Merge branch 'for-2.6.40' into for-2.6.41

45 files changed, 1304 insertions, 252 deletions

diff --git a/MAINTAINERS b/MAINTAINERS
index 9f926c0229d..75318bfb165 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -5840,7 +5840,7 @@ F:	include/sound/
 F:	sound/
 
 SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
-M:	Liam Girdwood <lrg@slimlogic.co.uk>
+M:	Liam Girdwood <lrg@ti.com>
 M:	Mark Brown <broonie@opensource.wolfsonmicro.com>
 T:	git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
@@ -6093,7 +6093,7 @@ F:	drivers/mmc/host/tifm_sd.c
 F:	include/linux/tifm.h
 
 TI TWL4030 SERIES SOC CODEC DRIVER
-M:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+M:	Peter Ujfalusi <peter.ujfalusi@ti.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
 F:	sound/soc/codecs/twl4030*
@@ -6736,7 +6736,7 @@ F:	drivers/scsi/vmw_pvscsi.c
 F:	drivers/scsi/vmw_pvscsi.h
 
 VOLTAGE AND CURRENT REGULATOR FRAMEWORK
-M:	Liam Girdwood <lrg@slimlogic.co.uk>
+M:	Liam Girdwood <lrg@ti.com>
 M:	Mark Brown <broonie@opensource.wolfsonmicro.com>
 W:	http://opensource.wolfsonmicro.com/node/15
 W:	http://www.slimlogic.co.uk/?p=48
diff --git a/include/sound/ak4641.h b/include/sound/ak4641.h
new file mode 100644
index 00000000000..96d1991c811
--- /dev/null
+++ b/include/sound/ak4641.h
@@ -0,0 +1,26 @@
+/*
+ * AK4641 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * 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 __AK4641_H
+#define __AK4641_H
+
+/**
+ * struct ak4641_platform_data - platform specific AK4641 configuration
+ * @gpio_power:	GPIO to control external power to AK4641
+ * @gpio_npdn:	GPIO connected to AK4641 nPDN pin
+ *
+ * Both GPIO parameters are optional.
+ */
+struct ak4641_platform_data {
+	int gpio_power;
+	int gpio_npdn;
+};
+
+#endif /* __AK4641_H */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index b27c7a2d3bb..f1de3e0c75b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -702,6 +702,8 @@ struct snd_soc_aux_dev {
 /* SoC card */
 struct snd_soc_card {
 	const char *name;
+	const char *long_name;
+	const char *driver_name;
 	struct device *dev;
 	struct snd_card *snd_card;
 	struct module *owner;
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
index 6c664965679..0b94192a8cd 100644
--- a/include/sound/tlv320dac33-plat.h
+++ b/include/sound/tlv320dac33-plat.h
@@ -1,7 +1,7 @@
 /*
  * Platform header for Texas Instruments TLV320DAC33 codec driver
  *
- * Author:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * Copyright:   (C) 2009 Nokia Corporation
  *
diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h
index e29fde6b5cb..89beccb57ed 100644
--- a/include/sound/tpa6130a2-plat.h
+++ b/include/sound/tpa6130a2-plat.h
@@ -3,7 +3,7 @@
  *
  * Copyright (C) Nokia Corporation
  *
- * Written by Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 28afbbf69ce..95572d290c2 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -146,7 +146,7 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd)
 			"at91sam9g20ek_wm8731 "
 			": at91sam9g20ek_wm8731_init() called\n");
 
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_MCLK,
 		MCLK_RATE, SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret);
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2a6971891d3..98175a096df 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -20,6 +20,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
+	select SND_SOC_AK4641 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
 	select SND_SOC_ALC5623 if I2C
@@ -139,6 +140,9 @@ config SND_SOC_AK4104
 config SND_SOC_AK4535
 	tristate
 
+config SND_SOC_AK4641
+	tristate
+
 config SND_SOC_AK4642
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4cb2f42dbff..fd8558406ef 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -7,6 +7,7 @@ snd-soc-ad73311-objs := ad73311.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
+snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
 snd-soc-cq93vc-objs := cq93vc.o
@@ -97,6 +98,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
+obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c
new file mode 100644
index 00000000000..ed96f247c2d
--- /dev/null
+++ b/sound/soc/codecs/ak4641.c
@@ -0,0 +1,664 @@
+/*
+ * ak4641.c  --  AK4641 ALSA Soc Audio driver
+ *
+ * Copyright (C) 2008 Harald Welte <laforge@gnufiish.org>
+ * Copyright (C) 2011 Dmitry Artamonow <mad_soft@inbox.ru>
+ *
+ * Based on ak4535.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.
