Merge branch 'topic/asoc' into for-linus

arch/arm/plat-mxc/Makefile

@@ -13,3 +13,7 @@ obj-$(CONFIG_USB_EHCI_MXC) += ehci.o
 obj-$(CONFIG_MXC_ULPI) += ulpi.o
 obj-$(CONFIG_ARCH_MXC_AUDMUX_V1) += audmux-v1.o
 obj-$(CONFIG_ARCH_MXC_AUDMUX_V2) += audmux-v2.o
+ifdef CONFIG_SND_IMX_SOC
+obj-y += ssi-fiq.o
+obj-y += ssi-fiq-ksym.o
+endif

arch/arm/plat-mxc/ssi-fiq-ksym.c

+/*
+ * Exported ksyms for the SSI FIQ handler
+ *
+ * Copyright (C) 2009, Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+
+#include <mach/ssi.h>
+
+EXPORT_SYMBOL(imx_ssi_fiq_tx_buffer);
+EXPORT_SYMBOL(imx_ssi_fiq_rx_buffer);
+EXPORT_SYMBOL(imx_ssi_fiq_start);
+EXPORT_SYMBOL(imx_ssi_fiq_end);
+EXPORT_SYMBOL(imx_ssi_fiq_base);
+

arch/arm/plat-mxc/ssi-fiq.S

+/*
+ *  Copyright (C) 2009 Sascha Hauer <s.hauer@pengutronix.de>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/linkage.h>
+#include <asm/assembler.h>
+
+/*
+ * r8  = bit 0-15: tx offset, bit 16-31: tx buffer size
+ * r9  = bit 0-15: rx offset, bit 16-31: rx buffer size
+ */
+
+#define SSI_STX0	0x00
+#define SSI_SRX0	0x08
+#define SSI_SISR	0x14
+#define SSI_SIER	0x18
+#define SSI_SACNT	0x38
+
+#define SSI_SACNT_AC97EN	(1 << 0)
+
+#define SSI_SIER_TFE0_EN	(1 << 0)
+#define SSI_SISR_TFE0		(1 << 0)
+#define SSI_SISR_RFF0		(1 << 2)
+#define SSI_SIER_RFF0_EN	(1 << 2)
+
+		.text
+		.global	imx_ssi_fiq_start
+		.global	imx_ssi_fiq_end
+		.global imx_ssi_fiq_base
+		.global imx_ssi_fiq_rx_buffer
+		.global imx_ssi_fiq_tx_buffer
+
+imx_ssi_fiq_start:
+		ldr r12, imx_ssi_fiq_base
+
+		/* TX */
+		ldr r11, imx_ssi_fiq_tx_buffer
+
+		/* shall we send? */
+		ldr r13, [r12, #SSI_SIER]
+		tst r13, #SSI_SIER_TFE0_EN
+		beq 1f
+
+		/* TX FIFO empty? */
+		ldr r13, [r12, #SSI_SISR]
+		tst r13, #SSI_SISR_TFE0
+		beq 1f
+
+		mov r10, #0x10000
+		sub r10, #1
+		and r10, r10, r8	/* r10: current buffer offset */
+
+		add r11, r11, r10
+
+		ldrh r13, [r11]
+		strh r13, [r12, #SSI_STX0]
+
+		ldrh r13, [r11, #2]
+		strh r13, [r12, #SSI_STX0]
+
+		ldrh r13, [r11, #4]
+		strh r13, [r12, #SSI_STX0]
+
+		ldrh r13, [r11, #6]
+		strh r13, [r12, #SSI_STX0]
+
+		add r10, #8
+		lsr r13, r8, #16	/* r13: buffer size */
+		cmp r10, r13
+		lslgt r8, r13, #16
+		addle r8, #8
+1:
+		/* RX */
+
+		/* shall we receive? */
+		ldr r13, [r12, #SSI_SIER]
+		tst r13, #SSI_SIER_RFF0_EN
+		beq 1f
+
+		/* RX FIFO full? */
+		ldr r13, [r12, #SSI_SISR]
+		tst r13, #SSI_SISR_RFF0
+		beq 1f
+
+		ldr r11, imx_ssi_fiq_rx_buffer
+
+		mov r10, #0x10000
+		sub r10, #1
+		and r10, r10, r9	/* r10: current buffer offset */
+
+		add r11, r11, r10
+
+		ldr r13, [r12, #SSI_SACNT]
+		tst r13, #SSI_SACNT_AC97EN
+
+		ldr r13, [r12, #SSI_SRX0]
+		strh r13, [r11]
+
+		ldr r13, [r12, #SSI_SRX0]
+		strh r13, [r11, #2]
+
+		/* dummy read to skip slot 12 */
+		ldrne r13, [r12, #SSI_SRX0]
+
+		ldr r13, [r12, #SSI_SRX0]
+		strh r13, [r11, #4]
+
+		ldr r13, [r12, #SSI_SRX0]
+		strh r13, [r11, #6]
+
+		/* dummy read to skip slot 12 */
+		ldrne r13, [r12, #SSI_SRX0]
+
+		add r10, #8
+		lsr r13, r9, #16	/* r13: buffer size */
+		cmp r10, r13
+		lslgt r9, r13, #16
+		addle r9, #8
+
+1:
+		@ return from FIQ
+		subs	pc, lr, #4
+imx_ssi_fiq_base:
+		.word 0x0
+imx_ssi_fiq_rx_buffer:
+		.word 0x0
+imx_ssi_fiq_tx_buffer:
+		.word 0x0
+imx_ssi_fiq_end:
+

arch/sh/include/asm/siu.h

+/*
+ * platform header for the SIU ASoC driver
+ *
+ * Copyright (C) 2009-2010 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+ *
+ * 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 ASM_SIU_H
+#define ASM_SIU_H
+
+#include <asm/dma-sh.h>
+
+struct device;
+
+struct siu_platform {
+	struct device *dma_dev;
+	enum sh_dmae_slave_chan_id dma_slave_tx_a;
+	enum sh_dmae_slave_chan_id dma_slave_rx_a;
+	enum sh_dmae_slave_chan_id dma_slave_tx_b;
+	enum sh_dmae_slave_chan_id dma_slave_rx_b;
+};
+
+#endif /* ASM_SIU_H */

drivers/mfd/twl-core.c

@@ -115,7 +115,8 @@
 #define twl_has_watchdog()        false
 #endif
 
-#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE)
+#if defined(CONFIG_TWL4030_CODEC) || defined(CONFIG_TWL4030_CODEC_MODULE) ||\
+	defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE)
 #define twl_has_codec()	true
 #else
 #define twl_has_codec()	false
@@ -711,8 +712,19 @@ add_children(struct twl4030_platform_data *pdata, unsigned long features)
 			return PTR_ERR(child);
 	}
 
-	if (twl_has_codec() && pdata->codec) {
-		child = add_child(1, "twl4030_codec",
+	if (twl_has_codec() && pdata->codec && twl_class_is_4030()) {
+		sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid;
+		child = add_child(sub_chip_id, "twl4030_codec",
+				pdata->codec, sizeof(*pdata->codec),
+				false, 0, 0);
+		if (IS_ERR(child))
+			return PTR_ERR(child);
+	}
+
+	/* Phoenix*/
+	if (twl_has_codec() && pdata->codec && twl_class_is_6030()) {
+		sub_chip_id = twl_map[TWL_MODULE_AUDIO_VOICE].sid;
+		child = add_child(sub_chip_id, "twl6030_codec",
 				pdata->codec, sizeof(*pdata->codec),
 				false, 0, 0);
 		if (IS_ERR(child))

include/linux/i2c/twl.h

@@ -547,6 +547,10 @@ struct twl4030_codec_data {
 	unsigned int	audio_mclk;
 	struct twl4030_codec_audio_data		*audio;
 	struct twl4030_codec_vibra_data		*vibra;
+
+	/* twl6030 */
+	int audpwron_gpio;      /* audio power-on gpio */
+	int naudint_irq;        /* audio interrupt */
 };
 
 struct twl4030_platform_data {

include/sound/soc-dai.h

@@ -16,6 +16,8 @@
 
 #include <linux/list.h>
 
+#include <sound/soc.h>
+
 struct snd_pcm_substream;
 
 /*

include/sound/soc-dapm.h

@@ -95,6 +95,21 @@
 	.shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
 	.num_kcontrols = 1}
 
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\
+	 wcontrols) \
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_ARRAY(wname, wreg, wshift, winvert, \
+	 wcontrols)\
+{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols)}
+#define SOC_MIXER_NAMED_CTL_ARRAY(wname, wreg, wshift, winvert, \
+	 wcontrols)\
+{       .id = snd_soc_dapm_mixer_named_ctl, .name = wname, .reg = wreg, \
+	.shift = wshift, .invert = winvert, .kcontrols = wcontrols, \
+	.num_kcontrols = ARRAY_SIZE(wcontrols)}
+
 /* path domain with event - event handler must return 0 for success */
 #define SND_SOC_DAPM_PGA_E(wname, wreg, wshift, winvert, wcontrols, \
 	wncontrols, wevent, wflags) \
@@ -126,6 +141,23 @@
 	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = 1, \
 	.event = wevent, .event_flags = wflags}
 
+/* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */
+#define SOC_PGA_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+	wevent, wflags) \
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = wreg, .shift = wshift, \
+	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+	.event = wevent, .event_flags = wflags}
+#define SOC_MIXER_E_ARRAY(wname, wreg, wshift, winvert, wcontrols, \
+	wevent, wflags) \
+{	.id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+	.invert = winvert, .kcontrols = wcontrols, .num_kcontrols = ARRAY_SIZE(wcontrols), \
+	.event = wevent, .event_flags = wflags}
+#define SOC_MIXER_NAMED_CTL_E_ARRAY(wname, wreg, wshift, winvert, \
+	wcontrols, wevent, wflags) \
+{       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
+	.invert = winvert, .kcontrols = wcontrols, \
+	.num_kcontrols = ARRAY_SIZE(wcontrols), .event = wevent, .event_flags = wflags}
+
 /* events that are pre and post DAPM */
 #define SND_SOC_DAPM_PRE(wname, wevent) \
 {	.id = snd_soc_dapm_pre, .name = wname, .kcontrols = NULL, \

include/sound/soc.h

@@ -168,6 +168,23 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&xenum }
 
+/*
+ * Simplified versions of above macros, declaring a struct and calculating
+ * ARRAY_SIZE internally
+ */
+#define SOC_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xtexts) \
+	struct soc_enum name = SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, \
+						ARRAY_SIZE(xtexts), xtexts)
+#define SOC_ENUM_SINGLE_DECL(name, xreg, xshift, xtexts) \
+	SOC_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xtexts)
+#define SOC_ENUM_SINGLE_EXT_DECL(name, xtexts) \
+	struct soc_enum name = SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(xtexts), xtexts)
+#define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \
+	struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \
+							ARRAY_SIZE(xtexts), xtexts, xvalues)
+#define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \
+	SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues)
+
 /*
  * Bias levels
  *
@@ -253,6 +270,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 /* codec register bit access */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
 				unsigned int mask, unsigned int value);
+int snd_soc_update_bits_locked(struct snd_soc_codec *codec,
+			       unsigned short reg, unsigned int mask,
+			       unsigned int value);
 int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
 				unsigned int mask, unsigned int value);
 
@@ -402,6 +422,10 @@ struct snd_soc_codec {
 	short reg_cache_size;
 	short reg_cache_step;
 
+	unsigned int idle_bias_off:1; /* Use BIAS_OFF instead of STANDBY */
+	unsigned int cache_only:1;  /* Suppress writes to hardware */
+	unsigned int cache_sync:1; /* Cache needs to be synced to hardware */
+
 	/* dapm */
 	u32 pop_time;
 	struct list_head dapm_widgets;
@@ -497,6 +521,8 @@ struct snd_soc_card {
 	int (*set_bias_level)(struct snd_soc_card *,
 			      enum snd_soc_bias_level level);
 
+	long pmdown_time;
+
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
 	int num_links;

include/sound/tlv320dac33-plat.h

@@ -15,6 +15,7 @@
 
 struct tlv320dac33_platform_data {
 	int power_gpio;
+	u8 burst_bclkdiv;
 };
 
 #endif /* __TLV320DAC33_PLAT_H */

include/sound/tpa6130a2-plat.h

@@ -23,7 +23,13 @@
 #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/wm2000.h

+/*
+ * linux/sound/wm2000.h -- Platform data for WM2000
+ *
+ * Copyright 2010 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_WM2000_H
+#define __LINUX_SND_WM2000_H
+
+struct wm2000_platform_data {
+	/** Filename for system-specific image to download to device. */
+	const char *download_file;
+
+	/** Divide MCLK by 2 for system clock? */
+	unsigned int mclkdiv2:1;
+
+	/** Disable speech clarity enhancement, for use when an
+	 * external algorithm is used. */
+	unsigned int speech_enh_disable:1;
+};
+
+#endif

include/sound/wm8904.h

+/*
+ * Platform data for WM8904
+ *
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef __MFD_WM8994_PDATA_H__
+#define __MFD_WM8994_PDATA_H__
+
+#define WM8904_DRC_REGS 4
+#define WM8904_EQ_REGS  25
+
+/**
+ * DRC configurations are specified with a label and a set of register
+ * values to write (the enable bits will be ignored).  At runtime an
+ * enumerated control will be presented for each DRC block allowing
+ * the user to choose the configration to use.
+ *
+ * Configurations may be generated by hand or by using the DRC control
+ * panel provided by the WISCE - see  http://www.wolfsonmicro.com/wisce/
+ * for details.
+ */
+struct wm8904_drc_cfg {
+	const char *name;
+	u16 regs[WM8904_DRC_REGS];
+};
+
+/**
+ * ReTune Mobile configurations are specified with a label, sample
+ * rate and set of values to write (the enable bits will be ignored).
+ *
+ * Configurations are expected to be generated using the ReTune Mobile
+ * control panel in WISCE - see http://www.wolfsonmicro.com/wisce/
+ */
+struct wm8904_retune_mobile_cfg {
+	const char *name;
+	unsigned int rate;
+	u16 regs[WM8904_EQ_REGS];
+};
+
+struct wm8904_pdata {
+	int num_drc_cfgs;
+	struct wm8904_drc_cfg *drc_cfgs;
+
+	int num_retune_mobile_cfgs;
+	struct wm8904_retune_mobile_cfg *retune_mobile_cfgs;
+};
+
+#endif

sound/soc/s3c24xx/s3c24xx-ac97.h → include/sound/wm8955.h

 /*
- * s3c24xx-ac97.c  --  ALSA Soc Audio Layer
+ * Platform data for WM8955
  *
- * (c) 2007 Wolfson Microelectronics PLC.
- * Author: Graeme Gregory
- *         graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Copyright 2009 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
  *  Free Software Foundation;  either version 2 of the  License, or (at your
  *  option) any later version.
  *
- *  Revision history
- *    10th Nov 2006   Initial version.
  */
 
-#ifndef S3C24XXAC97_H_
-#define S3C24XXAC97_H_
+#ifndef __WM8955_PDATA_H__
+#define __WM8955_PDATA_H__
 
