Merge branch 'fix/hda' into topic/hda

25 files changed, 1235 insertions, 53 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index e0b9869df0f..38a0e3b620a 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -3,9 +3,11 @@ config SND_SOC_ALL_CODECS
 	depends on I2C
 	select SPI
 	select SPI_MASTER
+	select SND_SOC_AD73311
 	select SND_SOC_AK4535
 	select SND_SOC_CS4270
 	select SND_SOC_SSM2602
+	select SND_SOC_TLV320AIC23
 	select SND_SOC_TLV320AIC26
 	select SND_SOC_TLV320AIC3X
 	select SND_SOC_UDA1380
@@ -34,6 +36,9 @@ config SND_SOC_AC97_CODEC
 config SND_SOC_AD1980
 	tristate
 
+config SND_SOC_AD73311
+	tristate
+
 config SND_SOC_AK4535
 	tristate
 
@@ -58,9 +63,13 @@ config SND_SOC_CS4270_VD33_ERRATA
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_TLV320AIC23
+	tristate
+	depends on I2C
+
 config SND_SOC_TLV320AIC26
-	tristate "TI TLV320AIC26 Codec support"
-	depends on SND_SOC && SPI
+	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
+	depends on SPI
 
 config SND_SOC_TLV320AIC3X
 	tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f977978a340..90f0a585fc7 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,8 +1,10 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
+snd-soc-ad73311-objs := ad73311.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
 snd-soc-uda1380-objs := uda1380.o
@@ -20,9 +22,11 @@ snd-soc-wm9713-objs := wm9713.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
+obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index 61fd96ca7bc..bd1ebdc6c86 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -2,8 +2,7 @@
  * ac97.c  --  ALSA Soc AC97 codec support
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 4e09c1f2c06..1397b8e06c0 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -13,7 +13,6 @@
 
 #include <linux/init.h>
 #include <linux/module.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/device.h>
 #include <sound/core.h>
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
new file mode 100644
index 00000000000..37af8607b00
--- /dev/null
+++ b/sound/soc/codecs/ad73311.c
@@ -0,0 +1,107 @@
+/*
+ * ad73311.c  --  ALSA Soc AD73311 codec support
+ *
+ * Copyright:	Analog Device Inc.
+ * Author:	Cliff Cai <cliff.cai@analog.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
+ *    25th Sep 2008   Initial version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/version.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include "ad73311.h"
+
+struct snd_soc_dai ad73311_dai = {
+	.name = "AD73311",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = SNDRV_PCM_RATE_8000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+};
+EXPORT_SYMBOL_GPL(ad73311_dai);
+
+static int ad73311_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 = "AD73311";
+	codec->owner = THIS_MODULE;
+	codec->dai = &ad73311_dai;
+	codec->num_dai = 1;
+	socdev->codec = codec;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "ad73311: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "ad73311: failed to register card\n");
+		goto register_err;
+	}
+
+	return ret;
+
+register_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	kfree(socdev->codec);
+	socdev->codec = NULL;
+	return ret;
+}
+
+static int ad73311_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec == NULL)
+		return 0;
+	snd_soc_free_pcms(socdev);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_ad73311 = {
+	.probe = 	ad73311_soc_probe,
+	.remove = 	ad73311_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_ad73311);
+
+MODULE_DESCRIPTION("ASoC ad73311 driver");
+MODULE_AUTHOR("Cliff Cai ");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
new file mode 100644
index 00000000000..507ce0c30ed
--- /dev/null
+++ b/sound/soc/codecs/ad73311.h
@@ -0,0 +1,90 @@
+/*
+ * File:         sound/soc/codec/ad73311.h
+ * Based on:
+ * Author:       Cliff Cai <cliff.cai@analog.com>
+ *
+ * Created:      Thur Sep 25, 2008
+ * Description:  definitions for AD73311 registers
+ *
+ *
+ * Modified:
+ *               Copyright 2006 Analog Devices Inc.
+ *
+ * Bugs:         Enter bugs at http://blackfin.uclinux.org/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, see the file COPYING, or write
+ * to the Free Software Foundation, Inc.,
+ * 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+ */
+
+#ifndef __AD73311_H__
+#define __AD73311_H__
+
+#define AD_CONTROL	0x8000
+#define AD_DATA		0x0000
+#define AD_READ		0x4000
+#define AD_WRITE	0x0000
+
+/* Control register A */
+#define CTRL_REG_A	(0 << 8)
+
+#define REGA_MODE_PRO	0x00
+#define REGA_MODE_DATA	0x01
+#define REGA_MODE_MIXED	0x03
+#define REGA_DLB		0x04
+#define REGA_SLB		0x08
+#define REGA_DEVC(x)		((x & 0x7) << 4)
+#define REGA_RESET		0x80
+
+/* Control register B */
+#define CTRL_REG_B	(1 << 8)
+
+#define REGB_DIRATE(x)	(x & 0x3)
+#define REGB_SCDIV(x)	((x & 0x3) << 2)
+#define REGB_MCDIV(x)	((x & 0x7) << 4)
+#define REGB_CEE		(1 << 7)
+
+/* Control register C */
+#define CTRL_REG_C	(2 << 8)
+
+#define REGC_PUDEV		(1 << 0)
+#define REGC_PUADC		(1 << 3)
+#define REGC_PUDAC		(1 << 4)
+#define REGC_PUREF		(1 << 5)
+#define REGC_REFUSE		(1 << 6)
+
+/* Control register D */
+#define CTRL_REG_D	(3 << 8)
+
+#define REGD_IGS(x)		(x & 0x7)
+#define REGD_RMOD		(1 << 3)
+#define REGD_OGS(x)		((x & 0x7) << 4)
+#define REGD_MUTE		(x << 7)
+
+/* Control register E */
+#define CTRL_REG_E	(4 << 8)
+
+#define REGE_DA(x)		(x & 0x1f)
+#define REGE_IBYP		(1 << 5)
+
+/* Control register F */
+#define CTRL_REG_F	(5 << 8)
+
+#define REGF_SEEN		(1 << 5)
+#define REGF_INV		(1 << 6)
+#define REGF_ALB		(1 << 7)
+
+extern struct snd_soc_dai ad73311_dai;
+extern struct snd_soc_codec_device soc_codec_dev_ad73311;
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 088cf992772..2a89b5888e1 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -28,7 +28,6 @@
 