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/ak4641.h>
+
+#include "ak4641.h"
+
+/* codec private data */
+struct ak4641_priv {
+	struct snd_soc_codec *codec;
+	unsigned int sysclk;
+	int deemph;
+	int playback_fs;
+};
+
+/*
+ * ak4641 register cache
+ */
+static const u8 ak4641_reg[AK4641_CACHEREGNUM] = {
+	0x00, 0x80, 0x00, 0x80,
+	0x02, 0x00, 0x11, 0x05,
+	0x00, 0x00, 0x36, 0x10,
+	0x00, 0x00, 0x57, 0x00,
+	0x88, 0x88, 0x08, 0x08
+};
+
+static const int deemph_settings[] = {44100, 0, 48000, 32000};
+
+static int ak4641_set_deemph(struct snd_soc_codec *codec)
+{
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+	int i, best = 0;
+
+	for (i = 0 ; i < ARRAY_SIZE(deemph_settings); i++) {
+		/* if deemphasis is on, select the nearest available rate */
+		if (ak4641->deemph && deemph_settings[i] != 0 &&
+		    abs(deemph_settings[i] - ak4641->playback_fs) <
+		    abs(deemph_settings[best] - ak4641->playback_fs))
+			best = i;
+
+		if (!ak4641->deemph && deemph_settings[i] == 0)
+			best = i;
+	}
+
+	dev_dbg(codec->dev, "Set deemphasis %d\n", best);
+
+	return snd_soc_update_bits(codec, AK4641_DAC, 0x3, best);
+}
+
+static int ak4641_put_deemph(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+	int deemph = ucontrol->value.enumerated.item[0];
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	ak4641->deemph = deemph;
+
+	return ak4641_set_deemph(codec);
+}
+
+static int ak4641_get_deemph(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = ak4641->deemph;
+	return 0;
+};
+
+static const char *ak4641_mono_out[] = {"(L + R)/2", "Hi-Z"};
+static const char *ak4641_hp_out[] = {"Stereo", "Mono"};
+static const char *ak4641_mic_select[] = {"Internal", "External"};
+static const char *ak4641_mic_or_dac[] = {"Microphone", "Voice DAC"};
+
+
+static const DECLARE_TLV_DB_SCALE(mono_gain_tlv, -1700, 2300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, 0, 2000, 0);
+static const DECLARE_TLV_DB_SCALE(eq_tlv, -1050, 150, 0);
+static const DECLARE_TLV_DB_SCALE(master_tlv, -12750, 50, 0);
+static const DECLARE_TLV_DB_SCALE(mic_stereo_sidetone_tlv, -2700, 300, 0);
+static const DECLARE_TLV_DB_SCALE(mic_mono_sidetone_tlv, -400, 400, 0);
+static const DECLARE_TLV_DB_SCALE(capture_tlv, -800, 50, 0);
+static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0);
+static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0);
+
+
+static const struct soc_enum ak4641_mono_out_enum =
+	SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out);
+static const struct soc_enum ak4641_hp_out_enum =
+	SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out);
+static const struct soc_enum ak4641_mic_select_enum =
+	SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select);
+static const struct soc_enum ak4641_mic_or_dac_enum =
+	SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac);
+
+static const struct snd_kcontrol_new ak4641_snd_controls[] = {
+	SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum),
+	SOC_SINGLE_TLV("Mono 1 Gain Volume", AK4641_SIG1, 7, 1, 1,
+							mono_gain_tlv),
+	SOC_ENUM("Headphone Output", ak4641_hp_out_enum),