-#define AC_CMD_ADDR(x) (x << 16)
-#define AC_CMD_DATA(x) (x & 0xffff)
+struct wm8955_pdata {
+	/* Configure LOUT2/ROUT2 to drive a speaker */
+	unsigned int out2_speaker:1;
 
-extern struct snd_soc_dai s3c2443_ac97_dai[];
+	/* Configure MONOIN+/- in differential mode */
+	unsigned int monoin_diff:1;
+};
 
-#endif /*S3C24XXAC97_H_*/
+#endif

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

@@ -349,9 +349,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 			sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \
 				size, &sport_handle->tx_dma_phy, GFP_KERNEL);
 			if (!sport_handle->tx_dma_buf) {
-				pr_err("Failed to allocate memory for tx dma \
-					buf - Please increase uncached DMA \
-					memory region\n");
+				pr_err("Failed to allocate memory for tx dma buf - Please increase uncached DMA memory region\n");
 				return -ENOMEM;
 			} else
 				memset(sport_handle->tx_dma_buf, 0, size);
@@ -362,9 +360,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 			sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \
 				size, &sport_handle->rx_dma_phy, GFP_KERNEL);
 			if (!sport_handle->rx_dma_buf) {
-				pr_err("Failed to allocate memory for rx dma \
-					buf - Please increase uncached DMA \
-					memory region\n");
+				pr_err("Failed to allocate memory for rx dma buf - Please increase uncached DMA memory region\n");
 				return -ENOMEM;
 			} else
 				memset(sport_handle->rx_dma_buf, 0, size);

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

@@ -207,8 +207,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	buf->area = dma_alloc_coherent(pcm->card->dev, size,
 			&buf->addr, GFP_KERNEL);
 	if (!buf->area) {
-		pr_err("Failed to allocate dma memory \
-			Please increase uncached DMA memory region\n");
+		pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n");
 		return -ENOMEM;
 	}
 	buf->bytes = size;

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

@@ -244,8 +244,7 @@ static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	buf->area = dma_alloc_coherent(pcm->card->dev, size * 4,
 		&buf->addr, GFP_KERNEL);
 	if (!buf->area) {
-		pr_err("Failed to allocate dma memory \
-			Please increase uncached DMA memory region\n");
+		pr_err("Failed to allocate dma memory - Please increase uncached DMA memory region\n");
 		return -ENOMEM;
 	}
 	buf->bytes = size;

sound/soc/codecs/Kconfig

@@ -23,6 +23,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4671 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_MAX9877 if I2C
+	select SND_SOC_DA7210 if I2C
 	select SND_SOC_PCM3008
 	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
@@ -35,6 +36,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_TWL4030 if TWL4030_CORE
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
+	select SND_SOC_WM2000 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
@@ -49,14 +51,18 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8904 if I2C
 	select SND_SOC_WM8940 if I2C
+	select SND_SOC_WM8955 if I2C
 	select SND_SOC_WM8960 if I2C
 	select SND_SOC_WM8961 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8974 if I2C
+	select SND_SOC_WM8978 if I2C
 	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
 	select SND_SOC_WM8993 if I2C
+	select SND_SOC_WM8994 if MFD_WM8994
 	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
@@ -112,6 +118,9 @@ config SND_SOC_AK4671
 config SND_SOC_CS4270
 	tristate
 
+config SND_SOC_DA7210
+        tristate
+
 # Cirrus Logic CS4270 Codec VD = 3.3V Errata
 # Select if you are affected by the errata where the part will not function
 # if MCLK divide-by-1.5 is selected and VD is set to 3.3V.  The driver will
@@ -203,9 +212,15 @@ config SND_SOC_WM8900
 config SND_SOC_WM8903
 	tristate
 
+config SND_SOC_WM8904
+	tristate
+
 config SND_SOC_WM8940
         tristate
 
+config SND_SOC_WM8955
+	tristate
+
 config SND_SOC_WM8960
 	tristate
 
@@ -218,6 +233,9 @@ config SND_SOC_WM8971
 config SND_SOC_WM8974
 	tristate
 
+config SND_SOC_WM8978
+	tristate
+
 config SND_SOC_WM8988
 	tristate
 
@@ -227,6 +245,9 @@ config SND_SOC_WM8990
 config SND_SOC_WM8993
 	tristate
 
+config SND_SOC_WM8994
+	tristate
+
 config SND_SOC_WM9081
 	tristate
 
@@ -245,3 +266,6 @@ config SND_SOC_MAX9877
 
 config SND_SOC_TPA6130A2
 	tristate
+
+config SND_SOC_WM2000
+	tristate

sound/soc/codecs/Makefile

@@ -10,6 +10,7 @@ snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-cx20442-objs := cx20442.o
+snd-soc-da7210-objs := da7210.o
 snd-soc-l3-objs := l3.o
 snd-soc-pcm3008-objs := pcm3008.o
 snd-soc-spdif-objs := spdif_transciever.o
@@ -36,14 +37,18 @@ snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8776-objs := wm8776.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8904-objs := wm8904.o
 snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8955-objs := wm8955.o
 snd-soc-wm8960-objs := wm8960.o
 snd-soc-wm8961-objs := wm8961.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8974-objs := wm8974.o
+snd-soc-wm8978-objs := wm8978.o
 snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
 snd-soc-wm8993-objs := wm8993.o
+snd-soc-wm8994-objs := wm8994.o
 snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
@@ -53,6 +58,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o
 # Amp
 snd-soc-max9877-objs := max9877.o
 snd-soc-tpa6130a2-objs := tpa6130a2.o
+snd-soc-wm2000-objs := wm2000.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
@@ -66,6 +72,7 @@ obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_CX20442)	+= snd-soc-cx20442.o
+obj-$(CONFIG_SND_SOC_DA7210)	+= snd-soc-da7210.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
 obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
@@ -92,14 +99,18 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8776)	+= snd-soc-wm8776.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
-obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
-obj-$(CONFIG_SND_SOC_WM8974)	+= snd-soc-wm8974.o
+obj-$(CONFIG_SND_SOC_WM8904)	+= snd-soc-wm8904.o
 obj-$(CONFIG_SND_SOC_WM8940)	+= snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8955)	+= snd-soc-wm8955.o
 obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
 obj-$(CONFIG_SND_SOC_WM8961)	+= snd-soc-wm8961.o
+obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8974)	+= snd-soc-wm8974.o
+obj-$(CONFIG_SND_SOC_WM8978)	+= snd-soc-wm8978.o
 obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
 obj-$(CONFIG_SND_SOC_WM8993)	+= snd-soc-wm8993.o
+obj-$(CONFIG_SND_SOC_WM8994)	+= snd-soc-wm8994.o
 obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
@@ -109,3 +120,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS)	+= snd-soc-wm-hubs.o
 # Amp
 obj-$(CONFIG_SND_SOC_MAX9877)	+= snd-soc-max9877.o
 obj-$(CONFIG_SND_SOC_TPA6130A2)	+= snd-soc-tpa6130a2.o
+obj-$(CONFIG_SND_SOC_WM2000)	+= snd-soc-wm2000.o

sound/soc/codecs/ad1836.c

@@ -171,57 +171,35 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-
-/*
- * interface to read/write ad1836 register
- */
-#define AD1836_SPI_REG_SHFT 12
-#define AD1836_SPI_READ     (1 << 11)
-#define AD1836_SPI_VAL_MSK  0x3FF
-
-/*
- * write to the ad1836 register space
- */
-
-static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg,
-		unsigned int value)
+#ifdef CONFIG_PM
+static int ad1836_soc_suspend(struct platform_device *pdev,
+		pm_message_t state)
 {
-	u16 *reg_cache = codec->reg_cache;
-	int ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
 
-	if (value != reg_cache[reg]) {
-		unsigned short buf;
-		struct spi_transfer t = {
-			.tx_buf = &buf,
-			.len = 2,
-		};
-		struct spi_message m;
-
-		buf = (reg << AD1836_SPI_REG_SHFT) |
-			(value & AD1836_SPI_VAL_MSK);
-		spi_message_init(&m);
-		spi_message_add_tail(&t, &m);
-		ret = spi_sync(codec->control_data, &m);
-		if (ret == 0)
-			reg_cache[reg] = value;
-	}
+	/* reset clock control mode */
+	u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
+	adc_ctrl2 &= ~AD1836_ADC_SERFMT_MASK;
 
-	return ret;
+	return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
 }
 
-/*
- * read from the ad1836 register space cache
- */
-static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec,
-					  unsigned int reg)
+static int ad1836_soc_resume(struct platform_device *pdev)
 {
-	u16 *reg_cache = codec->reg_cache;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
 
-	if (reg >= codec->reg_cache_size)
-		return -EINVAL;
+	/* restore clock control mode */
+	u16 adc_ctrl2 = snd_soc_read(codec, AD1836_ADC_CTRL2);
+	adc_ctrl2 |= AD1836_ADC_AUX;
 
-	return reg_cache[reg];
+	return snd_soc_write(codec, AD1836_ADC_CTRL2, adc_ctrl2);
 }
+#else
+#define ad1836_soc_suspend NULL
+#define ad1836_soc_resume  NULL
+#endif
 
 static int __devinit ad1836_spi_probe(struct spi_device *spi)
 {
@@ -306,32 +284,38 @@ static int ad1836_register(struct ad1836_priv *ad1836)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ad1836_dai;
 	codec->num_dai = 1;
-	codec->write = ad1836_write_reg;
-	codec->read = ad1836_read_reg_cache;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	ad1836_dai.dev = codec->dev;
 	ad1836_codec = codec;
 
+	ret = snd_soc_codec_set_cache_io(codec, 4, 12, SND_SOC_SPI);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n",
+				ret);
+		kfree(ad1836);
+		return ret;
+	}
+
 	/* default setting for ad1836 */
 	/* de-emphasis: 48kHz, power-on dac */
-	codec->write(codec, AD1836_DAC_CTRL1, 0x300);
+	snd_soc_write(codec, AD1836_DAC_CTRL1, 0x300);
 	/* unmute dac channels */
-	codec->write(codec, AD1836_DAC_CTRL2, 0x0);
+	snd_soc_write(codec, AD1836_DAC_CTRL2, 0x0);
 	/* high-pass filter enable, power-on adc */
-	codec->write(codec, AD1836_ADC_CTRL1, 0x100);
+	snd_soc_write(codec, AD1836_ADC_CTRL1, 0x100);
 	/* unmute adc channles, adc aux mode */
-	codec->write(codec, AD1836_ADC_CTRL2, 0x180);
+	snd_soc_write(codec, AD1836_ADC_CTRL2, 0x180);
 	/* left/right diff:PGA/MUX */
-	codec->write(codec, AD1836_ADC_CTRL3, 0x3A);
+	snd_soc_write(codec, AD1836_ADC_CTRL3, 0x3A);
 	/* volume */
-	codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF);
-	codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF);
-	codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF);
-	codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF);
-	codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF);
-	codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_L1_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_R1_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_L2_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_R2_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_L3_VOL, 0x3FF);
+	snd_soc_write(codec, AD1836_DAC_R3_VOL, 0x3FF);
 
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
@@ -404,6 +388,8 @@ static int ad1836_remove(struct platform_device *pdev)
 struct snd_soc_codec_device soc_codec_dev_ad1836 = {
 	.probe = 	ad1836_probe,
 	.remove = 	ad1836_remove,
+	.suspend =      ad1836_soc_suspend,
+	.resume =       ad1836_soc_resume,
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836);
 

sound/soc/codecs/ad1836.h

@@ -54,6 +54,7 @@
 #define AD1836_ADC_SERFMT_MASK	       (7 << 6)
 #define AD1836_ADC_SERFMT_PCK256       (0x4 << 6)
 #define AD1836_ADC_SERFMT_PCK128       (0x5 << 6)
+#define AD1836_ADC_AUX                 (0x6 << 6)
 
 #define AD1836_ADC_CTRL3               14
 

sound/soc/codecs/ad1938.c

@@ -46,6 +46,11 @@ struct ad1938_priv {
 	u8 reg_cache[AD1938_NUM_REGS];
 };
 
+/* ad1938 register cache & default register settings */
+static const u8 ad1938_reg[AD1938_NUM_REGS] = {
+	0, 0, 0, 0, 0, 0, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0, 0, 0,
+};
+
 static struct snd_soc_codec *ad1938_codec;
 struct snd_soc_codec_device soc_codec_dev_ad1938;
 static int ad1938_register(struct ad1938_priv *ad1938);
@@ -97,6 +102,7 @@ static const struct snd_kcontrol_new ad1938_snd_controls[] = {
 static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = {
 	SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1),
 	SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_SUPPLY("PLL_PWR", AD1938_PLL_CLK_CTRL0, 0, 1, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0),
 	SND_SOC_DAPM_OUTPUT("DAC1OUT"),
 	SND_SOC_DAPM_OUTPUT("DAC2OUT"),
@@ -107,6 +113,8 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_route audio_paths[] = {
+	{ "DAC", NULL, "PLL_PWR" },
+	{ "ADC", NULL, "PLL_PWR" },
 	{ "DAC", NULL, "ADC_PWR" },
 	{ "ADC", NULL, "ADC_PWR" },
 	{ "DAC1OUT", "DAC1 Switch", "DAC" },
@@ -126,30 +134,20 @@ static int ad1938_mute(struct snd_soc_dai *dai, int mute)
 	struct snd_soc_codec *codec = dai->codec;
 	int reg;
 
-	reg = codec->read(codec, AD1938_DAC_CTRL2);
+	reg = snd_soc_read(codec, AD1938_DAC_CTRL2);
 	reg = (mute > 0) ? reg | AD1938_DAC_MASTER_MUTE : reg &
 		(~AD1938_DAC_MASTER_MUTE);
-	codec->write(codec, AD1938_DAC_CTRL2, reg);
-
-	return 0;
-}
-
-static inline int ad1938_pll_powerctrl(struct snd_soc_codec *codec, int cmd)
-{
-	int reg = codec->read(codec, AD1938_PLL_CLK_CTRL0);
-	reg = (cmd > 0) ? reg & (~AD1938_PLL_POWERDOWN) : reg |
-		AD1938_PLL_POWERDOWN;
-	codec->write(codec, AD1938_PLL_CLK_CTRL0, reg);
+	snd_soc_write(codec, AD1938_DAC_CTRL2, reg);
 
 	return 0;
 }
 
 static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
-			       unsigned int mask, int slots, int width)
+			       unsigned int rx_mask, int slots, int width)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	int dac_reg = codec->read(codec, AD1938_DAC_CTRL1);
-	int adc_reg = codec->read(codec, AD1938_ADC_CTRL2);
+	int dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1);
+	int adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2);
 
 	dac_reg &= ~AD1938_DAC_CHAN_MASK;
 	adc_reg &= ~AD1938_ADC_CHAN_MASK;
@@ -175,8 +173,8 @@ static int ad1938_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
 		return -EINVAL;
 	}
 
-	codec->write(codec, AD1938_DAC_CTRL1, dac_reg);
-	codec->write(codec, AD1938_ADC_CTRL2, adc_reg);
+	snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg);
+	snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg);
 
 	return 0;
 }
@@ -187,8 +185,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	int adc_reg, dac_reg;
 