 #include "ak4535.h"
 
-#define AUDIO_NAME "ak4535"
 #define AK4535_VERSION "0.3"
 
 struct snd_soc_codec_device soc_codec_dev_ak4535;
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 940ce1c3522..44ef0dacd56 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -42,7 +42,6 @@
 
 #include "ssm2602.h"
 
-#define AUDIO_NAME "ssm2602"
 #define SSM2602_VERSION "0.1"
 
 struct snd_soc_codec_device soc_codec_dev_ssm2602;
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
new file mode 100644
index 00000000000..44308dac9e1
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -0,0 +1,714 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd.,
+ *
+ * Based on sound/soc/codecs/wm8731.c by Richard Purdie
+ *
+ * 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.
+ *
+ * Notes:
+ *  The AIC23 is a driver for a low power stereo audio
+ *  codec tlv320aic23
+ *
+ *  The machine layer should disable unsupported inputs/outputs by
+ *  snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <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 "tlv320aic23.h"
+
+#define AIC23_VERSION "0.1"
+
+struct tlv320aic23_srate_reg_info {
+	u32 sample_rate;
+	u8 control;		/* SR3, SR2, SR1, SR0 and BOSR */
+	u8 divider;		/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */
+};
+
+/*
+ * AIC23 register cache
+ */
+static const u16 tlv320aic23_reg[] = {
+	0x0097, 0x0097, 0x00F9, 0x00F9,	/* 0 */
+	0x001A, 0x0004, 0x0007, 0x0001,	/* 4 */
+	0x0020, 0x0000, 0x0000, 0x0000,	/* 8 */
+	0x0000, 0x0000, 0x0000, 0x0000,	/* 12 */
+};
+
+/*
+ * read tlv320aic23 register cache
+ */
+static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
+						      *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return -1;
+	return cache[reg];
+}
+
+/*
+ * write tlv320aic23 register cache
+ */
+static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
+					       u8 reg, u16 value)
+{
+	u16 *cache = codec->reg_cache;
+	if (reg >= ARRAY_SIZE(tlv320aic23_reg))
+		return;
+	cache[reg] = value;
+}
+
+/*
+ * write to the tlv320aic23 register space
+ */
+static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
+			     unsigned int value)
+{
+
+	u8 data[2];
+
+	/* TLV320AIC23 has 7 bit address and 9 bits of data
+	 * so we need to switch one data bit into reg and rest
+	 * of data into val
+	 */
+
+	if ((reg < 0 || reg > 9) && (reg != 15)) {
+		printk(KERN_WARNING "%s Invalid register R%d\n", __func__, reg);
+		return -1;
+	}
+
+	data[0] = (reg << 1) | (value >> 8 & 0x01);
+	data[1] = value & 0xff;
+
+	tlv320aic23_write_reg_cache(codec, reg, value);
+
+	if (codec->hw_write(codec->control_data, data, 2) == 2)
+		return 0;
+
+	printk(KERN_ERR "%s cannot write %03x to register R%d\n", __func__,
+	       value, reg);
+
+	return -EIO;
+}
+
+static const char *rec_src_text[] = { "Line", "Mic" };
+static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
+
+static const struct soc_enum rec_src_enum =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+
+static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
+SOC_DAPM_ENUM("Input Select", rec_src_enum);
+
+static const struct soc_enum tlv320aic23_rec_src =
+	SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
+static const struct soc_enum tlv320aic23_deemph =
+	SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
+
+static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
+static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
+
+static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u16 val, reg;
+
+	val = (ucontrol->value.integer.value[0] & 0x07);
+
+	/* linear conversion to userspace
+	* 000	=	-6db
+	* 001	=	-9db
+	* 010	=	-12db
+	* 011	=	-18db (Min)
+	* 100	=	0db (Max)
+	*/
+	val = (val >= 4) ? 4  : (3 - val);
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
+	tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
+
+	return 0;
+}
+
+static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	u16 val;
+
+	val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
+	val = val >> 6;
+	val = (val >= 4) ? 4  : (3 -  val);
+	ucontrol->value.integer.value[0] = val;
+	return 0;
+
+}
+
+#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
+	.put = snd_soc_tlv320aic23_put_volsw, \
+	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
+			 TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
+	SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
+	SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
+		     TLV320AIC23_RINVOL, 7, 1, 0),
+	SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
+			 TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
+	SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
+	SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
+	SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
+				  6, 4, 0, sidetone_vol_tlv),
+	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
+};
+
+/* add non dapm controls */
+static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