+	SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0,
+					ak4641_get_deemph, ak4641_put_deemph),
+
+	SOC_SINGLE_TLV("Mic Boost Volume", AK4641_MIC, 0, 1, 0, mic_boost_tlv),
+
+	SOC_SINGLE("ALC Operation Time", AK4641_TIMER, 0, 3, 0),
+	SOC_SINGLE("ALC Recovery Time", AK4641_TIMER, 2, 3, 0),
+	SOC_SINGLE("ALC ZC Time", AK4641_TIMER, 4, 3, 0),
+
+	SOC_SINGLE("ALC 1 Switch", AK4641_ALC1, 5, 1, 0),
+
+	SOC_SINGLE_TLV("ALC Volume", AK4641_ALC2, 0, 71, 0, alc_tlv),
+	SOC_SINGLE("Left Out Enable Switch", AK4641_SIG2, 1, 1, 0),
+	SOC_SINGLE("Right Out Enable Switch", AK4641_SIG2, 0, 1, 0),
+
+	SOC_SINGLE_TLV("Capture Volume", AK4641_PGA, 0, 71, 0, capture_tlv),
+
+	SOC_DOUBLE_R_TLV("Master Playback Volume", AK4641_LATT,
+				AK4641_RATT, 0, 255, 1, master_tlv),
+
+	SOC_SINGLE_TLV("AUX In Volume", AK4641_VOL, 0, 15, 0, aux_in_tlv),
+
+	SOC_SINGLE("Equalizer Switch", AK4641_DAC, 2, 1, 0),
+	SOC_SINGLE_TLV("EQ1 100 Hz Volume", AK4641_EQLO, 0, 15, 1, eq_tlv),
+	SOC_SINGLE_TLV("EQ2 250 Hz Volume", AK4641_EQLO, 4, 15, 1, eq_tlv),
+	SOC_SINGLE_TLV("EQ3 1 kHz Volume", AK4641_EQMID, 0, 15, 1, eq_tlv),
+	SOC_SINGLE_TLV("EQ4 3.5 kHz Volume", AK4641_EQMID, 4, 15, 1, eq_tlv),
+	SOC_SINGLE_TLV("EQ5 10 kHz Volume", AK4641_EQHI, 0, 15, 1, eq_tlv),
+};
+
+/* Mono 1 Mixer */
+static const struct snd_kcontrol_new ak4641_mono1_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("Mic Mono Sidetone Volume", AK4641_VOL, 7, 1, 0,
+						mic_mono_sidetone_tlv),
+	SOC_DAPM_SINGLE("Mic Mono Sidetone Switch", AK4641_SIG1, 4, 1, 0),
+	SOC_DAPM_SINGLE("Mono Playback Switch", AK4641_SIG1, 5, 1, 0),
+};
+
+/* Stereo Mixer */
+static const struct snd_kcontrol_new ak4641_stereo_mixer_controls[] = {
+	SOC_DAPM_SINGLE_TLV("Mic Sidetone Volume", AK4641_VOL, 4, 7, 0,
+						mic_stereo_sidetone_tlv),
+	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4641_SIG2, 4, 1, 0),
+	SOC_DAPM_SINGLE("Playback Switch", AK4641_SIG2, 7, 1, 0),
+	SOC_DAPM_SINGLE("Aux Bypass Switch", AK4641_SIG2, 5, 1, 0),
+};
+
+/* Input Mixer */
+static const struct snd_kcontrol_new ak4641_input_mixer_controls[] = {
+	SOC_DAPM_SINGLE("Mic Capture Switch", AK4641_MIC, 2, 1, 0),
+	SOC_DAPM_SINGLE("Aux Capture Switch", AK4641_MIC, 5, 1, 0),
+};
+
+/* Mic mux */
+static const struct snd_kcontrol_new ak4641_mic_mux_control =
+	SOC_DAPM_ENUM("Mic Select", ak4641_mic_select_enum);
+
+/* Input mux */
+static const struct snd_kcontrol_new ak4641_input_mux_control =
+	SOC_DAPM_ENUM("Input Select", ak4641_mic_or_dac_enum);
+
+/* mono 2 switch */
+static const struct snd_kcontrol_new ak4641_mono2_control =
+	SOC_DAPM_SINGLE("Switch", AK4641_SIG1, 0, 1, 0);
+
+/* ak4641 dapm widgets */
+static const struct snd_soc_dapm_widget ak4641_dapm_widgets[] = {
+	SND_SOC_DAPM_MIXER("Stereo Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4641_stereo_mixer_controls[0],
+		ARRAY_SIZE(ak4641_stereo_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono1 Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4641_mono1_mixer_controls[0],
+		ARRAY_SIZE(ak4641_mono1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Input Mixer", SND_SOC_NOPM, 0, 0,