-	adc_reg = codec->read(codec, AD1938_ADC_CTRL2);
-	dac_reg = codec->read(codec, AD1938_DAC_CTRL1);
+	adc_reg = snd_soc_read(codec, AD1938_ADC_CTRL2);
+	dac_reg = snd_soc_read(codec, AD1938_DAC_CTRL1);
 
 	/* At present, the driver only support AUX ADC mode(SND_SOC_DAIFMT_I2S
 	 * with TDM) and ADC&DAC TDM mode(SND_SOC_DAIFMT_DSP_A)
@@ -265,8 +263,8 @@ static int ad1938_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	codec->write(codec, AD1938_ADC_CTRL2, adc_reg);
-	codec->write(codec, AD1938_DAC_CTRL1, dac_reg);
+	snd_soc_write(codec, AD1938_ADC_CTRL2, adc_reg);
+	snd_soc_write(codec, AD1938_DAC_CTRL1, dac_reg);
 
 	return 0;
 }
@@ -295,134 +293,13 @@ static int ad1938_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	reg = codec->read(codec, AD1938_DAC_CTRL2);
+	reg = snd_soc_read(codec, AD1938_DAC_CTRL2);
 	reg = (reg & (~AD1938_DAC_WORD_LEN_MASK)) | word_len;
-	codec->write(codec, AD1938_DAC_CTRL2, reg);
+	snd_soc_write(codec, AD1938_DAC_CTRL2, reg);
 
-	reg = codec->read(codec, AD1938_ADC_CTRL1);
+	reg = snd_soc_read(codec, AD1938_ADC_CTRL1);
 	reg = (reg & (~AD1938_ADC_WORD_LEN_MASK)) | word_len;
-	codec->write(codec, AD1938_ADC_CTRL1, reg);
-
-	return 0;
-}
-
-static int ad1938_set_bias_level(struct snd_soc_codec *codec,
-		enum snd_soc_bias_level level)
-{
-	switch (level) {
-	case SND_SOC_BIAS_ON:
-		ad1938_pll_powerctrl(codec, 1);
-		break;
-	case SND_SOC_BIAS_PREPARE:
-		break;
-	case SND_SOC_BIAS_STANDBY:
-	case SND_SOC_BIAS_OFF:
-		ad1938_pll_powerctrl(codec, 0);
-		break;
-	}
-	codec->bias_level = level;
-	return 0;
-}
-
-/*
- * interface to read/write ad1938 register
- */
-
-#define AD1938_SPI_ADDR    0x4
-#define AD1938_SPI_READ    0x1
-#define AD1938_SPI_BUFLEN  3
-
-/*
- * write to the ad1938 register space
- */
-
-static int ad1938_write_reg(struct snd_soc_codec *codec, unsigned int reg,
-		unsigned int value)
-{
-	u8 *reg_cache = codec->reg_cache;
-	int ret = 0;
-
-	if (value != reg_cache[reg]) {
-		uint8_t buf[AD1938_SPI_BUFLEN];
-		struct spi_transfer t = {
-			.tx_buf = buf,
-			.len = AD1938_SPI_BUFLEN,
-		};
-		struct spi_message m;
-
-		buf[0] = AD1938_SPI_ADDR << 1;
-		buf[1] = reg;
-		buf[2] = value;
-		spi_message_init(&m);
-		spi_message_add_tail(&t, &m);
-		ret = spi_sync(codec->control_data, &m);
-		if (ret == 0)
-			reg_cache[reg] = value;
-	}
-
-	return ret;
-}
-
-/*
- * read from the ad1938 register space cache
- */
-
-static unsigned int ad1938_read_reg_cache(struct snd_soc_codec *codec,
-					  unsigned int reg)
-{
-	u8 *reg_cache = codec->reg_cache;
-
-	if (reg >= codec->reg_cache_size)
-		return -EINVAL;
-
-	return reg_cache[reg];
-}
-
-/*
- * read from the ad1938 register space
- */
-
-static unsigned int ad1938_read_reg(struct snd_soc_codec *codec,
-						unsigned int reg)
-{
-	char w_buf[AD1938_SPI_BUFLEN];
-	char r_buf[AD1938_SPI_BUFLEN];
-	int ret;
-
-	struct spi_transfer t = {
-		.tx_buf = w_buf,
-		.rx_buf = r_buf,
-		.len = AD1938_SPI_BUFLEN,
-	};
-	struct spi_message m;
-
-	w_buf[0] = (AD1938_SPI_ADDR << 1) | AD1938_SPI_READ;
-	w_buf[1] = reg;
-	w_buf[2] = 0;
-
-	spi_message_init(&m);
-	spi_message_add_tail(&t, &m);
-	ret = spi_sync(codec->control_data, &m);
-	if (ret == 0)
-		return	r_buf[2];
-	else
-		return -EIO;
-}
-
-static int ad1938_fill_cache(struct snd_soc_codec *codec)
-{
-	int i;
-	u8 *reg_cache = codec->reg_cache;
-	struct spi_device *spi = codec->control_data;
-
-	for (i = 0; i < codec->reg_cache_size; i++) {
-		int ret = ad1938_read_reg(codec, i);
-		if (ret == -EIO) {
-			dev_err(&spi->dev, "AD1938 SPI read failure\n");
-			return ret;
-		}
-		reg_cache[i] = ret;
-	}
+	snd_soc_write(codec, AD1938_ADC_CTRL1, reg);
 
 	return 0;
 }
@@ -512,32 +389,37 @@ static int ad1938_register(struct ad1938_priv *ad1938)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ad1938_dai;
 	codec->num_dai = 1;
-	codec->write = ad1938_write_reg;
-	codec->read = ad1938_read_reg_cache;
-	codec->set_bias_level = ad1938_set_bias_level;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	ad1938_dai.dev = codec->dev;
 	ad1938_codec = codec;
 
+	memcpy(codec->reg_cache, ad1938_reg, AD1938_NUM_REGS);
+
+	ret = snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_SPI);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n",
+				ret);
+		kfree(ad1938);
+		return ret;
+	}
+
 	/* default setting for ad1938 */
 
 	/* unmute dac channels */
-	codec->write(codec, AD1938_DAC_CHNL_MUTE, 0x0);
+	snd_soc_write(codec, AD1938_DAC_CHNL_MUTE, 0x0);
 	/* de-emphasis: 48kHz, powedown dac */
-	codec->write(codec, AD1938_DAC_CTRL2, 0x1A);
+	snd_soc_write(codec, AD1938_DAC_CTRL2, 0x1A);
 	/* powerdown dac, dac in tdm mode */
-	codec->write(codec, AD1938_DAC_CTRL0, 0x41);
+	snd_soc_write(codec, AD1938_DAC_CTRL0, 0x41);
 	/* high-pass filter enable */
-	codec->write(codec, AD1938_ADC_CTRL0, 0x3);
+	snd_soc_write(codec, AD1938_ADC_CTRL0, 0x3);
 	/* sata delay=1, adc aux mode */
-	codec->write(codec, AD1938_ADC_CTRL1, 0x43);
+	snd_soc_write(codec, AD1938_ADC_CTRL1, 0x43);
 	/* pll input: mclki/xi */
-	codec->write(codec, AD1938_PLL_CLK_CTRL0, 0x9D);
-	codec->write(codec, AD1938_PLL_CLK_CTRL1, 0x04);
-
-	ad1938_fill_cache(codec);
+	snd_soc_write(codec, AD1938_PLL_CLK_CTRL0, 0x9D);
+	snd_soc_write(codec, AD1938_PLL_CLK_CTRL1, 0x04);
 
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
@@ -559,7 +441,6 @@ static int ad1938_register(struct ad1938_priv *ad1938)
 
 static void ad1938_unregister(struct ad1938_priv *ad1938)
 {
-	ad1938_set_bias_level(&ad1938->codec, SND_SOC_BIAS_OFF);
 	snd_soc_unregister_dai(&ad1938_dai);
 	snd_soc_unregister_codec(&ad1938->codec);
 	kfree(ad1938);
@@ -593,7 +474,6 @@ static int ad1938_probe(struct platform_device *pdev)
 				  ARRAY_SIZE(ad1938_dapm_widgets));
 	snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
 
-	ad1938_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 pcm_err:
 	return ret;
@@ -610,37 +490,9 @@ static int ad1938_remove(struct platform_device *pdev)
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int ad1938_suspend(struct platform_device *pdev,
-		pm_message_t state)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	ad1938_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	return 0;
-}
-
-static int ad1938_resume(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
-
-	if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
-		ad1938_set_bias_level(codec, SND_SOC_BIAS_ON);
-
-	return 0;
-}
-#else
-#define ad1938_suspend NULL
-#define ad1938_resume NULL
-#endif
-
 struct snd_soc_codec_device soc_codec_dev_ad1938 = {
 	.probe = 	ad1938_probe,
 	.remove = 	ad1938_remove,
-	.suspend =      ad1938_suspend,
-	.resume =       ad1938_resume,
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_ad1938);
 

sound/soc/codecs/ak4104.c

@@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_44100 |
-			 SNDRV_PCM_RATE_48000 |
-			 SNDRV_PCM_RATE_32000,
+		.rates = SNDRV_PCM_RATE_8000_192000,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE  |
 			   SNDRV_PCM_FMTBIT_S24_3LE |
 			   SNDRV_PCM_FMTBIT_S24_LE

sound/soc/codecs/cs4270.c

@@ -28,6 +28,7 @@
 #include <sound/initval.h>
 #include <linux/i2c.h>
 #include <linux/delay.h>
+#include <linux/regulator/consumer.h>
 
 #include "cs4270.h"
 
@@ -106,6 +107,10 @@
 #define CS4270_MUTE_DAC_A	0x01
 #define CS4270_MUTE_DAC_B	0x02
 
+static const char *supply_names[] = {
+	"va", "vd", "vlc"
+};
+
 /* Private data for the CS4270 */
 struct cs4270_private {
 	struct snd_soc_codec codec;
@@ -114,6 +119,9 @@ struct cs4270_private {
 	unsigned int mode; /* The mode (I2S or left-justified) */
 	unsigned int slave_mode;
 	unsigned int manual_mute;
+
+	/* power domain regulators */
+	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
 /**
@@ -192,6 +200,11 @@ static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
  * This function must be called by the machine driver's 'startup' function,
  * otherwise the list of supported sample rates will not be available in
  * time for ALSA.
+ *
+ * For setups with variable MCLKs, pass 0 as 'freq' argument. This will cause
+ * theoretically possible sample rates to be enabled. Call it again with a
+ * proper value set one the external clock is set (most probably you would do
+ * that from a machine's driver 'hw_param' hook.
  */
 static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
@@ -205,20 +218,27 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 	cs4270->mclk = freq;
 
-	for (i = 0; i < NUM_MCLK_RATIOS; i++) {
-		unsigned int rate = freq / cs4270_mode_ratios[i].ratio;
-		rates |= snd_pcm_rate_to_rate_bit(rate);
-		if (rate < rate_min)
-			rate_min = rate;
-		if (rate > rate_max)
-			rate_max = rate;
-	}
-	/* FIXME: soc should support a rate list */
-	rates &= ~SNDRV_PCM_RATE_KNOT;
+	if (cs4270->mclk) {
+		for (i = 0; i < NUM_MCLK_RATIOS; i++) {
+			unsigned int rate = freq / cs4270_mode_ratios[i].ratio;
+			rates |= snd_pcm_rate_to_rate_bit(rate);
+			if (rate < rate_min)
+				rate_min = rate;
+			if (rate > rate_max)
+				rate_max = rate;
+		}
+		/* FIXME: soc should support a rate list */
+		rates &= ~SNDRV_PCM_RATE_KNOT;
 
-	if (!rates) {
-		dev_err(codec->dev, "could not find a valid sample rate\n");
-		return -EINVAL;
+		if (!rates) {
+			dev_err(codec->dev, "could not find a valid sample rate\n");
+			return -EINVAL;
+		}
+	} else {
+		/* enable all possible rates */
+		rates = SNDRV_PCM_RATE_8000_192000;
+		rate_min = 8000;
+		rate_max = 192000;
 	}
 
 	codec_dai->playback.rates = rates;
@@ -579,7 +599,8 @@ static int cs4270_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = cs4270_codec;
-	int ret;
+	struct cs4270_private *cs4270 = codec->private_data;
+	int i, ret;
 
 	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
 	socdev->card->codec = codec;
@@ -599,8 +620,26 @@ static int cs4270_probe(struct platform_device *pdev)
 		goto error_free_pcms;
 	}
 
+	/* get the power supply regulators */
+	for (i = 0; i < ARRAY_SIZE(supply_names); i++)
+		cs4270->supplies[i].supply = supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(cs4270->supplies),
+				 cs4270->supplies);
+	if (ret < 0)
+		goto error_free_pcms;
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
+				    cs4270->supplies);
+	if (ret < 0)
+		goto error_free_regulators;
+
 	return 0;
 
+error_free_regulators:
+	regulator_bulk_free(ARRAY_SIZE(cs4270->supplies),
+			    cs4270->supplies);
+
 error_free_pcms:
 	snd_soc_free_pcms(socdev);
 
@@ -616,8 +655,12 @@ static int cs4270_probe(struct platform_device *pdev)
 static int cs4270_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = cs4270_codec;
+	struct cs4270_private *cs4270 = codec->private_data;
 
 	snd_soc_free_pcms(socdev);
+	regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies), cs4270->supplies);
+	regulator_bulk_free(ARRAY_SIZE(cs4270->supplies), cs4270->supplies);
 
 	return 0;
 };
@@ -799,17 +842,33 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id);
 static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg)
 {
 	struct snd_soc_codec *codec = cs4270_codec;
-	int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+	struct cs4270_private *cs4270 = codec->private_data;
+	int reg, ret;
 
-	return snd_soc_write(codec, CS4270_PWRCTL, reg);
+	reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+	if (reg < 0)
+		return reg;
+
+	ret = snd_soc_write(codec, CS4270_PWRCTL, reg);
+	if (ret < 0)
+		return ret;
+
+	regulator_bulk_disable(ARRAY_SIZE(cs4270->supplies),
+			       cs4270->supplies);
+
+	return 0;
 }
 
 static int cs4270_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_codec *codec = cs4270_codec;
+	struct cs4270_private *cs4270 = codec->private_data;
 	struct i2c_client *i2c_client = codec->control_data;
 	int reg;
 