+{
+
+	int err, i;
+
+	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
+		err = snd_ctl_add(codec->card,
+				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
+					       codec, NULL));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+
+}
+
+/* PGA Mixer controls for Line and Mic switch */
+static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
+	SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
+	SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
+			 &tlv320aic23_rec_src_mux_controls),
+	SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
+			   &tlv320aic23_output_mixer_controls[0],
+			   ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
+	SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("LHPOUT"),
+	SND_SOC_DAPM_OUTPUT("RHPOUT"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+
+	SND_SOC_DAPM_INPUT("LLINEIN"),
+	SND_SOC_DAPM_INPUT("RLINEIN"),
+
+	SND_SOC_DAPM_INPUT("MICIN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	/* Output Mixer */
+	{"Output Mixer", "Line Bypass Switch", "Line Input"},
+	{"Output Mixer", "Playback Switch", "DAC"},
+	{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
+
+	/* Outputs */
+	{"RHPOUT", NULL, "Output Mixer"},
+	{"LHPOUT", NULL, "Output Mixer"},
+	{"LOUT", NULL, "Output Mixer"},
+	{"ROUT", NULL, "Output Mixer"},
+
+	/* Inputs */
+	{"Line Input", "NULL", "LLINEIN"},
+	{"Line Input", "NULL", "RLINEIN"},
+
+	{"Mic Input", "NULL", "MICIN"},
+
+	/* input mux */
+	{"Capture Source", "Line", "Line Input"},
+	{"Capture Source", "Mic", "Mic Input"},
+	{"ADC", NULL, "Capture Source"},
+
+};
+
+/* tlv320aic23 related */
+static const struct tlv320aic23_srate_reg_info srate_reg_info[] = {
+	{4000, 0x06, 1},	/*  4000 */
+	{8000, 0x06, 0},	/*  8000 */
+	{16000, 0x0C, 1},	/* 16000 */
+	{22050, 0x11, 1},	/* 22050 */
+	{24000, 0x00, 1},	/* 24000 */
+	{32000, 0x0C, 0},	/* 32000 */
+	{44100, 0x11, 0},	/* 44100 */
+	{48000, 0x00, 0},	/* 48000 */
+	{88200, 0x1F, 0},	/* 88200 */
+	{96000, 0x0E, 0},	/* 96000 */
+};
+
+static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* set up audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	u16 iface_reg, data;
+	u8 count = 0;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec,
+				       TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
+
+	/* Search for the right sample rate */
+	/* Verify what happens if the rate is not supported
+	 * now it goes to 96Khz */
+	while ((srate_reg_info[count].sample_rate != params_rate(params)) &&
+	       (count < ARRAY_SIZE(srate_reg_info))) {
+		count++;
+	}
+
+	data =  (srate_reg_info[count].divider << TLV320AIC23_CLKIN_SHIFT) |
+		(srate_reg_info[count]. control << TLV320AIC23_BOSR_SHIFT) |
+		TLV320AIC23_USB_CLK_ON;
+
+	tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		iface_reg |= (0x01 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iface_reg |= (0x02 << 2);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		iface_reg |= (0x03 << 2);
+		break;
+	}
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* set active */
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
+
+	return 0;
+}
+
+static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+
+	/* deactivate */
+	if (!codec->active) {
+		udelay(50);
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	}
+}
+
+static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 reg;
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
+	if (mute)
+		reg |= TLV320AIC23_DACM_MUTE;
+
+	else
+		reg &= ~TLV320AIC23_DACM_MUTE;
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				   unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 iface_reg;
+
+	iface_reg =
+	    tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		iface_reg |= TLV320AIC23_MS_MASTER;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface_reg |= TLV320AIC23_FOR_I2S;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		iface_reg |= TLV320AIC23_FOR_DSP;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iface_reg |= TLV320AIC23_FOR_LJUST;
+		break;
+	default:
+		return -EINVAL;
+
+	}
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
+
+	return 0;
+}
+
+static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+				      int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	switch (freq) {
+	case 12000000:
+		return 0;
+	}
+	return -EINVAL;
+}
+
+static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
+				      enum snd_soc_bias_level level)
+{
+	u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* vref/mid, osc on, dac unmute */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		/* everything off except vref/vmid, */
+		tlv320aic23_write(codec, TLV320AIC23_PWR, reg | 0x0040);
+		break;
+	case SND_SOC_BIAS_OFF:
+		/* everything off, dac mute, inactive */
+		tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+		tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+#define AIC23_RATES	SNDRV_PCM_RATE_8000_96000