+		&ak4641_input_mixer_controls[0],
+		ARRAY_SIZE(ak4641_input_mixer_controls)),
+	SND_SOC_DAPM_MUX("Mic Mux", SND_SOC_NOPM, 0, 0,
+		&ak4641_mic_mux_control),
+	SND_SOC_DAPM_MUX("Input Mux", SND_SOC_NOPM, 0, 0,
+		&ak4641_input_mux_control),
+	SND_SOC_DAPM_SWITCH("Mono 2 Enable", SND_SOC_NOPM, 0, 0,
+		&ak4641_mono2_control),
+
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("MOUT1"),
+	SND_SOC_DAPM_OUTPUT("MOUT2"),
+	SND_SOC_DAPM_OUTPUT("MICOUT"),
+
+	SND_SOC_DAPM_ADC("ADC", "HiFi Capture", AK4641_PM1, 0, 0),
+	SND_SOC_DAPM_PGA("Mic", AK4641_PM1, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX In", AK4641_PM1, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono Out", AK4641_PM1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line Out", AK4641_PM1, 4, 0, NULL, 0),
+
+	SND_SOC_DAPM_DAC("DAC", "HiFi Playback", AK4641_PM2, 0, 0),
+	SND_SOC_DAPM_PGA("Mono Out 2", AK4641_PM2, 3, 0, NULL, 0),
+
+	SND_SOC_DAPM_ADC("Voice ADC", "Voice Capture", AK4641_BTIF, 0, 0),
+	SND_SOC_DAPM_ADC("Voice DAC", "Voice Playback", AK4641_BTIF, 1, 0),
+
+	SND_SOC_DAPM_MICBIAS("Mic Int Bias", AK4641_MIC, 3, 0),
+	SND_SOC_DAPM_MICBIAS("Mic Ext Bias", AK4641_MIC, 4, 0),
+
+	SND_SOC_DAPM_INPUT("MICIN"),
+	SND_SOC_DAPM_INPUT("MICEXT"),
+	SND_SOC_DAPM_INPUT("AUX"),
+	SND_SOC_DAPM_INPUT("AIN"),
+};
+
+static const struct snd_soc_dapm_route ak4641_audio_map[] = {
+	/* Stereo Mixer */
+	{"Stereo Mixer", "Playback Switch", "DAC"},
+	{"Stereo Mixer", "Mic Sidetone Switch", "Input Mux"},
+	{"Stereo Mixer", "Aux Bypass Switch", "AUX In"},
+
+	/* Mono 1 Mixer */
+	{"Mono1 Mixer", "Mic Mono Sidetone Switch", "Input Mux"},
+	{"Mono1 Mixer", "Mono Playback Switch", "DAC"},
+
+	/* Mic */
+	{"Mic", NULL, "AIN"},
+	{"Mic Mux", "Internal", "Mic Int Bias"},
+	{"Mic Mux", "External", "Mic Ext Bias"},
+	{"Mic Int Bias", NULL, "MICIN"},
+	{"Mic Ext Bias", NULL, "MICEXT"},
+	{"MICOUT", NULL, "Mic Mux"},
+
+	/* Input Mux */
+	{"Input Mux", "Microphone", "Mic"},
+	{"Input Mux", "Voice DAC", "Voice DAC"},
+
+	/* Line Out */
+	{"LOUT", NULL, "Line Out"},
+	{"ROUT", NULL, "Line Out"},
+	{"Line Out", NULL, "Stereo Mixer"},
+
+	/* Mono 1 Out */
+	{"MOUT1", NULL, "Mono Out"},
+	{"Mono Out", NULL, "Mono1 Mixer"},
+
+	/* Mono 2 Out */
+	{"MOUT2", NULL, "Mono 2 Enable"},
+	{"Mono 2 Enable", "Switch", "Mono Out 2"},
+	{"Mono Out 2", NULL, "Stereo Mixer"},
+
+	{"Voice ADC", NULL, "Mono 2 Enable"},
+
+	/* Aux In */
+	{"AUX In", NULL, "AUX"},
+
+	/* ADC */
+	{"ADC", NULL, "Input Mixer"},
+	{"Input Mixer", "Mic Capture Switch", "Mic"},
+	{"Input Mixer", "Aux Capture Switch", "AUX In"},
+};
+
+static int ak4641_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+	int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+
+	ak4641->sysclk = freq;
+	return 0;
+}
+
+static int ak4641_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+	struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec);
+	int rate = params_rate(params), fs = 256;
+	u8 mode2;
+
+	if (rate)
+		fs = ak4641->sysclk / rate;
+	else
+		return -EINVAL;
+
+	/* set fs */
+	switch (fs) {
+	case 1024:
+		mode2 = (0x2 << 5);
+		break;
+	case 512:
+		mode2 = (0x1 << 5);
+		break;
+	case 256:
+		mode2 = (0x0 << 5);
+		break;
+	default:
+		dev_err(codec->dev, "Error: unsupported fs=%d\n", fs);
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, AK4641_MODE2, (0x3 << 5), mode2);
+
+	/* Update de-emphasis filter for the new rate */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		ak4641->playback_fs = rate;
+		ak4641_set_deemph(codec);
+	};
+
+	return 0;
+}
+
+static int ak4641_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				  unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 btif;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		btif = (0x3 << 5);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		btif = (0x2 << 5);
+		break;
+	case SND_SOC_DAIFMT_DSP_A:	/* MSB after FRM */
+		btif = (0x0 << 5);
+		break;
+	case SND_SOC_DAIFMT_DSP_B:	/* MSB during FRM */
+		btif = (0x1 << 5);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_update_bits(codec, AK4641_BTIF, (0x3 << 5), btif);
+}
+
+static int ak4641_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u8 mode1 = 0;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		mode1 = 0x02;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		mode1 = 0x01;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return snd_soc_write(codec, AK4641_MODE1, mode1);
+}
+
+static int ak4641_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+
+	return snd_soc_update_bits(codec, AK4641_DAC, 0x20, mute ? 0x20 : 0);
+}
+
+static int ak4641_set_bias_level(struct snd_soc_codec *codec,
+	enum snd_soc_bias_level level)
+{
+	struct ak4641_platform_data *pdata = codec->dev->platform_data;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		/* unmute */
+		snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0);
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		/* mute */
+		snd_soc_update_bits(codec, AK4641_DAC, 0x20, 0x20);
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) {
+			if (pdata && gpio_is_valid(pdata->gpio_power))
+				gpio_set_value(pdata->gpio_power, 1);
+			mdelay(1);
+			if (pdata && gpio_is_valid(pdata->gpio_npdn))
+				gpio_set_value(pdata->gpio_npdn, 1);
+			mdelay(1);
+
+			ret = snd_soc_cache_sync(codec);
+			if (ret) {
+				dev_err(codec->dev,
+					"Failed to sync cache: %d\n", ret);
+				return ret;
+			}
+		}
+		snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0x80);
+		snd_soc_update_bits(codec, AK4641_PM2, 0x80, 0);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, AK4641_PM1, 0x80, 0);
+		if (pdata && gpio_is_valid(pdata->gpio_npdn))
+			gpio_set_value(pdata->gpio_npdn, 0);
+		if (pdata && gpio_is_valid(pdata->gpio_power))
+			gpio_set_value(pdata->gpio_power, 0);
+		codec->cache_sync = 1;
+		break;
+	}
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+#define AK4641_RATES	(SNDRV_PCM_RATE_8000_48000)
+#define AK4641_RATES_BT (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
+			 SNDRV_PCM_RATE_16000)
+#define AK4641_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE)
+
+static struct snd_soc_dai_ops ak4641_i2s_dai_ops = {
