Merge branch 'for-2.6.40' into for-2.6.41

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

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 */

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;

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
  *

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

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);

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
 

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

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_resume,
+	.controls		= ak4641_snd_controls,
+	.num_controls		= ARRAY_SIZE(ak4641_snd_controls),
+	.dapm_widgets		= ak4641_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(ak4641_dapm_widgets),
+	.dapm_routes		= ak4641_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(ak4641_audio_map),
+	.set_bias_level		= ak4641_set_bias_level,
+	.reg_cache_size		= ARRAY_SIZE(ak4641_reg),
+	.reg_word_size		= sizeof(u8),
+	.reg_cache_default	= ak4641_reg,
+	.reg_cache_step		= 1,
+};
+
+
+static int __devinit ak4641_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct ak4641_priv *ak4641;
+	int ret;
+
+	ak4641 = kzalloc(sizeof(struct ak4641_priv), GFP_KERNEL);
+	if (!ak4641)
+		return -ENOMEM;
+
+	i2c_set_clientdata(i2c, ak4641);
+
+	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_ak4641,
+				ak4641_dai, ARRAY_SIZE(ak4641_dai));
+	if (ret < 0)
+		kfree(ak4641);
+
+	return ret;
+}
+
+static int __devexit ak4641_i2c_remove(struct i2c_client *i2c)
+{
+	snd_soc_unregister_codec(&i2c->dev);
+	kfree(i2c_get_clientdata(i2c));
+	return 0;
+}
+
+static const struct i2c_device_id ak4641_i2c_id[] = {
+	{ "ak4641", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ak4641_i2c_id);
+
+static struct i2c_driver ak4641_i2c_driver = {
+	.driver = {
+		.name = "ak4641",
+		.owner = THIS_MODULE,
+	},
+	.probe =    ak4641_i2c_probe,
+	.remove =   __devexit_p(ak4641_i2c_remove),
+	.id_table = ak4641_i2c_id,
+};
+
+static int __init ak4641_modinit(void)
+{
+	int ret;
+
+	ret = i2c_add_driver(&ak4641_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register AK4641 I2C driver: %d\n", ret);
+
+	return ret;
+}
+module_init(ak4641_modinit);
+
+static void __exit ak4641_exit(void)
+{
+	i2c_del_driver(&ak4641_i2c_driver);
+}
+module_exit(ak4641_exit);
+
+MODULE_DESCRIPTION("SoC AK4641 driver");
+MODULE_AUTHOR("Harald Welte <laforge@gnufiish.org>");
+MODULE_LICENSE("GPL");

sound/soc/codecs/ak4641.h

@@ -0,0 +1,47 @@
+/*
+ * ak4641.h  --  AK4641 SoC Audio driver
+ *
+ * Copyright 2008 Harald Welte <laforge@gnufiish.org>
+ *
+ * Based on ak4535.h
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _AK4641_H
+#define _AK4641_H
+
+/* AK4641 register space */
+
+#define AK4641_PM1		0x00
+#define AK4641_PM2		0x01
+#define AK4641_SIG1		0x02
+#define AK4641_SIG2		0x03
+#define AK4641_MODE1		0x04
+#define AK4641_MODE2		0x05
+#define AK4641_DAC		0x06
+#define AK4641_MIC		0x07
+#define AK4641_TIMER		0x08
+#define AK4641_ALC1		0x09
+#define AK4641_ALC2		0x0a
+#define AK4641_PGA		0x0b
+#define AK4641_LATT		0x0c
+#define AK4641_RATT		0x0d
+#define AK4641_VOL		0x0e
+#define AK4641_STATUS		0x0f
+#define AK4641_EQLO		0x10
+#define AK4641_EQMID		0x11
+#define AK4641_EQHI		0x12
+#define AK4641_BTIF		0x13
+
+#define AK4641_CACHEREGNUM	0x14
+
+
+
+#define AK4641_DAI_HIFI		0
+#define AK4641_DAI_VOICE	1
+
+
+#endif

sound/soc/codecs/dmic.c

@@ -39,7 +39,31 @@ static struct snd_soc_dai_driver dmic_dai = {
 	},
 };
 
-static struct snd_soc_codec_driver soc_dmic = {};
+static const struct snd_soc_dapm_widget dmic_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_OUT("DMIC AIF", "Capture", 0,
+			     SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_INPUT("DMic"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+	{"DMIC AIF", NULL, "DMic"},
+};
+
+static int dmic_probe(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
+				  ARRAY_SIZE(dmic_dapm_widgets));
+        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
+	snd_soc_dapm_new_widgets(dapm);
+
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_dmic = {
+	.probe	= dmic_probe,
+};
 
 static int __devinit dmic_dev_probe(struct platform_device *pdev)
 {

sound/soc/codecs/max98088.c

@@ -656,8 +656,6 @@ static const struct soc_enum max98088_exmode_enum =
                              ARRAY_SIZE(max98088_exmode_texts),
                              max98088_exmode_texts,
                              max98088_exmode_values);
-static const struct snd_kcontrol_new max98088_exmode_controls =
-       SOC_DAPM_VALUE_ENUM("Route", max98088_exmode_enum);
 
 static const char *max98088_ex_thresh[] = { /* volts PP */
        "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"};
@@ -783,6 +781,7 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
        SOC_SINGLE("EQ1 Switch", M98088_REG_49_CFG_LEVEL, 0, 1, 0),
        SOC_SINGLE("EQ2 Switch", M98088_REG_49_CFG_LEVEL, 1, 1, 0),
 