+#define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
+			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+struct snd_soc_dai tlv320aic23_dai = {
+	.name = "tlv320aic23",
+	.playback = {
+		     .stream_name = "Playback",
+		     .channels_min = 2,
+		     .channels_max = 2,
+		     .rates = AIC23_RATES,
+		     .formats = AIC23_FORMATS,},
+	.capture = {
+		    .stream_name = "Capture",
+		    .channels_min = 2,
+		    .channels_max = 2,
+		    .rates = AIC23_RATES,
+		    .formats = AIC23_FORMATS,},
+	.ops = {
+		.prepare = tlv320aic23_pcm_prepare,
+		.hw_params = tlv320aic23_hw_params,
+		.shutdown = tlv320aic23_shutdown,
+		},
+	.dai_ops = {
+		    .digital_mute = tlv320aic23_mute,
+		    .set_fmt = tlv320aic23_set_dai_fmt,
+		    .set_sysclk = tlv320aic23_set_dai_sysclk,
+		    }
+};
+EXPORT_SYMBOL_GPL(tlv320aic23_dai);
+
+static int tlv320aic23_suspend(struct platform_device *pdev,
+			       pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int tlv320aic23_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+	int i;
+	u16 reg;
+
+	/* Sync reg_cache with the hardware */
+	for (reg = 0; reg < ARRAY_SIZE(tlv320aic23_reg); i++) {
+		u16 val = tlv320aic23_read_reg_cache(codec, reg);
+		tlv320aic23_write(codec, reg, val);
+	}
+
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	tlv320aic23_set_bias_level(codec, codec->suspend_bias_level);
+
+	return 0;
+}
+
+/*
+ * initialise the AIC23 driver
+ * register the mixer and dsp interfaces with the kernel
+ */
+static int tlv320aic23_init(struct snd_soc_device *socdev)
+{
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret = 0;
+	u16 reg;
+
+	codec->name = "tlv320aic23";
+	codec->owner = THIS_MODULE;
+	codec->read = tlv320aic23_read_reg_cache;
+	codec->write = tlv320aic23_write;
+	codec->set_bias_level = tlv320aic23_set_bias_level;
+	codec->dai = &tlv320aic23_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = ARRAY_SIZE(tlv320aic23_reg);
+	codec->reg_cache =
+	    kmemdup(tlv320aic23_reg, sizeof(tlv320aic23_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL)
+		return -ENOMEM;
+
+	/* Reset codec */
+	tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	/* power on device */
+	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
+
+	/* Unmute input */
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_LINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  (TLV320AIC23_LRS_ENABLED));
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
+	tlv320aic23_write(codec, TLV320AIC23_RINVOL,
+			  (reg & (~TLV320AIC23_LIM_MUTED)) |
+			  TLV320AIC23_LRS_ENABLED);
+
+	reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
+	tlv320aic23_write(codec, TLV320AIC23_ANLG,
+			 (reg) & (~TLV320AIC23_BYPASS_ON) &
+			 (~TLV320AIC23_MICM_MUTED));
+
+	/* Default output volume */
+	tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+	tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
+			  TLV320AIC23_DEFAULT_OUT_VOL &
+			  TLV320AIC23_OUT_VOL_MASK);
+
+	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
+
+	tlv320aic23_add_controls(codec);
+	tlv320aic23_add_widgets(codec);
+	ret = snd_soc_register_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	kfree(codec->reg_cache);
+	return ret;
+}
+static struct snd_soc_device *tlv320aic23_socdev;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+/*
+ * If the i2c layer weren't so broken, we could pass this kind of data
+ * around
+ */
+static int tlv320aic23_codec_probe(struct i2c_client *i2c,
+				   const struct i2c_device_id *i2c_id)
+{
+	struct snd_soc_device *socdev = tlv320aic23_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
+		return -EINVAL;
+
+	i2c_set_clientdata(i2c, codec);
+	codec->control_data = i2c;
+
+	ret = tlv320aic23_init(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "tlv320aic23: failed to initialise AIC23\n");
+		goto err;
+	}
+	return ret;
+
+err:
+	kfree(codec);
+	kfree(i2c);
+	return ret;
+}
+static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
+{
+	put_device(&i2c->dev);
+	return 0;
+}
+
+static const struct i2c_device_id tlv320aic23_id[] = {
+	{"tlv320aic23", 0},
+	{}
+};
+
+MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
+
+static struct i2c_driver tlv320aic23_i2c_driver = {
+	.driver = {
+		   .name = "tlv320aic23",
+		   },
+	.probe = tlv320aic23_codec_probe,
+	.remove = __exit_p(tlv320aic23_i2c_remove),
+	.id_table = tlv320aic23_id,
+};
+
+#endif
+
+static int tlv320aic23_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
+
+	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (codec == NULL)
+		return -ENOMEM;
+
+	socdev->codec = codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	tlv320aic23_socdev = socdev;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	codec->hw_write = (hw_write_t) i2c_master_send;
+	codec->hw_read = NULL;
+	ret = i2c_add_driver(&tlv320aic23_i2c_driver);