+	regulator_bulk_enable(ARRAY_SIZE(cs4270->supplies),
+			      cs4270->supplies);
+
 	/* In case the device was put to hard reset during sleep, we need to
 	 * wait 500ns here before any I2C communication. */
 	ndelay(500);

sound/soc/codecs/da7210.c

+/*
+ * DA7210 ALSA Soc codec driver
+ *
+ * Copyright (c) 2009 Dialog Semiconductor
+ * Written by David Chen <Dajun.chen@diasemi.com>
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * Tested on SuperH Ecovec24 board with S16/S24 LE in 48KHz using I2S
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <asm/div64.h>
+
+#include "da7210.h"
+
+/* DA7210 register space */
+#define DA7210_STATUS			0x02
+#define DA7210_STARTUP1			0x03
+#define DA7210_MIC_L			0x07
+#define DA7210_MIC_R			0x08
+#define DA7210_INMIX_L			0x0D
+#define DA7210_INMIX_R			0x0E
+#define DA7210_ADC_HPF			0x0F
+#define DA7210_ADC			0x10
+#define DA7210_DAC_HPF			0x14
+#define DA7210_DAC_L			0x15
+#define DA7210_DAC_R			0x16
+#define DA7210_DAC_SEL			0x17
+#define DA7210_OUTMIX_L			0x1C
+#define DA7210_OUTMIX_R			0x1D
+#define DA7210_HP_L_VOL			0x21
+#define DA7210_HP_R_VOL			0x22
+#define DA7210_HP_CFG			0x23
+#define DA7210_DAI_SRC_SEL		0x25
+#define DA7210_DAI_CFG1			0x26
+#define DA7210_DAI_CFG3			0x28
+#define DA7210_PLL_DIV3			0x2B
+#define DA7210_PLL			0x2C
+
+/* STARTUP1 bit fields */
+#define DA7210_SC_MST_EN		(1 << 0)
+
+/* MIC_L bit fields */
+#define DA7210_MICBIAS_EN		(1 << 6)
+#define DA7210_MIC_L_EN			(1 << 7)
+
+/* MIC_R bit fields */
+#define DA7210_MIC_R_EN			(1 << 7)
+
+/* INMIX_L bit fields */
+#define DA7210_IN_L_EN			(1 << 7)
+
+/* INMIX_R bit fields */
+#define DA7210_IN_R_EN			(1 << 7)
+
+/* ADC_HPF bit fields */
+#define DA7210_ADC_VOICE_EN		(1 << 7)
+
+/* ADC bit fields */
+#define DA7210_ADC_L_EN			(1 << 3)
+#define DA7210_ADC_R_EN			(1 << 7)
+
+/* DAC_HPF fields */
+#define DA7210_DAC_VOICE_EN		(1 << 7)
+
+/* DAC_SEL bit fields */
+#define DA7210_DAC_L_SRC_DAI_L		(4 << 0)
+#define DA7210_DAC_L_EN			(1 << 3)
+#define DA7210_DAC_R_SRC_DAI_R		(5 << 4)
+#define DA7210_DAC_R_EN			(1 << 7)
+
+/* OUTMIX_L bit fields */
+#define DA7210_OUT_L_EN			(1 << 7)
+
+/* OUTMIX_R bit fields */
+#define DA7210_OUT_R_EN			(1 << 7)
+
+/* HP_CFG bit fields */
+#define DA7210_HP_2CAP_MODE		(1 << 1)
+#define DA7210_HP_SENSE_EN		(1 << 2)
+#define DA7210_HP_L_EN			(1 << 3)
+#define DA7210_HP_MODE			(1 << 6)
+#define DA7210_HP_R_EN			(1 << 7)
+
+/* DAI_SRC_SEL bit fields */
+#define DA7210_DAI_OUT_L_SRC		(6 << 0)
+#define DA7210_DAI_OUT_R_SRC		(7 << 4)
+
+/* DAI_CFG1 bit fields */
+#define DA7210_DAI_WORD_S16_LE		(0 << 0)
+#define DA7210_DAI_WORD_S24_LE		(2 << 0)
+#define DA7210_DAI_FLEN_64BIT		(1 << 2)
+#define DA7210_DAI_MODE_MASTER		(1 << 7)
+
+/* DAI_CFG3 bit fields */
+#define DA7210_DAI_FORMAT_I2SMODE	(0 << 0)
+#define DA7210_DAI_OE			(1 << 3)
+#define DA7210_DAI_EN			(1 << 7)
+
+/*PLL_DIV3 bit fields */
+#define DA7210_MCLK_RANGE_10_20_MHZ	(1 << 4)
+#define DA7210_PLL_BYP			(1 << 6)
+
+/* PLL bit fields */
+#define DA7210_PLL_FS_48000		(11 << 0)
+
+#define DA7210_VERSION "0.0.1"
+
+/* Codec private data */
+struct da7210_priv {
+	struct snd_soc_codec codec;
+};
+
+static struct snd_soc_codec *da7210_codec;
+
+/*
+ * Register cache
+ */
+static const u8 da7210_reg[] = {
+	0x00, 0x11, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R0  - R7  */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x08,	/* R8  - RF  */
+	0x00, 0x00, 0x00, 0x00, 0x08, 0x10, 0x10, 0x54,	/* R10 - R17 */
+	0x40, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R18 - R1F */
+	0x00, 0x00, 0x00, 0x02, 0x00, 0x76, 0x00, 0x00,	/* R20 - R27 */
+	0x04, 0x00, 0x00, 0x30, 0x2A, 0x00, 0x40, 0x00,	/* R28 - R2F */
+	0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x40, 0x00,	/* R30 - R37 */
+	0x40, 0x00, 0x40, 0x00, 0x40, 0x00, 0x00, 0x00,	/* R38 - R3F */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R40 - R4F */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R48 - R4F */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R50 - R57 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R58 - R5F */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R60 - R67 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R68 - R6F */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R70 - R77 */
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x54, 0x54, 0x00,	/* R78 - R7F */
+	0x00, 0x00, 0x2C, 0x00, 0x00, 0x00, 0x00, 0x00,	/* R80 - R87 */
+	0x00,						/* R88       */
+};
+
+/*
+ * Read da7210 register cache
+ */
+static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg)
+{
+	u8 *cache = codec->reg_cache;
+	BUG_ON(reg > ARRAY_SIZE(da7210_reg));
+	return cache[reg];
+}
+
+/*
+ * 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->volatile_register);
+
+	data[0] = reg & 0xff;
+	data[1] = value & 0xff;
+
+	if (reg >= codec->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) {
+		/* PlayBack Volume 40 */
+		snd_soc_update_bits(codec, DA7210_HP_L_VOL, 0x3F, 40);
+		snd_soc_update_bits(codec, DA7210_HP_R_VOL, 0x3F, 40);
+
+		/* 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;
+}
+
+/*
+ * Set PCM DAI word length.
+ */
+static int da7210_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u32 dai_cfg1;
+	u32 reg, mask;
+
+	/* set DAI source to Left and Right ADC */
+	da7210_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);
+
+	dai_cfg1 = 0xFC & da7210_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_S24_LE:
+		dai_cfg1 |= DA7210_DAI_WORD_S24_LE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1);
+
+	/* FIXME
+	 *
+	 * It support 48K only now
+	 */
+	switch (params_rate(params)) {
+	case 48000:
+		if (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) {
+			reg  = DA7210_DAC_HPF;
+			mask = DA7210_DAC_VOICE_EN;
+		} else {
+			reg  = DA7210_ADC_HPF;
+			mask = DA7210_ADC_VOICE_EN;
+		}
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, reg, mask, 0);
+
+	return 0;
+}
+
+/*
+ * Set DAI mode and Format
+ */
+static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u32 dai_cfg1;
+	u32 dai_cfg3;
+
+	dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1);
+	dai_cfg3 = 0xfc & da7210_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;
+	default:
+		return -EINVAL;
+	}
+
+	/* FIXME
+	 *
+	 * It support I2S only now
+	 */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		dai_cfg3 |= DA7210_DAI_FORMAT_I2SMODE;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* FIXME
+	 *
+	 * It support 64bit data transmission only now
+	 */
+	dai_cfg1 |= DA7210_DAI_FLEN_64BIT;
+
+	da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1);
+	da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3);
+
+	return 0;
+}
+
+#define DA7210_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_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,
+};
+
+struct snd_soc_dai da7210_dai = {
+	.name = "DA7210 IIS",
+	.id = 0,
+	/* playback capabilities */
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = DA7210_FORMATS,
+	},
+	/* capture capabilities */
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.formats = DA7210_FORMATS,
+	},
+	.ops = &da7210_dai_ops,
+};
+EXPORT_SYMBOL_GPL(da7210_dai);
+
+/*
+ * Initialize the DA7210 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int da7210_init(struct da7210_priv *da7210)
+{
+	struct snd_soc_codec *codec = &da7210->codec;
+	int ret = 0;
+
+	if (da7210_codec) {
+		dev_err(codec->dev, "Another da7210 is registered\n");
+		return -EINVAL;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->private_data	= da7210;
+	codec->name		= "DA7210";
+	codec->owner		= THIS_MODULE;
+	codec->read		= da7210_read;
+	codec->write		= da7210_write;
+	codec->dai		= &da7210_dai;
+	codec->num_dai		= 1;
+	codec->hw_write		= (hw_write_t)i2c_master_send;
+	codec->reg_cache_size	= ARRAY_SIZE(da7210_reg);
+	codec->reg_cache	= kmemdup(da7210_reg,
+					  sizeof(da7210_reg), GFP_KERNEL);
+
+	if (!codec->reg_cache)
+		return -ENOMEM;
+
+	da7210_dai.dev = codec->dev;
+	da7210_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register CODEC: %d\n", ret);
+		goto init_err;
+	}
+
+	ret = snd_soc_register_dai(&da7210_dai);
+	if (ret) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto init_err;
+	}
+
+	/* FIXME
+	 *
+	 * This driver use fixed value here
+	 */
+
+	/*
+	 * ADC settings
+	 */
+
+	/* Enable Left & Right MIC PGA and Mic Bias */
+	da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN);
+	da7210_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);
+	da7210_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);
+
+	/*
+	 * DAC settings
+	 */
+
+	/* Enable Left and Right DAC */
+	da7210_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);
+	da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN);
+
+	/* Enable Left and Right HeadPhone PGA */
+	da7210_write(codec, DA7210_HP_CFG,
+		     DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN |
+		     DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN);
+
+	/* Diable PLL and bypass it */
+	da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000);
+
+	/* Bypass PLL and set MCLK freq rang to 10-20MHz */
+	da7210_write(codec, DA7210_PLL_DIV3,
+		     DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP);
+
+	/* Activate all enabled subsystem */
+	da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN);
+
+	return ret;
+
+init_err:
+	kfree(codec->reg_cache);
+	codec->reg_cache = NULL;
+
+	return ret;
+
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static int __devinit da7210_i2c_probe(struct i2c_client *i2c,
+			   	      const struct i2c_device_id *id)
+{
+	struct da7210_priv *da7210;
+	struct snd_soc_codec *codec;
+	int ret;
+
+	da7210 = kzalloc(sizeof(struct da7210_priv), GFP_KERNEL);
+	if (!da7210)
+		return -ENOMEM;
+
+	codec = &da7210->codec;
+	codec->dev = &i2c->dev;
+
+	i2c_set_clientdata(i2c, da7210);
+	codec->control_data = i2c;
+
+	ret = da7210_init(da7210);
+	if (ret < 0)
+		pr_err("Failed to initialise da7210 audio codec\n");
+
+	return ret;
+}
+
+static int __devexit da7210_i2c_remove(struct i2c_client *client)
+{
+	struct da7210_priv *da7210 = i2c_get_clientdata(client);
+
+	snd_soc_unregister_dai(&da7210_dai);
+	kfree(da7210->codec.reg_cache);
+	kfree(da7210);
+	da7210_codec = NULL;
+
+	return 0;
+}
+
+static const struct i2c_device_id da7210_i2c_id[] = {
+	{ "da7210", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, da7210_i2c_id);
+
+/* I2C codec control layer */
+static struct i2c_driver da7210_i2c_driver = {
+	.driver = {
+		.name = "DA7210 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = da7210_i2c_probe,
+	.remove =  __devexit_p(da7210_i2c_remove),
+	.id_table = da7210_i2c_id,
+};
+#endif
+
+static int da7210_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (!da7210_codec) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
+
+	socdev->card->codec = da7210_codec;
+	codec = da7210_codec;
+
+	/* Register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	dev_info(&pdev->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION);
+
+pcm_err:
+	return ret;
+}
+
+static int da7210_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_da7210 = {
+	.probe =	da7210_probe,
+	.remove =	da7210_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_da7210);
+
+static int __init da7210_modinit(void)
+{
+	int ret = 0;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&da7210_i2c_driver);
+#endif
+	return ret;
+}
+module_init(da7210_modinit);
+
+static void __exit da7210_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&da7210_i2c_driver);
+#endif
+}
+module_exit(da7210_exit);
+
+MODULE_DESCRIPTION("ASoC DA7210 driver");
+MODULE_AUTHOR("David Chen, Kuninori Morimoto");
+MODULE_LICENSE("GPL");

sound/soc/codecs/da7210.h

+/*
+ * da7210.h  --  audio driver for da7210
+ *
+ * Copyright (c) 2009 Dialog Semiconductor
+ * Written by David Chen <Dajun.chen@diasemi.com>
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Cleanups by Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ *
+ */
+
+#ifndef _DA7210_H
+#define _DA7210_H
+
+extern struct snd_soc_dai da7210_dai;
+extern struct snd_soc_codec_device soc_codec_dev_da7210;
+
+#endif
+

sound/soc/codecs/tlv320aic3x.c

@@ -765,9 +765,10 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
-	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
-	u16 pll_d = 1;
+	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
+	u16 d, pll_d = 1;
 	u8 reg;
+	int clk;
 
 	/* select data word length */
 	data =
@@ -833,48 +834,70 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	if (bypass_pll)
 		return 0;
 
-	/* Use PLL
-	 * find an apropriate setup for j, d, r and p by iterating over
-	 * p and r - j and d are calculated for each fraction.
-	 * Up to 128 values are probed, the closest one wins the game.
+	/* Use PLL, compute apropriate setup for j, d, r and p, the closest
+	 * one wins the game. Try with d==0 first, next with d!=0.
+	 * Constraints for j are according to the datasheet.
 	 * The sysclk is divided by 1000 to prevent integer overflows.
 	 */
+
 	codec_clk = (2048 * fsref) / (aic3x->sysclk / 1000);
 
 	for (r = 1; r <= 16; r++)
 		for (p = 1; p <= 8; p++) {
-			int clk, tmp = (codec_clk * pll_r * 10) / pll_p;
-			u8 j = tmp / 10000;
-			u16 d = tmp % 10000;
+			for (j = 4; j <= 55; j++) {
+				/* This is actually 1000*((j+(d/10000))*r)/p
+				 * The term had to be converted to get
+				 * rid of the division by 10000; d = 0 here
+				 */
+				int tmp_clk = (1000 * j * r) / p;
+
+				/* Check whether this values get closer than
+				 * the best ones we had before
+				 */
+				if (abs(codec_clk - tmp_clk) <
+					abs(codec_clk - last_clk)) {
+					pll_j = j; pll_d = 0;
+					pll_r = r; pll_p = p;
+					last_clk = tmp_clk;
+				}
+
+				/* Early exit for exact matches */
+				if (tmp_clk == codec_clk)
+					goto found;
+			}
+		}
 
-			if (j > 63)
-				continue;
+	/* try with d != 0 */
+	for (p = 1; p <= 8; p++) {
+		j = codec_clk * p / 1000;
 
-			if (d != 0 && aic3x->sysclk < 10000000)
-				continue;
+		if (j < 4 || j > 11)
+			continue;
 
-			/* This is actually 1000 * ((j + (d/10000)) * r) / p
-			 * The term had to be converted to get rid of the
-			 * division by 10000 */
-			clk = ((10000 * j * r) + (d * r)) / (10 * p);
+		/* do not use codec_clk here since we'd loose precision */
+		d = ((2048 * p * fsref) - j * aic3x->sysclk)
+			* 100 / (aic3x->sysclk/100);
 