+	.hw_params    = ak4641_i2s_hw_params,
+	.set_fmt      = ak4641_i2s_set_dai_fmt,
+	.digital_mute = ak4641_mute,
+	.set_sysclk   = ak4641_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops ak4641_pcm_dai_ops = {
+	.hw_params    = NULL, /* rates are controlled by BT chip */
+	.set_fmt      = ak4641_pcm_set_dai_fmt,
+	.digital_mute = ak4641_mute,
+	.set_sysclk   = ak4641_set_dai_sysclk,
+};
+
+struct snd_soc_dai_driver ak4641_dai[] = {
+{
+	.name = "ak4641-hifi",
+	.id = 1,
+	.playback = {
+		.stream_name = "HiFi Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4641_RATES,
+		.formats = AK4641_FORMATS,
+	},
+	.capture = {
+		.stream_name = "HiFi Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = AK4641_RATES,
+		.formats = AK4641_FORMATS,
+	},
+	.ops = &ak4641_i2s_dai_ops,
+	.symmetric_rates = 1,
+},
+{
+	.name = "ak4641-voice",
+	.id = 1,
+	.playback = {
+		.stream_name = "Voice Playback",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = AK4641_RATES_BT,
+		.formats = AK4641_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Voice Capture",
+		.channels_min = 1,
+		.channels_max = 1,
+		.rates = AK4641_RATES_BT,
+		.formats = AK4641_FORMATS,
+	},
+	.ops = &ak4641_pcm_dai_ops,
+	.symmetric_rates = 1,
+},
+};
+
+static int ak4641_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static int ak4641_resume(struct snd_soc_codec *codec)
+{
+	ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	return 0;
+}
+
+static int ak4641_probe(struct snd_soc_codec *codec)
+{
+	struct ak4641_platform_data *pdata = codec->dev->platform_data;
+	int ret;
+
+
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power)) {
+			ret = gpio_request_one(pdata->gpio_power,
+					GPIOF_OUT_INIT_LOW, "ak4641 power");
+			if (ret)
+				goto err_out;
+		}
+		if (gpio_is_valid(pdata->gpio_npdn)) {
+			ret = gpio_request_one(pdata->gpio_npdn,
+					GPIOF_OUT_INIT_LOW, "ak4641 npdn");
+			if (ret)
+				goto err_gpio;
+
+			udelay(1); /* > 150 ns */
+			gpio_set_value(pdata->gpio_npdn, 1);
+		}
+	}
+
+	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
+		goto err_register;
+	}
+
+	/* power on device */
+	ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+
+err_register:
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power))
+			gpio_set_value(pdata->gpio_power, 0);
+		if (gpio_is_valid(pdata->gpio_npdn))
+			gpio_free(pdata->gpio_npdn);
+	}
+err_gpio:
+	if (pdata && gpio_is_valid(pdata->gpio_power))
+		gpio_free(pdata->gpio_power);
+err_out:
+	return ret;
+}
+
+static int ak4641_remove(struct snd_soc_codec *codec)
+{
+	struct ak4641_platform_data *pdata = codec->dev->platform_data;
+
+	ak4641_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	if (pdata) {
+		if (gpio_is_valid(pdata->gpio_power)) {
+			gpio_set_value(pdata->gpio_power, 0);
+			gpio_free(pdata->gpio_power);
+		}
+		if (gpio_is_valid(pdata->gpio_npdn))
+			gpio_free(pdata->gpio_npdn);
+	}
+	return 0;
+}
+
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4641 = {
+	.probe			= ak4641_probe,
+	.remove			= ak4641_remove,
+	.suspend		= ak4641_suspend,
+	.resume			= ak4641