+	if (ret != 0)
+		printk(KERN_ERR "can't add i2c driver");
+#endif
+	return ret;
+}
+
+static int tlv320aic23_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->codec;
+
+	if (codec->control_data)
+		tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&tlv320aic23_i2c_driver);
+#endif
+	kfree(codec->reg_cache);
+	kfree(codec);
+
+	return 0;
+}
+struct snd_soc_codec_device soc_codec_dev_tlv320aic23 = {
+	.probe = tlv320aic23_probe,
+	.remove = tlv320aic23_remove,
+	.suspend = tlv320aic23_suspend,
+	.resume = tlv320aic23_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320aic23);
+
+MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h
new file mode 100644
index 00000000000..79d1faf8e57
--- /dev/null
+++ b/sound/soc/codecs/tlv320aic23.h
@@ -0,0 +1,122 @@
+/*
+ * ALSA SoC TLV320AIC23 codec driver
+ *
+ * Author:      Arun KS, <arunks@mistralsolutions.com>
+ * Copyright:   (C) 2008 Mistral Solutions Pvt Ltd
+ *
+ * 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 _TLV320AIC23_H
+#define _TLV320AIC23_H
+
+/* Codec TLV320AIC23 */
+#define TLV320AIC23_LINVOL		0x00
+#define TLV320AIC23_RINVOL		0x01
+#define TLV320AIC23_LCHNVOL		0x02
+#define TLV320AIC23_RCHNVOL		0x03
+#define TLV320AIC23_ANLG		0x04
+#define TLV320AIC23_DIGT		0x05
+#define TLV320AIC23_PWR			0x06
+#define TLV320AIC23_DIGT_FMT		0x07
+#define TLV320AIC23_SRATE		0x08
+#define TLV320AIC23_ACTIVE		0x09
+#define TLV320AIC23_RESET		0x0F
+
+/* Left (right) line input volume control register */
+#define TLV320AIC23_LRS_ENABLED		0x0100
+#define TLV320AIC23_LIM_MUTED		0x0080
+#define TLV320AIC23_LIV_DEFAULT		0x0017
+#define TLV320AIC23_LIV_MAX		0x001f
+#define TLV320AIC23_LIV_MIN		0x0000
+
+/* Left (right) channel headphone volume control register */
+#define TLV320AIC23_LZC_ON		0x0080
+#define TLV320AIC23_LHV_DEFAULT		0x0079
+#define TLV320AIC23_LHV_MAX		0x007f
+#define TLV320AIC23_LHV_MIN		0x0000
+
+/* Analog audio path control register */
+#define TLV320AIC23_STA_REG(x)		((x)<<6)
+#define TLV320AIC23_STE_ENABLED		0x0020
+#define TLV320AIC23_DAC_SELECTED	0x0010
+#define TLV320AIC23_BYPASS_ON		0x0008
+#define TLV320AIC23_INSEL_MIC		0x0004
+#define TLV320AIC23_MICM_MUTED		0x0002
+#define TLV320AIC23_MICB_20DB		0x0001
+
+/* Digital audio path control register */
+#define TLV320AIC23_DACM_MUTE		0x0008
+#define TLV320AIC23_DEEMP_32K		0x0002
+#define TLV320AIC23_DEEMP_44K		0x0004
+#define TLV320AIC23_DEEMP_48K		0x0006
+#define TLV320AIC23_ADCHP_ON		0x0001
+
+/* Power control down register */
+#define TLV320AIC23_DEVICE_PWR_OFF  	0x0080
+#define TLV320AIC23_CLK_OFF		0x0040
+#define TLV320AIC23_OSC_OFF		0x0020
+#define TLV320AIC23_OUT_OFF		0x0010
+#define TLV320AIC23_DAC_OFF		0x0008
+#define TLV320AIC23_ADC_OFF		0x0004
+#define TLV320AIC23_MIC_OFF		0x0002
+#define TLV320AIC23_LINE_OFF		0x0001
+
+/* Digital audio interface register */
+#define TLV320AIC23_MS_MASTER		0x0040
+#define TLV320AIC23_LRSWAP_ON		0x0020
+#define TLV320AIC23_LRP_ON		0x0010
+#define TLV320AIC23_IWL_16		0x0000
+#define TLV320AIC23_IWL_20		0x0004
+#define TLV320AIC23_IWL_24		0x0008
+#define TLV320AIC23_IWL_32		0x000C
+#define TLV320AIC23_FOR_I2S		0x0002
+#define TLV320AIC23_FOR_DSP		0x0003
+#define TLV320AIC23_FOR_LJUST		0x0001
+
+/* Sample rate control register */
+#define TLV320AIC23_CLKOUT_HALF		0x0080
+#define TLV320AIC23_CLKIN_HALF		0x0040
+#define TLV320AIC23_BOSR_384fs		0x0002	/* BOSR_272fs in USB mode */
+#define TLV320AIC23_USB_CLK_ON		0x0001
+#define TLV320AIC23_SR_MASK             0xf
+#define TLV320AIC23_CLKOUT_SHIFT        7
+#define TLV320AIC23_CLKIN_SHIFT         6
+#define TLV320AIC23_SR_SHIFT            2
+#define TLV320AIC23_BOSR_SHIFT          1
+
+/* Digital interface register */
+#define TLV320AIC23_ACT_ON		0x0001
+
+/*
+ * AUDIO related MACROS
+ */
+
+#define TLV320AIC23_DEFAULT_OUT_VOL	0x70
+#define TLV320AIC23_DEFAULT_IN_VOLUME	0x10
+
+#define TLV320AIC23_OUT_VOL_MIN		TLV320AIC23_LHV_MIN
+#define TLV320AIC23_OUT_VOL_MAX		TLV320AIC23_LHV_MAX
+#define TLV320AIC23_OUT_VO_RANGE	(TLV320AIC23_OUT_VOL_MAX - \
+					TLV320AIC23_OUT_VOL_MIN)
+#define TLV320AIC23_OUT_VOL_MASK	TLV320AIC23_OUT_VOL_MAX
+
+#define TLV320AIC23_IN_VOL_MIN		TLV320AIC23_LIV_MIN
+#define TLV320AIC23_IN_VOL_MAX		TLV320AIC23_LIV_MAX
+#define TLV320AIC23_IN_VOL_RANGE	(TLV320AIC23_IN_VOL_MAX - \
+					TLV320AIC23_IN_VOL_MIN)
+#define TLV320AIC23_IN_VOL_MASK		TLV320AIC23_IN_VOL_MAX
+
+#define TLV320AIC23_SIDETONE_MASK	0x1c0
+#define TLV320AIC23_SIDETONE_0		0x100
+#define TLV320AIC23_SIDETONE_6		0x000
+#define TLV320AIC23_SIDETONE_9		0x040
+#define TLV320AIC23_SIDETONE_12		0x080
+#define TLV320AIC23_SIDETONE_18		0x0c0
+
+extern struct snd_soc_dai tlv320aic23_dai;
+extern struct snd_soc_codec_device soc_codec_dev_tlv320aic23;
+
+#endif /* _TLV320AIC23_H */
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 566a427c928..cff276ee261 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -48,7 +48,6 @@
 
