg == mc->rreg && mc->shift == mc->rshift)
+		return 0;
+	/*
+	 * mc->reg == mc->rreg && mc->shift != mc->rshift, or
+	 * mc->reg != mc->rreg means that the control is
+	 * stereo (bits in one register or in two registers)
+	 */
+	return 1;
+}
+
 int snd_soc_util_init(void);
 void snd_soc_util_exit(void);
 

include/sound/tpa6130a2-plat.h

@@ -23,13 +23,7 @@
 #ifndef TPA6130A2_PLAT_H
 #define TPA6130A2_PLAT_H
 
-enum tpa_model {
-	TPA6130A2,
-	TPA6140A2,
-};
-
 struct tpa6130a2_platform_data {
-	enum tpa_model id;
 	int power_gpio;
 };
 

include/sound/wm1250-ev1.h

@@ -0,0 +1,27 @@
+/*
+ * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1
+ *
+ * Copyright 2011 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM1250_EV1_H
+#define __LINUX_SND_WM1250_EV1_H
+
+#define WM1250_EV1_NUM_GPIOS 5
+
+#define WM1250_EV1_GPIO_CLK_ENA  0
+#define WM1250_EV1_GPIO_CLK_SEL0 1
+#define WM1250_EV1_GPIO_CLK_SEL1 2
+#define WM1250_EV1_GPIO_OSR      3
+#define WM1250_EV1_GPIO_MASTER   4
+
+
+struct wm1250_ev1_pdata {
+	int gpios[WM1250_EV1_NUM_GPIOS];
+};
+
+#endif

include/sound/wm5100.h

@@ -0,0 +1,59 @@
+/*
+ * linux/sound/wm5100.h -- Platform data for WM5100
+ *
+ * Copyright 2011 Wolfson Microelectronics. PLC.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __LINUX_SND_WM5100_H
+#define __LINUX_SND_WM5100_H
+
+enum wm5100_in_mode {
+	WM5100_IN_SE = 0,
+	WM5100_IN_DIFF = 1,
+	WM5100_IN_DMIC = 2,
+};
+
+enum wm5100_dmic_sup {
+	WM5100_DMIC_SUP_MICVDD = 0,
+	WM5100_DMIC_SUP_MICBIAS1 = 1,
+	WM5100_DMIC_SUP_MICBIAS2 = 2,
+	WM5100_DMIC_SUP_MICBIAS3 = 3,
+};
+
+enum wm5100_micdet_bias {
+	WM5100_MICDET_MICBIAS1 = 0,
+	WM5100_MICDET_MICBIAS2 = 1,
+	WM5100_MICDET_MICBIAS3 = 2,
+};
+
+struct wm5100_jack_mode {
+	enum wm5100_micdet_bias bias;
+	int hp_pol;
+	int micd_src;
+};
+
+#define WM5100_GPIO_SET 0x10000
+
+struct wm5100_pdata {
+	int reset;      /** GPIO controlling /RESET, if any */
+	int ldo_ena;    /** GPIO controlling LODENA, if any */
+	int hp_pol;     /** GPIO controlling headset polarity, if any */
+	int irq_flags;
+	int gpio_base;
+
+	struct wm5100_jack_mode jack_modes[2];
+
+	/* Input pin mode selection */
+	enum wm5100_in_mode in_mode[4];
+
+	/* DMIC supply selection */
+	enum wm5100_dmic_sup dmic_sup[4];
+
+	int gpio_defaults[6];
+};
+
+#endif

include/trace/events/asoc.h

@@ -216,6 +216,31 @@ DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_done,
 
 );
 
+TRACE_EVENT(snd_soc_dapm_walk_done,
+
+	TP_PROTO(struct snd_soc_card *card),
+
+	TP_ARGS(card),
+
+	TP_STRUCT__entry(
+		__string(	name,	card->name		)
+		__field(	int,	power_checks		)
+		__field(	int,	path_checks		)
+		__field(	int,	neighbour_checks	)
+	),
+
+	TP_fast_assign(
+		__assign_str(name, card->name);
+		__entry->power_checks = card->dapm_stats.power_checks;
+		__entry->path_checks = card->dapm_stats.path_checks;
+		__entry->neighbour_checks = card->dapm_stats.neighbour_checks;
+	),
+
+	TP_printk("%s: checks %d power, %d path, %d neighbour",
+		  __get_str(name), (int)__entry->power_checks,
+		  (int)__entry->path_checks, (int)__entry->neighbour_checks)
+);
+
 TRACE_EVENT(snd_soc_jack_irq,
 
 	TP_PROTO(const char *name),

sound/soc/atmel/playpaq_wm8510.c

@@ -338,7 +338,6 @@ static int playpaq_wm8510_init(struct snd_soc_pcm_runtime *rtd)
 	/* always connected pins */
 	snd_soc_dapm_enable_pin(dapm, "Int Mic");
 	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
-	snd_soc_dapm_sync(dapm);
 
 
 

sound/soc/atmel/sam9g20_wm8731.c

@@ -173,8 +173,6 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 	/* always connected */
 	snd_soc_dapm_enable_pin(dapm, "Ext Spk");
 
-	snd_soc_dapm_sync(dapm);
-
 	return 0;
 }
 

sound/soc/atmel/snd-soc-afeb9260.c

@@ -117,8 +117,6 @@ static int afeb9260_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd)
 	snd_soc_dapm_enable_pin(dapm, "Line In");
 	snd_soc_dapm_enable_pin(dapm, "Mic Jack");
 
-	snd_soc_dapm_sync(dapm);
-
 	return 0;
 }
 

sound/soc/au1x/ac97c.c

@@ -226,7 +226,7 @@ static struct snd_soc_dai_driver au1xac97c_dai_driver = {
 static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
 {
 	int ret;
-	struct resource *r;
+	struct resource *iores, *dmares;
 	struct au1xpsc_audio_data *ctx;
 
 	ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
@@ -235,29 +235,30 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
 
 	mutex_init(&ctx->lock);
 
-	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!r) {
+	if (!iores) {
 		ret = -ENODEV;
 		goto out0;
 	}
 
 	ret = -EBUSY;
-	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+	if (!request_mem_region(iores->start, resource_size(iores),
+				pdev->name))
 		goto out0;
 
-	ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+	ctx->mmio = ioremap_nocache(iores->start, resource_size(iores));
 	if (!ctx->mmio)
 		goto out1;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r)
+	if (!dmares)
-		goto out1;
+		goto out2;
-	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r)
+	if (!dmares)
-		goto out1;
+		goto out2;
-	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
 
 	/* switch it on */
 	WR(ctx, AC97_ENABLE, EN_D | EN_CE);
@@ -270,13 +271,15 @@ static int __devinit au1xac97c_drvprobe(struct platform_device *pdev)
 
 	ret = snd_soc_register_dai(&pdev->dev, &au1xac97c_dai_driver);
 	if (ret)
-		goto out1;
+		goto out2;
 
 	ac97c_workdata = ctx;
 	return 0;
 
+out2:
+	iounmap(ctx->mmio);
 out1:
-	release_mem_region(r->start, resource_size(r));
+	release_mem_region(iores->start, resource_size(iores));
 out0:
 	kfree(ctx);
 	return ret;

sound/soc/au1x/i2sc.c

@@ -228,47 +228,50 @@ static struct snd_soc_dai_driver au1xi2s_dai_driver = {
 static int __devinit au1xi2s_drvprobe(struct platform_device *pdev)
 {
 	int ret;
-	struct resource *r;
+	struct resource *iores, *dmares;
 	struct au1xpsc_audio_data *ctx;
 
 	ctx = kzalloc(sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)
 		return -ENOMEM;
 
-	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!r) {
+	if (!iores) {
 		ret = -ENODEV;
 		goto out0;
 	}
 
 	ret = -EBUSY;
-	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+	if (!request_mem_region(iores->start, resource_size(iores),
+				pdev->name))
 		goto out0;
 
-	ctx->mmio = ioremap_nocache(r->start, resource_size(r));
+	ctx->mmio = ioremap_nocache(iores->start, resource_size(iores));
 	if (!ctx->mmio)
 		goto out1;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r)
+	if (!dmares)
-		goto out1;
+		goto out2;
-	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+	ctx->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r)
+	if (!dmares)
-		goto out1;
+		goto out2;
-	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+	ctx->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
 
 	platform_set_drvdata(pdev, ctx);
 
 	ret = snd_soc_register_dai(&pdev->dev, &au1xi2s_dai_driver);
 	if (ret)
-		goto out1;
+		goto out2;
 
 	return 0;
 
+out2:
+	iounmap(ctx->mmio);
 out1:
-	release_mem_region(r->start, resource_size(r));
+	release_mem_region(iores->start, resource_size(iores));
 out0:
 	kfree(ctx);
 	return ret;

sound/soc/au1x/psc-ac97.c

@@ -364,7 +364,7 @@ static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = {
 static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 {
 	int ret;
-	struct resource *r;
+	struct resource *iores, *dmares;
 	unsigned long sel;
 	struct au1xpsc_audio_data *wd;
 
@@ -374,29 +374,30 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 
 	mutex_init(&wd->lock);
 
-	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!r) {
+	if (!iores) {
 		ret = -ENODEV;
 		goto out0;
 	}
 
 	ret = -EBUSY;
-	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+	if (!request_mem_region(iores->start, resource_size(iores),
+				pdev->name))
 		goto out0;
 
-	wd->mmio = ioremap(r->start, resource_size(r));
+	wd->mmio = ioremap(iores->start, resource_size(iores));
 	if (!wd->mmio)
 		goto out1;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r)
+	if (!dmares)
 		goto out2;
-	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r)
+	if (!dmares)
 		goto out2;
-	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
 
 	/* configuration: max dma trigger threshold, enable ac97 */
 	wd->cfg = PSC_AC97CFG_RT_FIFO8 | PSC_AC97CFG_TT_FIFO8 |
@@ -428,7 +429,7 @@ static int __devinit au1xpsc_ac97_drvprobe(struct platform_device *pdev)
 out2:
 	iounmap(wd->mmio);
 out1:
-	release_mem_region(r->start, resource_size(r));
+	release_mem_region(iores->start, resource_size(iores));
 out0:
 	kfree(wd);
 	return ret;

sound/soc/au1x/psc-i2s.c

@@ -290,7 +290,7 @@ static const struct snd_soc_dai_driver au1xpsc_i2s_dai_template = {
 
 static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 {
-	struct resource *r;
+	struct resource *iores, *dmares;
 	unsigned long sel;
 	int ret;
 	struct au1xpsc_audio_data *wd;
@@ -299,29 +299,30 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 	if (!wd)
 		return -ENOMEM;
 
-	r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	iores = platform_get_resource(pdev, IORESOURCE_MEM, 0);
-	if (!r) {
+	if (!iores) {
 		ret = -ENODEV;
 		goto out0;
 	}
 
 	ret = -EBUSY;
-	if (!request_mem_region(r->start, resource_size(r), pdev->name))
+	if (!request_mem_region(iores->start, resource_size(iores),
+				pdev->name))
 		goto out0;
 
-	wd->mmio = ioremap(r->start, resource_size(r));
+	wd->mmio = ioremap(iores->start, resource_size(iores));
 	if (!wd->mmio)
 		goto out1;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0);
-	if (!r)
+	if (!dmares)
 		goto out2;
-	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_PLAYBACK] = dmares->start;
 
-	r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	dmares = platform_get_resource(pdev, IORESOURCE_DMA, 1);
-	if (!r)
+	if (!dmares)
 		goto out2;
-	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = r->start;
+	wd->dmaids[SNDRV_PCM_STREAM_CAPTURE] = dmares->start;
 
 	/* preserve PSC clock source set up by platform (dev.platform_data
 	 * is already occupied by soc layer)
@@ -355,7 +356,7 @@ static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 out2:
 	iounmap(wd->mmio);
 out1:
-	release_mem_region(r->start, resource_size(r));
+	release_mem_region(iores->start, resource_size(iores));
 out0:
 	kfree(wd);
 	return ret;

sound/soc/blackfin/bf5xx-ac97-pcm.c

@@ -418,7 +418,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
+static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
 	struct snd_soc_dai *dai = rtd->cpu_dai;

sound/soc/blackfin/bf5xx-i2s-pcm.c

@@ -257,7 +257,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 
 static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32);
 
-int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
+static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_card *card = rtd->card->snd_card;
 	struct snd_soc_dai *dai = rtd->cpu_dai;

sound/soc/codecs/88pm860x-codec.c

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

sound/soc/codecs/Kconfig

@@ -40,6 +40,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX9850 if I2C
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_PCM3008
+	select SND_SOC_RT5631 if I2C
 	select SND_SOC_SGTL5000 if I2C
 	select SND_SOC_SN95031 if INTEL_SCU_IPC
 	select SND_SOC_SPDIF
@@ -48,7 +49,7 @@ config SND_SOC_ALL_CODECS
 	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_TLV320AIC32X4 if I2C
 	select SND_SOC_TLV320AIC3X if I2C
 	select SND_SOC_TPA6130A2 if I2C
 	select SND_SOC_TLV320DAC33 if I2C
@@ -59,6 +60,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WL1273 if MFD_WL1273_CORE
 	select SND_SOC_WM1250_EV1 if I2C
 	select SND_SOC_WM2000 if I2C
+	select SND_SOC_WM5100 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
 	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
@@ -218,6 +220,9 @@ config SND_SOC_MAX9850
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_RT5631
+	tristate
+
 #Freescale sgtl5000 codec
 config SND_SOC_SGTL5000
 	tristate
@@ -244,7 +249,7 @@ config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
 	depends on SPI
 
-config SND_SOC_TVL320AIC32X4
+config SND_SOC_TLV320AIC32X4
 	tristate
 
 config SND_SOC_TLV320AIC3X
@@ -273,6 +278,9 @@ config SND_SOC_WL1273
 config SND_SOC_WM1250_EV1
 	tristate
 
+config SND_SOC_WM5100
+	tristate
+
 config SND_SOC_WM8350
 	tristate
 

sound/soc/codecs/Makefile

@@ -26,6 +26,7 @@ snd-soc-max98088-objs := max98088.o
 snd-soc-max98095-objs := max98095.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-rt5631-objs := rt5631.o
 snd-soc-sgtl5000-objs := sgtl5000.o
 snd-soc-alc5623-objs := alc5623.o
 snd-soc-sn95031-objs := sn95031.o
@@ -44,6 +45,7 @@ snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wl1273-objs := wl1273.o
 snd-soc-wm1250-ev1-objs := wm1250-ev1.o
+snd-soc-wm5100-objs := wm5100.o wm5100-tables.o
 snd-soc-wm8350-objs := wm8350.o
 snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
@@ -125,6 +127,7 @@ obj-$(CONFIG_SND_SOC_MAX98088)	+= snd-soc-max98088.o
 obj-$(CONFIG_SND_SOC_MAX98095)	+= snd-soc-max98095.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_RT5631)	+= snd-soc-rt5631.o
 obj-$(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
@@ -134,7 +137,7 @@ obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
-obj-$(CONFIG_SND_SOC_TVL320AIC32X4)     += snd-soc-tlv320aic32x4.o
+obj-$(CONFIG_SND_SOC_TLV320AIC32X4)     += snd-soc-tlv320aic32x4.o
 obj-$(CONFIG_SND_SOC_TLV320DAC33)	+= snd-soc-tlv320dac33.o
 obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_TWL6040)	+= snd-soc-twl6040.o
@@ -142,6 +145,7 @@ obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WL1273)	+= snd-soc-wl1273.o
 obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o
+obj-$(CONFIG_SND_SOC_WM5100)	+= snd-soc-wm5100.o
 obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
 obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o

sound/soc/codecs/ad193x.c

@@ -103,12 +103,14 @@ static const struct snd_soc_dapm_route audio_paths[] = {
 static int ad193x_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	int reg;
 
-	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
+	if (mute)
-	reg = (mute > 0) ? reg | AD193X_DAC_MASTER_MUTE : reg &
+		snd_soc_update_bits(codec, AD193X_DAC_CTRL2,
-		(~AD193X_DAC_MASTER_MUTE);
+				    AD193X_DAC_MASTER_MUTE,
-	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
+				    AD193X_DAC_MASTER_MUTE);
+	else
+		snd_soc_update_bits(codec, AD193X_DAC_CTRL2,
+				    AD193X_DAC_MASTER_MUTE, 0);
 
 	return 0;
 }
@@ -262,7 +264,7 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 		struct snd_pcm_hw_params *params,
 		struct snd_soc_dai *dai)
 {
-	int word_len = 0, reg = 0, master_rate = 0;
+	int word_len = 0, master_rate = 0;
 
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
@@ -297,18 +299,15 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	reg = snd_soc_read(codec, AD193X_PLL_CLK_CTRL0);
+	snd_soc_update_bits(codec, AD193X_PLL_CLK_CTRL0,
-	reg = (reg & AD193X_PLL_INPUT_MASK) | master_rate;
+			    AD193X_PLL_INPUT_MASK, master_rate);
-	snd_soc_write(codec, AD193X_PLL_CLK_CTRL0, reg);
 
-	reg = snd_soc_read(codec, AD193X_DAC_CTRL2);
+	snd_soc_update_bits(codec, AD193X_DAC_CTRL2,
-	reg = (reg & (~AD193X_DAC_WORD_LEN_MASK))
+			    AD193X_DAC_WORD_LEN_MASK,
-		| (word_len << AD193X_DAC_WORD_LEN_SHFT);
+			    word_len << AD193X_DAC_WORD_LEN_SHFT);
-	snd_soc_write(codec, AD193X_DAC_CTRL2, reg);
 
-	reg = snd_soc_read(codec, AD193X_ADC_CTRL1);
+	snd_soc_update_bits(codec, AD193X_ADC_CTRL1,
-	reg = (reg & (~AD193X_ADC_WORD_LEN_MASK)) | word_len;
+			    AD193X_ADC_WORD_LEN_MASK, word_len);
-	snd_soc_write(codec, AD193X_ADC_CTRL1, reg);
 
 	return 0;
 }

sound/soc/codecs/ad193x.h

@@ -11,7 +11,7 @@
 
 #define AD193X_PLL_CLK_CTRL0    0x00
 #define AD193X_PLL_POWERDOWN           0x01
-#define AD193X_PLL_INPUT_MASK   (~0x6)
+#define AD193X_PLL_INPUT_MASK   0x6
 #define AD193X_PLL_INPUT_256    (0 << 1)
 #define AD193X_PLL_INPUT_384    (1 << 1)
 #define AD193X_PLL_INPUT_512    (2 << 1)

sound/soc/codecs/ad1980.c

@@ -148,7 +148,6 @@ static struct snd_soc_dai_driver ad1980_dai = {
 		.rates = SNDRV_PCM_RATE_48000,
 		.formats = SND_SOC_STD_AC97_FMTS, },
 };
-EXPORT_SYMBOL_GPL(ad1980_dai);
 
 static int ad1980_reset(struct snd_soc_codec *codec, int try_warm)
 {

sound/soc/codecs/adau1373.c

@@ -974,7 +974,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		ctrl = 0;
-		adau1373_dai->master = true;
+		adau1373_dai->master = false;
 		break;
 	default:
 		return -EINVAL;

sound/soc/codecs/adau1701.c

@@ -401,7 +401,7 @@ static int adau1701_digital_mute(struct snd_soc_dai *dai, int mute)
 }
 
 static int adau1701_set_sysclk(struct snd_soc_codec *codec, int clk_id,
-	unsigned int freq, int dir)
+	int source, unsigned int freq, int dir)
 {
 	unsigned int val;
 
@@ -458,6 +458,7 @@ static int adau1701_probe(struct snd_soc_codec *codec)
 	int ret;
 
 	codec->dapm.idle_bias_off = 1;
+	codec->control_data = to_i2c_client(codec->dev);
 
 	ret = adau1701_load_firmware(codec);
 	if (ret)

sound/soc/codecs/ads117x.h

@@ -1,13 +0,0 @@
-/*
- * ads117x.h  --  Driver for ads1174/8 ADC chips
- *
- * Copyright 2009 ShotSpotter Inc.
- * Author: Graeme Gregory <gg@slimlogic.co.uk>
- *
- *  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.
- */
-extern struct snd_soc_dai_driver ads117x_dai;
-extern struct snd_soc_codec_driver soc_codec_dev_ads117x;

sound/soc/codecs/ak4104.c

@@ -247,7 +247,7 @@ static struct snd_soc_codec_driver soc_codec_device_ak4104 = {
 	.probe =	ak4104_probe,
 	.remove =	ak4104_remove,
 	.reg_cache_size = AK4104_NUM_REGS,
-	.reg_word_size = sizeof(u16),
+	.reg_word_size = sizeof(u8),
 };
 
 static int ak4104_spi_probe(struct spi_device *spi)

sound/soc/codecs/ak4535.c

@@ -34,74 +34,16 @@
 struct ak4535_priv {
 	unsigned int sysclk;
 	enum snd_soc_control_type control_type;
-	void *control_data;
 };
 
 /*
  * ak4535 register cache
  */
-static const u16 ak4535_reg[AK4535_CACHEREGNUM] = {
+static const u8 ak4535_reg[AK4535_CACHEREGNUM] = {
-    0x0000, 0x0080, 0x0000, 0x0003,
+	0x00, 0x80, 0x00, 0x03,
-    0x0002, 0x0000, 0x0011, 0x0001,
+	0x02, 0x00, 0x11, 0x01,
-    0x0000, 0x0040, 0x0036, 0x0010,
+	0x00, 0x40, 0x36, 0x10,
-    0x0000, 0x0000, 0x0057, 0x0000,
+	0x00, 0x00, 0x57, 0x00,
-};
-
-/*
- * read ak4535 register cache
- */
-static inline unsigned int ak4535_read_reg_cache(struct snd_soc_codec *codec,
-	unsigned int reg)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= AK4535_CACHEREGNUM)
-		return -1;
-	return cache[reg];
-}
-
-/*
- * write ak4535 register cache
- */
-static inline void ak4535_write_reg_cache(struct snd_soc_codec *codec,
-	u16 reg, unsigned int value)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= AK4535_CACHEREGNUM)
-		return;
-	cache[reg] = value;
-}
-
-/*
- * write to the AK4535 register space
- */
-static int ak4535_write(struct snd_soc_codec *codec, unsigned int reg,
-	unsigned int value)
-{
-	u8 data[2];
-
-	/* data is
-	 *   D15..D8 AK4535 register offset
-	 *   D7...D0 register data
-	 */
-	data[0] = reg & 0xff;
-	data[1] = value & 0xff;
-
-	ak4535_write_reg_cache(codec, reg, value);
-	if (codec->hw_write(codec->control_data, data, 2) == 2)
-		return 0;
-	else
-		return -EIO;
-}
-
-static int ak4535_sync(struct snd_soc_codec *codec)
-{
-	u16 *cache = codec->reg_cache;
-	int i, r = 0;
-
-	for (i = 0; i < AK4535_CACHEREGNUM; i++)
-		r |= ak4535_write(codec, i, cache[i]);
-
-	return r;
 };
 
 static const char *ak4535_mono_gain[] = {"+6dB", "-17dB"};
@@ -304,7 +246,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
-	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
+	u8 mode2 = snd_soc_read(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
 
 	if (rate)
@@ -323,7 +265,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	/* set rate */
-	ak4535_write(codec, AK4535_MODE2, mode2);
+	snd_soc_write(codec, AK4535_MODE2, mode2);
 	return 0;
 }
 
@@ -348,44 +290,37 @@ static int ak4535_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	/* use 32 fs for BCLK to save power */
 	mode1 |= 0x4;
 
-	ak4535_write(codec, AK4535_MODE1, mode1);
+	snd_soc_write(codec, AK4535_MODE1, mode1);
 	return 0;
 }
 
 static int ak4535_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
+	u16 mute_reg = snd_soc_read(codec, AK4535_DAC);
 	if (!mute)
-		ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20);
+		snd_soc_write(codec, AK4535_DAC, mute_reg & ~0x20);
 	else
-		ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
+		snd_soc_write(codec, AK4535_DAC, mute_reg | 0x20);
 	return 0;
 }
 
 static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
-	u16 i, mute_reg;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
+		snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0);
-		ak4535_write(codec, AK4535_DAC, mute_reg & ~0x20);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		mute_reg = ak4535_read_reg_cache(codec, AK4535_DAC);
+		snd_soc_update_bits(codec, AK4535_DAC, 0x20, 0x20);
-		ak4535_write(codec, AK4535_DAC, mute_reg | 0x20);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		i = ak4535_read_reg_cache(codec, AK4535_PM1);
+		snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0x80);
-		ak4535_write(codec, AK4535_PM1, i | 0x80);
+		snd_soc_update_bits(codec, AK4535_PM2, 0x80, 0);
-		i = ak4535_read_reg_cache(codec, AK4535_PM2);
-		ak4535_write(codec, AK4535_PM2, i & (~0x80));
 		break;
 	case SND_SOC_BIAS_OFF:
-		i = ak4535_read_reg_cache(codec, AK4535_PM1);
+		snd_soc_update_bits(codec, AK4535_PM1, 0x80, 0);
-		ak4535_write(codec, AK4535_PM1, i & (~0x80));
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -428,7 +363,7 @@ static int ak4535_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int ak4535_resume(struct snd_soc_codec *codec)
 {
-	ak4535_sync(codec);
+	snd_soc_cache_sync(codec);
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	return 0;
 }
@@ -436,11 +371,15 @@ static int ak4535_resume(struct snd_soc_codec *codec)
 static int ak4535_probe(struct snd_soc_codec *codec)
 {
 	struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec);
+	int ret;
 
 	printk(KERN_INFO "AK4535 Audio Codec %s", AK4535_VERSION);
 
-	codec->control_data = ak4535->control_data;
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4535->control_type);
-
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
 	/* power on device */
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -461,8 +400,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4535 = {
 	.remove =	ak4535_remove,
 	.suspend =	ak4535_suspend,
 	.resume =	ak4535_resume,
-	.read = ak4535_read_reg_cache,
-	.write = ak4535_write,
 	.set_bias_level = ak4535_set_bias_level,
 	.reg_cache_size = ARRAY_SIZE(ak4535_reg),
 	.reg_word_size = sizeof(u8),
@@ -485,7 +422,6 @@ static __devinit int ak4535_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, ak4535);
-	ak4535->control_data = i2c;
 	ak4535->control_type = SND_SOC_I2C;
 
 	ret = snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/ak4641.c

@@ -31,7 +31,6 @@
 
 /* codec private data */
 struct ak4641_priv {
-	struct snd_soc_codec *codec;
 	unsigned int sysclk;
 	int deemph;
 	int playback_fs;
@@ -226,7 +225,7 @@ static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0),
 
 	SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0),
-	SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
+	SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
 
 	SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0),
 	SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0),

sound/soc/codecs/ak4642.c

@@ -156,81 +156,22 @@ static const struct snd_kcontrol_new ak4642_snd_controls[] = {
 struct ak4642_priv {
 	unsigned int sysclk;
 	enum snd_soc_control_type control_type;
-	void *control_data;
 };
 
 /*
  * ak4642 register cache
  */
-static const u16 ak4642_reg[AK4642_CACHEREGNUM] = {
+static const u8 ak4642_reg[AK4642_CACHEREGNUM] = {
-	0x0000, 0x0000, 0x0001, 0x0000,
+	0x00, 0x00, 0x01, 0x00,
-	0x0002, 0x0000, 0x0000, 0x0000,
+	0x02, 0x00, 0x00, 0x00,
-	0x00e1, 0x00e1, 0x0018, 0x0000,
+	0xe1, 0xe1, 0x18, 0x00,
-	0x00e1, 0x0018, 0x0011, 0x0008,
+	0xe1, 0x18, 0x11, 0x08,
-	0x0000, 0x0000, 0x0000, 0x0000,
+	0x00, 0x00, 0x00, 0x00,
-	0x0000, 0x0000, 0x0000, 0x0000,
+	0x00, 0x00, 0x00, 0x00,
-	0x0000, 0x0000, 0x0000, 0x0000,
+	0x00, 0x00, 0x00, 0x00,
-	0x0000, 0x0000, 0x0000, 0x0000,
+	0x00, 0x00, 0x00, 0x00,
-	0x0000, 0x0000, 0x0000, 0x0000,
+	0x00, 0x00, 0x00, 0x00,
-	0x0000,
+	0x00,
-};
-
-/*
- * read ak4642 register cache
- */
-static inline unsigned int ak4642_read_reg_cache(struct snd_soc_codec *codec,
-	unsigned int reg)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= AK4642_CACHEREGNUM)
-		return -1;
-	return cache[reg];
-}
-
-/*
- * write ak4642 register cache
- */
-static inline void ak4642_write_reg_cache(struct snd_soc_codec *codec,
-	u16 reg, unsigned int value)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= AK4642_CACHEREGNUM)
-		return;
-
-	cache[reg] = value;
-}
-
-/*
- * write to the AK4642 register space
- */
-static int ak4642_write(struct snd_soc_codec *codec, unsigned int reg,
-	unsigned int value)
-{
-	u8 data[2];
-
-	/* data is
-	 *   D15..D8 AK4642 register offset
-	 *   D7...D0 register data
-	 */
-	data[0] = reg & 0xff;
-	data[1] = value & 0xff;
-
-	if (codec->hw_write(codec->control_data, data, 2) == 2) {
-		ak4642_write_reg_cache(codec, reg, value);
-		return 0;
-	} else
-		return -EIO;
-}
-
-static int ak4642_sync(struct snd_soc_codec *codec)
-{
-	u16 *cache = codec->reg_cache;
-	int i, r = 0;
-
-	for (i = 0; i < AK4642_CACHEREGNUM; i++)
-		r |= ak4642_write(codec, i, cache[i]);
-
-	return r;
 };
 
 static int ak4642_dai_startup(struct snd_pcm_substream *substream,
@@ -252,8 +193,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 */
 		snd_soc_update_bits(codec, MD_CTL4, DACH, DACH);
 		snd_soc_update_bits(codec, MD_CTL3, BST1, BST1);
-		ak4642_write(codec, L_IVC, 0x91); /* volume */
+		snd_soc_write(codec, L_IVC, 0x91); /* volume */
-		ak4642_write(codec, R_IVC, 0x91); /* volume */
+		snd_soc_write(codec, R_IVC, 0x91); /* volume */
 		snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC,
 						     PMVCM | PMMIN | PMDAC);
 		snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK,	PMHP);
@@ -272,9 +213,9 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
 		 * This operation came from example code of
 		 * "ASAHI KASEI AK4642" (japanese) manual p94.
 		 */
-		ak4642_write(codec, SG_SL1, PMMP | MGAIN0);
+		snd_soc_write(codec, SG_SL1, PMMP | MGAIN0);
-		ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
+		snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3));
-		ak4642_write(codec, ALC_CTL1, ALC | LMTH0);
+		snd_soc_write(codec, ALC_CTL1, ALC | LMTH0);
 		snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL,
 						     PMVCM | PMADL);
 		snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR);
@@ -462,7 +403,7 @@ static struct snd_soc_dai_driver ak4642_dai = {
 
 static int ak4642_resume(struct snd_soc_codec *codec)
 {
-	ak4642_sync(codec);
+	snd_soc_cache_sync(codec);
 	return 0;
 }
 