+       SOC_ENUM("EX Limiter Mode", max98088_exmode_enum),
        SOC_ENUM("EX Limiter Threshold", max98088_ex_thresh_enum),
 
        SOC_ENUM("DAI1 Filter Mode", max98088_filter_mode_enum),
@@ -808,10 +807,10 @@ static const struct snd_kcontrol_new max98088_snd_controls[] = {
 
 /* Left speaker mixer switch */
 static const struct snd_kcontrol_new max98088_left_speaker_mixer_controls[] = {
-       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
-       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 7, 1, 0),
        SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 5, 1, 0),
        SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_2B_MIX_SPK_LEFT, 6, 1, 0),
        SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_2B_MIX_SPK_LEFT, 1, 1, 0),
@@ -836,10 +835,10 @@ static const struct snd_kcontrol_new max98088_right_speaker_mixer_controls[] = {
 
 /* Left headphone mixer switch */
 static const struct snd_kcontrol_new max98088_left_hp_mixer_controls[] = {
-       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
-       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_25_MIX_HP_LEFT, 7, 1, 0),
        SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_25_MIX_HP_LEFT, 5, 1, 0),
        SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_25_MIX_HP_LEFT, 6, 1, 0),
        SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_25_MIX_HP_LEFT, 1, 1, 0),
@@ -864,10 +863,10 @@ static const struct snd_kcontrol_new max98088_right_hp_mixer_controls[] = {
 
 /* Left earpiece/receiver mixer switch */
 static const struct snd_kcontrol_new max98088_left_rec_mixer_controls[] = {
-       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
-       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
-       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC1 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
+       SOC_DAPM_SINGLE("Left DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 0, 1, 0),
+       SOC_DAPM_SINGLE("Right DAC2 Switch", M98088_REG_28_MIX_REC_LEFT, 7, 1, 0),
        SOC_DAPM_SINGLE("MIC1 Switch", M98088_REG_28_MIX_REC_LEFT, 5, 1, 0),
        SOC_DAPM_SINGLE("MIC2 Switch", M98088_REG_28_MIX_REC_LEFT, 6, 1, 0),
        SOC_DAPM_SINGLE("INA1 Switch", M98088_REG_28_MIX_REC_LEFT, 1, 1, 0),
@@ -1094,9 +1093,6 @@ static const struct snd_soc_dapm_widget max98088_dapm_widgets[] = {
 
        SND_SOC_DAPM_MICBIAS("MICBIAS", M98088_REG_4C_PWR_EN_IN, 3, 0),
 
-       SND_SOC_DAPM_MUX("EX Limiter Mode", SND_SOC_NOPM, 0, 0,
-               &max98088_exmode_controls),
-
        SND_SOC_DAPM_OUTPUT("HPL"),
        SND_SOC_DAPM_OUTPUT("HPR"),
        SND_SOC_DAPM_OUTPUT("SPKL"),
@@ -1568,6 +1564,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai,
        return 0;
 }
 
+static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+       struct snd_soc_codec *codec = codec_dai->codec;
+       int reg;
+
+       if (mute)
+               reg = M98088_DAI_MUTE;
+       else
+               reg = 0;
+
+       snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY,
+                           M98088_DAI_MUTE_MASK, reg);
+       return 0;
+}
+
+static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+       struct snd_soc_codec *codec = codec_dai->codec;
+       int reg;
+
+       if (mute)
+               reg = M98088_DAI_MUTE;
+       else
+               reg = 0;
+
+       snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY,
+                           M98088_DAI_MUTE_MASK, reg);
+       return 0;
+}
+
 static void max98088_sync_cache(struct snd_soc_codec *codec)
 {
        u16 *reg_cache = codec->reg_cache;
@@ -1629,12 +1655,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai1_set_fmt,
        .hw_params = max98088_dai1_hw_params,
+       .digital_mute = max98088_dai1_digital_mute,
 };
 
 static struct snd_soc_dai_ops max98088_dai2_ops = {
        .set_sysclk = max98088_dai_set_sysclk,
        .set_fmt = max98088_dai2_set_fmt,
        .hw_params = max98088_dai2_hw_params,
+       .digital_mute = max98088_dai2_digital_mute,
 };
 
 static struct snd_soc_dai_driver max98088_dai[] = {

sound/soc/codecs/max98088.h

@@ -133,6 +133,19 @@
        #define M98088_REC_LINEMODE             (1<<7)
        #define M98088_REC_LINEMODE_MASK        (1<<7)
 