-			/* check whether this values get closer than the best
-			 * ones we had before */
-			if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
-				pll_j = j; pll_d = d; pll_r = r; pll_p = p;
-				last_clk = clk;
-			}
+		clk = (10000 * j + d) / (10 * p);
 
-			/* Early exit for exact matches */
-			if (clk == codec_clk)
-				break;
+		/* check whether this values get closer than the best
+		 * ones we had before */
+		if (abs(codec_clk - clk) < abs(codec_clk - last_clk)) {
+			pll_j = j; pll_d = d; pll_r = 1; pll_p = p;
+			last_clk = clk;
 		}
 
+		/* Early exit for exact matches */
+		if (clk == codec_clk)
+			goto found;
+	}
+
 	if (last_clk == 0) {
 		printk(KERN_ERR "%s(): unable to setup PLL\n", __func__);
 		return -EINVAL;
 	}
 
+found:
 	data = aic3x_read_reg_cache(codec, AIC3X_PLL_PROGA_REG);
 	aic3x_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT));
 	aic3x_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT);

sound/soc/codecs/tlv320dac33.c

@@ -30,6 +30,7 @@
 #include <linux/platform_device.h>
 #include <linux/interrupt.h>
 #include <linux/gpio.h>
+#include <linux/regulator/consumer.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -58,11 +59,26 @@ enum dac33_state {
 	DAC33_FLUSH,
 };
 
+enum dac33_fifo_modes {
+	DAC33_FIFO_BYPASS = 0,
+	DAC33_FIFO_MODE1,
+	DAC33_FIFO_MODE7,
+	DAC33_FIFO_LAST_MODE,
+};
+
+#define DAC33_NUM_SUPPLIES 3
+static const char *dac33_supply_names[DAC33_NUM_SUPPLIES] = {
+	"AVDD",
+	"DVDD",
+	"IOVDD",
+};
+
 struct tlv320dac33_priv {
 	struct mutex mutex;
 	struct workqueue_struct *dac33_wq;
 	struct work_struct work;
 	struct snd_soc_codec codec;
+	struct regulator_bulk_data supplies[DAC33_NUM_SUPPLIES];
 	int power_gpio;
 	int chip_power;
 	int irq;
@@ -73,8 +89,9 @@ struct tlv320dac33_priv {
 					 * this */
 	unsigned int nsample_max;	/* nsample should not be higher than
 					 * this */
-	unsigned int nsample_switch;	/* Use FIFO or bypass FIFO switch */
+	enum dac33_fifo_modes fifo_mode;/* FIFO mode selection */
 	unsigned int nsample;		/* burst read amount from host */
+	u8 burst_bclkdiv;		/* BCLK divider value in burst mode */
 
 	enum dac33_state state;
 };
@@ -297,28 +314,49 @@ static inline void dac33_soft_power(struct snd_soc_codec *codec, int power)
 	dac33_write(codec, DAC33_PWR_CTRL, reg);
 }
 
-static void dac33_hard_power(struct snd_soc_codec *codec, int power)
+static int dac33_hard_power(struct snd_soc_codec *codec, int power)
 {
 	struct tlv320dac33_priv *dac33 = codec->private_data;
+	int ret;
 
 	mutex_lock(&dac33->mutex);
 	if (power) {
-		if (dac33->power_gpio >= 0) {
-			gpio_set_value(dac33->power_gpio, 1);
-			dac33->chip_power = 1;
-			/* Restore registers */
-			dac33_restore_regs(codec);
+		ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
+					  dac33->supplies);
+		if (ret != 0) {
+			dev_err(codec->dev,
+				"Failed to enable supplies: %d\n", ret);
+				goto exit;
 		}
+
+		if (dac33->power_gpio >= 0)
+			gpio_set_value(dac33->power_gpio, 1);
+
+		dac33->chip_power = 1;
+
+		/* Restore registers */
+		dac33_restore_regs(codec);
+
 		dac33_soft_power(codec, 1);
 	} else {
 		dac33_soft_power(codec, 0);
-		if (dac33->power_gpio >= 0) {
+		if (dac33->power_gpio >= 0)
 			gpio_set_value(dac33->power_gpio, 0);
-			dac33->chip_power = 0;
+
+		ret = regulator_bulk_disable(ARRAY_SIZE(dac33->supplies),
+					     dac33->supplies);
+		if (ret != 0) {
+			dev_err(codec->dev,
+				"Failed to disable supplies: %d\n", ret);
+			goto exit;
 		}
+
+		dac33->chip_power = 0;
 	}
-	mutex_unlock(&dac33->mutex);
 
+exit:
+	mutex_unlock(&dac33->mutex);
+	return ret;
 }
 
 static int dac33_get_nsample(struct snd_kcontrol *kcontrol,
@@ -351,39 +389,48 @@ static int dac33_set_nsample(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 
-static int dac33_get_nsample_switch(struct snd_kcontrol *kcontrol,
+static int dac33_get_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct tlv320dac33_priv *dac33 = codec->private_data;
 
-	ucontrol->value.integer.value[0] = dac33->nsample_switch;
+	ucontrol->value.integer.value[0] = dac33->fifo_mode;
 
 	return 0;
 }
 
-static int dac33_set_nsample_switch(struct snd_kcontrol *kcontrol,
+static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct tlv320dac33_priv *dac33 = codec->private_data;
 	int ret = 0;
 
-	if (dac33->nsample_switch == ucontrol->value.integer.value[0])
+	if (dac33->fifo_mode == ucontrol->value.integer.value[0])
 		return 0;
 	/* Do not allow changes while stream is running*/
 	if (codec->active)
 		return -EPERM;
 
 	if (ucontrol->value.integer.value[0] < 0 ||
-	    ucontrol->value.integer.value[0] > 1)
+	    ucontrol->value.integer.value[0] >= DAC33_FIFO_LAST_MODE)
 		ret = -EINVAL;
 	else
-		dac33->nsample_switch = ucontrol->value.integer.value[0];
+		dac33->fifo_mode = ucontrol->value.integer.value[0];
 
 	return ret;
 }
 
+/* Codec operation modes */
+static const char *dac33_fifo_mode_texts[] = {
+	"Bypass", "Mode 1", "Mode 7"
+};
+
+static const struct soc_enum dac33_fifo_mode_enum =
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts),
+			    dac33_fifo_mode_texts);
+
 /*
  * DACL/R digital volume control:
  * from 0 dB to -63.5 in 0.5 dB steps
@@ -406,8 +453,8 @@ static const struct snd_kcontrol_new dac33_snd_controls[] = {
 static const struct snd_kcontrol_new dac33_nsample_snd_controls[] = {
 	SOC_SINGLE_EXT("nSample", 0, 0, 5900, 0,
 		 dac33_get_nsample, dac33_set_nsample),
-	SOC_SINGLE_EXT("nSample Switch", 0, 0, 1, 0,
-		 dac33_get_nsample_switch, dac33_set_nsample_switch),
+	SOC_ENUM_EXT("FIFO Mode", dac33_fifo_mode_enum,
+		 dac33_get_fifo_mode, dac33_set_fifo_mode),
 };
 
 /* Analog bypass */
@@ -469,6 +516,8 @@ static int dac33_add_widgets(struct snd_soc_codec *codec)
 static int dac33_set_bias_level(struct snd_soc_codec *codec,
 				enum snd_soc_bias_level level)
 {
+	int ret;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 		dac33_soft_power(codec, 1);
@@ -476,12 +525,19 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		if (codec->bias_level == SND_SOC_BIAS_OFF)
-			dac33_hard_power(codec, 1);
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = dac33_hard_power(codec, 1);
+			if (ret != 0)
+				return ret;
+		}
+
 		dac33_soft_power(codec, 0);
 		break;
 	case SND_SOC_BIAS_OFF:
-		dac33_hard_power(codec, 0);
+		ret = dac33_hard_power(codec, 0);
+		if (ret != 0)
+			return ret;
+
 		break;
 	}
 	codec->bias_level = level;
@@ -489,6 +545,51 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+static inline void dac33_prefill_handler(struct tlv320dac33_priv *dac33)
+{
+	struct snd_soc_codec *codec;
+
+	codec = &dac33->codec;
+
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
+		dac33_write16(codec, DAC33_NSAMPLE_MSB,
+				DAC33_THRREG(dac33->nsample));
+		dac33_write16(codec, DAC33_PREFILL_MSB,
+				DAC33_THRREG(dac33->alarm_threshold));
+		break;
+	case DAC33_FIFO_MODE7:
+		dac33_write16(codec, DAC33_PREFILL_MSB,
+				DAC33_THRREG(10));
+		break;
+	default:
+		dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
+							dac33->fifo_mode);
+		break;
+	}
+}
+
+static inline void dac33_playback_handler(struct tlv320dac33_priv *dac33)
+{
+	struct snd_soc_codec *codec;
+
+	codec = &dac33->codec;
+
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
+		dac33_write16(codec, DAC33_NSAMPLE_MSB,
+				DAC33_THRREG(dac33->nsample));
+		break;
+	case DAC33_FIFO_MODE7:
+		/* At the moment we are not using interrupts in mode7 */
+		break;
+	default:
+		dev_warn(codec->dev, "Unhandled FIFO mode: %d\n",
+							dac33->fifo_mode);
+		break;
+	}
+}
+
 static void dac33_work(struct work_struct *work)
 {
 	struct snd_soc_codec *codec;
@@ -502,14 +603,10 @@ static void dac33_work(struct work_struct *work)
 	switch (dac33->state) {
 	case DAC33_PREFILL:
 		dac33->state = DAC33_PLAYBACK;
-		dac33_write16(codec, DAC33_NSAMPLE_MSB,
-				DAC33_THRREG(dac33->nsample));
-		dac33_write16(codec, DAC33_PREFILL_MSB,
-				DAC33_THRREG(dac33->alarm_threshold));
+		dac33_prefill_handler(dac33);
 		break;
 	case DAC33_PLAYBACK:
-		dac33_write16(codec, DAC33_NSAMPLE_MSB,
-				DAC33_THRREG(dac33->nsample));
+		dac33_playback_handler(dac33);
 		break;
 	case DAC33_IDLE:
 		break;
@@ -547,7 +644,7 @@ static void dac33_shutdown(struct snd_pcm_substream *substream,
 	unsigned int pwr_ctrl;
 
 	/* Stop pending workqueue */
-	if (dac33->nsample_switch)
+	if (dac33->fifo_mode)
 		cancel_work_sync(&dac33->work);
 
 	mutex_lock(&dac33->mutex);
@@ -603,7 +700,7 @@ static int dac33_hw_params(struct snd_pcm_substream *substream,
 }
 
 #define CALC_OSCSET(rate, refclk) ( \
-	((((rate * 10000) / refclk) * 4096) + 5000) / 10000)
+	((((rate * 10000) / refclk) * 4096) + 7000) / 10000)
 #define CALC_RATIOSET(rate, refclk) ( \
 	((((refclk  * 100000) / rate) * 16384) + 50000) / 100000)
 
@@ -619,7 +716,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct tlv320dac33_priv *dac33 = codec->private_data;
 	unsigned int oscset, ratioset, pwr_ctrl, reg_tmp;
-	u8 aictrl_a, fifoctrl_a;
+	u8 aictrl_a, aictrl_b, fifoctrl_a;
 
 	switch (substream->runtime->rate) {
 	case 44100:
@@ -637,7 +734,10 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 
 	aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A);
 	aictrl_a &= ~(DAC33_NCYCL_MASK | DAC33_WLEN_MASK);
+	/* Read FIFO control A, and clear FIFO flush bit */
 	fifoctrl_a = dac33_read_reg_cache(codec, DAC33_FIFO_CTRL_A);
+	fifoctrl_a &= ~DAC33_FIFOFLUSH;
+
 	fifoctrl_a &= ~DAC33_WIDTH;
 	switch (substream->runtime->format) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -675,7 +775,8 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 
 	dac33_oscwait(codec);
 
-	if (dac33->nsample_switch) {
+	if (dac33->fifo_mode) {
+		/* Generic for all FIFO modes */
 		/* 50-51 : ASRC Control registers */
 		dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */
 		dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */
@@ -685,38 +786,101 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream)
 
 		/* Set interrupts to high active */
 		dac33_write(codec, DAC33_INTP_CTRL_A, DAC33_INTPM_AHIGH);
-
-		dac33_write(codec, DAC33_FIFO_IRQ_MODE_B,
-			    DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL));
-		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
 	} else {
+		/* FIFO bypass mode */
 		/* 50-51 : ASRC Control registers */
 		dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCBYP);
 		dac33_write(codec, DAC33_ASRC_CTRL_B, 0); /* ??? */
 	}
 
-	if (dac33->nsample_switch)
+	/* Interrupt behaviour configuration */
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
+		dac33_write(codec, DAC33_FIFO_IRQ_MODE_B,
+			    DAC33_ATM(DAC33_FIFO_IRQ_MODE_LEVEL));
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, DAC33_MAT);
+		break;
+	case DAC33_FIFO_MODE7:
+		/* Disable all interrupts */
+		dac33_write(codec, DAC33_FIFO_IRQ_MASK, 0);
+		break;
+	default:
+		/* in FIFO bypass mode, the interrupts are not used */
+		break;
+	}
+
+	aictrl_b = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B);
+
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
+		/*
+		 * For mode1:
+		 * Disable the FIFO bypass (Enable the use of FIFO)
+		 * Select nSample mode
+		 * BCLK is only running when data is needed by DAC33
+		 */
 		fifoctrl_a &= ~DAC33_FBYPAS;
-	else
+		fifoctrl_a &= ~DAC33_FAUTO;
+		aictrl_b &= ~DAC33_BCLKON;
+		break;
+	case DAC33_FIFO_MODE7:
+		/*
+		 * For mode1:
+		 * Disable the FIFO bypass (Enable the use of FIFO)
+		 * Select Threshold mode
+		 * BCLK is only running when data is needed by DAC33
+		 */
+		fifoctrl_a &= ~DAC33_FBYPAS;
+		fifoctrl_a |= DAC33_FAUTO;
+		aictrl_b &= ~DAC33_BCLKON;
+		break;
+	default:
+		/*
+		 * For FIFO bypass mode:
+		 * Enable the FIFO bypass (Disable the FIFO use)
+		 * Set the BCLK as continous
+		 */
 		fifoctrl_a |= DAC33_FBYPAS;
-	dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a);
+		aictrl_b |= DAC33_BCLKON;
+		break;
+	}
 
+	dac33_write(codec, DAC33_FIFO_CTRL_A, fifoctrl_a);
 	dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_A, aictrl_a);
-	reg_tmp = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_B);
-	if (dac33->nsample_switch)
-		reg_tmp &= ~DAC33_BCLKON;
-	else
-		reg_tmp |= DAC33_BCLKON;
-	dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, reg_tmp);
+	dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_B, aictrl_b);
 
-	if (dac33->nsample_switch) {
-		/* 20: BCLK divide ratio */
-		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 3);
+	/*
+	 * BCLK divide ratio
+	 * 0: 1.5
+	 * 1: 1
+	 * 2: 2
+	 * ...
+	 * 254: 254
+	 * 255: 255
+	 */
+	if (dac33->fifo_mode)
+		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C,
+							dac33->burst_bclkdiv);
+	else
+		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
 