@@ -470,11 +411,15 @@ static int ak4642_resume(struct snd_soc_codec *codec)
 static int ak4642_probe(struct snd_soc_codec *codec)
 {
 	struct ak4642_priv *ak4642 = snd_soc_codec_get_drvdata(codec);
+	int ret;
 
 	dev_info(codec->dev, "AK4642 Audio Codec %s", AK4642_VERSION);
 
-	codec->hw_write		= (hw_write_t)i2c_master_send;
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4642->control_type);
-	codec->control_data	= ak4642->control_data;
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
 
 	snd_soc_add_controls(codec, ak4642_snd_controls,
 			     ARRAY_SIZE(ak4642_snd_controls));
@@ -485,8 +430,6 @@ static int ak4642_probe(struct snd_soc_codec *codec)
 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
 	.probe			= ak4642_probe,
 	.resume			= ak4642_resume,
-	.read			= ak4642_read_reg_cache,
-	.write			= ak4642_write,
 	.reg_cache_size		= ARRAY_SIZE(ak4642_reg),
 	.reg_word_size		= sizeof(u8),
 	.reg_cache_default	= ak4642_reg,
@@ -504,7 +447,6 @@ static __devinit int ak4642_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, ak4642);
-	ak4642->control_data = i2c;
 	ak4642->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/ak4671.c

@@ -26,7 +26,6 @@
 /* codec private data */
 struct ak4671_priv {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 };
 
 /* ak4671 register cache & default register settings */
@@ -169,18 +168,15 @@ static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	u8 reg;
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+		snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT,
-		reg |= AK4671_MUTEN;
+				    AK4671_MUTEN, AK4671_MUTEN);
-		snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
-		reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT);
+		snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT,
-		reg &= ~AK4671_MUTEN;
+				    AK4671_MUTEN, 0);
-		snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg);
 		break;
 	}
 
@@ -576,15 +572,12 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 static int ak4671_set_bias_level(struct snd_soc_codec *codec,
 		enum snd_soc_bias_level level)
 {
-	u8 reg;
-
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
 	case SND_SOC_BIAS_STANDBY:
-		reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT);
+		snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT,
-		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT,
+				    AK4671_PMVCM, AK4671_PMVCM);
-				reg | AK4671_PMVCM);
 		break;
 	case SND_SOC_BIAS_OFF:
 		snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
@@ -629,8 +622,6 @@ static int ak4671_probe(struct snd_soc_codec *codec)
 	struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec);
 	int ret;
 
-	codec->hw_write = (hw_write_t)i2c_master_send;
-
 	ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
@@ -675,7 +666,6 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client,
 		return -ENOMEM;
 
 	i2c_set_clientdata(client, ak4671);
-	ak4671->control_data = client;
 	ak4671->control_type = SND_SOC_I2C;
 
 	ret = snd_soc_register_codec(&client->dev,

sound/soc/codecs/alc5623.c

@@ -40,7 +40,6 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
 /* codec private data */
 struct alc5623_priv {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	u8 id;
 	unsigned int sysclk;
 	u16 reg_cache[ALC5623_VENDOR_ID2+2];
@@ -54,8 +53,10 @@ static void alc5623_fill_cache(struct snd_soc_codec *codec)
 	u16 *cache = codec->reg_cache;
 
 	/* not really efficient ... */
+	codec->cache_bypass = 1;
 	for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
-		cache[i] = codec->hw_read(codec, i);
+		cache[i] = snd_soc_read(codec, i);
+	codec->cache_bypass = 0;
 }
 
 static inline int alc5623_reset(struct snd_soc_codec *codec)
@@ -1049,7 +1050,6 @@ static int alc5623_i2c_probe(struct i2c_client *client,
 	}
 
 	i2c_set_clientdata(client, alc5623);
-	alc5623->control_data = client;
 	alc5623->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&client->dev,

sound/soc/codecs/cs4270.c

@@ -128,7 +128,6 @@ static const char *supply_names[] = {
 /* Private data for the CS4270 */
 struct cs4270_private {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	unsigned int mclk; /* Input frequency of the MCLK pin */
 	unsigned int mode; /* The mode (I2S or left-justified) */
 	unsigned int slave_mode;
@@ -262,7 +261,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
 
 	/* set DAI format */
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -272,7 +270,7 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		break;
 	default:
 		dev_err(codec->dev, "invalid dai format\n");
-		ret = -EINVAL;
+		return -EINVAL;
 	}
 
 	/* set master/slave audio interface */
@@ -285,10 +283,11 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		break;
 	default:
 		/* all other modes are unsupported by the hardware */
-		ret = -EINVAL;
+		dev_err(codec->dev, "Unknown master/slave configuration\n");
+		return -EINVAL;
 	}
 
-	return ret;
+	return 0;
 }
 
 /**
@@ -490,8 +489,6 @@ static int cs4270_probe(struct snd_soc_codec *codec)
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
 	int i, ret;
 
-	codec->control_data = cs4270->control_data;
-
 	/* Tell ASoC what kind of I/O to use to read the registers.  ASoC will
 	 * then do the I2C transactions itself.
 	 */
@@ -604,7 +601,7 @@ static int cs4270_soc_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
 static int cs4270_soc_resume(struct snd_soc_codec *codec)
 {
 	struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec);
-	struct i2c_client *i2c_client = codec->control_data;
+	struct i2c_client *i2c_client = to_i2c_client(codec->dev);
 	int reg;
 
 	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
@@ -690,7 +687,6 @@ static int cs4270_i2c_probe(struct i2c_client *i2c_client,
 	}
 
 	i2c_set_clientdata(i2c_client, cs4270);
-	cs4270->control_data = i2c_client;
 	cs4270->control_type = SND_SOC_I2C;
 
 	ret = snd_soc_register_codec(&i2c_client->dev,

sound/soc/codecs/cs4271.c

@@ -156,7 +156,6 @@ static const u8 cs4271_dflt_reg[CS4271_NR_REGS] = {
 struct cs4271_private {
 	/* SND_SOC_I2C or SND_SOC_SPI */
 	enum snd_soc_control_type	bus_type;
-	void				*control_data;
 	unsigned int			mclk;
 	bool				master;
 	bool				deemph;
@@ -466,8 +465,6 @@ static int cs4271_probe(struct snd_soc_codec *codec)
 	int ret;
 	int gpio_nreset = -EINVAL;
 
-	codec->control_data = cs4271->control_data;
-
 	if (cs4271plat && gpio_is_valid(cs4271plat->gpio_nreset))
 		gpio_nreset = cs4271plat->gpio_nreset;
 
@@ -555,7 +552,6 @@ static int __devinit cs4271_spi_probe(struct spi_device *spi)
 		return -ENOMEM;
 
 	spi_set_drvdata(spi, cs4271);
-	cs4271->control_data = spi;
 	cs4271->bus_type = SND_SOC_SPI;
 
 	return snd_soc_register_codec(&spi->dev, &soc_codec_dev_cs4271,
@@ -595,7 +591,6 @@ static int __devinit cs4271_i2c_probe(struct i2c_client *client,
 		return -ENOMEM;
 
 	i2c_set_clientdata(client, cs4271);
-	cs4271->control_data = client;
 	cs4271->bus_type = SND_SOC_I2C;
 
 	return snd_soc_register_codec(&client->dev, &soc_codec_dev_cs4271,

sound/soc/codecs/cs42l51.c

@@ -42,7 +42,6 @@ enum master_slave_mode {
 
 struct cs42l51_private {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	unsigned int mclk;
 	unsigned int audio_mode;	/* The mode (I2S or left-justified) */
 	enum master_slave_mode func;
@@ -57,7 +56,7 @@ struct cs42l51_private {
 static int cs42l51_fill_cache(struct snd_soc_codec *codec)
 {
 	u8 *cache = codec->reg_cache + 1;
-	struct i2c_client *i2c_client = codec->control_data;
+	struct i2c_client *i2c_client = to_i2c_client(codec->dev);
 	s32 length;
 
 	length = i2c_smbus_read_i2c_block_data(i2c_client,
@@ -289,7 +288,6 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec);
-	int ret = 0;
 
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
@@ -299,7 +297,7 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		break;
 	default:
 		dev_err(codec->dev, "invalid DAI format\n");
-		ret = -EINVAL;
+		return -EINVAL;
 	}
 
 	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -310,11 +308,11 @@ static int cs42l51_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		cs42l51->func = MODE_SLAVE_AUTO;
 		break;
 	default:
-		ret = -EINVAL;
+		dev_err(codec->dev, "Unknown master/slave configuration\n");
-		break;
+		return -EINVAL;
 	}
 
-	return ret;
+	return 0;
 }
 
 struct cs42l51_ratios {
@@ -520,8 +518,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec)
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret, reg;
 
-	codec->control_data = cs42l51->control_data;
-
 	ret = cs42l51_fill_cache(codec);
 	if (ret < 0) {
 		dev_err(codec->dev, "failed to fill register cache\n");
@@ -593,7 +589,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client,
 	}
 
 	i2c_set_clientdata(i2c_client, cs42l51);
-	cs42l51->control_data = i2c_client;
 	cs42l51->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&i2c_client->dev,

sound/soc/codecs/da7210.c

@@ -26,23 +26,41 @@
 #include <sound/tlv.h>
 
 /* DA7210 register space */
+#define DA7210_CONTROL			0x01
 #define DA7210_STATUS			0x02
 #define DA7210_STARTUP1			0x03
+#define DA7210_STARTUP2			0x04
+#define DA7210_STARTUP3			0x05
 #define DA7210_MIC_L			0x07
 #define DA7210_MIC_R			0x08
+#define DA7210_AUX1_L			0x09
+#define DA7210_AUX1_R			0x0A
+#define DA7210_AUX2			0x0B
+#define DA7210_IN_GAIN			0x0C
 #define DA7210_INMIX_L			0x0D
 #define DA7210_INMIX_R			0x0E
 #define DA7210_ADC_HPF			0x0F
 #define DA7210_ADC			0x10
+#define DA7210_ADC_EQ1_2		0X11
+#define DA7210_ADC_EQ3_4		0x12
+#define DA7210_ADC_EQ5			0x13
 #define DA7210_DAC_HPF			0x14
 #define DA7210_DAC_L			0x15
 #define DA7210_DAC_R			0x16
 #define DA7210_DAC_SEL			0x17
+#define DA7210_SOFTMUTE			0x18
+#define DA7210_DAC_EQ1_2		0x19
+#define DA7210_DAC_EQ3_4		0x1A
+#define DA7210_DAC_EQ5			0x1B
 #define DA7210_OUTMIX_L			0x1C
 #define DA7210_OUTMIX_R			0x1D
+#define DA7210_OUT1_L			0x1E
+#define DA7210_OUT1_R			0x1F
+#define DA7210_OUT2			0x20
 #define DA7210_HP_L_VOL			0x21
 #define DA7210_HP_R_VOL			0x22
 #define DA7210_HP_CFG			0x23
+#define DA7210_ZERO_CROSS		0x24
 #define DA7210_DAI_SRC_SEL		0x25
 #define DA7210_DAI_CFG1			0x26
 #define DA7210_DAI_CFG3			0x28
@@ -50,6 +68,12 @@
 #define DA7210_PLL_DIV2			0x2A
 #define DA7210_PLL_DIV3			0x2B
 #define DA7210_PLL			0x2C
+#define DA7210_ALC_MAX			0x83
+#define DA7210_ALC_MIN			0x84
+#define DA7210_ALC_NOIS			0x85
+#define DA7210_ALC_ATT			0x86
+#define DA7210_ALC_REL			0x87
+#define DA7210_ALC_DEL			0x88
 #define DA7210_A_HID_UNLOCK		0x8A
 #define DA7210_A_TEST_UNLOCK		0x8B
 #define DA7210_A_PLL1			0x90
@@ -72,6 +96,7 @@
 #define DA7210_IN_R_EN			(1 << 7)
 
 /* ADC bit fields */
+#define DA7210_ADC_ALC_EN		(1 << 0)
 #define DA7210_ADC_L_EN			(1 << 3)
 #define DA7210_ADC_R_EN			(1 << 7)
 
@@ -105,12 +130,17 @@
 
 /* DAI_CFG1 bit fields */
 #define DA7210_DAI_WORD_S16_LE		(0 << 0)
+#define DA7210_DAI_WORD_S20_3LE		(1 << 0)
 #define DA7210_DAI_WORD_S24_LE		(2 << 0)
+#define DA7210_DAI_WORD_S32_LE		(3 << 0)
 #define DA7210_DAI_FLEN_64BIT		(1 << 2)
+#define DA7210_DAI_MODE_SLAVE		(0 << 7)
 #define DA7210_DAI_MODE_MASTER		(1 << 7)
 
 /* DAI_CFG3 bit fields */
 #define DA7210_DAI_FORMAT_I2SMODE	(0 << 0)
+#define DA7210_DAI_FORMAT_LEFT_J	(1 << 0)
+#define DA7210_DAI_FORMAT_RIGHT_J	(2 << 0)
 #define DA7210_DAI_OE			(1 << 3)
 #define DA7210_DAI_EN			(1 << 7)
 
@@ -133,6 +163,43 @@
 #define DA7210_PLL_FS_96000		(0xF << 0)
 #define DA7210_PLL_EN			(0x1 << 7)
 
+/* SOFTMUTE bit fields */
+#define DA7210_RAMP_EN			(1 << 6)
+
+/* CONTROL bit fields */
+#define DA7210_NOISE_SUP_EN		(1 << 3)
+
+/* IN_GAIN bit fields */
+#define DA7210_INPGA_L_VOL		(0x0F << 0)
+#define DA7210_INPGA_R_VOL		(0xF0 << 0)
+
+/* ZERO_CROSS bit fields */
+#define DA7210_AUX1_L_ZC		(1 << 0)
+#define DA7210_AUX1_R_ZC		(1 << 1)
+#define DA7210_HP_L_ZC			(1 << 6)
+#define DA7210_HP_R_ZC			(1 << 7)
+
+/* AUX1_L bit fields */
+#define DA7210_AUX1_L_VOL		(0x3F << 0)
+
+/* AUX1_R bit fields */
+#define DA7210_AUX1_R_VOL		(0x3F << 0)
+
+/* Minimum INPGA and AUX1 volume to enable noise suppression */
+#define DA7210_INPGA_MIN_VOL_NS		0x0A  /* 10.5dB */
+#define DA7210_AUX1_MIN_VOL_NS		0x35  /* 6dB */
+
+/* OUT1_L bit fields */
+#define DA7210_OUT1_L_EN		(1 << 7)
+
+/* OUT1_R bit fields */
+#define DA7210_OUT1_R_EN		(1 << 7)
+
+/* OUT2 bit fields */
+#define DA7210_OUT2_OUTMIX_R		(1 << 5)
+#define DA7210_OUT2_OUTMIX_L		(1 << 6)
+#define DA7210_OUT2_EN			(1 << 7)
+
 #define DA7210_VERSION "0.0.1"
 
 /*
@@ -144,24 +211,351 @@
  * mute		: 0x10
  * reserved	: 0x00 - 0x0F
  *
- * ** FIXME **
- *
  * Reserved area are considered as "mute".
- * -> min = -79.5 dB
  */
-static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1);
+static const unsigned int hp_out_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+	/* -54 dB to +15 dB */
+	0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0),
+};
+
+static const unsigned int lineout_vol_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+	/* -54dB to 15dB */
+	0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0)
+};
+
+static const unsigned int mono_vol_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1),
+	/* -18dB to 6dB */
+	0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0)
+};
+
+static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1);
+static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0);
+
+/* ADC and DAC high pass filter f0 value */
+static const char const *da7210_hpf_cutoff_txt[] = {
+	"Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi"
+};
+
+static const struct soc_enum da7210_dac_hpf_cutoff =
+	SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+
+static const struct soc_enum da7210_adc_hpf_cutoff =
+	SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt);
+
+/* ADC and DAC voice (8kHz) high pass cutoff value */
+static const char const *da7210_vf_cutoff_txt[] = {
+	"2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da7210_dac_vf_cutoff =
+	SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt);
+
+static const struct soc_enum da7210_adc_vf_cutoff =
+	SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt);
+
+static const char *da7210_hp_mode_txt[] = {
+	"Class H", "Class G"
+};
+
+static const struct soc_enum da7210_hp_mode_sel =
+	SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt);
+
+/* ALC can be enabled only if noise suppression is disabled */
+static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0]) {
+		/* Check if noise suppression is enabled */
+		if (snd_soc_read(codec, DA7210_CONTROL) & DA7210_NOISE_SUP_EN) {
+			dev_dbg(codec->dev,
+				"Disable noise suppression to enable ALC\n");
+			return -EINVAL;
+		}
+	}
+	/* If all conditions are met or we are actually disabling ALC */
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
+/* Noise suppression can be enabled only if following conditions are met
+ *  ALC disabled
+ *  ZC enabled for HP and AUX1 PGA
+ *  INPGA_L_VOL and INPGA_R_VOL >= 10.5 dB
+ *  AUX1_L_VOL and AUX1_R_VOL >= 6 dB
+ */
+static int da7210_put_noise_sup_sw(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u8 val;
+
+	if (ucontrol->value.integer.value[0]) {
+		/* Check if ALC is enabled */
+		if (snd_soc_read(codec, DA7210_ADC) & DA7210_ADC_ALC_EN)
+			goto err;
+
+		/* Check ZC for HP and AUX1 PGA */
+		if ((snd_soc_read(codec, DA7210_ZERO_CROSS) &
+			(DA7210_AUX1_L_ZC | DA7210_AUX1_R_ZC | DA7210_HP_L_ZC |
+			DA7210_HP_R_ZC)) != (DA7210_AUX1_L_ZC |
+			DA7210_AUX1_R_ZC | DA7210_HP_L_ZC | DA7210_HP_R_ZC))
+			goto err;
+
+		/* Check INPGA_L_VOL and INPGA_R_VOL */
+		val = snd_soc_read(codec, DA7210_IN_GAIN);
+		if (((val & DA7210_INPGA_L_VOL) < DA7210_INPGA_MIN_VOL_NS) ||
+			(((val & DA7210_INPGA_R_VOL) >> 4) <
+			DA7210_INPGA_MIN_VOL_NS))
+			goto err;
+
+		/* Check AUX1_L_VOL and AUX1_R_VOL */
+		if (((snd_soc_read(codec, DA7210_AUX1_L) & DA7210_AUX1_L_VOL) <
+		    DA7210_AUX1_MIN_VOL_NS) ||
+		    ((snd_soc_read(codec, DA7210_AUX1_R) & DA7210_AUX1_R_VOL) <
+		    DA7210_AUX1_MIN_VOL_NS))
+			goto err;
+	}
+	/* If all conditions are met or we are actually disabling Noise sup */
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+
+err:
+	return -EINVAL;
+}
 
 static const struct snd_kcontrol_new da7210_snd_controls[] = {
 
 	SOC_DOUBLE_R_TLV("HeadPhone Playback Volume",
 			 DA7210_HP_L_VOL, DA7210_HP_R_VOL,
 			 0, 0x3F, 0, hp_out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume",
+			 DA7210_DAC_L, DA7210_DAC_R,
+			 0, 0x77, 1, dac_gain_tlv),
+	SOC_DOUBLE_R_TLV("Lineout Playback Volume",
+			 DA7210_OUT1_L, DA7210_OUT1_R,
+			 0, 0x3f, 0, lineout_vol_tlv),
+	SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0,
+		       mono_vol_tlv),
+
+	/* DAC Equalizer  controls */
+	SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0),
+	SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1,
+		       eq_gain_tlv),
+
+	/* ADC Equalizer  controls */
+	SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0),
+	SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3,
+		       1, adc_eq_master_gain_tlv),
+	SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1,
+		       eq_gain_tlv),
+	SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1,
+		       eq_gain_tlv),
+
+	SOC_SINGLE("DAC HPF Switch", DA7210_DAC_HPF, 3, 1, 0),
+	SOC_ENUM("DAC HPF Cutoff", da7210_dac_hpf_cutoff),
+	SOC_SINGLE("DAC Voice Mode Switch", DA7210_DAC_HPF, 7, 1, 0),
+	SOC_ENUM("DAC Voice Cutoff", da7210_dac_vf_cutoff),
+
+	SOC_SINGLE("ADC HPF Switch", DA7210_ADC_HPF, 3, 1, 0),
+	SOC_ENUM("ADC HPF Cutoff", da7210_adc_hpf_cutoff),
+	SOC_SINGLE("ADC Voice Mode Switch", DA7210_ADC_HPF, 7, 1, 0),
+	SOC_ENUM("ADC Voice Cutoff", da7210_adc_vf_cutoff),
+
+	/* Mute controls */
+	SOC_DOUBLE_R("Mic Capture Switch", DA7210_MIC_L, DA7210_MIC_R, 3, 1, 0),
+	SOC_SINGLE("Aux2 Capture Switch", DA7210_AUX2, 2, 1, 0),
+	SOC_DOUBLE("ADC Capture Switch", DA7210_ADC, 2, 6, 1, 0),
+	SOC_SINGLE("Digital Soft Mute Switch", DA7210_SOFTMUTE, 7, 1, 0),
+	SOC_SINGLE("Digital Soft Mute Rate", DA7210_SOFTMUTE, 0, 0x7, 0),
+
+	/* Zero cross controls */
+	SOC_DOUBLE("Aux1 ZC Switch", DA7210_ZERO_CROSS, 0, 1, 1, 0),
+	SOC_DOUBLE("In PGA ZC Switch", DA7210_ZERO_CROSS, 2, 3, 1, 0),
+	SOC_DOUBLE("Lineout ZC Switch", DA7210_ZERO_CROSS, 4, 5, 1, 0),
+	SOC_DOUBLE("Headphone ZC Switch", DA7210_ZERO_CROSS, 6, 7, 1, 0),
+
+	SOC_ENUM("Headphone Class", da7210_hp_mode_sel),
+
+	/* ALC controls */
+	SOC_SINGLE_EXT("ALC Enable Switch", DA7210_ADC, 0, 1, 0,
+		       snd_soc_get_volsw, da7210_put_alc_sw),
+	SOC_SINGLE("ALC Capture Max Volume", DA7210_ALC_MAX, 0, 0x3F, 0),
+	SOC_SINGLE("ALC Capture Min Volume", DA7210_ALC_MIN, 0, 0x3F, 0),
+	SOC_SINGLE("ALC Capture Noise Volume", DA7210_ALC_NOIS, 0, 0x3F, 0),
+	SOC_SINGLE("ALC Capture Attack Rate", DA7210_ALC_ATT, 0, 0xFF, 0),
+	SOC_SINGLE("ALC Capture Release Rate", DA7210_ALC_REL, 0, 0xFF, 0),
+	SOC_SINGLE("ALC Capture Release Delay", DA7210_ALC_DEL, 0, 0xFF, 0),
+
+	SOC_SINGLE_EXT("Noise Suppression Enable Switch", DA7210_CONTROL, 3, 1,
+		       0, snd_soc_get_volsw, da7210_put_noise_sup_sw),
+};
+
+/*
+ * DAPM Controls
+ *
+ * Current DAPM implementation covers almost all codec components e.g. IOs,
+ * mixers, PGAs,ADC and DAC.
+ */
+/* In Mixer Left */
+static const struct snd_kcontrol_new da7210_dapm_inmixl_controls[] = {
+	SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_L, 0, 1, 0),
+	SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_L, 1, 1, 0),
+};
+
+/* In Mixer Right */
+static const struct snd_kcontrol_new da7210_dapm_inmixr_controls[] = {
+	SOC_DAPM_SINGLE("Mic Right Switch", DA7210_INMIX_R, 0, 1, 0),
+	SOC_DAPM_SINGLE("Mic Left Switch", DA7210_INMIX_R, 1, 1, 0),
+};
+
+/* Out Mixer Left */
+static const struct snd_kcontrol_new da7210_dapm_outmixl_controls[] = {
+	SOC_DAPM_SINGLE("DAC Left Switch", DA7210_OUTMIX_L, 4, 1, 0),
+};
+
+/* Out Mixer Right */
+static const struct snd_kcontrol_new da7210_dapm_outmixr_controls[] = {
+	SOC_DAPM_SINGLE("DAC Right Switch", DA7210_OUTMIX_R, 4, 1, 0),
+};
+
+/* Mono Mixer */
+static const struct snd_kcontrol_new da7210_dapm_monomix_controls[] = {
+	SOC_DAPM_SINGLE("Outmix Right Switch", DA7210_OUT2, 5, 1, 0),
+	SOC_DAPM_SINGLE("Outmix Left Switch", DA7210_OUT2, 6, 1, 0),
+};
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget da7210_dapm_widgets[] = {
+	/* Input Side */
+	/* Input Lines */
+	SND_SOC_DAPM_INPUT("MICL"),
+	SND_SOC_DAPM_INPUT("MICR"),
+
+	/* Input PGAs */
+	SND_SOC_DAPM_PGA("Mic Left", DA7210_STARTUP3, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic Right", DA7210_STARTUP3, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_PGA("INPGA Left", DA7210_INMIX_L, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("INPGA Right", DA7210_INMIX_R, 7, 0, NULL, 0),
+
+	/* Input Mixers */
+	SND_SOC_DAPM_MIXER("In Mixer Left", SND_SOC_NOPM, 0, 0,
+		&da7210_dapm_inmixl_controls[0],
+		ARRAY_SIZE(da7210_dapm_inmixl_controls)),
+
+	SND_SOC_DAPM_MIXER("In Mixer Right", SND_SOC_NOPM, 0, 0,
+		&da7210_dapm_inmixr_controls[0],
+		ARRAY_SIZE(da7210_dapm_inmixr_controls)),
+
+	/* ADCs */
+	SND_SOC_DAPM_ADC("ADC Left", "Capture", DA7210_STARTUP3, 5, 1),
+	SND_SOC_DAPM_ADC("ADC Right", "Capture", DA7210_STARTUP3, 6, 1),
+
+	/* Output Side */
+	/* DACs */
+	SND_SOC_DAPM_DAC("DAC Left", "Playback", DA7210_STARTUP2, 5, 1),
+	SND_SOC_DAPM_DAC("DAC Right", "Playback", DA7210_STARTUP2, 6, 1),
+
+	/* Output Mixers */
+	SND_SOC_DAPM_MIXER("Out Mixer Left", SND_SOC_NOPM, 0, 0,
+		&da7210_dapm_outmixl_controls[0],
+		ARRAY_SIZE(da7210_dapm_outmixl_controls)),
+
+	SND_SOC_DAPM_MIXER("Out Mixer Right", SND_SOC_NOPM, 0, 0,
+		&da7210_dapm_outmixr_controls[0],
+		ARRAY_SIZE(da7210_dapm_outmixr_controls)),
+
+	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0,
+		&da7210_dapm_monomix_controls[0],
+		ARRAY_SIZE(da7210_dapm_monomix_controls)),
+
+	/* Output PGAs */
+	SND_SOC_DAPM_PGA("OUTPGA Left Enable", DA7210_OUTMIX_L, 7, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("OUTPGA Right Enable", DA7210_OUTMIX_R, 7, 0, NULL, 0),
+
+	SND_SOC_DAPM_PGA("Out1 Left", DA7210_STARTUP2, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Out1 Right", DA7210_STARTUP2, 1, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Out2 Mono", DA7210_STARTUP2, 2, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Headphone Left", DA7210_STARTUP2, 3, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Headphone Right", DA7210_STARTUP2, 4, 1, NULL, 0),
+
+	/* Output Lines */
+	SND_SOC_DAPM_OUTPUT("OUT1L"),
+	SND_SOC_DAPM_OUTPUT("OUT1R"),
+	SND_SOC_DAPM_OUTPUT("HPL"),
+	SND_SOC_DAPM_OUTPUT("HPR"),
+	SND_SOC_DAPM_OUTPUT("OUT2"),
+};
+
+/* DAPM audio route definition */
+static const struct snd_soc_dapm_route da7210_audio_map[] = {
+	/* Dest       Connecting Widget    source */
+	/* Input path */
+	{"Mic Left", NULL, "MICL"},
+	{"Mic Right", NULL, "MICR"},
+
+	{"In Mixer Left", "Mic Left Switch", "Mic Left"},
+	{"In Mixer Left", "Mic Right Switch", "Mic Right"},
+
+	{"In Mixer Right", "Mic Right Switch", "Mic Right"},
+	{"In Mixer Right", "Mic Left Switch", "Mic Left"},
+
+	{"INPGA Left", NULL, "In Mixer Left"},
+	{"ADC Left", NULL, "INPGA Left"},
+
+	{"INPGA Right", NULL, "In Mixer Right"},
+	{"ADC Right", NULL, "INPGA Right"},
+
+	/* Output path */
+	{"Out Mixer Left", "DAC Left Switch", "DAC Left"},
+	{"Out Mixer Right", "DAC Right Switch", "DAC Right"},
+
+	{"Mono Mixer", "Outmix Right Switch", "Out Mixer Right"},
+	{"Mono Mixer", "Outmix Left Switch", "Out Mixer Left"},
+
+	{"OUTPGA Left Enable", NULL, "Out Mixer Left"},
+	{"OUTPGA Right Enable", NULL, "Out Mixer Right"},
+
+	{"Out1 Left", NULL, "OUTPGA Left Enable"},
+	{"OUT1L", NULL, "Out1 Left"},
+
+	{"Out1 Right", NULL, "OUTPGA Right Enable"},
+	{"OUT1R", NULL, "Out1 Right"},
+
+	{"Headphone Left", NULL, "OUTPGA Left Enable"},
+	{"HPL", NULL, "Headphone Left"},
+
+	{"Headphone Right", NULL, "OUTPGA Right Enable"},
+	{"HPR", NULL, "Headphone Right"},
+
+	{"Out2 Mono", NULL, "Mono Mixer"},
+	{"OUT2", NULL, "Out2 Mono"},
 };
 