 #include "tlv320aic3x.h"
 
-#define AUDIO_NAME "aic3x"
 #define AIC3X_VERSION "0.2"
 
 /* codec private data */
@@ -864,17 +863,21 @@ static int aic3x_set_dai_fmt(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	/* interface format */
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_I2S:
+	/*
+	 * match both interface format and signal polarities since they
+	 * are fixed
+	 */
+	switch (fmt & (SND_SOC_DAIFMT_FORMAT_MASK |
+		       SND_SOC_DAIFMT_INV_MASK)) {
+	case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF):
 		break;
-	case SND_SOC_DAIFMT_DSP_A:
+	case (SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF):
 		iface_breg |= (0x01 << 6);
 		break;
-	case SND_SOC_DAIFMT_RIGHT_J:
+	case (SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x02 << 6);
 		break;
-	case SND_SOC_DAIFMT_LEFT_J:
+	case (SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF):
 		iface_breg |= (0x03 << 6);
 		break;
 	default:
@@ -991,7 +994,7 @@ EXPORT_SYMBOL_GPL(aic3x_headset_detected);
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
 struct snd_soc_dai aic3x_dai = {
-	.name = "aic3x",
+	.name = "tlv320aic3x",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
@@ -1055,7 +1058,7 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
-	codec->name = "aic3x";
+	codec->name = "tlv320aic3x";
 	codec->owner = THIS_MODULE;
 	codec->read = aic3x_read_reg_cache;
 	codec->write = aic3x_write;
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index d206d7f892b..a69ee72a7af 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -36,7 +36,6 @@
 #include "uda1380.h"
 
 #define UDA1380_VERSION "0.6"
-#define AUDIO_NAME "uda1380"
 
 /*
  * uda1380 register cache
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 9a37c8d95ed..d8ca2da8d63 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -3,7 +3,7 @@
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
  *
- * Author: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  * This program is free software; you can redistribute it and/or modify
  * it under the terms of the GNU General Public License version 2 as
@@ -18,6 +18,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -27,7 +28,6 @@
 
 #include "wm8510.h"
 
-#define AUDIO_NAME "wm8510"
 #define WM8510_VERSION "0.6"
 
 struct snd_soc_codec_device soc_codec_dev_wm8510;
@@ -55,6 +55,9 @@ static const u16 wm8510_reg[WM8510_CACHEREGNUM] = {
 	0x0001,
 };
 