+	switch (dac33->fifo_mode) {
+	case DAC33_FIFO_MODE1:
 		dac33_write16(codec, DAC33_ATHR_MSB,
 			      DAC33_THRREG(dac33->alarm_threshold));
-	} else {
-		dac33_write(codec, DAC33_SER_AUDIOIF_CTRL_C, 32);
+		break;
+	case DAC33_FIFO_MODE7:
+		/*
+		 * Configure the threshold levels, and leave 10 sample space
+		 * at the bottom, and also at the top of the FIFO
+		 */
+		dac33_write16(codec, DAC33_UTHR_MSB,
+			DAC33_THRREG(DAC33_BUFFER_SIZE_SAMPLES - 10));
+		dac33_write16(codec, DAC33_LTHR_MSB,
+			DAC33_THRREG(10));
+		break;
+	default:
+		break;
 	}
 
 	mutex_unlock(&dac33->mutex);
@@ -789,7 +953,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (dac33->nsample_switch) {
+		if (dac33->fifo_mode) {
 			dac33->state = DAC33_PREFILL;
 			queue_work(dac33->dac33_wq, &dac33->work);
 		}
@@ -797,7 +961,7 @@ static int dac33_pcm_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (dac33->nsample_switch) {
+		if (dac33->fifo_mode) {
 			dac33->state = DAC33_FLUSH;
 			queue_work(dac33->dac33_wq, &dac33->work);
 		}
@@ -843,6 +1007,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			     unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct tlv320dac33_priv *dac33 = codec->private_data;
 	u8 aictrl_a, aictrl_b;
 
 	aictrl_a = dac33_read_reg_cache(codec, DAC33_SER_AUDIOIF_CTRL_A);
@@ -855,7 +1020,11 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		break;
 	case SND_SOC_DAIFMT_CBS_CFS:
 		/* Codec Slave */
-		aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK);
+		if (dac33->fifo_mode) {
+			dev_err(codec->dev, "FIFO mode requires master mode\n");
+			return -EINVAL;
+		} else
+			aictrl_a &= ~(DAC33_MSBCLK | DAC33_MSWCLK);
 		break;
 	default:
 		return -EINVAL;
@@ -959,6 +1128,9 @@ static int dac33_soc_probe(struct platform_device *pdev)
 	/* power on device */
 	dac33_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
+	/* Bias level configuration has enabled regulator an extra time */
+	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
+
 	return 0;
 
 pcm_err:
@@ -1033,13 +1205,13 @@ struct snd_soc_dai dac33_dai = {
 };
 EXPORT_SYMBOL_GPL(dac33_dai);
 
-static int dac33_i2c_probe(struct i2c_client *client,
-			   const struct i2c_device_id *id)
+static int __devinit dac33_i2c_probe(struct i2c_client *client,
+				     const struct i2c_device_id *id)
 {
 	struct tlv320dac33_platform_data *pdata;
 	struct tlv320dac33_priv *dac33;
 	struct snd_soc_codec *codec;
-	int ret = 0;
+	int ret, i;
 
 	if (client->dev.platform_data == NULL) {
 		dev_err(&client->dev, "Platform data not set\n");
@@ -1080,10 +1252,11 @@ static int dac33_i2c_probe(struct i2c_client *client,
 	i2c_set_clientdata(client, dac33);
 
 	dac33->power_gpio = pdata->power_gpio;
+	dac33->burst_bclkdiv = pdata->burst_bclkdiv;
 	dac33->irq = client->irq;
 	dac33->nsample = NSAMPLE_MAX;
 	/* Disable FIFO use by default */
-	dac33->nsample_switch = 0;
+	dac33->fifo_mode = DAC33_FIFO_BYPASS;
 
 	tlv320dac33_codec = codec;
 
@@ -1130,6 +1303,24 @@ static int dac33_i2c_probe(struct i2c_client *client,
 		}
 	}
 
+	for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++)
+		dac33->supplies[i].supply = dac33_supply_names[i];
+
+	ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(dac33->supplies),
+				 dac33->supplies);
+
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to request supplies: %d\n", ret);
+		goto err_get;
+	}
+
+	ret = regulator_bulk_enable(ARRAY_SIZE(dac33->supplies),
+				    dac33->supplies);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to enable supplies: %d\n", ret);
+		goto err_enable;
+	}
+
 	ret = snd_soc_register_codec(codec);
 	if (ret != 0) {
 		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
@@ -1149,6 +1340,10 @@ static int dac33_i2c_probe(struct i2c_client *client,
 	return ret;
 
 error_codec:
+	regulator_bulk_disable(ARRAY_SIZE(dac33->supplies), dac33->supplies);
+err_enable:
+	regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
+err_get:
 	if (dac33->irq >= 0) {
 		free_irq(dac33->irq, &dac33->codec);
 		destroy_workqueue(dac33->dac33_wq);
@@ -1165,7 +1360,7 @@ static int dac33_i2c_probe(struct i2c_client *client,
 	return ret;
 }
 
-static int dac33_i2c_remove(struct i2c_client *client)
+static int __devexit dac33_i2c_remove(struct i2c_client *client)
 {
 	struct tlv320dac33_priv *dac33;
 
@@ -1177,6 +1372,8 @@ static int dac33_i2c_remove(struct i2c_client *client)
 	if (dac33->irq >= 0)
 		free_irq(dac33->irq, &dac33->codec);
 
+	regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies);
+
 	destroy_workqueue(dac33->dac33_wq);
 	snd_soc_unregister_dai(&dac33_dai);
 	snd_soc_unregister_codec(&dac33->codec);

sound/soc/codecs/tpa6130a2.c

@@ -25,6 +25,7 @@
 #include <linux/device.h>
 #include <linux/i2c.h>
 #include <linux/gpio.h>
+#include <linux/regulator/consumer.h>
 #include <sound/tpa6130a2-plat.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
@@ -34,10 +35,22 @@
 
 static struct i2c_client *tpa6130a2_client;
 
+#define TPA6130A2_NUM_SUPPLIES 2
+static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
+	"CPVSS",
+	"Vdd",
+};
+
+static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = {
+	"HPVdd",
+	"AVdd",
+};
+
 /* This struct is used to save the context */
 struct tpa6130a2_data {
 	struct mutex mutex;
 	unsigned char regs[TPA6130A2_CACHEREGNUM];
+	struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES];
 	int power_gpio;
 	unsigned char power_state;
 };
@@ -106,10 +119,11 @@ static void tpa6130a2_initialize(void)
 		tpa6130a2_i2c_write(i, data->regs[i]);
 }
 
-static void tpa6130a2_power(int power)
+static int tpa6130a2_power(int power)
 {
 	struct	tpa6130a2_data *data;
 	u8	val;
+	int	ret;
 
 	BUG_ON(tpa6130a2_client == NULL);
 	data = i2c_get_clientdata(tpa6130a2_client);
@@ -117,11 +131,20 @@ static void tpa6130a2_power(int power)
 	mutex_lock(&data->mutex);
 	if (power) {
 		/* Power on */
-		if (data->power_gpio >= 0) {
+		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 1);
-			data->power_state = 1;
-			tpa6130a2_initialize();
+
+		ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies),
+					    data->supplies);
+		if (ret != 0) {
+			dev_err(&tpa6130a2_client->dev,
+				"Failed to enable supplies: %d\n", ret);
+			goto exit;
 		}
+
+		data->power_state = 1;
+		tpa6130a2_initialize();
+
 		/* Clear SWS */
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val &= ~TPA6130A2_SWS;
@@ -131,13 +154,25 @@ static void tpa6130a2_power(int power)
 		val = tpa6130a2_read(TPA6130A2_REG_CONTROL);
 		val |= TPA6130A2_SWS;
 		tpa6130a2_i2c_write(TPA6130A2_REG_CONTROL, val);
+
 		/* Power off */
-		if (data->power_gpio >= 0) {
+		if (data->power_gpio >= 0)
 			gpio_set_value(data->power_gpio, 0);
-			data->power_state = 0;
+
+		ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies),
+					     data->supplies);
+		if (ret != 0) {
+			dev_err(&tpa6130a2_client->dev,
+				"Failed to disable supplies: %d\n", ret);
+			goto exit;
 		}
+
+		data->power_state = 0;
 	}
+
+exit:
 	mutex_unlock(&data->mutex);
+	return ret;
 }
 
 static int tpa6130a2_get_reg(struct snd_kcontrol *kcontrol,
@@ -237,12 +272,8 @@ static const struct snd_kcontrol_new tpa6130a2_controls[] = {
  */
 static void tpa6130a2_channel_enable(u8 channel, int enable)
 {
-	struct	tpa6130a2_data *data;
 	u8	val;
 
-	BUG_ON(tpa6130a2_client == NULL);
-	data = i2c_get_clientdata(tpa6130a2_client);
-
 	if (enable) {
 		/* Enable channel */
 		/* Enable amplifier */
@@ -299,15 +330,17 @@ static int tpa6130a2_right_event(struct snd_soc_dapm_widget *w,
 static int tpa6130a2_supply_event(struct snd_soc_dapm_widget *w,
 		struct snd_kcontrol *kcontrol, int event)
 {
+	int ret = 0;
+
 	switch (event) {
 	case SND_SOC_DAPM_POST_PMU:
-		tpa6130a2_power(1);
+		ret = tpa6130a2_power(1);
 		break;
 	case SND_SOC_DAPM_POST_PMD:
-		tpa6130a2_power(0);
+		ret = tpa6130a2_power(0);
 		break;
 	}
-	return 0;
+	return ret;
 }
 
 static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = {
@@ -346,13 +379,13 @@ int tpa6130a2_add_controls(struct snd_soc_codec *codec)
 }
 EXPORT_SYMBOL_GPL(tpa6130a2_add_controls);
 
-static int tpa6130a2_probe(struct i2c_client *client,
-			   const struct i2c_device_id *id)
+static int __devinit tpa6130a2_probe(struct i2c_client *client,
+				     const struct i2c_device_id *id)
 {
 	struct device *dev;
 	struct tpa6130a2_data *data;
 	struct tpa6130a2_platform_data *pdata;
-	int ret;
+	int i, ret;
 
 	dev = &client->dev;
 
@@ -387,15 +420,38 @@ static int tpa6130a2_probe(struct i2c_client *client,
 		if (ret < 0) {
 			dev_err(dev, "Failed to request power GPIO (%d)\n",
 				data->power_gpio);
-			goto fail;
+			goto err_gpio;
 		}
 		gpio_direction_output(data->power_gpio, 0);
-	} else {
-		data->power_state = 1;
-		tpa6130a2_initialize();
 	}
 
-	tpa6130a2_power(1);
+	switch (pdata->id) {
+	case TPA6130A2:
+		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+			data->supplies[i].supply = tpa6130a2_supply_names[i];
+		break;
+	case TPA6140A2:
+		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+			data->supplies[i].supply = tpa6140a2_supply_names[i];;
+		break;
+	default:
+		dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n",
+			 pdata->id);
+		for (i = 0; i < ARRAY_SIZE(data->supplies); i++)
+			data->supplies[i].supply = tpa6130a2_supply_names[i];
+	}
+
+	ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies),
+				 data->supplies);
+	if (ret != 0) {
+		dev_err(dev, "Failed to request supplies: %d\n", ret);
+		goto err_regulator;
+	}
+
+	ret = tpa6130a2_power(1);
+	if (ret != 0)
+		goto err_power;
+
 
 	/* Read version */
 	ret = tpa6130a2_i2c_read(TPA6130A2_REG_VERSION) &
@@ -404,10 +460,18 @@ static int tpa6130a2_probe(struct i2c_client *client,
 		dev_warn(dev, "UNTESTED version detected (%d)\n", ret);
 
 	/* Disable the chip */
-	tpa6130a2_power(0);
+	ret = tpa6130a2_power(0);
+	if (ret != 0)
+		goto err_power;
 
 	return 0;
-fail:
+
+err_power:
+	regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies);
+err_regulator:
+	if (data->power_gpio >= 0)
+		gpio_free(data->power_gpio);
+err_gpio:
 	kfree(data);
 	i2c_set_clientdata(tpa6130a2_client, NULL);
 	tpa6130a2_client = NULL;
@@ -415,7 +479,7 @@ static int tpa6130a2_probe(struct i2c_client *client,
 	return ret;
 }
 
-static int tpa6130a2_remove(struct i2c_client *client)
+static int __devexit tpa6130a2_remove(struct i2c_client *client)
 {
 	struct tpa6130a2_data *data = i2c_get_clientdata(client);
 
@@ -423,6 +487,9 @@ static int tpa6130a2_remove(struct i2c_client *client)
 
 	if (data->power_gpio >= 0)
 		gpio_free(data->power_gpio);
+
+	regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies);
+
 	kfree(data);
 	tpa6130a2_client = NULL;
 

sound/soc/codecs/twl4030.c

@@ -55,7 +55,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x0c, /* REG_ATXR1PGA		(0xB)	*/
 	0x00, /* REG_AVTXL2PGA		(0xC)	*/
 	0x00, /* REG_AVTXR2PGA		(0xD)	*/
-	0x01, /* REG_AUDIO_IF		(0xE)	*/
+	0x00, /* REG_AUDIO_IF		(0xE)	*/
 	0x00, /* REG_VOICE_IF		(0xF)	*/
 	0x00, /* REG_ARXR1PGA		(0x10)	*/
 	0x00, /* REG_ARXL1PGA		(0x11)	*/
@@ -64,19 +64,19 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_VRXPGA		(0x14)	*/
 	0x00, /* REG_VSTPGA		(0x15)	*/
 	0x00, /* REG_VRX2ARXPGA		(0x16)	*/
-	0x0c, /* REG_AVDAC_CTL		(0x17)	*/
+	0x00, /* REG_AVDAC_CTL		(0x17)	*/
 	0x00, /* REG_ARX2VTXPGA		(0x18)	*/
 	0x00, /* REG_ARXL1_APGA_CTL	(0x19)	*/
 	0x00, /* REG_ARXR1_APGA_CTL	(0x1A)	*/
-	0x4b, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
-	0x4b, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
+	0x4a, /* REG_ARXL2_APGA_CTL	(0x1B)	*/
+	0x4a, /* REG_ARXR2_APGA_CTL	(0x1C)	*/
 	0x00, /* REG_ATX2ARXPGA		(0x1D)	*/
 	0x00, /* REG_BT_IF		(0x1E)	*/
 	0x00, /* REG_BTPGA		(0x1F)	*/
 	0x00, /* REG_BTSTPGA		(0x20)	*/
 	0x00, /* REG_EAR_CTL		(0x21)	*/
-	0x24, /* REG_HS_SEL		(0x22)	*/
-	0x0a, /* REG_HS_GAIN_SET	(0x23)	*/
+	0x00, /* REG_HS_SEL		(0x22)	*/
+	0x00, /* REG_HS_GAIN_SET	(0x23)	*/
 	0x00, /* REG_HS_POPN_SET	(0x24)	*/
 	0x00, /* REG_PREDL_CTL		(0x25)	*/
 	0x00, /* REG_PREDR_CTL		(0x26)	*/
@@ -99,7 +99,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_I2S_RX_SCRAMBLE_H	(0x37)	*/
 	0x00, /* REG_I2S_RX_SCRAMBLE_M	(0x38)	*/
 	0x00, /* REG_I2S_RX_SCRAMBLE_L	(0x39)	*/
-	0x16, /* REG_APLL_CTL		(0x3A)	*/
+	0x06, /* REG_APLL_CTL		(0x3A)	*/
 	0x00, /* REG_DTMF_CTL		(0x3B)	*/
 	0x00, /* REG_DTMF_PGA_CTL2	(0x3C)	*/
 	0x00, /* REG_DTMF_PGA_CTL1	(0x3D)	*/
@@ -1203,6 +1203,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("APLL Enable", SND_SOC_NOPM, 0, 0, apll_event,
 			    SND_SOC_DAPM_PRE_PMU|SND_SOC_DAPM_POST_PMD),
 