 /* Codec private data */
 struct da7210_priv {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 };
 
 /*
@@ -188,72 +582,15 @@ static const u8 da7210_reg[] = {
 	0x00,						/* R88       */
 };
 
-/*
+static int da7210_volatile_register(struct snd_soc_codec *codec,
- * Read da7210 register cache
+				    unsigned int reg)
- */
-static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg)
 {
-	u8 *cache = codec->reg_cache;
+	switch (reg) {
-	BUG_ON(reg >= ARRAY_SIZE(da7210_reg));
+	case DA7210_STATUS:
-	return cache[reg];
+		return 1;
-}
+	default:
-
-/*
- * Write to the da7210 register space
- */
-static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value)
-{
-	u8 *cache = codec->reg_cache;
-	u8 data[2];
-
-	BUG_ON(codec->driver->volatile_register);
-
-	data[0] = reg & 0xff;
-	data[1] = value & 0xff;
-
-	if (reg >= codec->driver->reg_cache_size)
-		return -EIO;
-
-	if (2 != codec->hw_write(codec->control_data, data, 2))
-		return -EIO;
-
-	cache[reg] = value;
 		return 0;
 	}
-
-/*
- * Read from the da7210 register space.
- */
-static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg)
-{
-	if (DA7210_STATUS == reg)
-		return i2c_smbus_read_byte_data(codec->control_data, reg);
-
-	return da7210_read_reg_cache(codec, reg);
-}
-
-static int da7210_startup(struct snd_pcm_substream *substream,
-			  struct snd_soc_dai *dai)
-{
-	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
-	struct snd_soc_codec *codec = dai->codec;
-
-	if (is_play) {
-		/* Enable Out */
-		snd_soc_update_bits(codec, DA7210_OUTMIX_L, 0x1F, 0x10);
-		snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10);
-
-	} else {
-		/* Volume 7 */
-		snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7);
-		snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7);
-
-		/* Enable Mic */
-		snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1);
-		snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1);
-	}
-
-	return 0;
 }
 
 /*
@@ -266,93 +603,75 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_codec *codec = rtd->codec;
 	u32 dai_cfg1;
-	u32 hpf_reg, hpf_mask, hpf_value;
 	u32 fs, bypass;
 
 	/* set DAI source to Left and Right ADC */
-	da7210_write(codec, DA7210_DAI_SRC_SEL,
+	snd_soc_write(codec, DA7210_DAI_SRC_SEL,
 		     DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC);
 
 	/* Enable DAI */
-	da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN);
+	snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN);
 
-	dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1);
+	dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		dai_cfg1 |= DA7210_DAI_WORD_S16_LE;
 		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		dai_cfg1 |= DA7210_DAI_WORD_S20_3LE;
+		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		dai_cfg1 |= DA7210_DAI_WORD_S24_LE;
 		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		dai_cfg1 |= DA7210_DAI_WORD_S32_LE;
+		break;
 	default:
 		return -EINVAL;
 	}
 
-	da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1);
+	snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1);
-
-	hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ?
-		DA7210_DAC_HPF : DA7210_ADC_HPF;
 
 	switch (params_rate(params)) {
 	case 8000:
 		fs		= DA7210_PLL_FS_8000;
-		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
-		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	case 11025:
 		fs		= DA7210_PLL_FS_11025;
-		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
-		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
 		bypass		= 0;
 		break;
 	case 12000:
 		fs		= DA7210_PLL_FS_12000;
-		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
-		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	case 16000:
 		fs		= DA7210_PLL_FS_16000;
-		hpf_mask	= DA7210_VOICE_F0_MASK	| DA7210_VOICE_EN;
-		hpf_value	= DA7210_VOICE_F0_25	| DA7210_VOICE_EN;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	case 22050:
 		fs		= DA7210_PLL_FS_22050;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= 0;
 		break;
 	case 32000:
 		fs		= DA7210_PLL_FS_32000;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	case 44100:
 		fs		= DA7210_PLL_FS_44100;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= 0;
 		break;
 	case 48000:
 		fs		= DA7210_PLL_FS_48000;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	case 88200:
 		fs		= DA7210_PLL_FS_88200;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= 0;
 		break;
 	case 96000:
 		fs		= DA7210_PLL_FS_96000;
-		hpf_mask	= DA7210_VOICE_EN;
-		hpf_value	= 0;
 		bypass		= DA7210_PLL_BYP;
 		break;
 	default:
@@ -362,7 +681,6 @@ static int da7210_hw_params(struct snd_pcm_substream *substream,
 	/* Disable active mode */
 	snd_soc_update_bits(codec, DA7210_STARTUP1, DA7210_SC_MST_EN, 0);
 
-	snd_soc_update_bits(codec, hpf_reg, hpf_mask, hpf_value);
 	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_FS_MASK, fs);
 	snd_soc_update_bits(codec, DA7210_PLL_DIV3, DA7210_PLL_BYP, bypass);
 
@@ -382,13 +700,16 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	u32 dai_cfg1;
 	u32 dai_cfg3;
 
-	dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1);
+	dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1);
-	dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3);
+	dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3);
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
 		dai_cfg1 |= DA7210_DAI_MODE_MASTER;
 		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		dai_cfg1 |= DA7210_DAI_MODE_SLAVE;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -401,6 +722,12 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	case SND_SOC_DAIFMT_I2S:
 		dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE;
 		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		dai_cfg3 |= DA7210_DAI_FORMAT_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		dai_cfg3 |= DA7210_DAI_FORMAT_RIGHT_J;
+		break;
 	default:
 		return -EINVAL;
 	}
@@ -411,19 +738,32 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
 	 */
 	dai_cfg1 |= DA7210_DAI_FLEN_64BIT;
 
-	da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1);
+	snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1);
-	da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3);
+	snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3);
 
 	return 0;
 }
 
-#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
+static int da7210_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u8 mute_reg = snd_soc_read(codec, DA7210_DAC_HPF) & 0xFB;
+
+	if (mute)
+		snd_soc_write(codec, DA7210_DAC_HPF, mute_reg | 0x4);
+	else
+		snd_soc_write(codec, DA7210_DAC_HPF, mute_reg);
+	return 0;
+}
+
+#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 /* DAI operations */
 static struct snd_soc_dai_ops da7210_dai_ops = {
-	.startup	= da7210_startup,
 	.hw_params	= da7210_hw_params,
 	.set_fmt	= da7210_set_dai_fmt,
+	.digital_mute	= da7210_mute,
 };
 
 static struct snd_soc_dai_driver da7210_dai = {
@@ -451,11 +791,15 @@ static struct snd_soc_dai_driver da7210_dai = {
 static int da7210_probe(struct snd_soc_codec *codec)
 {
 	struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec);
+	int ret;
 
 	dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
 
-	codec->control_data	= da7210->control_data;
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type);
-	codec->hw_write		= (hw_write_t)i2c_master_send;
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
 
 	/* FIXME
 	 *
@@ -472,8 +816,8 @@ static int da7210_probe(struct snd_soc_codec *codec)
 	/*
 	 * make sure that DA7210 use bypass mode before start up
 	 */
-	da7210_write(codec, DA7210_STARTUP1, 0);
+	snd_soc_write(codec, DA7210_STARTUP1, 0);
-	da7210_write(codec, DA7210_PLL_DIV3,
+	snd_soc_write(codec, DA7210_PLL_DIV3,
 		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
 
 	/*
@@ -481,36 +825,70 @@ static int da7210_probe(struct snd_soc_codec *codec)
 	 */
 
 	/* Enable Left & Right MIC PGA and Mic Bias */
-	da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN);
+	snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN);
-	da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN);
+	snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN);
 
 	/* Enable Left and Right input PGA */
-	da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN);
+	snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN);
-	da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN);
+	snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN);
 
 	/* Enable Left and Right ADC */
-	da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN);
+	snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN);
 
 	/*
 	 * DAC settings
 	 */
 
 	/* Enable Left and Right DAC */
-	da7210_write(codec, DA7210_DAC_SEL,
+	snd_soc_write(codec, DA7210_DAC_SEL,
 		     DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN |
 		     DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN);
 
 	/* Enable Left and Right out PGA */
-	da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN);
+	snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN);
-	da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN);
+	snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN);
 
 	/* Enable Left and Right HeadPhone PGA */
-	da7210_write(codec, DA7210_HP_CFG,
+	snd_soc_write(codec, DA7210_HP_CFG,
 		     DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN |
 		     DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN);
 
+	/* Enable ramp mode for DAC gain update */
+	snd_soc_write(codec, DA7210_SOFTMUTE, DA7210_RAMP_EN);
+
+	/*
+	 * For DA7210 codec, there are two ways to enable/disable analog IOs
+	 * and ADC/DAC,
+	 * (1) Using "Enable Bit" of register associated with that IO
+	 * (or ADC/DAC)
+	 *	e.g. Mic Left can be enabled using bit 7 of MIC_L(0x7) reg
+	 *
+	 * (2) Using "Standby Bit" of STARTUP2 or STARTUP3 register
+	 *	e.g. Mic left can be put to STANDBY using bit 0 of STARTUP3(0x5)
+	 *
+	 * Out of these two methods, the one using STANDBY bits is preferred
+	 * way to enable/disable individual blocks. This is because STANDBY
+	 * registers are part of system controller which allows system power
+	 * up/down in a controlled, pop-free manner. Also, as per application
+	 * note of DA7210, STANDBY register bits are only effective if a
+	 * particular IO (or ADC/DAC) is already enabled using enable/disable
+	 * register bits. Keeping these things in mind, current DAPM
+	 * implementation manipulates only STANDBY bits.
+	 *
+	 * Overall implementation can be outlined as below,
+	 *
+	 * - "Enable bit" of an IO or ADC/DAC is used to enable it in probe()
+	 * - "STANDBY bit" is controlled by DAPM
+	 */
+
+	/* Enable Line out amplifiers */
+	snd_soc_write(codec, DA7210_OUT1_L, DA7210_OUT1_L_EN);
+	snd_soc_write(codec, DA7210_OUT1_R, DA7210_OUT1_R_EN);
+	snd_soc_write(codec, DA7210_OUT2, DA7210_OUT2_EN |
+		     DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R);
+
 	/* Diable PLL and bypass it */
-	da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
+	snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
 
 	/*
 	 * If 48kHz sound came, it use bypass mode,
@@ -521,25 +899,22 @@ static int da7210_probe(struct snd_soc_codec *codec)
 	 * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit.
 	 *   see da7210_hw_params
 	 */
-	da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
+	snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */
-	da7210_write(codec, DA7210_PLL_DIV2, 0x99);
+	snd_soc_write(codec, DA7210_PLL_DIV2, 0x99);
-	da7210_write(codec, DA7210_PLL_DIV3, 0x0A |
+	snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A |
 		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
 	snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN);
 
 	/* As suggested by Dialog */
-	da7210_write(codec, DA7210_A_HID_UNLOCK,	0x8B); /* unlock */
+	snd_soc_write(codec, DA7210_A_HID_UNLOCK,	0x8B); /* unlock */
-	da7210_write(codec, DA7210_A_TEST_UNLOCK,	0xB4);
+	snd_soc_write(codec, DA7210_A_TEST_UNLOCK,	0xB4);
-	da7210_write(codec, DA7210_A_PLL1,		0x01);
+	snd_soc_write(codec, DA7210_A_PLL1,		0x01);
-	da7210_write(codec, DA7210_A_CP_MODE,		0x7C);
+	snd_soc_write(codec, DA7210_A_CP_MODE,		0x7C);
-	da7210_write(codec, DA7210_A_HID_UNLOCK,	0x00); /* re-lock */
+	snd_soc_write(codec, DA7210_A_HID_UNLOCK,	0x00); /* re-lock */
-	da7210_write(codec, DA7210_A_TEST_UNLOCK,	0x00);
+	snd_soc_write(codec, DA7210_A_TEST_UNLOCK,	0x00);
 
 	/* Activate all enabled subsystem */
-	da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
+	snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
-
-	snd_soc_add_controls(codec, da7210_snd_controls,
-			     ARRAY_SIZE(da7210_snd_controls));
 
 	dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
 
@@ -548,11 +923,18 @@ static int da7210_probe(struct snd_soc_codec *codec)
 
 static struct snd_soc_codec_driver soc_codec_dev_da7210 = {
 	.probe			= da7210_probe,
-	.read			= da7210_read,
-	.write			= da7210_write,
 	.reg_cache_size		= ARRAY_SIZE(da7210_reg),
 	.reg_word_size		= sizeof(u8),
 	.reg_cache_default	= da7210_reg,
+	.volatile_register	= da7210_volatile_register,
+
+	.controls		= da7210_snd_controls,
+	.num_controls		= ARRAY_SIZE(da7210_snd_controls),
+
+	.dapm_widgets		= da7210_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(da7210_dapm_widgets),
+	.dapm_routes		= da7210_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(da7210_audio_map),
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -567,7 +949,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, da7210);
-	da7210->control_data = i2c;
 	da7210->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/lm4857.c

@@ -3,7 +3,7 @@
  *
  * Copyright 2007 Wolfson Microelectronics PLC.
  * Author: Graeme Gregory
- *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ *         graeme.gregory@wolfsonmicro.com
  * Copyright 2011 Lars-Peter Clausen <lars@metafoo.de>
  *
  *  This program is free software; you can redistribute  it and/or modify it

sound/soc/codecs/max98088.c

@@ -40,7 +40,6 @@ struct max98088_cdata {
 
 struct max98088_priv {
        enum max98088_type devtype;
-       void *control_data;
        struct max98088_pdata *pdata;
        unsigned int sysclk;
        struct max98088_cdata dai[2];
@@ -1697,12 +1696,18 @@ static struct snd_soc_dai_driver max98088_dai[] = {
 }
 };
 
-static int max98088_get_channel(const char *name)
+static const char *eq_mode_name[] = {"EQ1 Mode", "EQ2 Mode"};
+
+static int max98088_get_channel(struct snd_soc_codec *codec, const char *name)
 {
-       if (strcmp(name, "EQ1 Mode") == 0)
+	int i;
-               return 0;
+
-       if (strcmp(name, "EQ2 Mode") == 0)
+	for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++)
-               return 1;
+		if (strcmp(name, eq_mode_name[i]) == 0)
+			return i;
+
+	/* Shouldn't happen */
+	dev_err(codec->dev, "Bad EQ channel name '%s'\n", name);
 	return -EINVAL;
 }
 
@@ -1807,10 +1812,13 @@ static int max98088_put_eq_enum(struct snd_kcontrol *kcontrol,
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
        struct max98088_pdata *pdata = max98088->pdata;
-       int channel = max98088_get_channel(kcontrol->id.name);
+       int channel = max98088_get_channel(codec, kcontrol->id.name);
        struct max98088_cdata *cdata;
        int sel = ucontrol->value.integer.value[0];
 
+       if (channel < 0)
+	       return channel;
+
        cdata = &max98088->dai[channel];
 
        if (sel >= pdata->eq_cfgcnt)
@@ -1835,9 +1843,12 @@ static int max98088_get_eq_enum(struct snd_kcontrol *kcontrol,
 {
        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
        struct max98088_priv *max98088 = snd_soc_codec_get_drvdata(codec);
-       int channel = max98088_get_channel(kcontrol->id.name);
+       int channel = max98088_get_channel(codec, kcontrol->id.name);
        struct max98088_cdata *cdata;
 
+       if (channel < 0)
+	       return channel;
+
        cdata = &max98088->dai[channel];
        ucontrol->value.enumerated.item[0] = cdata->eq_sel;
        return 0;
@@ -1852,17 +1863,17 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec)
        int i, j;
        const char **t;
        int ret;
-
        struct snd_kcontrol_new controls[] = {
-               SOC_ENUM_EXT("EQ1 Mode",
+               SOC_ENUM_EXT((char *)eq_mode_name[0],
                        max98088->eq_enum,
                        max98088_get_eq_enum,
                        max98088_put_eq_enum),
-               SOC_ENUM_EXT("EQ2 Mode",
+               SOC_ENUM_EXT((char *)eq_mode_name[1],
                        max98088->eq_enum,
                        max98088_get_eq_enum,
                        max98088_put_eq_enum),
        };
+       BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(eq_mode_name));
 
        cfg = pdata->eq_cfg;
        cfgcnt = pdata->eq_cfgcnt;
@@ -2066,7 +2077,6 @@ static int max98088_i2c_probe(struct i2c_client *i2c,
        max98088->devtype = id->driver_data;
 
        i2c_set_clientdata(i2c, max98088);
-       max98088->control_data = i2c;
        max98088->pdata = i2c->dev.platform_data;
 
        ret = snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/max98095.c

@@ -40,7 +40,6 @@ struct max98095_cdata {
 
 struct max98095_priv {
 	enum max98095_type devtype;
-	void *control_data;
 	struct max98095_pdata *pdata;
 	unsigned int sysclk;
 	struct max98095_cdata dai[3];
@@ -618,14 +617,13 @@ static int max98095_volatile(struct snd_soc_codec *codec, unsigned int reg)
 static int max98095_hw_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
-	u8 data[2];
+	int ret;
 
-	data[0] = reg;
+	codec->cache_bypass = 1;
-	data[1] = value;
+	ret = snd_soc_write(codec, reg, value);
-	if (codec->hw_write(codec->control_data, data, 2) == 2)
+	codec->cache_bypass = 0;
-		return 0;
+
-	else
+	return ret ? -EIO : 0;
-		return -EIO;
 }
 
 /*
@@ -1992,12 +1990,19 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec)
 		dev_err(codec->dev, "Failed to add EQ control: %d\n", ret);
 }
 
-static int max98095_get_bq_channel(const char *name)
+static const char *bq_mode_name[] = {"Biquad1 Mode", "Biquad2 Mode"};
+
+static int max98095_get_bq_channel(struct snd_soc_codec *codec,
+				   const char *name)
 {
-	if (strcmp(name, "Biquad1 Mode") == 0)
+	int i;
-		return 0;
+
-	if (strcmp(name, "Biquad2 Mode") == 0)
+	for (i = 0; i < ARRAY_SIZE(bq_mode_name); i++)
-		return 1;
+		if (strcmp(name, bq_mode_name[i]) == 0)
+			return i;
+
+	/* Shouldn't happen */
+	dev_err(codec->dev, "Bad biquad channel name '%s'\n", name);
 	return -EINVAL;
 }
 
@@ -2007,14 +2012,15 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
 	struct max98095_pdata *pdata = max98095->pdata;
-	int channel = max98095_get_bq_channel(kcontrol->id.name);
+	int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
 	struct max98095_cdata *cdata;
 	int sel = ucontrol->value.integer.value[0];
 	struct max98095_biquad_cfg *coef_set;
 	int fs, best, best_val, i;
 	int regmask, regsave;
 
-	BUG_ON(channel > 1);
+	if (channel < 0)
+		return channel;
 
 	if (!pdata || !max98095->bq_textcnt)
 		return 0;
@@ -2066,9 +2072,12 @@ static int max98095_get_bq_enum(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec);
-	int channel = max98095_get_bq_channel(kcontrol->id.name);
+	int channel = max98095_get_bq_channel(codec, kcontrol->id.name);
 	struct max98095_cdata *cdata;
 
+	if (channel < 0)
+		return channel;
+
 	cdata = &max98095->dai[channel];
 	ucontrol->value.enumerated.item[0] = cdata->bq_sel;
 
@@ -2086,15 +2095,16 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec)
 	int ret;
 
 	struct snd_kcontrol_new controls[] = {
-		SOC_ENUM_EXT("Biquad1 Mode",
+		SOC_ENUM_EXT((char *)bq_mode_name[0],
 			max98095->bq_enum,
 			max98095_get_bq_enum,
 			max98095_put_bq_enum),
-		SOC_ENUM_EXT("Biquad2 Mode",
+		SOC_ENUM_EXT((char *)bq_mode_name[1],
 			max98095->bq_enum,
 			max98095_get_bq_enum,
 			max98095_put_bq_enum),
 	};
+	BUILD_BUG_ON(ARRAY_SIZE(controls) != ARRAY_SIZE(bq_mode_name));
 
 	cfg = pdata->bq_cfg;
 	cfgcnt = pdata->bq_cfgcnt;
@@ -2337,7 +2347,6 @@ static int max98095_i2c_probe(struct i2c_client *i2c,
 
 	max98095->devtype = id->driver_data;
 	i2c_set_clientdata(i2c, max98095);
-	max98095->control_data = i2c;
 	max98095->pdata = i2c->dev.platform_data;
 