+#define WM8510_POWER1_BIASEN  0x08
+#define WM8510_POWER1_BUFIOEN 0x10
+
 /*
  * read wm8510 register cache
  */
@@ -224,9 +227,9 @@ SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0),
 SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0),
 SND_SOC_DAPM_PGA("Mono Out", WM8510_POWER3, 7, 0, NULL, 0),
 
-SND_SOC_DAPM_PGA("Mic PGA", WM8510_POWER2, 2, 0,
-		 &wm8510_micpga_controls[0],
-		 ARRAY_SIZE(wm8510_micpga_controls)),
+SND_SOC_DAPM_MIXER("Mic PGA", WM8510_POWER2, 2, 0,
+		   &wm8510_micpga_controls[0],
+		   ARRAY_SIZE(wm8510_micpga_controls)),
 SND_SOC_DAPM_MIXER("Boost Mixer", WM8510_POWER2, 4, 0,
 	&wm8510_boost_controls[0],
 	ARRAY_SIZE(wm8510_boost_controls)),
@@ -526,23 +529,35 @@ static int wm8510_mute(struct snd_soc_dai *dai, int mute)
 static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
+	u16 power1 = wm8510_read_reg_cache(codec, WM8510_POWER1) & ~0x3;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		wm8510_write(codec, WM8510_POWER1, 0x1ff);
-		wm8510_write(codec, WM8510_POWER2, 0x1ff);
-		wm8510_write(codec, WM8510_POWER3, 0x1ff);
-		break;
 	case SND_SOC_BIAS_PREPARE:
+		power1 |= 0x1;  /* VMID 50k */
+		wm8510_write(codec, WM8510_POWER1, power1);
+		break;
+
 	case SND_SOC_BIAS_STANDBY:
+		power1 |= WM8510_POWER1_BIASEN | WM8510_POWER1_BUFIOEN;
+
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Initial cap charge at VMID 5k */
+			wm8510_write(codec, WM8510_POWER1, power1 | 0x3);
+			mdelay(100);
+		}
+
+		power1 |= 0x2;  /* VMID 500k */
+		wm8510_write(codec, WM8510_POWER1, power1);
 		break;
+
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
-		wm8510_write(codec, WM8510_POWER1, 0x0);
-		wm8510_write(codec, WM8510_POWER2, 0x0);
-		wm8510_write(codec, WM8510_POWER3, 0x0);
+		wm8510_write(codec, WM8510_POWER1, 0);
+		wm8510_write(codec, WM8510_POWER2, 0);
+		wm8510_write(codec, WM8510_POWER3, 0);
 		break;
 	}
+
 	codec->bias_level = level;
 	return 0;
 }
@@ -640,6 +655,7 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	}
 
 	/* power on device */
+	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	wm8510_add_controls(codec);
 	wm8510_add_widgets(codec);
@@ -747,6 +763,62 @@ err_driver:
 }
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8510_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8510_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8510_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8510\n");
+
+	return ret;
+}
+
+static int __devexit wm8510_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8510_spi_driver = {
+	.driver = {
+		.name	= "wm8510",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8510_spi_probe,
+	.remove		= __devexit_p(wm8510_spi_remove),
+};
+
+static int wm8510_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif /* CONFIG_SPI_MASTER */
+
 static int wm8510_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -772,8 +844,14 @@ static int wm8510_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8510_add_i2c_device(pdev, setup);
 	}
-#else
-	/* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8510_spi_write;
+		ret = spi_register_driver(&wm8510_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
 #endif
 
 	if (ret != 0)
@@ -796,6 +874,9 @@ static int wm8510_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8510_i2c_driver);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8510_spi_driver);
+#endif
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/wm8510.h b/sound/soc/codecs/wm8510.h
index c5368396045..bdefcf5c69f 100644
--- a/sound/soc/codecs/wm8510.h
+++ b/sound/soc/codecs/wm8510.h
@@ -94,6 +94,7 @@
 #define WM8510_MCLKDIV_12	(7 << 5)
 
 struct wm8510_setup_data {
+	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index df1ffbe305b..627ebfb4209 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -18,7 +18,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
@@ -36,7 +35,6 @@
 
 #include "wm8580.h"
 
-#define AUDIO_NAME "wm8580"
 #define WM8580_VERSION "0.1"
 
 struct pll_state {
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 7b64d9a7ff7..7f8a7e36b33 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,7 +29,6 @@
 
 #include "wm8731.h"
 
-#define AUDIO_NAME "wm8731"
 #define WM8731_VERSION "0.13"
 
 struct snd_soc_codec_device soc_codec_dev_wm8731;
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 4892e398a59..9b7296ee5b0 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -29,7 +29,6 @@
 
 #include "wm8750.h"
 
-#define AUDIO_NAME "WM8750"
 #define WM8750_VERSION "0.12"
 
 /* codec private data */
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 8c4df44f334..d426eaa2218 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -2,8 +2,7 @@
  * wm8753.c  --  WM8753 ALSA Soc Audio driver
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -40,6 +39,7 @@
 #include <linux/pm.h>
 #include <linux/i2c.h>
 #include <linux/platform_device.h>
+#include <linux/spi/spi.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/pcm_params.h>
@@ -51,7 +51,6 @@
 