+	SND_SOC_DAPM_SUPPLY("AIF Enable", TWL4030_REG_AUDIO_IF, 0, 0, NULL, 0),
+
 	/* Output MIXER controls */
 	/* Earpiece */
 	SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
@@ -1337,6 +1339,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Digital L2 Playback Mixer", NULL, "APLL Enable"},
 	{"Digital Voice Playback Mixer", NULL, "APLL Enable"},
 
+	{"Digital R1 Playback Mixer", NULL, "AIF Enable"},
+	{"Digital L1 Playback Mixer", NULL, "AIF Enable"},
+	{"Digital R2 Playback Mixer", NULL, "AIF Enable"},
+	{"Digital L2 Playback Mixer", NULL, "AIF Enable"},
+
 	{"Analog L1 Playback Mixer", NULL, "Digital L1 Playback Mixer"},
 	{"Analog R1 Playback Mixer", NULL, "Digital R1 Playback Mixer"},
 	{"Analog L2 Playback Mixer", NULL, "Digital L2 Playback Mixer"},
@@ -1455,6 +1462,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "APLL Enable"},
 	{"ADC Virtual Right2", NULL, "APLL Enable"},
 
+	{"ADC Virtual Left1", NULL, "AIF Enable"},
+	{"ADC Virtual Right1", NULL, "AIF Enable"},
+	{"ADC Virtual Left2", NULL, "AIF Enable"},
+	{"ADC Virtual Right2", NULL, "AIF Enable"},
+
 	/* Analog bypass routes */
 	{"Right1 Analog Loopback", "Switch", "Analog Right"},
 	{"Left1 Analog Loopback", "Switch", "Analog Left"},
@@ -2152,8 +2164,6 @@ static int twl4030_soc_remove(struct platform_device *pdev)
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
-	kfree(codec->private_data);
-	kfree(codec);
 
 	return 0;
 }
@@ -2192,7 +2202,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev)
 	codec->write = twl4030_write;
 	codec->set_bias_level = twl4030_set_bias_level;
 	codec->dai = twl4030_dai;
-	codec->num_dai = ARRAY_SIZE(twl4030_dai),
+	codec->num_dai = ARRAY_SIZE(twl4030_dai);
 	codec->reg_cache_size = sizeof(twl4030_reg);
 	codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg),
 					GFP_KERNEL);
@@ -2237,6 +2247,9 @@ static int __devexit twl4030_codec_remove(struct platform_device *pdev)
 {
 	struct twl4030_priv *twl4030 = platform_get_drvdata(pdev);
 
+	snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai));
+	snd_soc_unregister_codec(&twl4030->codec);
+	kfree(twl4030->codec.reg_cache);
 	kfree(twl4030);
 
 	twl4030_codec = NULL;

sound/soc/codecs/twl4030.h

@@ -25,7 +25,7 @@
 /* Register descriptions are here */
 #include <linux/mfd/twl4030-codec.h>
 
-/* Sgadow register used by the audio driver */
+/* Shadow register used by the audio driver */
 #define TWL4030_REG_SW_SHADOW		0x4A
 #define TWL4030_CACHEREGNUM	(TWL4030_REG_SW_SHADOW + 1)
 

sound/soc/codecs/wm2000.h

+/*
+ * wm2000.h  --  WM2000 Soc Audio driver
+ *
+ * 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 _WM2000_H
+#define _WM2000_H
+
+struct wm2000_setup_data {
+	unsigned short i2c_address;
+	int mclk_div;   /* Set to a non-zero value if MCLK_DIV_2 required */
+};
+
+extern int wm2000_add_controls(struct snd_soc_codec *codec);
+
+extern struct snd_soc_dai wm2000_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm2000;
+
+#define WM2000_REG_SYS_START	    0x8000
+#define WM2000_REG_SPEECH_CLARITY   0x8fef
+#define WM2000_REG_SYS_WATCHDOG     0x8ff6
+#define WM2000_REG_ANA_VMID_PD_TIME 0x8ff7
+#define WM2000_REG_ANA_VMID_PU_TIME 0x8ff8
+#define WM2000_REG_CAT_FLTR_INDX    0x8ff9
+#define WM2000_REG_CAT_GAIN_0       0x8ffa
+#define WM2000_REG_SYS_STATUS       0x8ffc
+#define WM2000_REG_SYS_MODE_CNTRL   0x8ffd
+#define WM2000_REG_SYS_START0       0x8ffe
+#define WM2000_REG_SYS_START1       0x8fff
+#define WM2000_REG_ID1              0xf000
+#define WM2000_REG_ID2              0xf001
+#define WM2000_REG_REVISON          0xf002
+#define WM2000_REG_SYS_CTL1         0xf003
+#define WM2000_REG_SYS_CTL2         0xf004
+#define WM2000_REG_ANC_STAT         0xf005
+#define WM2000_REG_IF_CTL           0xf006
+
+/* SPEECH_CLARITY */
+#define WM2000_SPEECH_CLARITY   0x01
+
+/* SYS_STATUS */
+#define WM2000_STATUS_MOUSE_ACTIVE              0x40
+#define WM2000_STATUS_CAT_FREQ_COMPLETE         0x20
+#define WM2000_STATUS_CAT_GAIN_COMPLETE         0x10
+#define WM2000_STATUS_THERMAL_SHUTDOWN_COMPLETE 0x08
+#define WM2000_STATUS_ANC_DISABLED              0x04
+#define WM2000_STATUS_POWER_DOWN_COMPLETE       0x02
+#define WM2000_STATUS_BOOT_COMPLETE             0x01
+
+/* SYS_MODE_CNTRL */
+#define WM2000_MODE_ANA_SEQ_INCLUDE 0x80
+#define WM2000_MODE_MOUSE_ENABLE    0x40
+#define WM2000_MODE_CAT_FREQ_ENABLE 0x20
+#define WM2000_MODE_CAT_GAIN_ENABLE 0x10
+#define WM2000_MODE_BYPASS_ENTRY    0x08
+#define WM2000_MODE_STANDBY_ENTRY   0x04
+#define WM2000_MODE_THERMAL_ENABLE  0x02
+#define WM2000_MODE_POWER_DOWN      0x01
+
+/* SYS_CTL1 */
+#define WM2000_SYS_STBY          0x01
+
+/* SYS_CTL2 */
+#define WM2000_MCLK_DIV2_ENA_CLR 0x80
+#define WM2000_MCLK_DIV2_ENA_SET 0x40
+#define WM2000_ANC_ENG_CLR       0x20
+#define WM2000_ANC_ENG_SET       0x10
+#define WM2000_ANC_INT_N_CLR     0x08
+#define WM2000_ANC_INT_N_SET     0x04
+#define WM2000_RAM_CLR           0x02
+#define WM2000_RAM_SET           0x01
+
+/* ANC_STAT */
+#define WM2000_ANC_ENG_IDLE      0x01
+
+#endif

sound/soc/codecs/wm8727.c

@@ -44,23 +44,16 @@ struct snd_soc_dai wm8727_dai = {
 };
 EXPORT_SYMBOL_GPL(wm8727_dai);
 
+static struct snd_soc_codec *wm8727_codec;
+
 static int wm8727_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-	mutex_init(&codec->mutex);
-	codec->name = "WM8727";
-	codec->owner = THIS_MODULE;
-	codec->dai = &wm8727_dai;
-	codec->num_dai = 1;
-	socdev->card->codec = codec;
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+	BUG_ON(!wm8727_codec);
+
+	socdev->card->codec = wm8727_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -80,12 +73,9 @@ static int wm8727_soc_probe(struct platform_device *pdev)
 static int wm8727_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->card->codec;
 
-	if (codec == NULL)
-		return 0;
 	snd_soc_free_pcms(socdev);
-	kfree(codec);
+
 	return 0;
 }
 
@@ -98,13 +88,55 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_wm8727);
 
 static __devinit int wm8727_platform_probe(struct platform_device *pdev)
 {
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (wm8727_codec) {
+		dev_err(&pdev->dev, "Another WM8727 is registered\n");
+		return -EBUSY;
+	}
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+	wm8727_codec = codec;
+
+	platform_set_drvdata(pdev, codec);
+
+	mutex_init(&codec->mutex);
+	codec->dev = &pdev->dev;
+	codec->name = "WM8727";
+	codec->owner = THIS_MODULE;
+	codec->dai = &wm8727_dai;
+	codec->num_dai = 1;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
 	wm8727_dai.dev = &pdev->dev;
-	return snd_soc_register_dai(&wm8727_dai);
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register CODEC: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8727_dai);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(codec);
+	return ret;
 }
 
 static int __devexit wm8727_platform_remove(struct platform_device *pdev)
 {
 	snd_soc_unregister_dai(&wm8727_dai);
+	snd_soc_unregister_codec(platform_get_drvdata(pdev));
 	return 0;
 }
 

sound/soc/codecs/wm8731.c

@@ -456,6 +456,9 @@ static int wm8731_resume(struct platform_device *pdev)
 
 	/* Sync reg_cache with the hardware */
 	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);

sound/soc/codecs/wm8753.c

@@ -1507,10 +1507,6 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	/* we only need to suspend if we are a valid card */
-	if (!codec->card)
-		return 0;
-
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
@@ -1523,10 +1519,6 @@ static int wm8753_resume(struct platform_device *pdev)
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
 
-	/* we only need to resume if we are a valid card */
-	if (!codec->card)
-		return 0;
-
 	/* Sync reg_cache with the hardware */
 	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
 		if (i + 1 == WM8753_RESET)

sound/soc/codecs/wm8776.c

@@ -406,6 +406,8 @@ static int wm8776_resume(struct platform_device *pdev)
 
 	/* 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);

sound/soc/codecs/wm8961.c

@@ -1022,6 +1022,9 @@ static int wm8961_resume(struct platform_device *pdev)
 	int i;
 
 	for (i = 0; i < codec->reg_cache_size; i++) {
+		if (reg_cache[i] == wm8961_reg_defaults[i])
+			continue;
+
 		if (i == WM8961_SOFTWARE_RESET)
 			continue;
 

sound/soc/codecs/wm8974.c

@@ -170,6 +170,10 @@ SOC_ENUM("Aux Mode", wm8974_auxmode),
 
 SOC_SINGLE("Capture Boost(+20dB)", WM8974_ADCBOOST,  8, 1, 0),
 SOC_SINGLE("Mono Playback Switch", WM8974_MONOMIX, 6, 1, 1),
+
+/* DAC / ADC oversampling */
+SOC_SINGLE("DAC 128x Oversampling Switch", WM8974_DAC, 8, 1, 0),
+SOC_SINGLE("ADC 128x Oversampling Switch", WM8974_ADC, 8, 1, 0),
 };
 
 /* Speaker Output Mixer */
@@ -381,14 +385,6 @@ static int wm8974_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		reg = snd_soc_read(codec, WM8974_CLOCK) & 0x11f;
 		snd_soc_write(codec, WM8974_CLOCK, reg | div);
 		break;
-	case WM8974_ADCCLK:
-		reg = snd_soc_read(codec, WM8974_ADC) & 0x1f7;
-		snd_soc_write(codec, WM8974_ADC, reg | div);
-		break;
-	case WM8974_DACCLK:
-		reg = snd_soc_read(codec, WM8974_DAC) & 0x1f7;
-		snd_soc_write(codec, WM8974_DAC, reg | div);
-		break;
 	case WM8974_BCLKDIV:
 		reg = snd_soc_read(codec, WM8974_CLOCK) & 0x1e3;
 		snd_soc_write(codec, WM8974_CLOCK, reg | div);

sound/soc/codecs/wm8974.h

@@ -57,17 +57,7 @@
 /* Clock divider Id's */
 #define WM8974_OPCLKDIV		0
 #define WM8974_MCLKDIV		1
-#define WM8974_ADCCLK		2
-#define WM8974_DACCLK		3
-#define WM8974_BCLKDIV		4
-
-/* DAC clock dividers */
-#define WM8974_DACCLK_F2	(1 << 3)
-#define WM8974_DACCLK_F4	(0 << 3)
-
-/* ADC clock dividers */
-#define WM8974_ADCCLK_F2	(1 << 3)
-#define WM8974_ADCCLK_F4	(0 << 3)
+#define WM8974_BCLKDIV		2
 
 /* PLL Out dividers */
 #define WM8974_OPCLKDIV_1	(0 << 4)

sound/soc/codecs/wm8978.h

+/*
+ * wm8978.h		--  codec driver for WM8978
+ *
+ * Copyright 2009 Guennadi Liakhovetski <g.liakhovetski@gmx.de>
+ *
+ * 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 __WM8978_H__
+#define __WM8978_H__
+
+/*
+ * Register values.
+ */
+#define WM8978_RESET				0x00
+#define WM8978_POWER_MANAGEMENT_1		0x01
+#define WM8978_POWER_MANAGEMENT_2		0x02
+#define WM8978_POWER_MANAGEMENT_3		0x03
+#define WM8978_AUDIO_INTERFACE			0x04
+#define WM8978_COMPANDING_CONTROL		0x05
+#define WM8978_CLOCKING				0x06
+#define WM8978_ADDITIONAL_CONTROL		0x07
+#define WM8978_GPIO_CONTROL			0x08
+#define WM8978_JACK_DETECT_CONTROL_1		0x09
+#define WM8978_DAC_CONTROL			0x0A
+#define WM8978_LEFT_DAC_DIGITAL_VOLUME		0x0B
+#define WM8978_RIGHT_DAC_DIGITAL_VOLUME		0x0C
+#define WM8978_JACK_DETECT_CONTROL_2		0x0D
+#define WM8978_ADC_CONTROL			0x0E
+#define WM8978_LEFT_ADC_DIGITAL_VOLUME		0x0F
+#define WM8978_RIGHT_ADC_DIGITAL_VOLUME		0x10
+#define WM8978_EQ1				0x12
+#define WM8978_EQ2				0x13
+#define WM8978_EQ3				0x14
+#define WM8978_EQ4				0x15
+#define WM8978_EQ5				0x16
+#define WM8978_DAC_LIMITER_1			0x18
+#define WM8978_DAC_LIMITER_2			0x19
+#define WM8978_NOTCH_FILTER_1			0x1b
+#define WM8978_NOTCH_FILTER_2			0x1c
+#define WM8978_NOTCH_FILTER_3			0x1d
+#define WM8978_NOTCH_FILTER_4			0x1e
+#define WM8978_ALC_CONTROL_1			0x20
+#define WM8978_ALC_CONTROL_2			0x21
+#define WM8978_ALC_CONTROL_3			0x22
+#define WM8978_NOISE_GATE			0x23
+#define WM8978_PLL_N				0x24
+#define WM8978_PLL_K1				0x25
+#define WM8978_PLL_K2				0x26
+#define WM8978_PLL_K3				0x27
+#define WM8978_3D_CONTROL			0x29
+#define WM8978_BEEP_CONTROL			0x2b
+#define WM8978_INPUT_CONTROL			0x2c
+#define WM8978_LEFT_INP_PGA_CONTROL		0x2d
+#define WM8978_RIGHT_INP_PGA_CONTROL		0x2e
+#define WM8978_LEFT_ADC_BOOST_CONTROL		0x2f
+#define WM8978_RIGHT_ADC_BOOST_CONTROL		0x30
+#define WM8978_OUTPUT_CONTROL			0x31
+#define WM8978_LEFT_MIXER_CONTROL		0x32
+#define WM8978_RIGHT_MIXER_CONTROL		0x33
+#define WM8978_LOUT1_HP_CONTROL			0x34
+#define WM8978_ROUT1_HP_CONTROL			0x35
+#define WM8978_LOUT2_SPK_CONTROL		0x36
+#define WM8978_ROUT2_SPK_CONTROL		0x37
+#define WM8978_OUT3_MIXER_CONTROL		0x38
+#define WM8978_OUT4_MIXER_CONTROL		0x39
+
+#define WM8978_CACHEREGNUM			58
+
+/* Clock divider Id's */
+enum wm8978_clk_id {
+	WM8978_OPCLKRATE,
+	WM8978_BCLKDIV,
+};
+
+enum wm8978_sysclk_src {
+	WM8978_PLL,
+	WM8978_MCLK
+};
+
+extern struct snd_soc_dai wm8978_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8978;
+
+#endif	/* __WM8978_H__ */

sound/soc/codecs/wm8990.c

@@ -1319,10 +1319,6 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
 
-	/* we only need to suspend if we are a valid card */
-	if (!codec->card)
-		return 0;
-
 	wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
 }
@@ -1335,10 +1331,6 @@ static int wm8990_resume(struct platform_device *pdev)
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
 