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98095,

sound/soc/codecs/rt5631.h

@@ -0,0 +1,701 @@
+#ifndef __RTCODEC5631_H__
+#define __RTCODEC5631_H__
+
+
+#define RT5631_RESET				0x00
+#define RT5631_SPK_OUT_VOL			0x02
+#define RT5631_HP_OUT_VOL			0x04
+#define RT5631_MONO_AXO_1_2_VOL		0x06
+#define RT5631_AUX_IN_VOL			0x0A
+#define RT5631_STEREO_DAC_VOL_1		0x0C
+#define RT5631_MIC_CTRL_1			0x0E
+#define RT5631_STEREO_DAC_VOL_2		0x10
+#define RT5631_ADC_CTRL_1			0x12
+#define RT5631_ADC_REC_MIXER			0x14
+#define RT5631_ADC_CTRL_2			0x16
+#define RT5631_VDAC_DIG_VOL			0x18
+#define RT5631_OUTMIXER_L_CTRL			0x1A
+#define RT5631_OUTMIXER_R_CTRL			0x1C
+#define RT5631_AXO1MIXER_CTRL			0x1E
+#define RT5631_AXO2MIXER_CTRL			0x20
+#define RT5631_MIC_CTRL_2			0x22
+#define RT5631_DIG_MIC_CTRL			0x24
+#define RT5631_MONO_INPUT_VOL			0x26
+#define RT5631_SPK_MIXER_CTRL			0x28
+#define RT5631_SPK_MONO_OUT_CTRL		0x2A
+#define RT5631_SPK_MONO_HP_OUT_CTRL		0x2C
+#define RT5631_SDP_CTRL				0x34
+#define RT5631_MONO_SDP_CTRL			0x36
+#define RT5631_STEREO_AD_DA_CLK_CTRL		0x38
+#define RT5631_PWR_MANAG_ADD1		0x3A
+#define RT5631_PWR_MANAG_ADD2		0x3B
+#define RT5631_PWR_MANAG_ADD3		0x3C
+#define RT5631_PWR_MANAG_ADD4		0x3E
+#define RT5631_GEN_PUR_CTRL_REG		0x40
+#define RT5631_GLOBAL_CLK_CTRL			0x42
+#define RT5631_PLL_CTRL				0x44
+#define RT5631_INT_ST_IRQ_CTRL_1		0x48
+#define RT5631_INT_ST_IRQ_CTRL_2		0x4A
+#define RT5631_GPIO_CTRL			0x4C
+#define RT5631_MISC_CTRL			0x52
+#define RT5631_DEPOP_FUN_CTRL_1		0x54
+#define RT5631_DEPOP_FUN_CTRL_2		0x56
+#define RT5631_JACK_DET_CTRL			0x5A
+#define RT5631_SOFT_VOL_CTRL			0x5C
+#define RT5631_ALC_CTRL_1			0x64
+#define RT5631_ALC_CTRL_2			0x65
+#define RT5631_ALC_CTRL_3			0x66
+#define RT5631_PSEUDO_SPATL_CTRL		0x68
+#define RT5631_INDEX_ADD			0x6A
+#define RT5631_INDEX_DATA			0x6C
+#define RT5631_EQ_CTRL				0x6E
+#define RT5631_VENDOR_ID			0x7A
+#define RT5631_VENDOR_ID1			0x7C
+#define RT5631_VENDOR_ID2			0x7E
+
+/* Index of Codec Private Register definition */
+#define RT5631_EQ_BW_LOP			0x00
+#define RT5631_EQ_GAIN_LOP			0x01
+#define RT5631_EQ_FC_BP1			0x02
+#define RT5631_EQ_BW_BP1			0x03
+#define RT5631_EQ_GAIN_BP1			0x04
+#define RT5631_EQ_FC_BP2			0x05
+#define RT5631_EQ_BW_BP2			0x06
+#define RT5631_EQ_GAIN_BP2			0x07
+#define RT5631_EQ_FC_BP3			0x08
+#define RT5631_EQ_BW_BP3			0x09
+#define RT5631_EQ_GAIN_BP3			0x0a
+#define RT5631_EQ_BW_HIP			0x0b
+#define RT5631_EQ_GAIN_HIP			0x0c
+#define RT5631_EQ_HPF_A1			0x0d
+#define RT5631_EQ_HPF_A2			0x0e
+#define RT5631_EQ_HPF_GAIN			0x0f
+#define RT5631_EQ_PRE_VOL_CTRL			0x11
+#define RT5631_EQ_POST_VOL_CTRL		0x12
+#define RT5631_TEST_MODE_CTRL			0x39
+#define RT5631_CP_INTL_REG2			0x45
+#define RT5631_ADDA_MIXER_INTL_REG3		0x52
+#define RT5631_SPK_INTL_CTRL			0x56
+
+
+/* global definition */
+#define RT5631_L_MUTE					(0x1 << 15)
+#define RT5631_L_MUTE_SHIFT				15
+#define RT5631_L_EN					(0x1 << 14)
+#define RT5631_L_EN_SHIFT				14
+#define RT5631_R_MUTE					(0x1 << 7)
+#define RT5631_R_MUTE_SHIFT				7
+#define RT5631_R_EN					(0x1 << 6)
+#define RT5631_R_EN_SHIFT				6
+#define RT5631_VOL_MASK				0x1f
+#define RT5631_L_VOL_SHIFT				8
+#define RT5631_R_VOL_SHIFT				0
+
+/* Speaker Output Control(0x02) */
+#define RT5631_SPK_L_VOL_SEL_MASK			(0x1 << 14)
+#define RT5631_SPK_L_VOL_SEL_VMID			(0x0 << 14)
+#define RT5631_SPK_L_VOL_SEL_SPKMIX_L			(0x1 << 14)
+#define RT5631_SPK_R_VOL_SEL_MASK			(0x1 << 6)
+#define RT5631_SPK_R_VOL_SEL_VMID			(0x0 << 6)
+#define RT5631_SPK_R_VOL_SEL_SPKMIX_R			(0x1 << 6)
+
+/* Headphone Output Control(0x04) */
+#define RT5631_HP_L_VOL_SEL_MASK			(0x1 << 14)
+#define RT5631_HP_L_VOL_SEL_VMID			(0x0 << 14)
+#define RT5631_HP_L_VOL_SEL_OUTMIX_L			(0x1 << 14)
+#define RT5631_HP_R_VOL_SEL_MASK			(0x1 << 6)
+#define RT5631_HP_R_VOL_SEL_VMID			(0x0 << 6)
+#define RT5631_HP_R_VOL_SEL_OUTMIX_R			(0x1 << 6)
+
+/* Output Control for AUXOUT/MONO(0x06) */
+#define RT5631_AUXOUT_1_VOL_SEL_MASK			(0x1 << 14)
+#define RT5631_AUXOUT_1_VOL_SEL_VMID			(0x0 << 14)
+#define RT5631_AUXOUT_1_VOL_SEL_OUTMIX_L		(0x1 << 14)
+#define RT5631_MUTE_MONO				(0x1 << 13)
+#define RT5631_MUTE_MONO_SHIFT			13
+#define RT5631_AUXOUT_2_VOL_SEL_MASK			(0x1 << 6)
+#define RT5631_AUXOUT_2_VOL_SEL_VMID			(0x0 << 6)
+#define RT5631_AUXOUT_2_VOL_SEL_OUTMIX_R		(0x1 << 6)
+
+/* Microphone Input Control 1(0x0E) */
+#define RT5631_MIC1_DIFF_INPUT_CTRL			(0x1 << 15)
+#define RT5631_MIC1_DIFF_INPUT_SHIFT			15
+#define RT5631_MIC2_DIFF_INPUT_CTRL			(0x1 << 7)
+#define RT5631_MIC2_DIFF_INPUT_SHIFT			7
+
+/* Stereo DAC Digital Volume2(0x10) */
+#define RT5631_DAC_VOL_MASK				0xff
+
+/* ADC Recording Mixer Control(0x14) */
+#define RT5631_M_OUTMIXER_L_TO_RECMIXER_L		(0x1 << 15)
+#define RT5631_M_OUTMIXL_RECMIXL_BIT			15
+#define RT5631_M_MIC1_TO_RECMIXER_L			(0x1 << 14)
+#define RT5631_M_MIC1_RECMIXL_BIT			14
+#define RT5631_M_AXIL_TO_RECMIXER_L			(0x1 << 13)
+#define RT5631_M_AXIL_RECMIXL_BIT			13
+#define RT5631_M_MONO_IN_TO_RECMIXER_L		(0x1 << 12)
+#define RT5631_M_MONO_IN_RECMIXL_BIT			12
+#define RT5631_M_OUTMIXER_R_TO_RECMIXER_R		(0x1 << 7)
+#define RT5631_M_OUTMIXR_RECMIXR_BIT			7
+#define RT5631_M_MIC2_TO_RECMIXER_R			(0x1 << 6)
+#define RT5631_M_MIC2_RECMIXR_BIT			6
+#define RT5631_M_AXIR_TO_RECMIXER_R			(0x1 << 5)
+#define RT5631_M_AXIR_RECMIXR_BIT			5
+#define RT5631_M_MONO_IN_TO_RECMIXER_R		(0x1 << 4)
+#define RT5631_M_MONO_IN_RECMIXR_BIT			4
+
+/* Left Output Mixer Control(0x1A) */
+#define RT5631_M_RECMIXER_L_TO_OUTMIXER_L		(0x1 << 15)
+#define RT5631_M_RECMIXL_OUTMIXL_BIT			15
+#define RT5631_M_RECMIXER_R_TO_OUTMIXER_L		(0x1 << 14)
+#define RT5631_M_RECMIXR_OUTMIXL_BIT			14
+#define RT5631_M_DAC_L_TO_OUTMIXER_L			(0x1 << 13)
+#define RT5631_M_DACL_OUTMIXL_BIT			13
+#define RT5631_M_MIC1_TO_OUTMIXER_L			(0x1 << 12)
+#define RT5631_M_MIC1_OUTMIXL_BIT			12
+#define RT5631_M_MIC2_TO_OUTMIXER_L			(0x1 << 11)
+#define RT5631_M_MIC2_OUTMIXL_BIT			11
+#define RT5631_M_MONO_IN_P_TO_OUTMIXER_L		(0x1 << 10)
+#define RT5631_M_MONO_INP_OUTMIXL_BIT		10
+#define RT5631_M_AXIL_TO_OUTMIXER_L			(0x1 << 9)
+#define RT5631_M_AXIL_OUTMIXL_BIT			9
+#define RT5631_M_AXIR_TO_OUTMIXER_L			(0x1 << 8)
+#define RT5631_M_AXIR_OUTMIXL_BIT			8
+#define RT5631_M_VDAC_TO_OUTMIXER_L			(0x1 << 7)
+#define RT5631_M_VDAC_OUTMIXL_BIT			7
+
+/* Right Output Mixer Control(0x1C) */
+#define RT5631_M_RECMIXER_L_TO_OUTMIXER_R		(0x1 << 15)
+#define RT5631_M_RECMIXL_OUTMIXR_BIT			15
+#define RT5631_M_RECMIXER_R_TO_OUTMIXER_R		(0x1 << 14)
+#define RT5631_M_RECMIXR_OUTMIXR_BIT			14
+#define RT5631_M_DAC_R_TO_OUTMIXER_R			(0x1 << 13)
+#define RT5631_M_DACR_OUTMIXR_BIT			13
+#define RT5631_M_MIC1_TO_OUTMIXER_R			(0x1 << 12)
+#define RT5631_M_MIC1_OUTMIXR_BIT			12
+#define RT5631_M_MIC2_TO_OUTMIXER_R			(0x1 << 11)
+#define RT5631_M_MIC2_OUTMIXR_BIT			11
+#define RT5631_M_MONO_IN_N_TO_OUTMIXER_R		(0x1 << 10)
+#define RT5631_M_MONO_INN_OUTMIXR_BIT		10
+#define RT5631_M_AXIL_TO_OUTMIXER_R			(0x1 << 9)
+#define RT5631_M_AXIL_OUTMIXR_BIT			9
+#define RT5631_M_AXIR_TO_OUTMIXER_R			(0x1 << 8)
+#define RT5631_M_AXIR_OUTMIXR_BIT			8
+#define RT5631_M_VDAC_TO_OUTMIXER_R			(0x1 << 7)
+#define RT5631_M_VDAC_OUTMIXR_BIT			7
+
+/* Lout Mixer Control(0x1E) */
+#define RT5631_M_MIC1_TO_AXO1MIXER			(0x1 << 15)
+#define RT5631_M_MIC1_AXO1MIX_BIT			15
+#define RT5631_M_MIC2_TO_AXO1MIXER			(0x1 << 11)
+#define RT5631_M_MIC2_AXO1MIX_BIT			11
+#define RT5631_M_OUTMIXER_L_TO_AXO1MIXER		(0x1 << 7)
+#define RT5631_M_OUTMIXL_AXO1MIX_BIT			7
+#define RT5631_M_OUTMIXER_R_TO_AXO1MIXER		(0x1 << 6)
+#define RT5631_M_OUTMIXR_AXO1MIX_BIT			6
+
+/* Rout Mixer Control(0x20) */
+#define RT5631_M_MIC1_TO_AXO2MIXER			(0x1 << 15)
+#define RT5631_M_MIC1_AXO2MIX_BIT			15
+#define RT5631_M_MIC2_TO_AXO2MIXER			(0x1 << 11)
+#define RT5631_M_MIC2_AXO2MIX_BIT			11
+#define RT5631_M_OUTMIXER_L_TO_AXO2MIXER		(0x1 << 7)
+#define RT5631_M_OUTMIXL_AXO2MIX_BIT			7
+#define RT5631_M_OUTMIXER_R_TO_AXO2MIXER		(0x1 << 6)
+#define RT5631_M_OUTMIXR_AXO2MIX_BIT			6
+
+/* Micphone Input Control 2(0x22) */
+#define RT5631_MIC_BIAS_90_PRECNET_AVDD 1
+#define RT5631_MIC_BIAS_75_PRECNET_AVDD 2
+
+#define RT5631_MIC1_BOOST_CTRL_MASK			(0xf << 12)
+#define RT5631_MIC1_BOOST_CTRL_BYPASS		(0x0 << 12)
+#define RT5631_MIC1_BOOST_CTRL_20DB			(0x1 << 12)
+#define RT5631_MIC1_BOOST_CTRL_24DB			(0x2 << 12)
+#define RT5631_MIC1_BOOST_CTRL_30DB			(0x3 << 12)
+#define RT5631_MIC1_BOOST_CTRL_35DB			(0x4 << 12)
+#define RT5631_MIC1_BOOST_CTRL_40DB			(0x5 << 12)
+#define RT5631_MIC1_BOOST_CTRL_34DB			(0x6 << 12)
+#define RT5631_MIC1_BOOST_CTRL_50DB			(0x7 << 12)
+#define RT5631_MIC1_BOOST_CTRL_52DB			(0x8 << 12)
+#define RT5631_MIC1_BOOST_SHIFT			12
+
+#define RT5631_MIC2_BOOST_CTRL_MASK			(0xf << 8)
+#define RT5631_MIC2_BOOST_CTRL_BYPASS		(0x0 << 8)
+#define RT5631_MIC2_BOOST_CTRL_20DB			(0x1 << 8)
+#define RT5631_MIC2_BOOST_CTRL_24DB			(0x2 << 8)
+#define RT5631_MIC2_BOOST_CTRL_30DB			(0x3 << 8)
+#define RT5631_MIC2_BOOST_CTRL_35DB			(0x4 << 8)
+#define RT5631_MIC2_BOOST_CTRL_40DB			(0x5 << 8)
+#define RT5631_MIC2_BOOST_CTRL_34DB			(0x6 << 8)
+#define RT5631_MIC2_BOOST_CTRL_50DB			(0x7 << 8)
+#define RT5631_MIC2_BOOST_CTRL_52DB			(0x8 << 8)
+#define RT5631_MIC2_BOOST_SHIFT			8
+
+#define RT5631_MICBIAS1_VOLT_CTRL_MASK		(0x1 << 7)
+#define RT5631_MICBIAS1_VOLT_CTRL_90P			(0x0 << 7)
+#define RT5631_MICBIAS1_VOLT_CTRL_75P			(0x1 << 7)
+
+#define RT5631_MICBIAS1_S_C_DET_MASK			(0x1 << 6)
+#define RT5631_MICBIAS1_S_C_DET_DIS			(0x0 << 6)
+#define RT5631_MICBIAS1_S_C_DET_ENA			(0x1 << 6)
+
+#define RT5631_MICBIAS1_SHORT_CURR_DET_MASK		(0x3 << 4)
+#define RT5631_MICBIAS1_SHORT_CURR_DET_600UA	(0x0 << 4)
+#define RT5631_MICBIAS1_SHORT_CURR_DET_1500UA	(0x1 << 4)
+#define RT5631_MICBIAS1_SHORT_CURR_DET_2000UA	(0x2 << 4)
+
+#define RT5631_MICBIAS2_VOLT_CTRL_MASK		(0x1 << 3)
+#define RT5631_MICBIAS2_VOLT_CTRL_90P			(0x0 << 3)
+#define RT5631_MICBIAS2_VOLT_CTRL_75P			(0x1 << 3)
+
+#define RT5631_MICBIAS2_S_C_DET_MASK			(0x1 << 2)
+#define RT5631_MICBIAS2_S_C_DET_DIS			(0x0 << 2)
+#define RT5631_MICBIAS2_S_C_DET_ENA			(0x1 << 2)
+
+#define RT5631_MICBIAS2_SHORT_CURR_DET_MASK		(0x3)
+#define RT5631_MICBIAS2_SHORT_CURR_DET_600UA	(0x0)
+#define RT5631_MICBIAS2_SHORT_CURR_DET_1500UA	(0x1)
+#define RT5631_MICBIAS2_SHORT_CURR_DET_2000UA	(0x2)
+
+
+/* Digital Microphone Control(0x24) */
+#define RT5631_DMIC_ENA_MASK				(0x1 << 15)
+#define RT5631_DMIC_ENA_SHIFT				15
+/* DMIC_ENA: DMIC to ADC Digital filter */
+#define RT5631_DMIC_ENA				(0x1 << 15)
+/* DMIC_DIS: ADC mixer to ADC Digital filter */
+#define RT5631_DMIC_DIS					(0x0 << 15)
+#define RT5631_DMIC_L_CH_MUTE				(0x1 << 13)
+#define RT5631_DMIC_L_CH_MUTE_SHIFT			13
+#define RT5631_DMIC_R_CH_MUTE				(0x1 << 12)
+#define RT5631_DMIC_R_CH_MUTE_SHIFT			12
+#define RT5631_DMIC_L_CH_LATCH_MASK			(0x1 << 9)
+#define RT5631_DMIC_L_CH_LATCH_RISING			(0x1 << 9)
+#define RT5631_DMIC_L_CH_LATCH_FALLING		(0x0 << 9)
+#define RT5631_DMIC_R_CH_LATCH_MASK			(0x1 << 8)
+#define RT5631_DMIC_R_CH_LATCH_RISING			(0x1 << 8)
+#define RT5631_DMIC_R_CH_LATCH_FALLING		(0x0 << 8)
+#define RT5631_DMIC_CLK_CTRL_MASK			(0x3 << 4)
+#define RT5631_DMIC_CLK_CTRL_TO_128FS			(0x0 << 4)
+#define RT5631_DMIC_CLK_CTRL_TO_64FS			(0x1 << 4)
+#define RT5631_DMIC_CLK_CTRL_TO_32FS			(0x2 << 4)
+
+/* Microphone Input Volume(0x26) */
+#define RT5631_MONO_DIFF_INPUT_SHIFT			15
+
+/* Speaker Mixer Control(0x28) */
+#define RT5631_M_RECMIXER_L_TO_SPKMIXER_L		(0x1 << 15)
+#define RT5631_M_RECMIXL_SPKMIXL_BIT			15
+#define RT5631_M_MIC1_P_TO_SPKMIXER_L		(0x1 << 14)
+#define RT5631_M_MIC1P_SPKMIXL_BIT			14
+#define RT5631_M_DAC_L_TO_SPKMIXER_L			(0x1 << 13)
+#define RT5631_M_DACL_SPKMIXL_BIT			13
+#define RT5631_M_OUTMIXER_L_TO_SPKMIXER_L		(0x1 << 12)
+#define RT5631_M_OUTMIXL_SPKMIXL_BIT			12
+
+#define RT5631_M_RECMIXER_R_TO_SPKMIXER_R		(0x1 << 7)
+#define RT5631_M_RECMIXR_SPKMIXR_BIT			7
+#define RT5631_M_MIC2_P_TO_SPKMIXER_R		(0x1 << 6)
+#define RT5631_M_MIC2P_SPKMIXR_BIT			6
+#define RT5631_M_DAC_R_TO_SPKMIXER_R			(0x1 << 5)
+#define RT5631_M_DACR_SPKMIXR_BIT			5
+#define RT5631_M_OUTMIXER_R_TO_SPKMIXER_R		(0x1 << 4)
+#define RT5631_M_OUTMIXR_SPKMIXR_BIT			4
+
+/* Speaker/Mono Output Control(0x2A) */
+#define RT5631_M_SPKVOL_L_TO_SPOL_MIXER		(0x1 << 15)
+#define RT5631_M_SPKVOLL_SPOLMIX_BIT			15
+#define RT5631_M_SPKVOL_R_TO_SPOL_MIXER		(0x1 << 14)
+#define RT5631_M_SPKVOLR_SPOLMIX_BIT			14
+#define RT5631_M_SPKVOL_L_TO_SPOR_MIXER		(0x1 << 13)
+#define RT5631_M_SPKVOLL_SPORMIX_BIT			13
+#define RT5631_M_SPKVOL_R_TO_SPOR_MIXER		(0x1 << 12)
+#define RT5631_M_SPKVOLR_SPORMIX_BIT			12
+#define RT5631_M_OUTVOL_L_TO_MONOMIXER		(0x1 << 11)
+#define RT5631_M_OUTVOLL_MONOMIX_BIT			11
+#define RT5631_M_OUTVOL_R_TO_MONOMIXER		(0x1 << 10)
+#define RT5631_M_OUTVOLR_MONOMIX_BIT			10
+
+/* Speaker/Mono/HP Output Control(0x2C) */
+#define RT5631_SPK_L_MUX_SEL_MASK			(0x3 << 14)
+#define RT5631_SPK_L_MUX_SEL_SPKMIXER_L		(0x0 << 14)
+#define RT5631_SPK_L_MUX_SEL_MONO_IN			(0x1 << 14)
+#define RT5631_SPK_L_MUX_SEL_DAC_L			(0x3 << 14)
+#define RT5631_SPK_L_MUX_SEL_SHIFT			14
+
+#define RT5631_SPK_R_MUX_SEL_MASK			(0x3 << 10)
+#define RT5631_SPK_R_MUX_SEL_SPKMIXER_R		(0x0 << 10)
+#define RT5631_SPK_R_MUX_SEL_MONO_IN			(0x1 << 10)
+#define RT5631_SPK_R_MUX_SEL_DAC_R			(0x3 << 10)
+#define RT5631_SPK_R_MUX_SEL_SHIFT			10
+
+#define RT5631_MONO_MUX_SEL_MASK			(0x3 << 6)
+#define RT5631_MONO_MUX_SEL_MONOMIXER		(0x0 << 6)
+#define RT5631_MONO_MUX_SEL_MONO_IN			(0x1 << 6)
+#define RT5631_MONO_MUX_SEL_SHIFT			6
+
+#define RT5631_HP_L_MUX_SEL_MASK			(0x1 << 3)
+#define RT5631_HP_L_MUX_SEL_HPVOL_L			(0x0 << 3)
+#define RT5631_HP_L_MUX_SEL_DAC_L			(0x1 << 3)
+#define RT5631_HP_L_MUX_SEL_SHIFT			3
+
+#define RT5631_HP_R_MUX_SEL_MASK			(0x1 << 2)
+#define RT5631_HP_R_MUX_SEL_HPVOL_R			(0x0 << 2)
+#define RT5631_HP_R_MUX_SEL_DAC_R			(0x1 << 2)
+#define RT5631_HP_R_MUX_SEL_SHIFT			2
+
+/* Stereo I2S Serial Data Port Control(0x34) */
+#define RT5631_SDP_MODE_SEL_MASK			(0x1 << 15)
+#define RT5631_SDP_MODE_SEL_MASTER			(0x0 << 15)
+#define RT5631_SDP_MODE_SEL_SLAVE			(0x1 << 15)
+
+#define RT5631_SDP_ADC_CPS_SEL_MASK			(0x3 << 10)
+#define RT5631_SDP_ADC_CPS_SEL_OFF			(0x0 << 10)
+#define RT5631_SDP_ADC_CPS_SEL_U_LAW			(0x1 << 10)
+#define RT5631_SDP_ADC_CPS_SEL_A_LAW			(0x2 << 10)
+
+#define RT5631_SDP_DAC_CPS_SEL_MASK			(0x3 << 8)
+#define RT5631_SDP_DAC_CPS_SEL_OFF			(0x0 << 8)
+#define RT5631_SDP_DAC_CPS_SEL_U_LAW			(0x1 << 8)
+#define RT5631_SDP_DAC_CPS_SEL_A_LAW			(0x2 << 8)
+/* 0:Normal 1:Invert */
+#define RT5631_SDP_I2S_BCLK_POL_CTRL			(0x1 << 7)
+/* 0:Normal 1:Invert */
+#define RT5631_SDP_DAC_R_INV				(0x1 << 6)
+/* 0:ADC data appear at left phase of LRCK
+ * 1:ADC data appear at right phase of LRCK
+ */
+#define RT5631_SDP_ADC_DATA_L_R_SWAP			(0x1 << 5)
+/* 0:DAC data appear at left phase of LRCK
+ * 1:DAC data appear at right phase of LRCK
+ */
+#define RT5631_SDP_DAC_DATA_L_R_SWAP			(0x1 << 4)
+
+/* Data Length Slection */
+#define RT5631_SDP_I2S_DL_MASK				(0x3 << 2)
+#define RT5631_SDP_I2S_DL_16				(0x0 << 2)
+#define RT5631_SDP_I2S_DL_20				(0x1 << 2)
+#define RT5631_SDP_I2S_DL_24				(0x2 << 2)
+#define RT5631_SDP_I2S_DL_8				(0x3 << 2)
+
+/* PCM Data Format Selection */
+#define RT5631_SDP_I2S_DF_MASK				(0x3)
+#define RT5631_SDP_I2S_DF_I2S				(0x0)
+#define RT5631_SDP_I2S_DF_LEFT				(0x1)
+#define RT5631_SDP_I2S_DF_PCM_A			(0x2)
+#define RT5631_SDP_I2S_DF_PCM_B			(0x3)
+
+/* Stereo AD/DA Clock Control(0x38h) */
+#define RT5631_I2S_PRE_DIV_MASK			(0x7 << 13)
+#define RT5631_I2S_PRE_DIV_1				(0x0 << 13)
+#define RT5631_I2S_PRE_DIV_2				(0x1 << 13)
+#define RT5631_I2S_PRE_DIV_4				(0x2 << 13)
+#define RT5631_I2S_PRE_DIV_8				(0x3 << 13)
+#define RT5631_I2S_PRE_DIV_16				(0x4 << 13)
+#define RT5631_I2S_PRE_DIV_32				(0x5 << 13)
+/* CLOCK RELATIVE OF BCLK AND LCRK */
+#define RT5631_I2S_LRCK_SEL_N_BCLK_MASK		(0x1 << 12)
+#define RT5631_I2S_LRCK_SEL_64_BCLK			(0x0 << 12) /* 64FS */
+#define RT5631_I2S_LRCK_SEL_32_BCLK			(0x1 << 12) /* 32FS */
+
+#define RT5631_DAC_OSR_SEL_MASK			(0x3 << 10)
+#define RT5631_DAC_OSR_SEL_128FS			(0x3 << 10)
+#define RT5631_DAC_OSR_SEL_64FS			(0x3 << 10)
+#define RT5631_DAC_OSR_SEL_32FS			(0x3 << 10)
+#define RT5631_DAC_OSR_SEL_16FS			(0x3 << 10)
+
+#define RT5631_ADC_OSR_SEL_MASK			(0x3 << 8)
+#define RT5631_ADC_OSR_SEL_128FS			(0x3 << 8)
+#define RT5631_ADC_OSR_SEL_64FS			(0x3 << 8)
+#define RT5631_ADC_OSR_SEL_32FS			(0x3 << 8)
+#define RT5631_ADC_OSR_SEL_16FS			(0x3 << 8)
+
+#define RT5631_ADDA_FILTER_CLK_SEL_256FS		(0 << 7) /* 256FS */
+#define RT5631_ADDA_FILTER_CLK_SEL_384FS		(1 << 7) /* 384FS */
+
+/* Power managment addition 1 (0x3A) */
+#define RT5631_PWR_MAIN_I2S_EN			(0x1 << 15)
+#define RT5631_PWR_MAIN_I2S_BIT			15
+#define RT5631_PWR_CLASS_D				(0x1 << 12)
+#define RT5631_PWR_CLASS_D_BIT			12
+#define RT5631_PWR_ADC_L_CLK				(0x1 << 11)
+#define RT5631_PWR_ADC_L_CLK_BIT			11
+#define RT5631_PWR_ADC_R_CLK				(0x1 << 10)
+#define RT5631_PWR_ADC_R_CLK_BIT			10
+#define RT5631_PWR_DAC_L_CLK				(0x1 << 9)
+#define RT5631_PWR_DAC_L_CLK_BIT			9
+#define RT5631_PWR_DAC_R_CLK				(0x1 << 8)
+#define RT5631_PWR_DAC_R_CLK_BIT			8
+#define RT5631_PWR_DAC_REF				(0x1 << 7)
+#define RT5631_PWR_DAC_REF_BIT			7
+#define RT5631_PWR_DAC_L_TO_MIXER			(0x1 << 6)
+#define RT5631_PWR_DAC_L_TO_MIXER_BIT		6
+#define RT5631_PWR_DAC_R_TO_MIXER			(0x1 << 5)
+#define RT5631_PWR_DAC_R_TO_MIXER_BIT		5
+
+/* Power managment addition 2 (0x3B) */
+#define RT5631_PWR_OUTMIXER_L				(0x1 << 15)
+#define RT5631_PWR_OUTMIXER_L_BIT			15
+#define RT5631_PWR_OUTMIXER_R				(0x1 << 14)
+#define RT5631_PWR_OUTMIXER_R_BIT			14
+#define RT5631_PWR_SPKMIXER_L				(0x1 << 13)
+#define RT5631_PWR_SPKMIXER_L_BIT			13
+#define RT5631_PWR_SPKMIXER_R				(0x1 << 12)
+#define RT5631_PWR_SPKMIXER_R_BIT			12
+#define RT5631_PWR_RECMIXER_L				(0x1 << 11)
+#define RT5631_PWR_RECMIXER_L_BIT			11
+#define RT5631_PWR_RECMIXER_R				(0x1 << 10)
+#define RT5631_PWR_RECMIXER_R_BIT			10
+#define RT5631_PWR_MIC1_BOOT_GAIN			(0x1 << 5)
+#define RT5631_PWR_MIC1_BOOT_GAIN_BIT		5
+#define RT5631_PWR_MIC2_BOOT_GAIN			(0x1 << 4)
+#define RT5631_PWR_MIC2_BOOT_GAIN_BIT		4
+#define RT5631_PWR_MICBIAS1_VOL			(0x1 << 3)
+#define RT5631_PWR_MICBIAS1_VOL_BIT			3
+#define RT5631_PWR_MICBIAS2_VOL			(0x1 << 2)
+#define RT5631_PWR_MICBIAS2_VOL_BIT			2
+#define RT5631_PWR_PLL1				(0x1 << 1)
+#define RT5631_PWR_PLL1_BIT				1
+#define RT5631_PWR_PLL2				(0x1 << 0)
+#define RT5631_PWR_PLL2_BIT				0
+
+/* Power managment addition 3(0x3C) */
+#define RT5631_PWR_VREF				(0x1 << 15)
+#define RT5631_PWR_VREF_BIT				15
+#define RT5631_PWR_FAST_VREF_CTRL			(0x1 << 14)
+#define RT5631_PWR_FAST_VREF_CTRL_BIT			14
+#define RT5631_PWR_MAIN_BIAS				(0x1 << 13)
+#define RT5631_PWR_MAIN_BIAS_BIT			13
+#define RT5631_PWR_AXO1MIXER				(0x1 << 11)
+#define RT5631_PWR_AXO1MIXER_BIT			11
+#define RT5631_PWR_AXO2MIXER				(0x1 << 10)
+#define RT5631_PWR_AXO2MIXER_BIT			10
+#define RT5631_PWR_MONOMIXER				(0x1 << 9)
+#define RT5631_PWR_MONOMIXER_BIT			9
+#define RT5631_PWR_MONO_DEPOP_DIS			(0x1 << 8)
+#define RT5631_PWR_MONO_DEPOP_DIS_BIT		8
+#define RT5631_PWR_MONO_AMP_EN			(0x1 << 7)
+#define RT5631_PWR_MONO_AMP_EN_BIT			7
+#define RT5631_PWR_CHARGE_PUMP			(0x1 << 4)
+#define RT5631_PWR_CHARGE_PUMP_BIT			4
+#define RT5631_PWR_HP_L_AMP				(0x1 << 3)
+#define RT5631_PWR_HP_L_AMP_BIT			3
+#define RT5631_PWR_HP_R_AMP				(0x1 << 2)
+#define RT5631_PWR_HP_R_AMP_BIT			2
+#define RT5631_PWR_HP_DEPOP_DIS			(0x1 << 1)
+#define RT5631_PWR_HP_DEPOP_DIS_BIT			1
+#define RT5631_PWR_HP_AMP_DRIVING			(0x1 << 0)
+#define RT5631_PWR_HP_AMP_DRIVING_BIT		0
+
+/* Power managment addition 4(0x3E) */
+#define RT5631_PWR_SPK_L_VOL				(0x1 << 15)
+#define RT5631_PWR_SPK_L_VOL_BIT			15
+#define RT5631_PWR_SPK_R_VOL				(0x1 << 14)
+#define RT5631_PWR_SPK_R_VOL_BIT			14
+#define RT5631_PWR_LOUT_VOL				(0x1 << 13)
+#define RT5631_PWR_LOUT_VOL_BIT			13
+#define RT5631_PWR_ROUT_VOL				(0x1 << 12)
+#define RT5631_PWR_ROUT_VOL_BIT			12
+#define RT5631_PWR_HP_L_OUT_VOL			(0x1 << 11)
+#define RT5631_PWR_HP_L_OUT_VOL_BIT			11
+#define RT5631_PWR_HP_R_OUT_VOL			(0x1 << 10)
+#define RT5631_PWR_HP_R_OUT_VOL_BIT			10
+#define RT5631_PWR_AXIL_IN_VOL				(0x1 << 9)
+#define RT5631_PWR_AXIL_IN_VOL_BIT			9
+#define RT5631_PWR_AXIR_IN_VOL			(0x1 << 8)
+#define RT5631_PWR_AXIR_IN_VOL_BIT			8
+#define RT5631_PWR_MONO_IN_P_VOL			(0x1 << 7)
+#define RT5631_PWR_MONO_IN_P_VOL_BIT			7
+#define RT5631_PWR_MONO_IN_N_VOL			(0x1 << 6)
+#define RT5631_PWR_MONO_IN_N_VOL_BIT			6
+
+/* General Purpose Control Register(0x40) */
+#define RT5631_SPK_AMP_AUTO_RATIO_EN			(0x1 << 15)
+
+#define RT5631_SPK_AMP_RATIO_CTRL_MASK		(0x7 << 12)
+#define RT5631_SPK_AMP_RATIO_CTRL_2_34		(0x0 << 12) /* 7.40DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_99		(0x1 << 12) /* 5.99DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_68		(0x2 << 12) /* 4.50DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_56		(0x3 << 12) /* 3.86DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_44		(0x4 << 12) /* 3.16DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_27		(0x5 << 12) /* 2.10DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_09		(0x6 << 12) /* 0.80DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_1_00		(0x7 << 12) /* 0.00DB */
+#define RT5631_SPK_AMP_RATIO_CTRL_SHIFT		12
+
+#define RT5631_STEREO_DAC_HI_PASS_FILT_EN		(0x1 << 11)
+#define RT5631_STEREO_ADC_HI_PASS_FILT_EN		(0x1 << 10)
+/* Select ADC Wind Filter Clock type */
+#define RT5631_ADC_WIND_FILT_MASK			(0x3 << 4)
+#define RT5631_ADC_WIND_FILT_8_16_32K			(0x0 << 4) /*8/16/32k*/
+#define RT5631_ADC_WIND_FILT_11_22_44K		(0x1 << 4) /*11/22/44k*/
+#define RT5631_ADC_WIND_FILT_12_24_48K		(0x2 << 4) /*12/24/48k*/
+#define RT5631_ADC_WIND_FILT_EN			(0x1 << 3)
+/* SelectADC Wind Filter Corner Frequency */
+#define RT5631_ADC_WIND_CNR_FREQ_MASK	(0x7 << 0)
+#define RT5631_ADC_WIND_CNR_FREQ_82_113_122	(0x0 << 0) /* 82/113/122 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_102_141_153 (0x1 << 0) /* 102/141/153 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_131_180_156 (0x2 << 0) /* 131/180/156 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_163_225_245 (0x3 << 0) /* 163/225/245 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_204_281_306 (0x4 << 0) /* 204/281/306 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_261_360_392 (0x5 << 0) /* 261/360/392 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_327_450_490 (0x6 << 0) /* 327/450/490 Hz */
+#define RT5631_ADC_WIND_CNR_FREQ_408_563_612 (0x7 << 0) /* 408/563/612 Hz */
+
+/* Global Clock Control Register(0x42) */
+#define RT5631_SYSCLK_SOUR_SEL_MASK			(0x3 << 14)
+#define RT5631_SYSCLK_SOUR_SEL_MCLK			(0x0 << 14)
+#define RT5631_SYSCLK_SOUR_SEL_PLL			(0x1 << 14)
+#define RT5631_SYSCLK_SOUR_SEL_PLL_TCK		(0x2 << 14)
+
+#define RT5631_PLLCLK_SOUR_SEL_MASK			(0x3 << 12)
+#define RT5631_PLLCLK_SOUR_SEL_MCLK			(0x0 << 12)
+#define RT5631_PLLCLK_SOUR_SEL_BCLK			(0x1 << 12)
+#define RT5631_PLLCLK_SOUR_SEL_VBCLK			(0x2 << 12)
+
+#define RT5631_PLLCLK_PRE_DIV1				(0x0 << 11)
+#define RT5631_PLLCLK_PRE_DIV2				(0x1 << 11)
+
+/* PLL Control(0x44) */
+#define RT5631_PLL_CTRL_M_VAL(m)			((m)&0xf)
+#define RT5631_PLL_CTRL_K_VAL(k)			(((k)&0x7) << 4)
+#define RT5631_PLL_CTRL_N_VAL(n)			(((n)&0xff) << 8)
+
+/* Internal Status and IRQ Control2(0x4A) */
+#define RT5631_ADC_DATA_SEL_MASK			(0x3 << 14)
+#define RT5631_ADC_DATA_SEL_Disable			(0x0 << 14)
+#define RT5631_ADC_DATA_SEL_MIC1			(0x1 << 14)
+#define RT5631_ADC_DATA_SEL_MIC1_SHIFT		14
+#define RT5631_ADC_DATA_SEL_MIC2			(0x2 << 14)
+#define RT5631_ADC_DATA_SEL_MIC2_SHIFT		15
+#define RT5631_ADC_DATA_SEL_STO			(0x3 << 14)
+#define RT5631_ADC_DATA_SEL_SHIFT			14
+
+/* GPIO Pin Configuration(0x4C) */
+#define RT5631_GPIO_PIN_FUN_SEL_MASK			(0x1 << 15)
+#define RT5631_GPIO_PIN_FUN_SEL_IRQ			(0x1 << 15)
+#define RT5631_GPIO_PIN_FUN_SEL_GPIO_DIMC		(0x0 << 15)
+
+#define RT5631_GPIO_DMIC_FUN_SEL_MASK		(0x1 << 3)
+#define RT5631_GPIO_DMIC_FUN_SEL_DIMC		(0x1 << 3)
+#define RT5631_GPIO_DMIC_FUN_SEL_GPIO			(0x0 << 3)
+
+#define RT5631_GPIO_PIN_CON_MASK			(0x1 << 2)
+#define RT5631_GPIO_PIN_SET_INPUT			(0x0 << 2)
+#define RT5631_GPIO_PIN_SET_OUTPUT			(0x1 << 2)
+
+/* De-POP function Control 1(0x54) */
+#define RT5631_POW_ON_SOFT_GEN			(0x1 << 15)
+#define RT5631_EN_MUTE_UNMUTE_DEPOP			(0x1 << 14)
+#define RT5631_EN_DEPOP2_FOR_HP			(0x1 << 7)
+/* Power Down HPAMP_L Starts Up Signal */
+#define RT5631_PD_HPAMP_L_ST_UP			(0x1 << 5)
+/* Power Down HPAMP_R Starts Up Signal */
+#define RT5631_PD_HPAMP_R_ST_UP			(0x1 << 4)
+/* Enable left HP mute/unmute depop */
+#define RT5631_EN_HP_L_M_UN_MUTE_DEPOP		(0x1 << 1)
+/* Enable right HP mute/unmute depop */
+#define RT5631_EN_HP_R_M_UN_MUTE_DEPOP		(0x1 << 0)
+
+/* De-POP Fnction Control(0x56) */
+#define RT5631_EN_ONE_BIT_DEPOP			(0x1 << 15)
+#define RT5631_EN_CAP_FREE_DEPOP			(0x1 << 14)
+
+/* Jack Detect Control Register(0x5A) */
+#define RT5631_JD_USE_MASK				(0x3 << 14)
+#define RT5631_JD_USE_JD2				(0x3 << 14)
+#define RT5631_JD_USE_JD1				(0x2 << 14)
+#define RT5631_JD_USE_GPIO				(0x1 << 14)
+#define RT5631_JD_OFF					(0x0 << 14)
+/* JD trigger enable for HP */
+#define RT5631_JD_HP_EN					(0x1 << 11)
+#define RT5631_JD_HP_TRI_MASK				(0x1 << 10)
+#define RT5631_JD_HP_TRI_HI				(0x1 << 10)
+#define RT5631_JD_HP_TRI_LO				(0x1 << 10)
+/* JD trigger enable for speaker LP/LN */
+#define RT5631_JD_SPK_L_EN				(0x1 << 9)
+#define RT5631_JD_SPK_L_TRI_MASK			(0x1 << 8)
+#define RT5631_JD_SPK_L_TRI_HI				(0x1 << 8)
+#define RT5631_JD_SPK_L_TRI_LO				(0x0 << 8)
+/* JD trigger enable for speaker RP/RN */
+#define RT5631_JD_SPK_R_EN				(0x1 << 7)
+#define RT5631_JD_SPK_R_TRI_MASK			(0x1 << 6)
+#define RT5631_JD_SPK_R_TRI_HI				(0x1 << 6)
+#define RT5631_JD_SPK_R_TRI_LO				(0x0 << 6)
+/* JD trigger enable for monoout */
+#define RT5631_JD_MONO_EN				(0x1 << 5)
+#define RT5631_JD_MONO_TRI_MASK			(0x1 << 4)
+#define RT5631_JD_MONO_TRI_HI				(0x1 << 4)
+#define RT5631_JD_MONO_TRI_LO				(0x0 << 4)
+/* JD trigger enable for Lout */
+#define RT5631_JD_AUX_1_EN				(0x1 << 3)
+#define RT5631_JD_AUX_1_MASK				(0x1 << 2)
+#define RT5631_JD_AUX_1_TRI_HI				(0x1 << 2)
+#define RT5631_JD_AUX_1_TRI_LO				(0x0 << 2)
+/* JD trigger enable for Rout */
+#define RT5631_JD_AUX_2_EN				(0x1 << 1)
+#define RT5631_JD_AUX_2_MASK				(0x1 << 0)
+#define RT5631_JD_AUX_2_TRI_HI				(0x1 << 0)
+#define RT5631_JD_AUX_2_TRI_LO				(0x0 << 0)
+
+/* ALC CONTROL 1(0x64) */
+#define RT5631_ALC_ATTACK_RATE_MASK			(0x1F << 8)
+#define RT5631_ALC_RECOVERY_RATE_MASK		(0x1F << 0)
+
+/* ALC CONTROL 2(0x65) */
+/* select Compensation gain for Noise gate function */
+#define RT5631_ALC_COM_NOISE_GATE_MASK		(0xF << 0)
+
+/* ALC CONTROL 3(0x66) */
+#define RT5631_ALC_FUN_MASK				(0x3 << 14)
+#define RT5631_ALC_FUN_DIS				(0x0 << 14)
+#define RT5631_ALC_ENA_DAC_PATH			(0x1 << 14)
+#define RT5631_ALC_ENA_ADC_PATH			(0x3 << 14)
+#define RT5631_ALC_PARA_UPDATE			(0x1 << 13)
+#define RT5631_ALC_LIMIT_LEVEL_MASK			(0x1F << 8)
+#define RT5631_ALC_NOISE_GATE_FUN_MASK		(0x1 << 7)
+#define RT5631_ALC_NOISE_GATE_FUN_DIS			(0x0 << 7)
+#define RT5631_ALC_NOISE_GATE_FUN_ENA		(0x1 << 7)
+/* ALC noise gate hold data function */
+#define RT5631_ALC_NOISE_GATE_H_D_MASK		(0x1 << 6)
+#define RT5631_ALC_NOISE_GATE_H_D_DIS			(0x0 << 6)
+#define RT5631_ALC_NOISE_GATE_H_D_ENA		(0x1 << 6)
+
+/* Psedueo Stereo & Spatial Effect Block Control(0x68) */
+#define RT5631_SPATIAL_CTRL_EN				(0x1 << 15)
+#define RT5631_ALL_PASS_FILTER_EN			(0x1 << 14)
+#define RT5631_PSEUDO_STEREO_EN			(0x1 << 13)
+#define RT5631_STEREO_EXPENSION_EN			(0x1 << 12)
+/* 3D gain parameter */
+#define RT5631_GAIN_3D_PARA_MASK		(0x3 << 6)
+#define RT5631_GAIN_3D_PARA_1_00		(0x0 << 6) /* 3D gain 1.0 */
+#define RT5631_GAIN_3D_PARA_1_50		(0x1 << 6) /* 3D gain 1.5 */
+#define RT5631_GAIN_3D_PARA_2_00		(0x2 << 6) /* 3D gain 2.0 */
+/* 3D ratio parameter */
+#define RT5631_RATIO_3D_MASK			(0x3 << 4)
+#define RT5631_RATIO_3D_0_0			(0x0 << 4) /* 3D ratio 0.0 */
+#define RT5631_RATIO_3D_0_66			(0x1 << 4) /* 3D ratio 0.66 */
+#define RT5631_RATIO_3D_1_0			(0x2 << 4) /* 3D ratio 1.0 */
+/* select samplerate for all pass filter */
+#define RT5631_APF_FUN_SLE_MASK			(0x3 << 0)
+#define RT5631_APF_FUN_SEL_48K				(0x3 << 0)
+#define RT5631_APF_FUN_SEL_44_1K			(0x2 << 0)
+#define RT5631_APF_FUN_SEL_32K				(0x1 << 0)
+#define RT5631_APF_FUN_DIS				(0x0 << 0)
+
+/* EQ CONTROL 1(0x6E) */
+#define RT5631_HW_EQ_PATH_SEL_MASK			(0x1 << 15)
+#define RT5631_HW_EQ_PATH_SEL_DAC			(0x0 << 15)
+#define RT5631_HW_EQ_PATH_SEL_ADC			(0x1 << 15)
+#define RT5631_HW_EQ_UPDATE_CTRL			(0x1 << 14)
+
+#define RT5631_EN_HW_EQ_HPF2				(0x1 << 5)
+#define RT5631_EN_HW_EQ_HPF1				(0x1 << 4)
+#define RT5631_EN_HW_EQ_BP3				(0x1 << 3)
+#define RT5631_EN_HW_EQ_BP2				(0x1 << 2)
+#define RT5631_EN_HW_EQ_BP1				(0x1 << 1)
+#define RT5631_EN_HW_EQ_LPF				(0x1 << 0)
+
+
+#endif /* __RTCODEC5631_H__ */