 #include "wm8753.h"
 
-#define AUDIO_NAME "wm8753"
 #define WM8753_VERSION "0.16"
 
 static int caps_charge = 2000;
@@ -1719,6 +1718,63 @@ err_driver:
 }
 #endif
 
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_device *socdev = wm8753_socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	int ret;
+
+	codec->control_data = spi;
+
+	ret = wm8753_init(socdev);
+	if (ret < 0)
+		dev_err(&spi->dev, "failed to initialise WM8753\n");
+
+	return ret;
+}
+
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
+{
+	return 0;
+}
+
+static struct spi_driver wm8753_spi_driver = {
+	.driver = {
+		.name	= "wm8753",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8753_spi_probe,
+	.remove		= __devexit_p(wm8753_spi_remove),
+};
+
+static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
+{
+	struct spi_transfer t;
+	struct spi_message m;
+	u8 msg[2];
+
+	if (len <= 0)
+		return 0;
+
+	msg[0] = data[0];
+	msg[1] = data[1];
+
+	spi_message_init(&m);
+	memset(&t, 0, (sizeof t));
+
+	t.tx_buf = &msg[0];
+	t.len = len;
+
+	spi_message_add_tail(&t, &m);
+	spi_sync(spi, &m);
+
+	return len;
+}
+#endif
+
+
 static int wm8753_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -1753,8 +1809,14 @@ static int wm8753_probe(struct platform_device *pdev)
 		codec->hw_write = (hw_write_t)i2c_master_send;
 		ret = wm8753_add_i2c_device(pdev, setup);
 	}
-#else
-		/* Add other interfaces here */
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	if (setup->spi) {
+		codec->hw_write = (hw_write_t)wm8753_spi_write;
+		ret = spi_register_driver(&wm8753_spi_driver);
+		if (ret != 0)
+			printk(KERN_ERR "can't add spi driver");
+	}
 #endif
 
 	if (ret != 0) {
@@ -1798,6 +1860,9 @@ static int wm8753_remove(struct platform_device *pdev)
 	i2c_unregister_device(codec->control_data);
 	i2c_del_driver(&wm8753_i2c_driver);
 #endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8753_spi_driver);
+#endif
 	kfree(codec->private_data);
 	kfree(codec);
 
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index 7defde069f1..f55704ce931 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -2,8 +2,7 @@
  * wm8753.h  --  audio driver for WM8753
  *
  * Copyright 2003 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -79,6 +78,7 @@
 #define WM8753_ADCTL2		0x3f
 
 struct wm8753_setup_data {
+	int spi;
 	int i2c_bus;
 	unsigned short i2c_address;
 };
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 0b8c6d38b48..3b326c9b558 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -18,7 +18,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/init.h>
 #include <linux/delay.h>
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index a3f54ec4226..ce40d787760 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -653,14 +653,14 @@ static const struct snd_kcontrol_new wm8903_snd_controls[] = {
 
 /* Input PGAs - No TLV since the scale depends on PGA mode */
 SOC_SINGLE("Left Input PGA Switch", WM8903_ANALOGUE_LEFT_INPUT_0,
-	   7, 1, 0),
+	   7, 1, 1),
 SOC_SINGLE("Left Input PGA Volume", WM8903_ANALOGUE_LEFT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Left Input PGA Common Mode Switch", WM8903_ANALOGUE_LEFT_INPUT_1,
 	   6, 1, 0),
 
 SOC_SINGLE("Right Input PGA Switch", WM8903_ANALOGUE_RIGHT_INPUT_0,
-	   7, 1, 0),
+	   7, 1, 1),
 SOC_SINGLE("Right Input PGA Volume", WM8903_ANALOGUE_RIGHT_INPUT_0,
 	   0, 31, 0),
 SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 974a4cd0f3f..f41a578ddd4 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -29,7 +29,6 @@
 
 #include "wm8971.h"
 
-#define AUDIO_NAME "wm8971"
 #define WM8971_VERSION "0.9"
 
 #define	WM8971_REG_COUNT		43
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 63410d7b5ef..572d22b0880 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -30,7 +30,6 @@
 
 #include "wm8990.h"
 
-#define AUDIO_NAME "wm8990"
 #define WM8990_VERSION "0.2"
 
 /* codec private data */
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index 2f1c91b1d55..ffb471e420e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -2,8 +2,7 @@
  * wm9712.c  --  ALSA Soc WM9712 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index 441d0580db1..945b32ed988 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -2,8 +2,7 @@
  * wm9713.c  --  ALSA Soc WM9713 codec support
  *
  * Copyright 2006 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -141,7 +140,7 @@ SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
 
 SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
 SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
-SOC_SINGLE("ALC Decay Time ", AC97_CODEC_CLASS_REV, 4, 15, 0),
+SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
 SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
 SOC_ENUM("ALC Function", wm9713_enum[6]),
 SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),