-	/* we only need to resume if we are a valid card */
-	if (!codec->card)
-		return 0;
-
 	/* Sync reg_cache with the hardware */
 	for (i = 0; i < ARRAY_SIZE(wm8990_reg); i++) {
 		if (i + 1 == WM8990_RESET)

sound/soc/codecs/wm8994.h

+/*
+ * wm8994.h  --  WM8994 Soc Audio driver
+ *
+ * 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 _WM8994_H
+#define _WM8994_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_codec_device soc_codec_dev_wm8994;
+extern struct snd_soc_dai wm8994_dai[];
+
+/* Sources for AIF1/2 SYSCLK - use with set_dai_sysclk() */
+#define WM8994_SYSCLK_MCLK1 1
+#define WM8994_SYSCLK_MCLK2 2
+#define WM8994_SYSCLK_FLL1  3
+#define WM8994_SYSCLK_FLL2  4
+
+#define WM8994_FLL1 1
+#define WM8994_FLL2 2
+
+#endif

sound/soc/codecs/wm9713.c

@@ -23,13 +23,12 @@
 #include <sound/ac97_codec.h>
 #include <sound/initval.h>
 #include <sound/pcm_params.h>
+#include <sound/tlv.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
 #include "wm9713.h"
 
-#define WM9713_VERSION "0.15"
-
 struct wm9713_priv {
 	u32 pll_in; /* PLL input frequency */
 };
@@ -115,15 +114,27 @@ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18
 SOC_ENUM_SINGLE(MICB_MUX, 0, 2, wm9713_micb_select), /* mic selection 19 */
 };
 
+static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(misc_tlv, -1500, 300, 0);
+static unsigned int mic_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
+	3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0),
+};
+
 static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
-SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+SOC_DOUBLE_TLV("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1, out_tlv),
 SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
-SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+SOC_DOUBLE_TLV("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+	       out_tlv),
 SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1),
-SOC_DOUBLE("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1),
-SOC_DOUBLE("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1),
-SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
-SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
+SOC_DOUBLE_TLV("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1, main_tlv),
+SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv),
+SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
+SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv),
+SOC_SINGLE_TLV("Mic 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv),
+SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv),
 
 SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
 SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
@@ -133,7 +144,7 @@ SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
 SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
 SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
 
-SOC_SINGLE("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1),
+SOC_SINGLE_TLV("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1, misc_tlv),
 SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
 SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
 
@@ -154,28 +165,43 @@ SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
 
 SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
 SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
-SOC_SINGLE("Out4 Playback Volume", AC97_MASTER_MONO, 8, 63, 1),
+SOC_SINGLE_TLV("Out4 Playback Volume", AC97_MASTER_MONO, 8, 31, 1, out_tlv),
 
 SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
 SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
-SOC_SINGLE("Out3 Playback Volume", AC97_MASTER_MONO, 0, 63, 1),
+SOC_SINGLE_TLV("Out3 Playback Volume", AC97_MASTER_MONO, 0, 31, 1, out_tlv),
 
-SOC_SINGLE("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1),
+SOC_SINGLE_TLV("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1, main_tlv),
 SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
 SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
-SOC_SINGLE("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1),
+SOC_SINGLE_TLV("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1, out_tlv),
 
-SOC_SINGLE("Beep Playback Headphone Volume", AC97_AUX, 12, 7, 1),
-SOC_SINGLE("Beep Playback Speaker Volume", AC97_AUX, 8, 7, 1),
-SOC_SINGLE("Beep Playback Mono Volume", AC97_AUX, 4, 7, 1),
+SOC_SINGLE_TLV("Headphone Mixer Beep Playback Volume", AC97_AUX, 12, 7, 1,
+	       misc_tlv),
+SOC_SINGLE_TLV("Speaker Mixer Beep Playback Volume", AC97_AUX, 8, 7, 1,
+	       misc_tlv),
+SOC_SINGLE_TLV("Mono Mixer Beep Playback Volume", AC97_AUX, 4, 7, 1, misc_tlv),
 
-SOC_SINGLE("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1),
+SOC_SINGLE_TLV("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1,
+	       misc_tlv),
 SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
 SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
 
+SOC_SINGLE_TLV("Headphone Mixer Aux Playback Volume", AC97_REC_SEL, 12, 7, 1,
+	       misc_tlv),
+
+SOC_SINGLE_TLV("Speaker Mixer Voice Playback Volume", AC97_PCM, 8, 7, 1,
+	       misc_tlv),
+SOC_SINGLE_TLV("Speaker Mixer Aux Playback Volume", AC97_REC_SEL, 8, 7, 1,
+	       misc_tlv),
+
+SOC_SINGLE_TLV("Mono Mixer Voice Playback Volume", AC97_PCM, 4, 7, 1,
+	       misc_tlv),
+SOC_SINGLE_TLV("Mono Mixer Aux Playback Volume", AC97_REC_SEL, 4, 7, 1,
+	       misc_tlv),
+
 SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
 SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
-SOC_SINGLE("Aux Playback Mono Volume", AC97_REC_SEL, 4, 7, 1),
 
 SOC_ENUM("Bass Control", wm9713_enum[16]),
 SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
@@ -1186,8 +1212,6 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret = 0, reg;
 
-	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
-
 	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
 				      GFP_KERNEL);
 	if (socdev->card->codec == NULL)

sound/soc/codecs/wm_hubs.c

@@ -68,24 +68,77 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec)
 	int count = 0;
 
 	dev_dbg(codec->dev, "Waiting for DC servo...\n");
+
 	do {
 		count++;
 		msleep(1);
 		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0);
-		dev_dbg(codec->dev, "DC servo status: %x\n", reg);
-	} while ((reg & WM8993_DCS_CAL_COMPLETE_MASK)
-		 != WM8993_DCS_CAL_COMPLETE_MASK && count < 1000);
+		dev_dbg(codec->dev, "DC servo: %x\n", reg);
+	} while (reg & WM8993_DCS_DATAPATH_BUSY);
 
-	if ((reg & WM8993_DCS_CAL_COMPLETE_MASK)
-	    != WM8993_DCS_CAL_COMPLETE_MASK)
+	if (reg & WM8993_DCS_DATAPATH_BUSY)
 		dev_err(codec->dev, "Timed out waiting for DC Servo\n");
 }
 
+/*
+ * Startup calibration of the DC servo
+ */
+static void calibrate_dc_servo(struct snd_soc_codec *codec)
+{
+	struct wm_hubs_data *hubs = codec->private_data;
+	u16 reg, dcs_cfg;
+
+	/* Set for 32 series updates */
+	snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
+			    WM8993_DCS_SERIES_NO_01_MASK,
+			    32 << WM8993_DCS_SERIES_NO_01_SHIFT);
+
+	/* Enable the DC servo.  Write all bits to avoid triggering startup
+	 * or write calibration.
+	 */
+	snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
+			    0xFFFF,
+			    WM8993_DCS_ENA_CHAN_0 |
+			    WM8993_DCS_ENA_CHAN_1 |
+			    WM8993_DCS_TRIG_SERIES_1 |
+			    WM8993_DCS_TRIG_SERIES_0);
+
+	wait_for_dc_servo(codec);
+
+	/* Apply correction to DC servo result */
+	if (hubs->dcs_codes) {
+		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
+			hubs->dcs_codes);
+
+		/* HPOUT1L */
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) &
+			WM8993_DCS_INTEG_CHAN_0_MASK;;
+		reg += hubs->dcs_codes;
+		dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+
+		/* HPOUT1R */
+		reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) &
+			WM8993_DCS_INTEG_CHAN_1_MASK;
+		reg += hubs->dcs_codes;
+		dcs_cfg |= reg;
+
+		/* Do it */
+		snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
+		snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
+				    WM8993_DCS_TRIG_DAC_WR_0 |
+				    WM8993_DCS_TRIG_DAC_WR_1,
+				    WM8993_DCS_TRIG_DAC_WR_0 |
+				    WM8993_DCS_TRIG_DAC_WR_1);
+
+		wait_for_dc_servo(codec);
+	}
+}
+
 /*
  * Update the DC servo calibration on gain changes
  */
 static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol)
+			       struct snd_ctl_elem_value *ucontrol)
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	int ret;
@@ -251,6 +304,47 @@ SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1,
 	       line_tlv),
 };
 
+static int hp_supply_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct wm_hubs_data *hubs = codec->private_data;
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		switch (hubs->hp_startup_mode) {
+		case 0:
+			break;
+		case 1:
+			/* Enable the headphone amp */
+			snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
+					    WM8993_HPOUT1L_ENA |
+					    WM8993_HPOUT1R_ENA,
+					    WM8993_HPOUT1L_ENA |
+					    WM8993_HPOUT1R_ENA);
+
+			/* Enable the second stage */
+			snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
+					    WM8993_HPOUT1L_DLY |
+					    WM8993_HPOUT1R_DLY,
+					    WM8993_HPOUT1L_DLY |
+					    WM8993_HPOUT1R_DLY);
+			break;
+		default:
+			dev_err(codec->dev, "Unknown HP startup mode %d\n",
+				hubs->hp_startup_mode);
+			break;
+		}
+
+	case SND_SOC_DAPM_PRE_PMD:
+		snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
+				    WM8993_CP_ENA, 0);
+		break;
+	}
+
+	return 0;
+}
+
 static int hp_event(struct snd_soc_dapm_widget *w,
 		    struct snd_kcontrol *kcontrol, int event)
 {
@@ -271,14 +365,11 @@ static int hp_event(struct snd_soc_dapm_widget *w,
 		reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY;
 		snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
 
-		/* Start the DC servo */
-		snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-				    0xFFFF,
-				    WM8993_DCS_ENA_CHAN_0 |
-				    WM8993_DCS_ENA_CHAN_1 |
-				    WM8993_DCS_TRIG_STARTUP_1 |
-				    WM8993_DCS_TRIG_STARTUP_0);
-		wait_for_dc_servo(codec);
+		/* Smallest supported update interval */
+		snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
+				    WM8993_DCS_TIMER_PERIOD_01_MASK, 1);
+
+		calibrate_dc_servo(codec);
 
 		reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT |
 			WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT;
@@ -286,23 +377,19 @@ static int hp_event(struct snd_soc_dapm_widget *w,
 		break;
 
 	case SND_SOC_DAPM_PRE_PMD:
-		reg &= ~(WM8993_HPOUT1L_RMV_SHORT |
-			 WM8993_HPOUT1L_DLY |
-			 WM8993_HPOUT1L_OUTP |
-			 WM8993_HPOUT1R_RMV_SHORT |
-			 WM8993_HPOUT1R_DLY |
-			 WM8993_HPOUT1R_OUTP);
+		snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
+				    WM8993_HPOUT1L_DLY |
+				    WM8993_HPOUT1R_DLY |
+				    WM8993_HPOUT1L_RMV_SHORT |
+				    WM8993_HPOUT1R_RMV_SHORT, 0);
 
-		snd_soc_update_bits(codec, WM8993_DC_SERVO_0,
-				    0xffff, 0);
+		snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
+				    WM8993_HPOUT1L_OUTP |
+				    WM8993_HPOUT1R_OUTP, 0);
 
-		snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
 		snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
 				    WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
 				    0);
-
-		snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
-				    WM8993_CP_ENA, 0);
 		break;
 	}
 
@@ -473,6 +560,8 @@ SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0,
 SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
 
+SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event, 
+		    SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
 SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0,
 		   NULL, 0,
 		   hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
@@ -626,6 +715,7 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 	{ "Headphone PGA", NULL, "Left Headphone Mux" },
 	{ "Headphone PGA", NULL, "Right Headphone Mux" },
 	{ "Headphone PGA", NULL, "CLK_SYS" },
+	{ "Headphone PGA", NULL, "Headphone Supply" },
 
 	{ "HPOUT1L", NULL, "Headphone PGA" },
 	{ "HPOUT1R", NULL, "Headphone PGA" },
@@ -753,6 +843,12 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
 				    WM8993_LINEOUT2_MODE,
 				    WM8993_LINEOUT2_MODE);
 
+	/* If the line outputs are differential then we aren't presenting
+	 * VMID as an output and can disable it.
+	 */
+	if (lineout1_diff && lineout2_diff)
+		codec->idle_bias_off = 1;
+
 	if (lineout1fb)
 		snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
 				    WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);

sound/soc/codecs/wm_hubs.h

@@ -18,6 +18,12 @@ struct snd_soc_codec;
 
 extern const unsigned int wm_hubs_spkmix_tlv[];
 
+/* This *must* be the first element of the codec->private_data struct */
+struct wm_hubs_data {
+	int dcs_codes;
+	int hp_startup_mode;
+};
+
 extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *);
 extern int wm_hubs_add_analogue_routes(struct snd_soc_codec *, int, int);
 extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *,

sound/soc/davinci/davinci-mcasp.h

@@ -44,6 +44,7 @@ struct davinci_audio_dev {
 	int sample_rate;
 	struct clk *clk;
 	unsigned int codec_fmt;
+	u8 clk_active;
 
 	/* McASP specific data */
 	int	tdm_slots;