sound/soc/codecs/sgtl5000.c

@@ -131,16 +131,13 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w,
 	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);
+				SGTL5000_BIAS_R_MASK,
+				SGTL5000_BIAS_R_4k << SGTL5000_BIAS_R_SHIFT);
 		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);
+				SGTL5000_BIAS_R_MASK, 0);
 		break;
 	}
 	return 0;
@@ -726,7 +723,9 @@ static int sgtl5000_pcm_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL, i2s_ctl, i2s_ctl);
+	snd_soc_update_bits(codec, SGTL5000_CHIP_I2S_CTRL,
+			    SGTL5000_I2S_DLEN_MASK | SGTL5000_I2S_SCLKFREQ_MASK,
+			    i2s_ctl);
 
 	return 0;
 }
@@ -757,7 +756,7 @@ static int ldo_regulator_enable(struct regulator_dev *dev)
 
 	/* set voltage to register */
 	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
-				(0x1 << 4) - 1, reg);
+				SGTL5000_LINREG_VDDD_MASK, reg);
 
 	snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER,
 				SGTL5000_LINEREG_D_POWERUP,
@@ -783,7 +782,7 @@ static int ldo_regulator_disable(struct regulator_dev *dev)
 
 	/* clear voltage info */
 	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
-				(0x1 << 4) - 1, 0);
+				SGTL5000_LINREG_VDDD_MASK, 0);
 
 	ldo->enabled = 0;
 
@@ -809,6 +808,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
 				int voltage)
 {
 	struct ldo_regulator *ldo;
+	struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec);
 
 	ldo = kzalloc(sizeof(struct ldo_regulator), GFP_KERNEL);
 
@@ -843,6 +843,7 @@ static int ldo_regulator_register(struct snd_soc_codec *codec,
 
 		return ret;
 	}
+	sgtl5000->ldo = ldo;
 
 	return 0;
 }
@@ -1116,7 +1117,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 
 	/* set voltage to register */
 	snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL,
-				(0x1 << 4) - 1, 0x8);
+				SGTL5000_LINREG_VDDD_MASK, 0x8);
 
 	/*
 	 * if vddd linear reg has been enabled,
@@ -1147,8 +1148,7 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 		vag = (vag - SGTL5000_ANA_GND_BASE) / SGTL5000_ANA_GND_STP;
 
 	snd_soc_update_bits(codec, SGTL5000_CHIP_REF_CTRL,
-			vag << SGTL5000_ANA_GND_SHIFT,
+			SGTL5000_ANA_GND_MASK, 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;
@@ -1162,9 +1162,8 @@ static int sgtl5000_set_power_regs(struct snd_soc_codec *codec)
 		    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_MASK |
-			SGTL5000_LINE_OUT_CURRENT_360u <<
+			SGTL5000_LINE_OUT_GND_MASK,
-				SGTL5000_LINE_OUT_CURRENT_SHIFT,
 			vag << SGTL5000_LINE_OUT_GND_SHIFT |
 			SGTL5000_LINE_OUT_CURRENT_360u <<
 				SGTL5000_LINE_OUT_CURRENT_SHIFT);

sound/soc/codecs/sgtl5000.h

@@ -280,7 +280,7 @@
 /*
  * SGTL5000_CHIP_MIC_CTRL
  */
-#define SGTL5000_BIAS_R_MASK			0x0200
+#define SGTL5000_BIAS_R_MASK			0x0300
 #define SGTL5000_BIAS_R_SHIFT			8
 #define SGTL5000_BIAS_R_WIDTH			2
 #define SGTL5000_BIAS_R_off			0x0

sound/soc/codecs/sn95031.c

@@ -102,7 +102,7 @@ static int sn95031_initialize_adc(struct snd_soc_codec *sn95031_codec)
 {
 	int base_addr, chnl_addr;
 	int value;
-	static int channel_index;
+	int channel_index;
 
 	/* Index of the first channel in which the stop bit is set */
 	channel_index = find_free_channel(sn95031_codec);
@@ -161,7 +161,6 @@ static unsigned int sn95031_get_mic_bias(struct snd_soc_codec *codec)
 	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,
@@ -716,7 +715,7 @@ static struct snd_soc_dai_ops sn95031_vib2_dai_ops = {
 	.hw_params	= sn95031_pcm_hw_params,
 };
 
-struct snd_soc_dai_driver sn95031_dais[] = {
+static struct snd_soc_dai_driver sn95031_dais[] = {
 {
 	.name = "SN95031 Headset",
 	.playback = {
@@ -827,7 +826,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec)
 {
 	pr_debug("codec_probe called\n");
 
-	codec->dapm.bias_level = SND_SOC_BIAS_OFF;
 	codec->dapm.idle_bias_off = 1;
 
 	/* PCM interface config

sound/soc/codecs/ssm2602.c

@@ -59,6 +59,7 @@ struct ssm2602_priv {
 	struct snd_pcm_substream *slave_substream;
 
 	enum ssm2602_type type;
+	unsigned int clk_out_pwr;
 };
 
 /*
@@ -342,12 +343,14 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 static int ssm2602_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 mute_reg = snd_soc_read(codec, SSM2602_APDIGI) & ~APDIGI_ENABLE_DAC_MUTE;
+
 	if (mute)
-		snd_soc_write(codec, SSM2602_APDIGI,
+		snd_soc_update_bits(codec, SSM2602_APDIGI,
-				mute_reg | APDIGI_ENABLE_DAC_MUTE);
+				    APDIGI_ENABLE_DAC_MUTE,
+				    APDIGI_ENABLE_DAC_MUTE);
 	else
-		snd_soc_write(codec, SSM2602_APDIGI, mute_reg);
+		snd_soc_update_bits(codec, SSM2602_APDIGI,
+				    APDIGI_ENABLE_DAC_MUTE, 0);
 	return 0;
 }
 
@@ -356,6 +359,11 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
+
+	if (dir == SND_SOC_CLOCK_IN) {
+		if (clk_id != SSM2602_SYSCLK)
+			return -EINVAL;
+
 		switch (freq) {
 		case 11289600:
 		case 12000000:
@@ -363,10 +371,35 @@ static int ssm2602_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 		case 16934400:
 		case 18432000:
 			ssm2602->sysclk = freq;
-		return 0;
+			break;
-	}
+		default:
 			return -EINVAL;
 		}
+	} else {
+		unsigned int mask;
+
+		switch (clk_id) {
+		case SSM2602_CLK_CLKOUT:
+			mask = PWR_CLK_OUT_PDN;
+			break;
+		case SSM2602_CLK_XTO:
+			mask = PWR_OSC_PDN;
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		if (freq == 0)
+			ssm2602->clk_out_pwr |= mask;
+		else
+			ssm2602->clk_out_pwr &= ~mask;
+
+		snd_soc_update_bits(codec, SSM2602_PWR,
+			PWR_CLK_OUT_PDN | PWR_OSC_PDN, ssm2602->clk_out_pwr);
+	}
+
+	return 0;
+}
 
 static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
@@ -430,23 +463,27 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg = snd_soc_read(codec, SSM2602_PWR);
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN);
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* vref/mid, osc on, dac unmute */
+		/* vref/mid on, osc and clkout on if enabled */
-		snd_soc_write(codec, SSM2602_PWR, reg);
+		snd_soc_update_bits(codec, SSM2602_PWR,
+			PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+			ssm2602->clk_out_pwr);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
-		snd_soc_write(codec, SSM2602_PWR, reg | PWR_CLK_OUT_PDN);
+		snd_soc_update_bits(codec, SSM2602_PWR,
+			PWR_POWER_OFF | PWR_CLK_OUT_PDN | PWR_OSC_PDN,
+			PWR_CLK_OUT_PDN | PWR_OSC_PDN);
 		break;
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
+		/* everything off */
-		snd_soc_write(codec, SSM2602_PWR, 0xffff);
+		snd_soc_update_bits(codec, SSM2602_PWR,
+			PWR_POWER_OFF, PWR_POWER_OFF);
 		break;
 
 	}
@@ -505,12 +542,12 @@ static int ssm2602_resume(struct snd_soc_codec *codec)
 static int ssm2602_probe(struct snd_soc_codec *codec)
 {
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
-	int ret, reg;
+	int ret;
 
-	reg = snd_soc_read(codec, SSM2602_LOUT1V);
+	snd_soc_update_bits(codec, SSM2602_LOUT1V,
-	snd_soc_write(codec, SSM2602_LOUT1V, reg | LOUT1V_LRHP_BOTH);
+			    LOUT1V_LRHP_BOTH, LOUT1V_LRHP_BOTH);
-	reg = snd_soc_read(codec, SSM2602_ROUT1V);
+	snd_soc_update_bits(codec, SSM2602_ROUT1V,
-	snd_soc_write(codec, SSM2602_ROUT1V, reg | ROUT1V_RLHP_BOTH);
+			    ROUT1V_RLHP_BOTH, ROUT1V_RLHP_BOTH);
 
 	ret = snd_soc_add_controls(codec, ssm2602_snd_controls,
 			ARRAY_SIZE(ssm2602_snd_controls));
@@ -543,7 +580,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec)
 static int ssm260x_probe(struct snd_soc_codec *codec)
 {
 	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
-	int ret, reg;
+	int ret;
 
 	pr_info("ssm2602 Audio Codec %s", SSM2602_VERSION);
 
@@ -560,10 +597,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
 	}
 
 	/* set the update bits */
-	reg = snd_soc_read(codec, SSM2602_LINVOL);
+	snd_soc_update_bits(codec, SSM2602_LINVOL,
-	snd_soc_write(codec, SSM2602_LINVOL, reg | LINVOL_LRIN_BOTH);
+			    LINVOL_LRIN_BOTH, LINVOL_LRIN_BOTH);
-	reg = snd_soc_read(codec, SSM2602_RINVOL);
+	snd_soc_update_bits(codec, SSM2602_RINVOL,
-	snd_soc_write(codec, SSM2602_RINVOL, reg | RINVOL_RLIN_BOTH);
+			    RINVOL_RLIN_BOTH, RINVOL_RLIN_BOTH);
 	/*select Line in as default input*/
 	snd_soc_write(codec, SSM2602_APANA, APANA_SELECT_DAC |
 			APANA_ENABLE_MIC_BOOST);
@@ -577,7 +614,12 @@ static int ssm260x_probe(struct snd_soc_codec *codec)
 		break;
 	}
 
+	if (ret)
 		return ret;
+
+	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
 }
 
 /* remove everything here */

sound/soc/codecs/ssm2602.h

@@ -116,6 +116,10 @@
 
 #define SSM2602_CACHEREGNUM 	10
 
-#define SSM2602_SYSCLK	0
+enum ssm2602_clk {
+	SSM2602_SYSCLK,
+	SSM2602_CLK_CLKOUT,
+	SSM2602_CLK_XTO
+};
 
 #endif

sound/soc/codecs/sta32x.c

@@ -756,25 +756,19 @@ static int sta32x_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	/* read reg reset values into cache */
+	/* Chip documentation explicitly requires that the reset values
-	for (i = 0; i < STA32X_REGISTER_COUNT; i++)
+	 * of reserved register bits are left untouched.
-		snd_soc_cache_write(codec, i, sta32x_regs[i]);
+	 * Write the register default value to cache for reserved registers,
-
+	 * so the write to the these registers are suppressed by the cache
-	/* preserve reset values of reserved register bits */
+	 * restore code when it skips writes of default registers.
-	snd_soc_cache_write(codec, STA32X_CONFC,
+	 */
-			    codec->hw_read(codec, STA32X_CONFC));
+	snd_soc_cache_write(codec, STA32X_CONFC, 0xc2);
-	snd_soc_cache_write(codec, STA32X_CONFE,
+	snd_soc_cache_write(codec, STA32X_CONFE, 0xc2);
-			    codec->hw_read(codec, STA32X_CONFE));
+	snd_soc_cache_write(codec, STA32X_CONFF, 0x5c);
-	snd_soc_cache_write(codec, STA32X_CONFF,
+	snd_soc_cache_write(codec, STA32X_MMUTE, 0x10);
-			    codec->hw_read(codec, STA32X_CONFF));
+	snd_soc_cache_write(codec, STA32X_AUTO1, 0x60);
-	snd_soc_cache_write(codec, STA32X_MMUTE,
+	snd_soc_cache_write(codec, STA32X_AUTO3, 0x00);
-			    codec->hw_read(codec, STA32X_MMUTE));
+	snd_soc_cache_write(codec, STA32X_C3CFG, 0x40);
-	snd_soc_cache_write(codec, STA32X_AUTO1,
-			    codec->hw_read(codec, STA32X_AUTO1));
-	snd_soc_cache_write(codec, STA32X_AUTO3,
-			    codec->hw_read(codec, STA32X_AUTO3));
-	snd_soc_cache_write(codec, STA32X_C3CFG,
-			    codec->hw_read(codec, STA32X_C3CFG));
 
 	/* FIXME enable thermal warning adjustment and recovery  */
 	snd_soc_update_bits(codec, STA32X_CONFA,
@@ -837,6 +831,7 @@ static const struct snd_soc_codec_driver sta32x_codec = {
 	.resume =		sta32x_resume,
 	.reg_cache_size =	STA32X_REGISTER_COUNT,
 	.reg_word_size =	sizeof(u8),
+	.reg_cache_default =	sta32x_regs,
 	.volatile_register =	sta32x_reg_is_volatile,
 	.set_bias_level =	sta32x_set_bias_level,
 	.controls =		sta32x_snd_controls,

sound/soc/codecs/tlv320aic23.c

@@ -47,63 +47,6 @@ static const u16 tlv320aic23_reg[] = {
 	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
 };
 
-/*
- * read tlv320aic23 register cache
- */
-static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
-						      *codec, unsigned int reg)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
-		return -1;
-	return cache[reg];
-}
-
-/*
- * write tlv320aic23 register cache
- */
-static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
-					       u8 reg, u16 value)
-{
-	u16 *cache = codec->reg_cache;
-	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
-		return;
-	cache[reg] = value;
-}
-
-/*
- * write to the tlv320aic23 register space
- */
-static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
-			     unsigned int value)
-{
-
-	u8 data[2];
-
-	/* TLV320AIC23 has 7 bit address and 9 bits of data
-	 * so we need to switch one data bit into reg and rest
-	 * of data into val
-	 */
-
-	if (reg > 9 && reg != 15) {
-		printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
-		return -1;
-	}
-
-	data[0] = (reg << 1) | (value >> 8 & 0x01);
-	data[1] = value & 0xff;
-
-	tlv320aic23_write_reg_cache(codec, reg, value);
-
-	if (codec->hw_write(codec->control_data, data, 2) == 2)
-		return 0;
-
-	printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
-	       value, reg);
-
-	return -EIO;
-}
-
 static const char *rec_src_text[] = { "Line", "Mic" };
 static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
 
@@ -139,8 +82,8 @@ static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
 	*/
 	val = (val >= 4) ? 4  : (3 - val);
 
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
+	reg = snd_soc_read(codec, TLV320AIC23_ANLG) & (~0x1C0);
-	tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+	snd_soc_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
 
 	return 0;
 }
@@ -151,7 +94,7 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	u16 val;
 
-	val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
+	val = snd_soc_read(codec, TLV320AIC23_ANLG) & (0x1C0);
 	val = val >> 6;
 	val = (val >= 4) ? 4  : (3 -  val);
 	ucontrol->value.integer.value[0] = val;
@@ -159,15 +102,6 @@ static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
 
 }
 
-#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
-	.put = snd_soc_tlv320aic23_put_volsw, \
-	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
-
 static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
 	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
 			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
@@ -178,8 +112,9 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
 			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
 	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
 	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
-	SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
+	SOC_SINGLE_EXT_TLV("Sidetone Volume", TLV320AIC23_ANLG, 6, 4, 0,
-				  6, 4, 0, sidetone_vol_tlv),
+			   snd_soc_tlv320aic23_get_volsw,
+			   snd_soc_tlv320aic23_put_volsw, sidetone_vol_tlv),
 	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
 };
 
@@ -240,7 +175,6 @@ static const struct snd_soc_dapm_route tlv320aic23_intercon[] = {
 /* AIC23 driver data */
 struct aic23 {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	int mclk;
 	int requested_adc;
 	int requested_dac;
@@ -352,7 +286,7 @@ static int find_rate(int mclk, u32 need_adc, u32 need_dac)
 static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
 		u32 *sample_rate_adc, u32 *sample_rate_dac)
 {
-	int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
+	int src = snd_soc_read(codec, TLV320AIC23_SRATE);
 	int sr = (src >> 2) & 0x0f;
 	int val = (mclk / bosr_usb_divisor_table[src & 3]);
 	int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
@@ -376,7 +310,7 @@ static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
 				__func__, sample_rate_adc, sample_rate_dac);
 		return -EINVAL;
 	}
-	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+	snd_soc_write(codec, TLV320AIC23_SRATE, data);
 #ifdef DEBUG
 	{
 		u32 adc, dac;
@@ -415,9 +349,8 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	iface_reg =
+	iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
-	    tlv320aic23_read_reg_cache(codec,
+
-				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		break;
@@ -431,7 +364,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 		iface_reg |= (0x03 << 2);
 		break;
 	}
-	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+	snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
 
 	return 0;
 }
@@ -443,7 +376,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = rtd->codec;
 
 	/* set active */
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0001);
 
 	return 0;
 }
@@ -458,7 +391,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
 	/* deactivate */
 	if (!codec->active) {
 		udelay(50);
-		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	}
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		aic23->requested_dac = 0;
@@ -471,14 +404,14 @@ static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
 	struct snd_soc_codec *codec = dai->codec;
 	u16 reg;
 
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+	reg = snd_soc_read(codec, TLV320AIC23_DIGT);
 	if (mute)
 		reg |= TLV320AIC23_DACM_MUTE;
 
 	else
 		reg &= ~TLV320AIC23_DACM_MUTE;
 
-	tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+	snd_soc_write(codec, TLV320AIC23_DIGT, reg);
 
 	return 0;
 }
@@ -489,8 +422,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	u16 iface_reg;
 
-	iface_reg =
+	iface_reg = snd_soc_read(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
-	    tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
 
 	/* set master/slave audio interface */
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
@@ -524,7 +456,7 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
 
 	}
 
-	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+	snd_soc_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
 
 	return 0;
 }
@@ -540,26 +472,26 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 				      enum snd_soc_bias_level level)
 {
-	u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+	u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* vref/mid, osc on, dac unmute */
 		reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
 			TLV320AIC23_DAC_OFF);
-		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
+		snd_soc_write(codec, TLV320AIC23_PWR, reg);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		/* everything off except vref/vmid, */
-		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
+		snd_soc_write(codec, TLV320AIC23_PWR,
-			TLV320AIC23_CLK_OFF);
+			      reg | TLV320AIC23_CLK_OFF);
 		break;
 	case SND_SOC_BIAS_OFF:
 		/* everything off, dac mute, inactive */
-		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0);
-		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+		snd_soc_write(codec, TLV320AIC23_PWR, 0xffff);
 		break;
 	}
 	codec->dapm.bias_level = level;
@@ -606,13 +538,7 @@ static int tlv320aic23_suspend(struct snd_soc_codec *codec,
 
 static int tlv320aic23_resume(struct snd_soc_codec *codec)
 {
-	u16 reg;
+	snd_soc_cache_sync(codec);
-
-	/* Sync reg_cache with the hardware */
-	for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
-		u16 val = tlv320aic23_read_reg_cache(codec, reg);
-		tlv320aic23_write(codec, reg, val);
-	}
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
@@ -621,46 +547,52 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec)
 static int tlv320aic23_probe(struct snd_soc_codec *codec)
 {
 	struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
-	int reg;
+	int ret;
 
 	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
-	codec->control_data = aic23->control_data;
+
-	codec->hw_write = (hw_write_t)i2c_master_send;
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, aic23->control_type);
-	codec->hw_read = NULL;
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
 
 	/* Reset codec */
-	tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+	snd_soc_write(codec, TLV320AIC23_RESET, 0);
+
+	/* Write the register default value to cache for reserved registers,
+	 * so the write to the these registers are suppressed by the cache
+	 * restore code when it skips writes of default registers.
+	 */
+	snd_soc_cache_write(codec, 0x0A, 0);
+	snd_soc_cache_write(codec, 0x0B, 0);
+	snd_soc_cache_write(codec, 0x0C, 0);
+	snd_soc_cache_write(codec, 0x0D, 0);
+	snd_soc_cache_write(codec, 0x0E, 0);
 
 	/* power on device */
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+	snd_soc_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
 
 	/* Unmute input */
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+	snd_soc_update_bits(codec, TLV320AIC23_LINVOL,
-	tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+			    TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
-			  (reg & (~TLV320AIC23_LIM_MUTED)) |
-			  (TLV320AIC23_LRS_ENABLED));
 
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+	snd_soc_update_bits(codec, TLV320AIC23_RINVOL,
-	tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+			    TLV320AIC23_LIM_MUTED, TLV320AIC23_LRS_ENABLED);
-			  (reg & (~TLV320AIC23_LIM_MUTED)) |
-			  TLV320AIC23_LRS_ENABLED);
 
-	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+	snd_soc_update_bits(codec, TLV320AIC23_ANLG,
-	tlv320aic23_write(codec, TLV320AIC23_ANLG,
+			    TLV320AIC23_BYPASS_ON | TLV320AIC23_MICM_MUTED,
-			 (reg) & (~TLV320AIC23_BYPASS_ON) &
+			    0);
-			 (~TLV320AIC23_MICM_MUTED));
 
 	/* Default output volume */
-	tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+	snd_soc_write(codec, TLV320AIC23_LCHNVOL,
-			  TLV320AIC23_DEFAULT_OUT_VOL &
+		      TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
-			  TLV320AIC23_OUT_VOL_MASK);
+	snd_soc_write(codec, TLV320AIC23_RCHNVOL,
-	tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+		      TLV320AIC23_DEFAULT_OUT_VOL & TLV320AIC23_OUT_VOL_MASK);
-			  TLV320AIC23_DEFAULT_OUT_VOL &
-			  TLV320AIC23_OUT_VOL_MASK);
 
-	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+	snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x1);
 
 	snd_soc_add_controls(codec, tlv320aic23_snd_controls,
 				ARRAY_SIZE(tlv320aic23_snd_controls));
@@ -682,8 +614,6 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
 	.remove = tlv320aic23_remove,
 	.suspend = tlv320aic23_suspend,
 	.resume = tlv320aic23_resume,
-	.read = tlv320aic23_read_reg_cache,
-	.write = tlv320aic23_write,
 	.set_bias_level = tlv320aic23_set_bias_level,
 	.dapm_widgets = tlv320aic23_dapm_widgets,
 	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
@@ -710,7 +640,6 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, aic23);
-	aic23->control_data = i2c;
 	aic23->control_type = SND_SOC_I2C;
 
 	ret =  snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/tlv320aic32x4.c

@@ -528,40 +528,33 @@ 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_update_bits(codec, AIC32X4_PLLPR,
-			snd_soc_write(codec, AIC32X4_PLLPR,
+					    AIC32X4_PLLEN, AIC32X4_PLLEN);
-				      (value | AIC32X4_PLLEN));
 
 			/* Switch on NDAC Divider */
-			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_update_bits(codec, AIC32X4_NDAC,
-			snd_soc_write(codec, AIC32X4_NDAC,
+					    AIC32X4_NDACEN, AIC32X4_NDACEN);
-				      value | AIC32X4_NDACEN);
 
 			/* Switch on MDAC Divider */
-			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_update_bits(codec, AIC32X4_MDAC,
-			snd_soc_write(codec, AIC32X4_MDAC,
+					    AIC32X4_MDACEN, AIC32X4_MDACEN);
-				      value | AIC32X4_MDACEN);
 
 			/* Switch on NADC Divider */
-			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_update_bits(codec, AIC32X4_NADC,
-			snd_soc_write(codec, AIC32X4_NADC,
+					    AIC32X4_NADCEN, AIC32X4_NADCEN);
-				      value | AIC32X4_MDACEN);
 
 			/* Switch on MADC Divider */
-			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_update_bits(codec, AIC32X4_MADC,
-			snd_soc_write(codec, AIC32X4_MADC,
+					    AIC32X4_MADCEN, AIC32X4_MADCEN);
-				      value | AIC32X4_MDACEN);
 
 			/* Switch on BCLK_N Divider */
-			value = snd_soc_read(codec, AIC32X4_BCLKN);
+			snd_soc_update_bits(codec, AIC32X4_BCLKN,
-			snd_soc_write(codec, AIC32X4_BCLKN,
+					    AIC32X4_BCLKEN, AIC32X4_BCLKEN);
-				      value | AIC32X4_BCLKEN);
 		}
 		break;
 	case SND_SOC_BIAS_PREPARE:
@@ -569,34 +562,28 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_STANDBY:
 		if (aic32x4->master) {
 			/* Switch off PLL */
-			value = snd_soc_read(codec, AIC32X4_PLLPR);
+			snd_soc_update_bits(codec, AIC32X4_PLLPR,
-			snd_soc_write(codec, AIC32X4_PLLPR,
+					    AIC32X4_PLLEN, 0);
-				      (value & ~AIC32X4_PLLEN));
 
 			/* Switch off NDAC Divider */
-			value = snd_soc_read(codec, AIC32X4_NDAC);
+			snd_soc_update_bits(codec, AIC32X4_NDAC,
-			snd_soc_write(codec, AIC32X4_NDAC,
+					    AIC32X4_NDACEN, 0);
-				      value & ~AIC32X4_NDACEN);
 
 			/* Switch off MDAC Divider */
-			value = snd_soc_read(codec, AIC32X4_MDAC);
+			snd_soc_update_bits(codec, AIC32X4_MDAC,
-			snd_soc_write(codec, AIC32X4_MDAC,
+					    AIC32X4_MDACEN, 0);
-				      value & ~AIC32X4_MDACEN);
 
 			/* Switch off NADC Divider */
-			value = snd_soc_read(codec, AIC32X4_NADC);
+			snd_soc_update_bits(codec, AIC32X4_NADC,
-			snd_soc_write(codec, AIC32X4_NADC,
+					    AIC32X4_NADCEN, 0);
-				      value & ~AIC32X4_NDACEN);
 
 			/* Switch off MADC Divider */
-			value = snd_soc_read(codec, AIC32X4_MADC);
+			snd_soc_update_bits(codec, AIC32X4_MADC,
-			snd_soc_write(codec, AIC32X4_MADC,
+					    AIC32X4_MADCEN, 0);
-				      value & ~AIC32X4_MDACEN);
-			value = snd_soc_read(codec, AIC32X4_BCLKN);
 
 			/* Switch off BCLK_N Divider */
-			snd_soc_write(codec, AIC32X4_BCLKN,
+			snd_soc_update_bits(codec, AIC32X4_BCLKN,
-				      value & ~AIC32X4_BCLKEN);
+					    AIC32X4_BCLKEN, 0);
 		}
 		break;
 	case SND_SOC_BIAS_OFF:
@@ -685,10 +672,10 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}
 
 	/* Mic PGA routing */
-	if (aic32x4->micpga_routing | AIC32X4_MICPGA_ROUTE_LMIC_IN2R_10K) {
+	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) {
+	if (aic32x4->micpga_routing & AIC32X4_MICPGA_ROUTE_RMIC_IN1L_10K) {
 		snd_soc_write(codec, AIC32X4_RMICPGANIN, AIC32X4_RMICPGANIN_IN1L_10K);
 	}
 

sound/soc/codecs/tlv320aic3x.c

@@ -76,7 +76,6 @@ struct aic3x_priv {
 	struct aic3x_disable_nb disable_nb[AIC3X_NUM_SUPPLIES];
 	enum snd_soc_control_type control_type;
 	struct aic3x_setup_data *setup;
-	void *control_data;
 	unsigned int sysclk;
 	struct list_head list;
 	int master;
@@ -138,7 +137,10 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
 	if (reg >= AIC3X_CACHEREGNUM)
 		return -1;
 
-	*value = codec->hw_read(codec, reg);
+	codec->cache_bypass = 1;
+	*value = snd_soc_read(codec, reg);
+	codec->cache_bypass = 0;
+
 	cache[reg] = *value;
 
 	return 0;
@@ -198,6 +200,10 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 			else
 				/* old connection must be powered down */
 				path->connect = invert ? 1 : 0;
+
+			dapm_mark_dirty(path->source, "tlv320aic3x source");
+			dapm_mark_dirty(path->sink, "tlv320aic3x sink");
+
 			break;
 		}
 
@@ -1383,7 +1389,6 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 	int ret, i;
 
 	INIT_LIST_HEAD(&aic3x->list);
-	codec->control_data = aic3x->control_data;
 	aic3x->codec = codec;
 	codec->dapm.idle_bias_off = 1;
 
@@ -1495,9 +1500,9 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
  */
 
 static const struct i2c_device_id aic3x_i2c_id[] = {
-	[AIC3X_MODEL_3X] = { "tlv320aic3x", 0 },
+	{ "tlv320aic3x", AIC3X_MODEL_3X },
-	[AIC3X_MODEL_33] = { "tlv320aic33", 0 },
+	{ "tlv320aic33", AIC3X_MODEL_33 },
-	[AIC3X_MODEL_3007] = { "tlv320aic3007", 0 },
+	{ "tlv320aic3007", AIC3X_MODEL_3007 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
@@ -1512,7 +1517,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 	struct aic3x_pdata *pdata = i2c->dev.platform_data;
 	struct aic3x_priv *aic3x;
 	int ret;
-	const struct i2c_device_id *tbl;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
 	if (aic3x == NULL) {
@@ -1520,7 +1524,6 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 	}
 
-	aic3x->control_data = i2c;
 	aic3x->control_type = SND_SOC_I2C;
 
 	i2c_set_clientdata(i2c, aic3x);
@@ -1531,11 +1534,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		aic3x->gpio_reset = -1;
 	}
 
-	for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) {
+	aic3x->model = id->driver_data;
-		if (!strcmp(tbl->name, id->name))
-			break;
-	}
-	aic3x->model = tbl - aic3x_i2c_id;
 
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_aic3x, &aic3x_dai, 1);

sound/soc/codecs/tlv320dac33.c

@@ -55,13 +55,13 @@
 #define BURST_BASEFREQ_HZ	49152000
 
 #define SAMPLES_TO_US(rate, samples) \
-	(1000000000 / ((rate * 1000) / samples))
+	(1000000000 / (((rate) * 1000) / (samples)))
 
 #define US_TO_SAMPLES(rate, us) \
-	(rate / (1000000 / (us < 1000000 ? us : 1000000)))
+	((rate) / (1000000 / ((us) < 1000000 ? (us) : 1000000)))
 
 #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \
-	((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate)))
+	(((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate))))
 
 static void dac33_calculate_times(struct snd_pcm_substream *substream);
 static int dac33_prepare_chip(struct snd_pcm_substream *substream);
@@ -627,18 +627,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"RIGHT_LO", NULL, "Codec Power"},
 };
 
-static int dac33_add_widgets(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dac33_dapm_widgets,
-				  ARRAY_SIZE(dac33_dapm_widgets));
-	/* set up audio path interconnects */
-	snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map));
-
-	return 0;
-}
-
 static int dac33_set_bias_level(struct snd_soc_codec *codec,
 				enum snd_soc_bias_level level)
 {
@@ -1451,15 +1439,11 @@ static int dac33_soc_probe(struct snd_soc_codec *codec)
 		}
 	}
 
-	snd_soc_add_controls(codec, dac33_snd_controls,
-			     ARRAY_SIZE(dac33_snd_controls));
 	/* Only add the FIFO controls, if we have valid IRQ number */
 	if (dac33->irq >= 0)
 		snd_soc_add_controls(codec, dac33_mode_snd_controls,
 				     ARRAY_SIZE(dac33_mode_snd_controls));
 
-	dac33_add_widgets(codec);
-
 err_power:
 	return ret;
 }
@@ -1502,6 +1486,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320dac33 = {
 	.remove = dac33_soc_remove,
 	.suspend = dac33_soc_suspend,
 	.resume = dac33_soc_resume,
+
+	.controls = dac33_snd_controls,
+	.num_controls = ARRAY_SIZE(dac33_snd_controls),
+	.dapm_widgets = dac33_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dac33_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
 };
 
 #define DAC33_RATES	(SNDRV_PCM_RATE_44100 | \

sound/soc/codecs/tpa6130a2.c

@@ -33,6 +33,11 @@
 
 #include "tpa6130a2.h"
 
+enum tpa_model {
+	TPA6130A2,
+	TPA6140A2,
+};
+
 static struct i2c_client *tpa6130a2_client;
 
 /* This struct is used to save the context */
@@ -383,7 +388,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 
 	pdata = client->dev.platform_data;
 	data->power_gpio = pdata->power_gpio;
-	data->id = pdata->id;
+	data->id = id->driver_data;
 
 	mutex_init(&data->mutex);
 
@@ -405,7 +410,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client,
 	switch (data->id) {
 	default:
 		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
-			 pdata->id);
+			 data->id);
 	case TPA6130A2:
 		regulator = "Vdd";
 		break;
@@ -469,7 +474,8 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client)
 }
 
 static const struct i2c_device_id tpa6130a2_id[] = {
-	{ "tpa6130a2", 0 },
+	{ "tpa6130a2", TPA6130A2 },
+	{ "tpa6140a2", TPA6140A2 },
 	{ }
 };
 MODULE_DEVICE_TABLE(i2c, tpa6130a2_id);

sound/soc/codecs/twl4030.c

@@ -863,34 +863,6 @@ static int digimic_event(struct snd_soc_dapm_widget *w,
  * Inverting not going to help with these.
  * Custom volsw and volsw_2r get/put functions to handle these gain bits.
  */
-#define SOC_DOUBLE_TLV_TWL4030(xname, xreg, shift_left, shift_right, xmax,\
-			       xinvert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw, \
-	.get = snd_soc_get_volsw_twl4030, \
-	.put = snd_soc_put_volsw_twl4030, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = xreg, .shift = shift_left, .rshift = shift_right,\
-		 .max = xmax, .invert = xinvert} }
-#define SOC_DOUBLE_R_TLV_TWL4030(xname, reg_left, reg_right, xshift, xmax,\
-				 xinvert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r, \
-	.get = snd_soc_get_volsw_r2_twl4030,\
-	.put = snd_soc_put_volsw_r2_twl4030, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		 .rshift = xshift, .max = xmax, .invert = xinvert} }
-#define SOC_SINGLE_TLV_TWL4030(xname, xreg, xshift, xmax, xinvert, tlv_array) \
-	SOC_DOUBLE_TLV_TWL4030(xname, xreg, xshift, xshift, xmax, \
-			       xinvert, tlv_array)
-
 static int snd_soc_get_volsw_twl4030(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
 {
@@ -1197,19 +1169,23 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		TWL4030_REG_VDL_APGA_CTL, 1, 1, 0),
 
 	/* Separate output gain controls */
-	SOC_DOUBLE_R_TLV_TWL4030("PreDriv Playback Volume",
+	SOC_DOUBLE_R_EXT_TLV("PreDriv Playback Volume",
 		TWL4030_REG_PREDL_CTL, TWL4030_REG_PREDR_CTL,
-		4, 3, 0, output_tvl),
+		4, 3, 0, snd_soc_get_volsw_r2_twl4030,
+		snd_soc_put_volsw_r2_twl4030, output_tvl),
 
-	SOC_DOUBLE_TLV_TWL4030("Headset Playback Volume",
+	SOC_DOUBLE_EXT_TLV("Headset Playback Volume",
-		TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, output_tvl),
+		TWL4030_REG_HS_GAIN_SET, 0, 2, 3, 0, snd_soc_get_volsw_twl4030,
+		snd_soc_put_volsw_twl4030, output_tvl),
 
-	SOC_DOUBLE_R_TLV_TWL4030("Carkit Playback Volume",
+	SOC_DOUBLE_R_EXT_TLV("Carkit Playback Volume",
 		TWL4030_REG_PRECKL_CTL, TWL4030_REG_PRECKR_CTL,
-		4, 3, 0, output_tvl),
+		4, 3, 0, snd_soc_get_volsw_r2_twl4030,
+		snd_soc_put_volsw_r2_twl4030, output_tvl),
 
-	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
+	SOC_SINGLE_EXT_TLV("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, snd_soc_get_volsw_twl4030,
+		snd_soc_put_volsw_twl4030, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
@@ -1633,17 +1609,6 @@ static const struct snd_soc_dapm_route intercon[] = {
 
 };
 
-static int twl4030_add_widgets(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, twl4030_dapm_widgets,
-				 ARRAY_SIZE(twl4030_dapm_widgets));
-	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-
-	return 0;
-}
-
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
@@ -2265,9 +2230,6 @@ static int twl4030_soc_probe(struct snd_soc_codec *codec)
 
 	twl4030_init_chip(codec);
 
-	snd_soc_add_controls(codec, twl4030_snd_controls,
-				ARRAY_SIZE(twl4030_snd_controls));
-	twl4030_add_widgets(codec);
 	return 0;
 }
 
@@ -2293,6 +2255,13 @@ static struct snd_soc_codec_driver soc_codec_dev_twl4030 = {
 	.reg_cache_size = sizeof(twl4030_reg),
 	.reg_word_size = sizeof(u8),
 	.reg_cache_default = twl4030_reg,
+
+	.controls = twl4030_snd_controls,
+	.num_controls = ARRAY_SIZE(twl4030_snd_controls),
+	.dapm_widgets = twl4030_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(twl4030_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 static int __devinit twl4030_codec_probe(struct platform_device *pdev)

sound/soc/codecs/twl6040.h

@@ -22,8 +22,21 @@
 #ifndef __TWL6040_H__
 #define __TWL6040_H__
 
+enum twl6040_trim {
+	TWL6040_TRIM_TRIM1 = 0,
+	TWL6040_TRIM_TRIM2,
+	TWL6040_TRIM_TRIM3,
+	TWL6040_TRIM_HSOTRIM,
+	TWL6040_TRIM_HFOTRIM,
+	TWL6040_TRIM_INVAL,
+};
+
+#define TWL6040_HSF_TRIM_LEFT(x)	(x & 0x0f)
+#define TWL6040_HSF_TRIM_RIGHT(x)	((x >> 4) & 0x0f)
+
 void twl6040_hs_jack_detect(struct snd_soc_codec *codec,
 			    struct snd_soc_jack *jack, int report);
 int twl6040_get_clk_id(struct snd_soc_codec *codec);
+int twl6040_get_trim_value(struct snd_soc_codec *codec, enum twl6040_trim trim);
 
 #endif /* End of __TWL6040_H__ */

sound/soc/codecs/wl1273.c

@@ -462,7 +462,6 @@ static int wl1273_probe(struct snd_soc_codec *codec)
 	wl1273->core = *core;
 
 	snd_soc_codec_set_drvdata(codec, wl1273);
-	mutex_init(&codec->mutex);
 
 	r = snd_soc_add_controls(codec, wl1273_controls,
 				 ARRAY_SIZE(wl1273_controls));

sound/soc/codecs/wm1250-ev1.c

@@ -12,10 +12,59 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
+#include <linux/slab.h>
 #include <linux/i2c.h>
+#include <linux/gpio.h>
 
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include <sound/wm1250-ev1.h>
+
+static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = {
+	"WM1250 CLK_ENA",
+	"WM1250 CLK_SEL0",
+	"WM1250 CLK_SEL1",
+	"WM1250 OSR",
+	"WM1250 MASTER",
+};
+
+struct wm1250_priv {
+	struct gpio gpios[WM1250_EV1_NUM_GPIOS];
+};
+
+static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec,
+				     enum snd_soc_bias_level level)
+{
+	struct wm1250_priv *wm1250 = dev_get_drvdata(codec->dev);
+	int ena;
+
+	if (wm1250)
+		ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio;
+	else
+		ena = -1;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (ena >= 0)
+			gpio_set_value_cansleep(ena, 1);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		if (ena >= 0)
+			gpio_set_value_cansleep(ena, 0);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+
+	return 0;
+}
 
 static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = {
 SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0),
@@ -53,12 +102,67 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = {
 	.num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets),
 	.dapm_routes = wm1250_ev1_dapm_routes,
 	.num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes),
+
+	.set_bias_level = wm1250_ev1_set_bias_level,
+	.idle_bias_off = true,
 };
 
+static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c)
+{
+	struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev);
+	struct wm1250_priv *wm1250;
+	int i, ret;
+
+	if (!pdata)
+		return 0;
+
+	wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL);
+	if (!wm1250) {
+		dev_err(&i2c->dev, "Unable to allocate private data\n");
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) {
+		wm1250->gpios[i].gpio = pdata->gpios[i];
+		wm1250->gpios[i].label = wm1250_gpio_names[i];
+		wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW;
+	}
+	wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH;
+	wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH;
+
+	ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios));
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret);
+		goto err_alloc;
+	}
+
+	dev_set_drvdata(&i2c->dev, wm1250);
+
+	return ret;
+
+err_alloc:
+	kfree(wm1250);
+err:
+	return ret;
+}
+
+static void wm1250_ev1_free(struct i2c_client *i2c)
+{
+	struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev);
+
+	if (wm1250) {
+		gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios));
+		kfree(wm1250);
+	}
+}
+
 static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
 				      const struct i2c_device_id *i2c_id)
 {
-	int id, board, rev;
+	int id, board, rev, ret;
+
+	dev_set_drvdata(&i2c->dev, NULL);
 
 	board = i2c_smbus_read_byte_data(i2c, 0);
 	if (board < 0) {
@@ -76,13 +180,25 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c,
 
 	dev_info(&i2c->dev, "revision %d\n", rev + 1);
 
-	return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
+	ret = wm1250_ev1_pdata(i2c);
+	if (ret != 0)
+		return ret;
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1,
 				     &wm1250_ev1_dai, 1);
+	if (ret != 0) {
+		dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret);
+		wm1250_ev1_free(i2c);
+		return ret;
+	}
+
+	return 0;
 }
 
 static int __devexit wm1250_ev1_remove(struct i2c_client *i2c)
 {
 	snd_soc_unregister_codec(&i2c->dev);
+	wm1250_ev1_free(i2c);
 
 	return 0;
 }

sound/soc/codecs/wm8350.c

@@ -355,7 +355,7 @@ static int wm8350_put_volsw_2r_vu(struct snd_kcontrol *kcontrol,
 			return 1;
 	}
 
-	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
 	if (ret < 0)
 		return ret;
 
@@ -392,23 +392,9 @@ static int wm8350_get_volsw_2r(struct snd_kcontrol *kcontrol,
 		break;
 	}
 
-	return snd_soc_get_volsw_2r(kcontrol, ucontrol);
+	return snd_soc_get_volsw(kcontrol, ucontrol);
 }
 
-/* double control with volume update */
-#define SOC_WM8350_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
-				xinvert, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
-		SNDRV_CTL_ELEM_ACCESS_READWRITE | \
-		SNDRV_CTL_ELEM_ACCESS_VOLATILE, \
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r, \
-	.get = wm8350_get_volsw_2r, .put = wm8350_put_volsw_2r_vu, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		 .rshift = xshift, .max = xmax, .invert = xinvert}, }
-
 static const char *wm8350_deemp[] = { "None", "32kHz", "44.1kHz", "48kHz" };
 static const char *wm8350_pol[] = { "Normal", "Inv R", "Inv L", "Inv L & R" };
 static const char *wm8350_dacmutem[] = { "Normal", "Soft" };
@@ -443,26 +429,29 @@ static const unsigned int capture_sd_tlv[] = {
 static const struct snd_kcontrol_new wm8350_snd_controls[] = {
 	SOC_ENUM("Playback Deemphasis", wm8350_enum[0]),
 	SOC_ENUM("Playback DAC Inversion", wm8350_enum[1]),
-	SOC_WM8350_DOUBLE_R_TLV("Playback PCM Volume",
+	SOC_DOUBLE_R_EXT_TLV("Playback PCM Volume",
 				WM8350_DAC_DIGITAL_VOLUME_L,
 				WM8350_DAC_DIGITAL_VOLUME_R,
-				0, 255, 0, dac_pcm_tlv),
+				0, 255, 0, wm8350_get_volsw_2r,
+				wm8350_put_volsw_2r_vu, dac_pcm_tlv),
 	SOC_ENUM("Playback PCM Mute Function", wm8350_enum[2]),
 	SOC_ENUM("Playback PCM Mute Speed", wm8350_enum[3]),
 	SOC_ENUM("Capture PCM Filter", wm8350_enum[4]),
 	SOC_ENUM("Capture PCM HP Filter", wm8350_enum[5]),
 	SOC_ENUM("Capture ADC Inversion", wm8350_enum[6]),
-	SOC_WM8350_DOUBLE_R_TLV("Capture PCM Volume",
+	SOC_DOUBLE_R_EXT_TLV("Capture PCM Volume",
 				WM8350_ADC_DIGITAL_VOLUME_L,
 				WM8350_ADC_DIGITAL_VOLUME_R,
-				0, 255, 0, adc_pcm_tlv),
+				0, 255, 0, wm8350_get_volsw_2r,
+				wm8350_put_volsw_2r_vu, adc_pcm_tlv),
 	SOC_DOUBLE_TLV("Capture Sidetone Volume",
 		       WM8350_ADC_DIVIDER,
 		       8, 4, 15, 1, capture_sd_tlv),
-	SOC_WM8350_DOUBLE_R_TLV("Capture Volume",
+	SOC_DOUBLE_R_EXT_TLV("Capture Volume",
 				WM8350_LEFT_INPUT_VOLUME,
 				WM8350_RIGHT_INPUT_VOLUME,
-				2, 63, 0, pre_amp_tlv),
+				2, 63, 0, wm8350_get_volsw_2r,
+				wm8350_put_volsw_2r_vu, pre_amp_tlv),
 	SOC_DOUBLE_R("Capture ZC Switch",
 		     WM8350_LEFT_INPUT_VOLUME,
 		     WM8350_RIGHT_INPUT_VOLUME, 13, 1, 0),
@@ -490,17 +479,19 @@ static const struct snd_kcontrol_new wm8350_snd_controls[] = {
 	SOC_SINGLE_TLV("Out4 Capture Volume",
 		       WM8350_INPUT_MIXER_VOLUME,
 		       1, 7, 0, out_mix_tlv),
-	SOC_WM8350_DOUBLE_R_TLV("Out1 Playback Volume",
+	SOC_DOUBLE_R_EXT_TLV("Out1 Playback Volume",
 				WM8350_LOUT1_VOLUME,
 				WM8350_ROUT1_VOLUME,
-				2, 63, 0, out_pga_tlv),
+				2, 63, 0, wm8350_get_volsw_2r,
+				wm8350_put_volsw_2r_vu, out_pga_tlv),
 	SOC_DOUBLE_R("Out1 Playback ZC Switch",
 		     WM8350_LOUT1_VOLUME,
 		     WM8350_ROUT1_VOLUME, 13, 1, 0),
-	SOC_WM8350_DOUBLE_R_TLV("Out2 Playback Volume",
+	SOC_DOUBLE_R_EXT_TLV("Out2 Playback Volume",
 				WM8350_LOUT2_VOLUME,
 				WM8350_ROUT2_VOLUME,
-				2, 63, 0, out_pga_tlv),
+				2, 63, 0, wm8350_get_volsw_2r,
+				wm8350_put_volsw_2r_vu, out_pga_tlv),
 	SOC_DOUBLE_R("Out2 Playback ZC Switch", WM8350_LOUT2_VOLUME,
 		     WM8350_ROUT2_VOLUME, 13, 1, 0),
 	SOC_SINGLE("Out2 Right Invert Switch", WM8350_ROUT2_VOLUME, 10, 1, 0),

sound/soc/codecs/wm8400.c

@@ -383,7 +383,7 @@ static int inmixer_event (struct snd_soc_dapm_widget *w,
 		(1 << WM8400_AINRMUX_PWR))) {
 		reg |= WM8400_AINR_ENA;
 	} else {
-		reg &= ~WM8400_AINL_ENA;
+		reg &= ~WM8400_AINR_ENA;
 	}
 	wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
 

sound/soc/codecs/wm8510.c

@@ -480,6 +480,8 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 		power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
 
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8510_POWER1, power1 | 0x3);
 			mdelay(100);
@@ -541,18 +543,7 @@ static int wm8510_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8510_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8510_reg); i++) {
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 

sound/soc/codecs/wm8523.c

@@ -406,7 +406,6 @@ static int wm8523_probe(struct snd_soc_codec *codec)
 	struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec);
 	int ret, i;
 
-	codec->hw_write = (hw_write_t)i2c_master_send;
 	wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0];
 	wm8523->rate_constraint.count =
 		ARRAY_SIZE(wm8523->rate_constraint_list);

sound/soc/codecs/wm8580.c

@@ -213,7 +213,7 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 	reg_cache[reg] = 0;
 	reg_cache[reg2] = 0;
 
-	ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
+	ret = snd_soc_put_volsw(kcontrol, ucontrol);
 	if (ret < 0)
 		return ret;
 
@@ -224,31 +224,19 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
-				    xinvert, tlv_array)			\
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
-		SNDRV_CTL_ELEM_ACCESS_READWRITE,  \
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r, \
-	.get = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
-		.max = xmax, .invert = xinvert} }
-
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
-SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
+SOC_DOUBLE_R_EXT_TLV("DAC1 Playback Volume",
 		     WM8580_DIGITAL_ATTENUATION_DACL1,
 		     WM8580_DIGITAL_ATTENUATION_DACR1,
-			    0, 0xff, 0, dac_tlv),
+		     0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv),
-SOC_WM8580_OUT_DOUBLE_R_TLV("DAC2 Playback Volume",
+SOC_DOUBLE_R_EXT_TLV("DAC2 Playback Volume",
 		     WM8580_DIGITAL_ATTENUATION_DACL2,
 		     WM8580_DIGITAL_ATTENUATION_DACR2,
-			    0, 0xff, 0, dac_tlv),
+		     0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv),
-SOC_WM8580_OUT_DOUBLE_R_TLV("DAC3 Playback Volume",
+SOC_DOUBLE_R_EXT_TLV("DAC3 Playback Volume",
 		     WM8580_DIGITAL_ATTENUATION_DACL3,
 		     WM8580_DIGITAL_ATTENUATION_DACR3,
-			    0, 0xff, 0, dac_tlv),
+		     0, 0xff, 0, snd_soc_get_volsw, wm8580_out_vu, dac_tlv),
 
 SOC_SINGLE("DAC1 Deemphasis Switch", WM8580_DAC_CONTROL3, 0, 1, 0),
 SOC_SINGLE("DAC2 Deemphasis Switch", WM8580_DAC_CONTROL3, 1, 1, 0),
@@ -442,8 +430,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	/* Always disable the PLL - it is not safe to leave it running
 	 * while reprogramming it.
 	 */
-	reg = snd_soc_read(codec, WM8580_PWRDN2);
+	snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, pwr_mask);
-	snd_soc_write(codec, WM8580_PWRDN2, reg | pwr_mask);
 
 	if (!freq_in || !freq_out)
 		return 0;
@@ -461,8 +448,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 	snd_soc_write(codec, WM8580_PLLA4 + offset, reg);
 
 	/* All done, turn it on */
-	reg = snd_soc_read(codec, WM8580_PWRDN2);
+	snd_soc_update_bits(codec, WM8580_PWRDN2, pwr_mask, 0);
-	snd_soc_write(codec, WM8580_PWRDN2, reg & ~pwr_mask);
 
 	return 0;
 }
@@ -760,7 +746,6 @@ static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
-	u16 reg;
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
@@ -769,20 +754,19 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
 			/* Power up and get individual control of the DACs */
-			reg = snd_soc_read(codec, WM8580_PWRDN1);
+			snd_soc_update_bits(codec, WM8580_PWRDN1,
-			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
+					    WM8580_PWRDN1_PWDN |
-			snd_soc_write(codec, WM8580_PWRDN1, reg);
+					    WM8580_PWRDN1_ALLDACPD, 0);
 
 			/* Make VMID high impedance */
-			reg = snd_soc_read(codec,  WM8580_ADC_CONTROL1);
+			snd_soc_update_bits(codec, WM8580_ADC_CONTROL1,
-			reg &= ~0x100;
+					    0x100, 0);
-			snd_soc_write(codec, WM8580_ADC_CONTROL1, reg);
 		}
 		break;
 
 	case SND_SOC_BIAS_OFF:
-		reg = snd_soc_read(codec, WM8580_PWRDN1);
+		snd_soc_update_bits(codec, WM8580_PWRDN1,
-		snd_soc_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
+				    WM8580_PWRDN1_PWDN, WM8580_PWRDN1_PWDN);
 		break;
 	}
 	codec->dapm.bias_level = level;

sound/soc/codecs/wm8711.c

@@ -3,7 +3,7 @@
  *
  * Copyright 2006 Wolfson Microelectronics
  *
- * Author: Mike Arthur <linux@wolfsonmicro.com>
+ * Author: Mike Arthur <Mike.Arthur@wolfsonmicro.com>
  *
  * Based on wm8731.c by Richard Purdie
  *
@@ -287,7 +287,6 @@ static int wm8711_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-
 static int wm8711_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
@@ -300,6 +299,9 @@ static int wm8711_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			snd_soc_cache_sync(codec);
+
 		snd_soc_write(codec, WM8711_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
@@ -346,25 +348,14 @@ static int wm8711_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8711_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8711_reg); i++) {
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
 	wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
 static int wm8711_probe(struct snd_soc_codec *codec)
 {
 	struct wm8711_priv *wm8711 = snd_soc_codec_get_drvdata(codec);
-	int ret, reg;
+	int ret;
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8711->bus_type);
 	if (ret < 0) {
@@ -381,10 +372,8 @@ static int wm8711_probe(struct snd_soc_codec *codec)
 	wm8711_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the update bits */
-	reg = snd_soc_read(codec, WM8711_LOUT1V);
+	snd_soc_update_bits(codec, WM8711_LOUT1V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8711_LOUT1V, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8711_ROUT1V, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8711_ROUT1V);
-	snd_soc_write(codec, WM8711_ROUT1V, reg | 0x0100);
 
 	snd_soc_add_controls(codec, wm8711_snd_controls,
 			     ARRAY_SIZE(wm8711_snd_controls));

sound/soc/codecs/wm8731.c

@@ -427,9 +427,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
-	int i, ret;
+	int ret;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
 	u16 reg;
 
 	switch (level) {
@@ -444,16 +442,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 			if (ret != 0)
 				return ret;
 
-			/* Sync reg_cache with the hardware */
+			snd_soc_cache_sync(codec);
-			for (i = 0; i < ARRAY_SIZE(wm8731_reg); i++) {
-				if (cache[i] == wm8731_reg[i])
-					continue;
-
-				data[0] = (i << 1) | ((cache[i] >> 8)
-						      & 0x0001);
-				data[1] = cache[i] & 0x00ff;
-				codec->hw_write(codec->control_data, data, 2);
-			}
 		}
 
 		/* Clear PWROFF, gate CLKOUT, everything else as-is */

sound/soc/codecs/wm8741.c

@@ -339,10 +339,10 @@ static int wm8741_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		iface |= 0x0004;
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
-		iface |= 0x0003;
+		iface |= 0x000C;
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
-		iface |= 0x0013;
+		iface |= 0x001C;
 		break;
 	default:
 		return -EINVAL;
@@ -404,15 +404,7 @@ static struct snd_soc_dai_driver wm8741_dai = {
 #ifdef CONFIG_PM
 static int wm8741_resume(struct snd_soc_codec *codec)
 {
-	u16 *cache = codec->reg_cache;
+	snd_soc_cache_sync(codec);
-	int i;
-
-	/* RESTORE REG Cache */
-	for (i = 0; i < WM8741_REGISTER_COUNT; i++) {
-		if (cache[i] == wm8741_reg_defaults[i] || WM8741_RESET == i)
-			continue;
-		snd_soc_write(codec, i, cache[i]);
-	}
 	return 0;
 }
 #else
@@ -462,7 +454,7 @@ static int wm8741_probe(struct snd_soc_codec *codec)
 			    WM8741_UPDATELM, WM8741_UPDATELM);
 	snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION,
 			    WM8741_UPDATERL, WM8741_UPDATERL);
-	snd_soc_update_bits(codec, WM8741_DACRLSB_ATTENUATION,
+	snd_soc_update_bits(codec, WM8741_DACRMSB_ATTENUATION,
 			    WM8741_UPDATERM, WM8741_UPDATERM);
 
 	snd_soc_add_controls(codec, wm8741_snd_controls,

sound/soc/codecs/wm8750.c

@@ -616,6 +616,8 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* Set VMID to 5k */
 			snd_soc_write(codec, WM8750_PWR1, pwr_reg | 0x01c1);
 
@@ -673,28 +675,14 @@ static int wm8750_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8750_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8750_reg); i++) {
-		if (i == WM8750_RESET)
-			continue;
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
-
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
 static int wm8750_probe(struct snd_soc_codec *codec)
 {
 	struct wm8750_priv *wm8750 = snd_soc_codec_get_drvdata(codec);
-	int reg, ret;
+	int ret;
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8750->control_type);
 	if (ret < 0) {
@@ -712,22 +700,14 @@ static int wm8750_probe(struct snd_soc_codec *codec)
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* set the update bits */
-	reg = snd_soc_read(codec, WM8750_LDAC);
+	snd_soc_update_bits(codec, WM8750_LDAC, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8750_LDAC, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8750_RDAC, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8750_RDAC);
+	snd_soc_update_bits(codec, WM8750_LOUT1V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8750_RDAC, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8750_ROUT1V, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8750_LOUT1V);
+	snd_soc_update_bits(codec, WM8750_LOUT2V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8750_LOUT1V, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8750_ROUT2V, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8750_ROUT1V);
+	snd_soc_update_bits(codec, WM8750_LINVOL, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8750_ROUT1V, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8750_RINVOL, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8750_LOUT2V);
-	snd_soc_write(codec, WM8750_LOUT2V, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8750_ROUT2V);
-	snd_soc_write(codec, WM8750_ROUT2V, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8750_LINVOL);
-	snd_soc_write(codec, WM8750_LINVOL, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8750_RINVOL);
-	snd_soc_write(codec, WM8750_RINVOL, reg | 0x0100);
 
 	snd_soc_add_controls(codec, wm8750_snd_controls,
 				ARRAY_SIZE(wm8750_snd_controls));

sound/soc/codecs/wm8753.c

@@ -1455,8 +1455,8 @@ static int wm8753_probe(struct snd_soc_codec *codec)
 	/* set the update bits */
 	snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100);
-	snd_soc_update_bits(codec, WM8753_LDAC, 0x0100, 0x0100);
+	snd_soc_update_bits(codec, WM8753_LADC, 0x0100, 0x0100);
-	snd_soc_update_bits(codec, WM8753_RDAC, 0x0100, 0x0100);
+	snd_soc_update_bits(codec, WM8753_RADC, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_LOUT1V, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_ROUT1V, 0x0100, 0x0100);
 	snd_soc_update_bits(codec, WM8753_LOUT2V, 0x0100, 0x0100);

sound/soc/codecs/wm8776.c

@@ -308,6 +308,8 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* Disable the global powerdown; DAPM does the rest */
 			snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0);
 		}
@@ -379,21 +381,7 @@ static int wm8776_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8776_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) {
-		if (cache[i] == wm8776_reg[i])
-			continue;
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
-
 	wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 #else

sound/soc/codecs/wm8782.c

@@ -60,7 +60,7 @@ static struct platform_driver wm8782_codec_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = wm8782_probe,
-	.remove = wm8782_remove,
+	.remove = __devexit_p(wm8782_remove),
 };
 
 static int __init wm8782_init(void)

sound/soc/codecs/wm8900.c

@@ -110,8 +110,8 @@
 
 #define WM8900_REG_CLOCKING1_BCLK_DIR   0x1
 #define WM8900_REG_CLOCKING1_MCLK_SRC   0x100
-#define WM8900_REG_CLOCKING1_BCLK_MASK  (~0x01e)
+#define WM8900_REG_CLOCKING1_BCLK_MASK  0x01e
-#define WM8900_REG_CLOCKING1_OPCLK_MASK (~0x7000)
+#define WM8900_REG_CLOCKING1_OPCLK_MASK 0x7000
 
 #define WM8900_REG_CLOCKING2_ADC_CLKDIV 0xe0
 #define WM8900_REG_CLOCKING2_DAC_CLKDIV 0x1c
@@ -135,7 +135,7 @@
 #define WM8900_REG_HPCTL1_HP_SHORT       0x08
 #define WM8900_REG_HPCTL1_HP_SHORT2      0x04
 
-#define WM8900_LRC_MASK 0xfc00
+#define WM8900_LRC_MASK 0x03ff
 
 struct wm8900_priv {
 	enum snd_soc_control_type control_type;
@@ -742,26 +742,20 @@ static int wm8900_set_fll(struct snd_soc_codec *codec,
 {
 	struct wm8900_priv *wm8900 = snd_soc_codec_get_drvdata(codec);
 	struct _fll_div fll_div;
-	unsigned int reg;
 
 	if (wm8900->fll_in == freq_in && wm8900->fll_out == freq_out)
 		return 0;
 
 	/* The digital side should be disabled during any change. */
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
+	snd_soc_update_bits(codec, WM8900_REG_POWER1,
-	snd_soc_write(codec, WM8900_REG_POWER1,
+			    WM8900_REG_POWER1_FLL_ENA, 0);
-		     reg & (~WM8900_REG_POWER1_FLL_ENA));
 
 	/* Disable the FLL? */
 	if (!freq_in || !freq_out) {
-		reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+		snd_soc_update_bits(codec, WM8900_REG_CLOCKING1,
-		snd_soc_write(codec, WM8900_REG_CLOCKING1,
+				    WM8900_REG_CLOCKING1_MCLK_SRC, 0);
-			     reg & (~WM8900_REG_CLOCKING1_MCLK_SRC));
+		snd_soc_update_bits(codec, WM8900_REG_FLLCTL1,
-
+				    WM8900_REG_FLLCTL1_OSC_ENA, 0);
-		reg = snd_soc_read(codec, WM8900_REG_FLLCTL1);
-		snd_soc_write(codec, WM8900_REG_FLLCTL1,
-			     reg & (~WM8900_REG_FLLCTL1_OSC_ENA));
-
 		wm8900->fll_in = freq_in;
 		wm8900->fll_out = freq_out;
 
@@ -796,15 +790,14 @@ static int wm8900_set_fll(struct snd_soc_codec *codec,
 	else
 		snd_soc_write(codec, WM8900_REG_FLLCTL6, 0);
 
-	reg = snd_soc_read(codec, WM8900_REG_POWER1);
+	snd_soc_update_bits(codec, WM8900_REG_POWER1,
-	snd_soc_write(codec, WM8900_REG_POWER1,
+			    WM8900_REG_POWER1_FLL_ENA,
-		     reg | WM8900_REG_POWER1_FLL_ENA);
+			    WM8900_REG_POWER1_FLL_ENA);
 
 reenable:
-	reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+	snd_soc_update_bits(codec, WM8900_REG_CLOCKING1,
-	snd_soc_write(codec, WM8900_REG_CLOCKING1,
+			    WM8900_REG_CLOCKING1_MCLK_SRC,
-		     reg | WM8900_REG_CLOCKING1_MCLK_SRC);
+			    WM8900_REG_CLOCKING1_MCLK_SRC);
-
 	return 0;
 }
 
@@ -818,43 +811,35 @@ static int wm8900_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 				 int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	unsigned int reg;
 
 	switch (div_id) {
 	case WM8900_BCLK_DIV:
-		reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+		snd_soc_update_bits(codec, WM8900_REG_CLOCKING1,
-		snd_soc_write(codec, WM8900_REG_CLOCKING1,
+				    WM8900_REG_CLOCKING1_BCLK_MASK, div);
-			     div | (reg & WM8900_REG_CLOCKING1_BCLK_MASK));
 		break;
 	case WM8900_OPCLK_DIV:
-		reg = snd_soc_read(codec, WM8900_REG_CLOCKING1);
+		snd_soc_update_bits(codec, WM8900_REG_CLOCKING1,
-		snd_soc_write(codec, WM8900_REG_CLOCKING1,
+				    WM8900_REG_CLOCKING1_OPCLK_MASK, div);
-			     div | (reg & WM8900_REG_CLOCKING1_OPCLK_MASK));
 		break;
 	case WM8900_DAC_LRCLK:
-		reg = snd_soc_read(codec, WM8900_REG_AUDIO4);
+		snd_soc_update_bits(codec, WM8900_REG_AUDIO4,
-		snd_soc_write(codec, WM8900_REG_AUDIO4,
+				    WM8900_LRC_MASK, div);
-			     div | (reg & WM8900_LRC_MASK));
 		break;
 	case WM8900_ADC_LRCLK:
-		reg = snd_soc_read(codec, WM8900_REG_AUDIO3);
+		snd_soc_update_bits(codec, WM8900_REG_AUDIO3,
-		snd_soc_write(codec, WM8900_REG_AUDIO3,
+				    WM8900_LRC_MASK, div);
-			     div | (reg & WM8900_LRC_MASK));
 		break;
 	case WM8900_DAC_CLKDIV:
-		reg = snd_soc_read(codec, WM8900_REG_CLOCKING2);
+		snd_soc_update_bits(codec, WM8900_REG_CLOCKING2,
-		snd_soc_write(codec, WM8900_REG_CLOCKING2,
+				    WM8900_REG_CLOCKING2_DAC_CLKDIV, div);
-			     div | (reg & WM8900_REG_CLOCKING2_DAC_CLKDIV));
 		break;
 	case WM8900_ADC_CLKDIV:
-		reg = snd_soc_read(codec, WM8900_REG_CLOCKING2);
+		snd_soc_update_bits(codec, WM8900_REG_CLOCKING2,
-		snd_soc_write(codec, WM8900_REG_CLOCKING2,
+				    WM8900_REG_CLOCKING2_ADC_CLKDIV, div);
-			     div | (reg & WM8900_REG_CLOCKING2_ADC_CLKDIV));
 		break;
 	case WM8900_LRCLK_MODE:
-		reg = snd_soc_read(codec, WM8900_REG_DACCTRL);
+		snd_soc_update_bits(codec, WM8900_REG_DACCTRL,
-		snd_soc_write(codec, WM8900_REG_DACCTRL,
+				    WM8900_REG_DACCTRL_AIF_LRCLKRATE, div);
-			     div | (reg & WM8900_REG_DACCTRL_AIF_LRCLKRATE));
 		break;
 	default:
 		return -EINVAL;
@@ -1037,12 +1022,12 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		/* Enable thermal shutdown */
-		reg = snd_soc_read(codec, WM8900_REG_GPIO);
+		snd_soc_update_bits(codec, WM8900_REG_GPIO,
-		snd_soc_write(codec, WM8900_REG_GPIO,
+				    WM8900_REG_GPIO_TEMP_ENA,
-			     reg | WM8900_REG_GPIO_TEMP_ENA);
+				    WM8900_REG_GPIO_TEMP_ENA);
-		reg = snd_soc_read(codec, WM8900_REG_ADDCTL);
+		snd_soc_update_bits(codec, WM8900_REG_ADDCTL,
-		snd_soc_write(codec, WM8900_REG_ADDCTL,
+				    WM8900_REG_ADDCTL_TEMP_SD,
-			     reg | WM8900_REG_ADDCTL_TEMP_SD);
+				    WM8900_REG_ADDCTL_TEMP_SD);
 		break;
 
 	case SND_SOC_BIAS_PREPARE:
@@ -1205,26 +1190,16 @@ static int wm8900_probe(struct snd_soc_codec *codec)
 	wm8900_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* Latch the volume update bits */
-	snd_soc_write(codec, WM8900_REG_LINVOL,
+	snd_soc_update_bits(codec, WM8900_REG_LINVOL, 0x100, 0x100);
-		      snd_soc_read(codec, WM8900_REG_LINVOL) | 0x100);
+	snd_soc_update_bits(codec, WM8900_REG_RINVOL, 0x100, 0x100);
-	snd_soc_write(codec, WM8900_REG_RINVOL,
+	snd_soc_update_bits(codec, WM8900_REG_LOUT1CTL, 0x100, 0x100);
-		      snd_soc_read(codec, WM8900_REG_RINVOL) | 0x100);
+	snd_soc_update_bits(codec, WM8900_REG_ROUT1CTL, 0x100, 0x100);
-	snd_soc_write(codec, WM8900_REG_LOUT1CTL,
+	snd_soc_update_bits(codec, WM8900_REG_LOUT2CTL, 0x100, 0x100);
-		      snd_soc_read(codec, WM8900_REG_LOUT1CTL) | 0x100);
+	snd_soc_update_bits(codec, WM8900_REG_ROUT2CTL, 0x100, 0x100);
-	snd_soc_write(codec, WM8900_REG_ROUT1CTL,
+	snd_soc_update_bits(codec, WM8900_REG_LDAC_DV, 0x100, 0x100);
-		      snd_soc_read(codec, WM8900_REG_ROUT1CTL) | 0x100);
+	snd_soc_update_bits(codec, WM8900_REG_RDAC_DV, 0x100, 0x100);
-	snd_soc_write(codec, WM8900_REG_LOUT2CTL,
+	snd_soc_update_bits(codec, WM8900_REG_LADC_DV, 0x100, 0x100);
-		      snd_soc_read(codec, WM8900_REG_LOUT2CTL) | 0x100);
+	snd_soc_update_bits(codec, WM8900_REG_RADC_DV, 0x100, 0x100);
-	snd_soc_write(codec, WM8900_REG_ROUT2CTL,
-		      snd_soc_read(codec, WM8900_REG_ROUT2CTL) | 0x100);
-	snd_soc_write(codec, WM8900_REG_LDAC_DV,
-		      snd_soc_read(codec, WM8900_REG_LDAC_DV) | 0x100);
-	snd_soc_write(codec, WM8900_REG_RDAC_DV,
-		      snd_soc_read(codec, WM8900_REG_RDAC_DV) | 0x100);
-	snd_soc_write(codec, WM8900_REG_LADC_DV,
-		      snd_soc_read(codec, WM8900_REG_LADC_DV) | 0x100);
-	snd_soc_write(codec, WM8900_REG_RADC_DV,
-		      snd_soc_read(codec, WM8900_REG_RADC_DV) | 0x100);
 
 	/* Set the DAC and mixer output bias */
 	snd_soc_write(codec, WM8900_REG_OUTBIASCTL, 0x81);

sound/soc/codecs/wm8904.c

@@ -50,7 +50,6 @@ static const char *wm8904_supply_names[WM8904_NUM_SUPPLIES] = {
 struct wm8904_priv {
 
 	enum wm8904_type devtype;
-	void *control_data;
 
 	struct regulator_bulk_data supplies[WM8904_NUM_SUPPLIES];
 
@@ -2540,7 +2539,6 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c,
 
 	wm8904->devtype = id->driver_data;
 	i2c_set_clientdata(i2c, wm8904);
-	wm8904->control_data = i2c;
 	wm8904->pdata = i2c->dev.platform_data;
 
 	ret = snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/wm8940.c

@@ -43,9 +43,19 @@
 struct wm8940_priv {
 	unsigned int sysclk;
 	enum snd_soc_control_type control_type;
-	void *control_data;
 };
 
+static int wm8940_volatile_register(struct snd_soc_codec *codec,
+				    unsigned int reg)
+{
+	switch (reg) {
+	case WM8940_SOFTRESET:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
 static u16 wm8940_reg_defaults[] = {
 	0x8940, /* Soft Reset */
 	0x0000, /* Power 1 */
@@ -460,6 +470,14 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		ret = snd_soc_write(codec, WM8940_POWER1, pwr_reg | 0x1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = snd_soc_cache_sync(codec);
+			if (ret < 0) {
+				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+
 		/* ensure bufioen and biasen */
 		pwr_reg |= (1 << 2) | (1 << 3);
 		/* set vmid to 300k for standby */
@@ -470,6 +488,8 @@ static int wm8940_set_bias_level(struct snd_soc_codec *codec,
 		break;
 	}
 
+	codec->dapm.bias_level = level;
+
 	return ret;
 }
 
@@ -660,30 +680,8 @@ static int wm8940_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8940_resume(struct snd_soc_codec *codec)
 {
-	int i;
+	wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	int ret;
+	return 0;
-	u8 data[3];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware
-	 * Could use auto incremented writes to speed this up
-	 */
-	for (i = 0; i < ARRAY_SIZE(wm8940_reg_defaults); i++) {
-		data[0] = i;
-		data[1] = (cache[i] & 0xFF00) >> 8;
-		data[2] = cache[i] & 0x00FF;
-		ret = codec->hw_write(codec->control_data, data, 3);
-		if (ret < 0)
-			goto error_ret;
-		else if (ret != 3) {
-			ret = -EIO;
-			goto error_ret;
-		}
-	}
-	ret = wm8940_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-error_ret:
-	return ret;
 }
 
 static int wm8940_probe(struct snd_soc_codec *codec)
@@ -693,7 +691,6 @@ static int wm8940_probe(struct snd_soc_codec *codec)
 	int ret;
 	u16 reg;
 
-	codec->control_data = wm8940->control_data;
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm8940->control_type);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
@@ -744,6 +741,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8940 = {
 	.reg_cache_size = ARRAY_SIZE(wm8940_reg_defaults),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8940_reg_defaults,
+	.volatile_register = wm8940_volatile_register,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
@@ -758,7 +756,6 @@ static __devinit int wm8940_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, wm8940);
-	wm8940->control_data = i2c;
 	wm8940->control_type = SND_SOC_I2C;
 
 	ret = snd_soc_register_codec(&i2c->dev,

sound/soc/codecs/wm8960.c

@@ -72,7 +72,6 @@ static const u16 wm8960_reg[WM8960_CACHEREGNUM] = {
 
 struct wm8960_priv {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	int (*set_bias_level)(struct snd_soc_codec *,
 			      enum snd_soc_bias_level level);
 	struct snd_soc_dapm_widget *lout1;
@@ -575,6 +574,8 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* Enable anti-pop features */
 			snd_soc_write(codec, WM8960_APOP1,
 				      WM8960_POBCTRL | WM8960_SOFT_ST |
@@ -677,6 +678,9 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec,
 					    WM8960_VREF | WM8960_VMID_MASK, 0);
 			break;
 
+		case SND_SOC_BIAS_OFF:
+			snd_soc_cache_sync(codec);
+			break;
 		default:
 			break;
 		}
@@ -902,16 +906,6 @@ static int wm8960_suspend(struct snd_soc_codec *codec, pm_message_t state)
 static int wm8960_resume(struct snd_soc_codec *codec)
 {
 	struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec);
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) {
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
 
 	wm8960->set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	return 0;
@@ -925,7 +919,6 @@ static int wm8960_probe(struct snd_soc_codec *codec)
 	u16 reg;
 
 	wm8960->set_bias_level = wm8960_set_bias_level_out3;
-	codec->control_data = wm8960->control_data;
 
 	if (!pdata) {
 		dev_warn(codec->dev, "No platform data supplied\n");
@@ -1015,7 +1008,6 @@ static __devinit int wm8960_i2c_probe(struct i2c_client *i2c,
 
 	i2c_set_clientdata(i2c, wm8960);
 	wm8960->control_type = SND_SOC_I2C;
-	wm8960->control_data = i2c;
 
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8960, &wm8960_dai, 1);

sound/soc/codecs/wm8961.c

@@ -974,7 +974,9 @@ static int wm8961_probe(struct snd_soc_codec *codec)
 	}
 
 	/* This isn't volatile - readback doesn't correspond to write */
-	reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME);
+	codec->cache_bypass = 1;
+	reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME);
+	codec->cache_bypass = 0;
 	dev_info(codec->dev, "WM8961 family %d revision %c\n",
 		 (reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT,
 		 ((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT)

sound/soc/codecs/wm8962.c

@@ -2139,7 +2139,6 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol,
 			    struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	u16 *reg_cache = codec->reg_cache;
 	int ret;
 
 	/* Apply the update (if any) */
@@ -2148,16 +2147,19 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	/* If the left PGA is enabled hit that VU bit... */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTL_PGA_ENA)
+	ret = snd_soc_read(codec, WM8962_PWR_MGMT_2);
-		return snd_soc_write(codec, WM8962_SPKOUTL_VOLUME,
+	if (ret & WM8962_SPKOUTL_PGA_ENA) {
-				     reg_cache[WM8962_SPKOUTL_VOLUME]);
+		snd_soc_write(codec, WM8962_SPKOUTL_VOLUME,
+			      snd_soc_read(codec, WM8962_SPKOUTL_VOLUME));
+		return 1;
+	}
 
 	/* ...otherwise the right.  The VU is stereo. */
-	if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTR_PGA_ENA)
+	if (ret & WM8962_SPKOUTR_PGA_ENA)
-		return snd_soc_write(codec, WM8962_SPKOUTR_VOLUME,
+		snd_soc_write(codec, WM8962_SPKOUTR_VOLUME,
-				     reg_cache[WM8962_SPKOUTR_VOLUME]);
+			      snd_soc_read(codec, WM8962_SPKOUTR_VOLUME));
 
-	return 0;
+	return 1;
 }
 
 static const char *cap_hpf_mode_text[] = {
@@ -2498,7 +2500,6 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event)
 {
 	struct snd_soc_codec *codec = w->codec;
-	u16 *reg_cache = codec->reg_cache;
 	int reg;
 
 	switch (w->shift) {
@@ -2521,7 +2522,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w,
 
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		return snd_soc_write(codec, reg, reg_cache[reg]);
+		return snd_soc_write(codec, reg, snd_soc_read(codec, reg));
 	default:
 		BUG();
 		return -EINVAL;
@@ -2667,7 +2668,7 @@ SND_SOC_DAPM_INPUT("IN4R"),
 SND_SOC_DAPM_INPUT("Beep"),
 SND_SOC_DAPM_INPUT("DMICDAT"),
 
-SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0),
+SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0),
 SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event,
@@ -2688,7 +2689,7 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0,
 SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0,
 		   mixinr, ARRAY_SIZE(mixinr)),
 
-SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0),
+SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0),
 
 SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0),
 SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0),
@@ -2767,18 +2768,18 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = {
 
 	{ "MICBIAS", NULL, "SYSCLK" },
 
-	{ "DMIC", NULL, "DMICDAT" },
+	{ "DMIC_ENA", NULL, "DMICDAT" },
 
 	{ "ADCL", NULL, "SYSCLK" },
 	{ "ADCL", NULL, "TOCLK" },
 	{ "ADCL", NULL, "MIXINL" },
-	{ "ADCL", NULL, "DMIC" },
+	{ "ADCL", NULL, "DMIC_ENA" },
 	{ "ADCL", NULL, "DSP2" },
 
 	{ "ADCR", NULL, "SYSCLK" },
 	{ "ADCR", NULL, "TOCLK" },
 	{ "ADCR", NULL, "MIXINR" },
-	{ "ADCR", NULL, "DMIC" },
+	{ "ADCR", NULL, "DMIC_ENA" },
 	{ "ADCR", NULL, "DSP2" },
 
 	{ "STL", "Left", "ADCL" },
@@ -3223,9 +3224,9 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	int aif0 = 0;
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_DSP_A:
-		aif0 |= WM8962_LRCLK_INV;
 	case SND_SOC_DAIFMT_DSP_B:
+		aif0 |= WM8962_LRCLK_INV | 3;
+	case SND_SOC_DAIFMT_DSP_A:
 		aif0 |= 3;
 
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {

sound/soc/codecs/wm8971.c

@@ -546,6 +546,9 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF)
+			snd_soc_cache_sync(codec);
+
 		/* mute dac and set vmid to 500k, enable VREF */
 		snd_soc_write(codec, WM8971_PWR1, pwr_reg | 0x0140);
 		break;
@@ -605,20 +608,8 @@ static int wm8971_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8971_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
 	u16 reg;
 
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8971_reg); i++) {
-		if (i + 1 == WM8971_RESET)
-			continue;
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
-
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* charge wm8971 caps */
@@ -660,25 +651,14 @@ static int wm8971_probe(struct snd_soc_codec *codec)
 		msecs_to_jiffies(1000));
 
 	/* set the update bits */
-	reg = snd_soc_read(codec, WM8971_LDAC);
+	snd_soc_update_bits(codec, WM8971_LDAC, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8971_LDAC, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8971_RDAC, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8971_RDAC);
+	snd_soc_update_bits(codec, WM8971_LOUT1V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8971_RDAC, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8971_ROUT1V, 0x0100, 0x0100);
-
+	snd_soc_update_bits(codec, WM8971_LOUT2V, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8971_LOUT1V);
+	snd_soc_update_bits(codec, WM8971_ROUT2V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8971_LOUT1V, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8971_LINVOL, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8971_ROUT1V);
+	snd_soc_update_bits(codec, WM8971_RINVOL, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8971_ROUT1V, reg | 0x0100);
-
-	reg = snd_soc_read(codec, WM8971_LOUT2V);
-	snd_soc_write(codec, WM8971_LOUT2V, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8971_ROUT2V);
-	snd_soc_write(codec, WM8971_ROUT2V, reg | 0x0100);
-
-	reg = snd_soc_read(codec, WM8971_LINVOL);
-	snd_soc_write(codec, WM8971_LINVOL, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8971_RINVOL);
-	snd_soc_write(codec, WM8971_RINVOL, reg | 0x0100);
 
 	snd_soc_add_controls(codec, wm8971_snd_controls,
 				ARRAY_SIZE(wm8971_snd_controls));

sound/soc/codecs/wm8974.c

@@ -3,7 +3,7 @@
  *
  * Copyright 2006-2009 Wolfson Microelectronics PLC.
  *
- * Author: Liam Girdwood <linux@wolfsonmicro.com>
+ * Author: Liam Girdwood <Liam.Girdwood@wolfsonmicro.com>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
@@ -530,6 +530,8 @@ static int wm8974_set_bias_level(struct snd_soc_codec *codec,
 		power1 |= WM8974_POWER1_BIASEN | WM8974_POWER1_BUFIOEN;
 
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* Initial cap charge at VMID 5k */
 			snd_soc_write(codec, WM8974_POWER1, power1 | 0x3);
 			mdelay(100);
@@ -589,18 +591,7 @@ static int wm8974_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8974_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8974_reg); i++) {
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
 	wm8974_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 

sound/soc/codecs/wm8978.c

@@ -52,7 +52,6 @@ static const u16 wm8978_reg[WM8978_CACHEREGNUM] = {
 /* codec private data */
 struct wm8978_priv {
 	enum snd_soc_control_type control_type;
-	void *control_data;
 	unsigned int f_pllout;
 	unsigned int f_mclk;
 	unsigned int f_256fs;
@@ -955,7 +954,6 @@ static int wm8978_probe(struct snd_soc_codec *codec)
 	 * default hardware setting
 	 */
 	wm8978->sysclk = WM8978_PLL;
-	codec->control_data = wm8978->control_data;
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_I2C);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
@@ -1016,7 +1014,6 @@ static __devinit int wm8978_i2c_probe(struct i2c_client *i2c,
 		return -ENOMEM;
 
 	i2c_set_clientdata(i2c, wm8978);
-	wm8978->control_data = i2c;
 
 	ret = snd_soc_register_codec(&i2c->dev,
 			&soc_codec_dev_wm8978, &wm8978_dai, 1);

sound/soc/codecs/wm8983.c

@@ -1007,7 +1007,7 @@ static int wm8983_probe(struct snd_soc_codec *codec)
 		return ret;
 	}
 
-	ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0x8983);
+	ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0);
 	if (ret < 0) {
 		dev_err(codec->dev, "Failed to issue reset: %d\n", ret);
 		return ret;

sound/soc/codecs/wm8988.c

@@ -55,7 +55,6 @@ struct wm8988_priv {
 	struct snd_pcm_hw_constraint_list *sysclk_constraints;
 };
 
-
 #define wm8988_reset(c)	snd_soc_write(c, WM8988_RESET, 0)
 
 /*
@@ -676,6 +675,8 @@ static int wm8988_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			snd_soc_cache_sync(codec);
+
 			/* VREF, VMID=2x5k */
 			snd_soc_write(codec, WM8988_PWR1, pwr_reg | 0x1c1);
 
@@ -736,21 +737,7 @@ static int wm8988_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8988_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < WM8988_NUM_REG; i++) {
-		if (i == WM8988_RESET)
-			continue;
-		data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
-
 	wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
 	return 0;
 }
 
@@ -759,7 +746,6 @@ static int wm8988_probe(struct snd_soc_codec *codec)
 	struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec);
 	struct snd_soc_dapm_context *dapm = &codec->dapm;
 	int ret = 0;
-	u16 reg;
 
 	ret = snd_soc_codec_set_cache_io(codec, 7, 9, wm8988->control_type);
 	if (ret < 0) {
@@ -774,16 +760,11 @@ static int wm8988_probe(struct snd_soc_codec *codec)
 	}
 
 	/* set the update bits (we always update left then right) */
-	reg = snd_soc_read(codec, WM8988_RADC);
+	snd_soc_update_bits(codec, WM8988_RADC, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8988_RADC, reg | 0x100);
+	snd_soc_update_bits(codec, WM8988_RDAC, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8988_RDAC);
+	snd_soc_update_bits(codec, WM8988_ROUT1V, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8988_RDAC, reg | 0x0100);
+	snd_soc_update_bits(codec, WM8988_ROUT2V, 0x0100, 0x0100);
-	reg = snd_soc_read(codec, WM8988_ROUT1V);
+	snd_soc_update_bits(codec, WM8988_RINVOL, 0x0100, 0x0100);
-	snd_soc_write(codec, WM8988_ROUT1V, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8988_ROUT2V);
-	snd_soc_write(codec, WM8988_ROUT2V, reg | 0x0100);
-	reg = snd_soc_read(codec, WM8988_RINVOL);
-	snd_soc_write(codec, WM8988_RINVOL, reg | 0x0100);
 
 	wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 

sound/soc/codecs/wm8990.c

@@ -36,10 +36,17 @@ struct wm8990_priv {
 	unsigned int pcmclk;
 };
 
-/*
+static int wm8990_volatile_register(struct snd_soc_codec *codec,
- * wm8990 register cache.  Note that register 0 is not included in the
+				    unsigned int reg)
- * cache.
+{
- */
+	switch (reg) {
+	case WM8990_RESET:
+		return 1;
+	default:
+		return 0;
+	}
+}
+
 static const u16 wm8990_reg[] = {
 	0x8990,     /* R0  - Reset */
 	0x0000,     /* R1  - Power Management (1) */
@@ -394,7 +401,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w,
 		(1 << WM8990_AINRMUX_PWR_BIT))) {
 		reg |= WM8990_AINR_ENA;
 	} else {
-		reg &= ~WM8990_AINL_ENA;
+		reg &= ~WM8990_AINR_ENA;
 	}
 	snd_soc_write(w->codec, WM8990_POWER_MANAGEMENT_2, reg);
 
@@ -974,7 +981,6 @@ static void pll_factors(struct _pll_div *pll_div, unsigned int target,
 static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		int source, unsigned int freq_in, unsigned int freq_out)
 {
-	u16 reg;
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct _pll_div pll_div;
 
@@ -982,13 +988,12 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		pll_factors(&pll_div, freq_out * 4, freq_in);
 
 		/* Turn on PLL */
-		reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
+		snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2,
-		reg |= WM8990_PLL_ENA;
+				    WM8990_PLL_ENA, WM8990_PLL_ENA);
-		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
 
 		/* sysclk comes from PLL */
-		reg = snd_soc_read(codec, WM8990_CLOCKING_2);
+		snd_soc_update_bits(codec, WM8990_CLOCKING_2,
-		snd_soc_write(codec, WM8990_CLOCKING_2, reg | WM8990_SYSCLK_SRC);
+				    WM8990_SYSCLK_SRC, WM8990_SYSCLK_SRC);
 
 		/* set up N , fractional mode and pre-divisor if necessary */
 		snd_soc_write(codec, WM8990_PLL1, pll_div.n | WM8990_SDM |
@@ -996,10 +1001,9 @@ static int wm8990_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
 		snd_soc_write(codec, WM8990_PLL2, (u8)(pll_div.k>>8));
 		snd_soc_write(codec, WM8990_PLL3, (u8)(pll_div.k & 0xFF));
 	} else {
-		/* Turn on PLL */
+		/* Turn off PLL */
-		reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
+		snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2,
-		reg &= ~WM8990_PLL_ENA;
+				    WM8990_PLL_ENA, 0);
-		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg);
 	}
 	return 0;
 }
@@ -1077,28 +1081,23 @@ static int wm8990_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		int div_id, int div)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 reg;
 
 	switch (div_id) {
 	case WM8990_MCLK_DIV:
-		reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
+		snd_soc_update_bits(codec, WM8990_CLOCKING_2,
-			~WM8990_MCLK_DIV_MASK;
+				    WM8990_MCLK_DIV_MASK, div);
-		snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
 		break;
 	case WM8990_DACCLK_DIV:
-		reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
+		snd_soc_update_bits(codec, WM8990_CLOCKING_2,
-			~WM8990_DAC_CLKDIV_MASK;
+				    WM8990_DAC_CLKDIV_MASK, div);
-		snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
 		break;
 	case WM8990_ADCCLK_DIV:
-		reg = snd_soc_read(codec, WM8990_CLOCKING_2) &
+		snd_soc_update_bits(codec, WM8990_CLOCKING_2,
-			~WM8990_ADC_CLKDIV_MASK;
+				    WM8990_ADC_CLKDIV_MASK, div);
-		snd_soc_write(codec, WM8990_CLOCKING_2, reg | div);
 		break;
 	case WM8990_BCLK_DIV:
-		reg = snd_soc_read(codec, WM8990_CLOCKING_1) &
+		snd_soc_update_bits(codec, WM8990_CLOCKING_1,
-			~WM8990_BCLK_DIV_MASK;
+				    WM8990_BCLK_DIV_MASK, div);
-		snd_soc_write(codec, WM8990_CLOCKING_1, reg | div);
 		break;
 	default:
 		return -EINVAL;
@@ -1156,7 +1155,7 @@ static int wm8990_mute(struct snd_soc_dai *dai, int mute)
 static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
-	u16 val;
+	int ret;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
@@ -1164,13 +1163,18 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 
 	case SND_SOC_BIAS_PREPARE:
 		/* VMID=2*50k */
-		val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) &
+		snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1,
-			~WM8990_VMID_MODE_MASK;
+				    WM8990_VMID_MODE_MASK, 0x2);
-		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x2);
 		break;
 
 	case SND_SOC_BIAS_STANDBY:
 		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			ret = snd_soc_cache_sync(codec);
+			if (ret < 0) {
+				dev_err(codec->dev, "Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+
 			/* Enable all output discharge bits */
 			snd_soc_write(codec, WM8990_ANTIPOP1, WM8990_DIS_LLINE |
 				WM8990_DIS_RLINE | WM8990_DIS_OUT3 |
@@ -1225,9 +1229,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 		}
 
 		/* VMID=2*250k */
-		val = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_1) &
+		snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_1,
-			~WM8990_VMID_MODE_MASK;
+				    WM8990_VMID_MODE_MASK, 0x4);
-		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, val | 0x4);
 		break;
 
 	case SND_SOC_BIAS_OFF:
@@ -1241,8 +1244,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
 			WM8990_BUFIOEN);
 
 		/* mute DAC */
-		val = snd_soc_read(codec, WM8990_DAC_CTRL);
+		snd_soc_update_bits(codec, WM8990_DAC_CTRL,
-		snd_soc_write(codec, WM8990_DAC_CTRL, val | WM8990_DAC_MUTE);
+				    WM8990_DAC_MUTE, WM8990_DAC_MUTE);
 
 		/* Enable any disabled outputs */
 		snd_soc_write(codec, WM8990_POWER_MANAGEMENT_1, 0x1f03);
@@ -1319,19 +1322,6 @@ static int wm8990_suspend(struct snd_soc_codec *codec, pm_message_t state)
 
 static int wm8990_resume(struct snd_soc_codec *codec)
 {
-	int i;
-	u8 data[2];
-	u16 *cache = codec->reg_cache;
-
-	/* Sync reg_cache with the hardware */
-	for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) {
-		if (i + 1 == WM8990_RESET)
-			continue;
-		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
-		data[1] = cache[i] & 0x00ff;
-		codec->hw_write(codec->control_data, data, 2);
-	}
-
 	wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	return 0;
 }
@@ -1343,7 +1333,6 @@ static int wm8990_resume(struct snd_soc_codec *codec)
 static int wm8990_probe(struct snd_soc_codec *codec)
 {
 	int ret;
-	u16 reg;
 
 	ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
 	if (ret < 0) {
@@ -1356,15 +1345,14 @@ static int wm8990_probe(struct snd_soc_codec *codec)
 	/* charge output caps */
 	wm8990_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	reg = snd_soc_read(codec, WM8990_AUDIO_INTERFACE_4);
+	snd_soc_update_bits(codec, WM8990_AUDIO_INTERFACE_4,
-	snd_soc_write(codec, WM8990_AUDIO_INTERFACE_4, reg | WM8990_ALRCGPIO1);
+			    WM8990_ALRCGPIO1, WM8990_ALRCGPIO1);
 
-	reg = snd_soc_read(codec, WM8990_GPIO1_GPIO2) &
+	snd_soc_update_bits(codec, WM8990_GPIO1_GPIO2,
-		~WM8990_GPIO1_SEL_MASK;
+			    WM8990_GPIO1_SEL_MASK, 1);
-	snd_soc_write(codec, WM8990_GPIO1_GPIO2, reg | 1);
 
-	reg = snd_soc_read(codec, WM8990_POWER_MANAGEMENT_2);
+	snd_soc_update_bits(codec, WM8990_POWER_MANAGEMENT_2,
-	snd_soc_write(codec, WM8990_POWER_MANAGEMENT_2, reg | WM8990_OPCLK_ENA);
+			    WM8990_OPCLK_ENA, WM8990_OPCLK_ENA);
 
 	snd_soc_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	snd_soc_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
@@ -1392,6 +1380,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8990 = {
 	.reg_cache_size = ARRAY_SIZE(wm8990_reg),
 	.reg_word_size = sizeof(u16),
 	.reg_cache_default = wm8990_reg,
+	.volatile_register = wm8990_volatile_register,
 };
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)

sound/soc/codecs/wm8991.c

@@ -3,7 +3,7 @@
  *
  * Copyright 2007-2010 Wolfson Microelectronics PLC.
  * Author: Graeme Gregory
- *         linux@wolfsonmicro.com
+ *         Graeme.Gregory@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
@@ -393,7 +393,7 @@ static int inmixer_event(struct snd_soc_dapm_widget *w,
 			 (1 << WM8991_AINRMUX_PWR_BIT)))
 		reg |= WM8991_AINR_ENA;
 	else
-		reg &= ~WM8991_AINL_ENA;
+		reg &= ~WM8991_AINR_ENA;
 
 	snd_soc_write(w->codec, WM8991_POWER_MANAGEMENT_2, reg);
 	return 0;
@@ -1264,7 +1264,6 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 {
 	struct wm8991_priv *wm8991;
 	int ret;
-	unsigned int reg;
 
 	wm8991 = snd_soc_codec_get_drvdata(codec);
 
@@ -1282,19 +1281,18 @@ static int wm8991_probe(struct snd_soc_codec *codec)
 
 	wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	reg = snd_soc_read(codec, WM8991_AUDIO_INTERFACE_4);
+	snd_soc_update_bits(codec, WM8991_AUDIO_INTERFACE_4,
-	snd_soc_write(codec, WM8991_AUDIO_INTERFACE_4, reg | WM8991_ALRCGPIO1);
+			    WM8991_ALRCGPIO1, WM8991_ALRCGPIO1);
 
-	reg = snd_soc_read(codec, WM8991_GPIO1_GPIO2) &
+	snd_soc_update_bits(codec, WM8991_GPIO1_GPIO2,
-	      ~WM8991_GPIO1_SEL_MASK;
+			    WM8991_GPIO1_SEL_MASK, 1);
-	snd_soc_write(codec, WM8991_GPIO1_GPIO2, reg | 1);
 
-	reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_1);
+	snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_1,
-	snd_soc_write(codec, WM8991_POWER_MANAGEMENT_1, reg | WM8991_VREF_ENA|
+			    WM8991_VREF_ENA | WM8991_VMID_MODE_MASK,
-		      WM8991_VMID_MODE_MASK);
+			    WM8991_VREF_ENA | WM8991_VMID_MODE_MASK);
 
-	reg = snd_soc_read(codec, WM8991_POWER_MANAGEMENT_2);
+	snd_soc_update_bits(codec, WM8991_POWER_MANAGEMENT_2,
-	snd_soc_write(codec, WM8991_POWER_MANAGEMENT_2, reg | WM8991_OPCLK_ENA);
+			    WM8991_OPCLK_ENA, WM8991_OPCLK_ENA);
 
 	snd_soc_write(codec, WM8991_DAC_CTRL, 0);
 	snd_soc_write(codec, WM8991_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));

sound/soc/codecs/wm8994.c

@@ -208,7 +208,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 static int configure_clock(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
-	int old, new;
+	int change, new;
 
 	/* Bring up the AIF clocks first */
 	configure_aif_clock(codec, 0);
@@ -229,14 +229,11 @@ static int configure_clock(struct snd_soc_codec *codec)
 	else
 		new = 0;
 
-	old = snd_soc_read(codec, WM8994_CLOCKING_1) & WM8994_SYSCLK_SRC;
+	change = snd_soc_update_bits(codec, WM8994_CLOCKING_1,
-
+				     WM8994_SYSCLK_SRC, new);
-	/* If there's no change then we're done. */
+	if (!change)
-	if (old == new)
 		return 0;
 
-	snd_soc_update_bits(codec, WM8994_CLOCKING_1, WM8994_SYSCLK_SRC, new);
-
 	snd_soc_dapm_sync(&codec->dapm);
 
 	return 0;
@@ -283,6 +280,7 @@ static const DECLARE_TLV_DB_SCALE(st_tlv, -3600, 300, 0);
 static const DECLARE_TLV_DB_SCALE(wm8994_3d_tlv, -1600, 183, 0);
 static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 static const DECLARE_TLV_DB_SCALE(ng_tlv, -10200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(mixin_boost_tlv, 0, 900, 0);
 
 #define WM8994_DRC_SWITCH(xname, reg, shift) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
@@ -703,6 +701,13 @@ SOC_SINGLE_TLV("AIF2DAC Noise Gate Threshold Volume",
 	       7, 1, ng_tlv),
 };
 
+static const struct snd_kcontrol_new wm1811_snd_controls[] = {
+SOC_SINGLE_TLV("MIXINL IN1LP Boost Volume", WM8994_INPUT_MIXER_1, 7, 1, 0,
+	       mixin_boost_tlv),
+SOC_SINGLE_TLV("MIXINL IN1RP Boost Volume", WM8994_INPUT_MIXER_1, 8, 1, 0,
+	       mixin_boost_tlv),
+};
+
 static int clk_sys_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event)
 {
@@ -1414,7 +1419,7 @@ SND_SOC_DAPM_MUX("AIF2DAC Mux", SND_SOC_NOPM, 0, 0, &aif2dac_mux),
 SND_SOC_DAPM_MUX("AIF2ADC Mux", SND_SOC_NOPM, 0, 0, &aif2adc_mux),
 
 SND_SOC_DAPM_AIF_IN("AIF3DACDAT", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0),
-SND_SOC_DAPM_AIF_IN("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0),
+SND_SOC_DAPM_AIF_OUT("AIF3ADCDAT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_SUPPLY("TOCLK", WM8994_CLOCKING_1, 4, 0, NULL, 0),
 
@@ -2053,6 +2058,15 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec,
 							    WM8958_CP_DISCH);
 				}
 				break;
+
+			case WM1811:
+				if (wm8994->revision < 2) {
+					snd_soc_write(codec, 0x102, 0x3);
+					snd_soc_write(codec, 0x5d, 0x7e);
+					snd_soc_write(codec, 0x5e, 0x0);
+					snd_soc_write(codec, 0x102, 0x0);
+				}
+				break;
 			}
 
 			/* Discharge LINEOUT1 & 2 */
@@ -2168,10 +2182,18 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 
 	/* The AIF2 format configuration needs to be mirrored to AIF3
 	 * on WM8958 if it's in use so just do it all the time. */
-	if (control->type == WM8958 && dai->id == 2)
+	switch (control->type) {
+	case WM1811:
+	case WM8958:
+		if (dai->id == 2)
 			snd_soc_update_bits(codec, WM8958_AIF3_CONTROL_1,
 					    WM8994_AIF1_LRCLK_INV |
 					    WM8958_AIF3_FMT_MASK, aif1);
+		break;
+
+	default:
+		break;
+	}
 
 	snd_soc_update_bits(codec, aif1_reg,
 			    WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV |
@@ -2213,7 +2235,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	struct wm8994 *control = codec->control_data;
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	int aif1_reg;
 	int aif2_reg;
@@ -2256,14 +2277,6 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream,
 			dev_dbg(codec->dev, "AIF2 using split LRCLK\n");
 		}
 		break;
-	case 3:
-		switch (control->type) {
-		case WM8958:
-			aif1_reg = WM8958_AIF3_CONTROL_1;
-			break;
-		default:
-			return 0;
-		}
 	default:
 		return -EINVAL;
 	}
@@ -2384,6 +2397,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream,
 	switch (dai->id) {
 	case 3:
 		switch (control->type) {
+		case WM1811:
 		case WM8958:
 			aif1_reg = WM8958_AIF3_CONTROL_1;
 			break;
@@ -2424,7 +2438,7 @@ static void wm8994_aif_shutdown(struct snd_pcm_substream *substream,
 		rate_reg = WM8994_AIF1_RATE;
 		break;
 	case 2:
-		rate_reg = WM8994_AIF1_RATE;
+		rate_reg = WM8994_AIF2_RATE;
 		break;
 	default:
 		break;
@@ -2614,6 +2628,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
 	case WM8994:
 		snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
 		break;
+	case WM1811:
 	case WM8958:
 		snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
 				    WM8958_MICD_ENA, 0);
@@ -2682,6 +2697,7 @@ static int wm8994_resume(struct snd_soc_codec *codec)
 			snd_soc_update_bits(codec, WM8994_MICBIAS,
 					    WM8994_MICD_ENA, WM8994_MICD_ENA);
 		break;
+	case WM1811:
 	case WM8958:
 		if (wm8994->jack_cb)
 			snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
@@ -2980,8 +2996,13 @@ int wm8958_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack,
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 	struct wm8994 *control = codec->control_data;
 
-	if (control->type != WM8958)
+	switch (control->type) {
+	case WM1811:
+	case WM8958:
+		break;
+	default:
 		return -EINVAL;
+	}
 
 	if (jack) {
 		if (!cb) {
@@ -3135,6 +3156,24 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		wm8994->hubs.dcs_readback_mode = 1;
 		break;
 
+	case WM1811:
+		wm8994->hubs.dcs_readback_mode = 2;
+		wm8994->hubs.no_series_update = 1;
+
+		switch (wm8994->revision) {
+		case 0:
+		case 1:
+			wm8994->hubs.dcs_codes_l = -9;
+			wm8994->hubs.dcs_codes_r = -5;
+			break;
+		default:
+			break;
+		}
+
+		snd_soc_update_bits(codec, WM8994_ANALOGUE_HP_1,
+				    WM1811_HPOUT1_ATTN, WM1811_HPOUT1_ATTN);
+		break;
+
 	default:
 		break;
 	}
@@ -3195,6 +3234,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 		break;
 
 	case WM8958:
+	case WM1811:
 		if (wm8994->micdet_irq) {
 			ret = request_threaded_irq(wm8994->micdet_irq, NULL,
 						   wm8958_mic_irq,
@@ -3357,6 +3397,19 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 						  ARRAY_SIZE(wm8994_dac_widgets));
 		}
 		break;
+
+	case WM1811:
+		snd_soc_add_controls(codec, wm8958_snd_controls,
+				     ARRAY_SIZE(wm8958_snd_controls));
+		snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets,
+					  ARRAY_SIZE(wm8958_dapm_widgets));
+		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;
 	}
 		
 
@@ -3393,6 +3446,12 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec)
 
 		wm8958_dsp2_init(codec);
 		break;
+	case WM1811:
+		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;
 	}
 
 	return 0;
@@ -3448,6 +3507,7 @@ static int  wm8994_codec_remove(struct snd_soc_codec *codec)
 				wm8994);
 		break;
 
+	case WM1811:
 	case WM8958:
 		if (wm8994->micdet_irq)
 			free_irq(wm8994->micdet_irq, wm8994);

sound/soc/codecs/wm8995.c

@@ -485,7 +485,7 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif)
 static int configure_clock(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
-	int old, new;
+	int change, new;
 
 	wm8995 = snd_soc_codec_get_drvdata(codec);
 
@@ -509,14 +509,10 @@ static int configure_clock(struct snd_soc_codec *codec)
 	else
 		new = 0;
 
-	old = snd_soc_read(codec, WM8995_CLOCKING_1) & WM8995_SYSCLK_SRC;
+	change = snd_soc_update_bits(codec, WM8995_CLOCKING_1,
-
-	/* If there's no change then we're done. */
-	if (old == new)
-		return 0;
-
-	snd_soc_update_bits(codec, WM8995_CLOCKING_1,
 				     WM8995_SYSCLK_SRC_MASK, new);
+	if (!change)
+		return 0;
 
 	snd_soc_dapm_sync(&codec->dapm);
 
@@ -1573,9 +1569,16 @@ static int wm8995_resume(struct snd_soc_codec *codec)
 static int wm8995_remove(struct snd_soc_codec *codec)
 {
 	struct wm8995_priv *wm8995;
+	int i;
 
 	wm8995 = snd_soc_codec_get_drvdata(codec);
 	wm8995_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	for (i = 0; i < ARRAY_SIZE(wm8995->supplies); ++i)
+		regulator_unregister_notifier(wm8995->supplies[i].consumer,
+					      &wm8995->disable_nb[i]);
+
+	regulator_bulk_free(ARRAY_SIZE(wm8995->supplies), wm8995->supplies);
 	return 0;
 }
 

sound/soc/codecs/wm8996.c

@@ -641,6 +641,14 @@ SOC_DOUBLE_R("Speaker ZC Switch", WM8996_LEFT_PDM_SPEAKER,
 
 SOC_SINGLE("DSP1 EQ Switch", WM8996_DSP1_RX_EQ_GAINS_1, 0, 1, 0),
 SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0),
+
+SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0),
+SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0),
+SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0),
+
+SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0),
+SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0),
+SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0),
 };
 
 static const struct snd_kcontrol_new wm8996_eq_controls[] = {
@@ -1105,9 +1113,9 @@ SND_SOC_DAPM_AIF_IN("AIF2RX1", "AIF2 Playback", 0,
 SND_SOC_DAPM_AIF_IN("AIF2RX0", "AIF2 Playback", 1,
 		    WM8996_POWER_MANAGEMENT_4, 8, 0),
 
-SND_SOC_DAPM_AIF_IN("AIF2TX1", "AIF2 Capture", 0,
+SND_SOC_DAPM_AIF_OUT("AIF2TX1", "AIF2 Capture", 0,
 		    WM8996_POWER_MANAGEMENT_6, 9, 0),
-SND_SOC_DAPM_AIF_IN("AIF2TX0", "AIF2 Capture", 1,
+SND_SOC_DAPM_AIF_OUT("AIF2TX0", "AIF2 Capture", 1,
 		    WM8996_POWER_MANAGEMENT_6, 8, 0),
 
 SND_SOC_DAPM_AIF_IN("AIF1RX5", "AIF1 Playback", 5,
@@ -1912,7 +1920,7 @@ static int wm8996_hw_params(struct snd_pcm_substream *substream,
 	snd_soc_update_bits(codec, lrclk_reg, WM8996_AIF1RX_RATE_MASK,
 			    lrclk);
 	snd_soc_update_bits(codec, WM8996_AIF_CLOCKING_2,
-			    WM8996_DSP1_DIV_SHIFT << dsp_shift, dsp);
+			    WM8996_DSP1_DIV_MASK << dsp_shift, dsp);
 
 	return 0;
 }
@@ -2698,7 +2706,6 @@ static int wm8996_probe(struct snd_soc_codec *codec)
 	init_completion(&wm8996->fll_lock);
 
 	dapm->idle_bias_off = true;
-	dapm->bias_level = SND_SOC_BIAS_OFF;
 
 	ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_I2C);
 	if (ret != 0) {