+/* M98088_REG_2D_MIX_SPK_CNTL */
+       #define M98088_MIX_SPKR_GAIN_MASK       (3<<2)
+       #define M98088_MIX_SPKR_GAIN_SHIFT      2
+       #define M98088_MIX_SPKL_GAIN_MASK       (3<<0)
+       #define M98088_MIX_SPKL_GAIN_SHIFT      0
+
+/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */
+       #define M98088_DAI_MUTE                 (1<<7)
+       #define M98088_DAI_MUTE_MASK            (1<<7)
+       #define M98088_DAI_VOICE_GAIN_MASK      (3<<4)
+       #define M98088_DAI_ATTENUATION_MASK     (0xF<<0)
+       #define M98088_DAI_ATTENUATION_SHIFT    0
+
 /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */
        #define M98088_MICPRE_MASK              (3<<5)
        #define M98088_MICPRE_SHIFT             5

sound/soc/codecs/max98095.c

@@ -1870,16 +1870,14 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol,
 
 	BUG_ON(channel > 1);
 
-	cdata = &max98095->dai[channel];
+	if (!pdata || !max98095->eq_textcnt)
+		return 0;
 
 	if (sel >= pdata->eq_cfgcnt)
 		return -EINVAL;
 
+	cdata = &max98095->dai[channel];
 	cdata->eq_sel = sel;
-
-	if (!pdata || !max98095->eq_textcnt)
-		return 0;
-
 	fs = cdata->rate;
 
 	/* Find the selected configuration with nearest sample rate */
@@ -2018,16 +2016,14 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol,
 
 	BUG_ON(channel > 1);
 
-	cdata = &max98095->dai[channel];
+	if (!pdata || !max98095->bq_textcnt)
+		return 0;
 
 	if (sel >= pdata->bq_cfgcnt)
 		return -EINVAL;
 
+	cdata = &max98095->dai[channel];
 	cdata->bq_sel = sel;
-
-	if (!pdata || !max98095->bq_textcnt)
-		return 0;
-
 	fs = cdata->rate;
 
 	/* Find the selected configuration with nearest sample rate */

sound/soc/codecs/spdif_transciever.c

@@ -21,7 +21,7 @@
 #include <sound/pcm.h>
 #include <sound/initval.h>
 
-MODULE_LICENSE("GPL");
+#define DRV_NAME "spdif-dit"
 
 #define STUB_RATES	SNDRV_PCM_RATE_8000_96000
 #define STUB_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
@@ -56,7 +56,7 @@ static struct platform_driver spdif_dit_driver = {
 	.probe		= spdif_dit_probe,
 	.remove		= spdif_dit_remove,
 	.driver		= {
-		.name	= "spdif-dit",
+		.name	= DRV_NAME,
 		.owner	= THIS_MODULE,
 	},
 };
@@ -74,3 +74,7 @@ static void __exit dit_exit(void)
 module_init(dit_modinit);
 module_exit(dit_exit);
 
+MODULE_AUTHOR("Steve Chen <schen@mvista.com>");
+MODULE_DESCRIPTION("SPDIF dummy codec driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/codecs/ssm2602.c

@@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1),
 SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1),
 SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0),
 
-SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0),
+SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LOUT"),
 SND_SOC_DAPM_OUTPUT("ROUT"),

sound/soc/codecs/tlv320aic3x.c

@@ -157,7 +157,8 @@ static int aic3x_read(struct snd_soc_codec *codec, unsigned int reg,
 static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol,
 					struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct soc_mixer_control *mc =
 		(struct soc_mixer_control *)kcontrol->private_value;
 	unsigned int reg = mc->reg;

sound/soc/codecs/tlv320dac33.c

@@ -1,7 +1,7 @@
 /*
  * ALSA SoC Texas Instruments TLV320DAC33 codec driver
  *
- * Author:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * Copyright:   (C) 2009 Nokia Corporation
  *
@@ -1658,5 +1658,5 @@ module_exit(dac33_module_exit);
 
 
 MODULE_DESCRIPTION("ASoC TLV320DAC33 codec driver");
-MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@nokia.com>");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
 MODULE_LICENSE("GPL");

sound/soc/codecs/tlv320dac33.h

@@ -1,7 +1,7 @@
 /*
  * ALSA SoC Texas Instruments TLV320DAC33 codec driver
  *
- * Author:	Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *
  * Copyright:   (C) 2009 Nokia Corporation
  *

sound/soc/codecs/tpa6130a2.c

@@ -3,7 +3,7 @@
  *
  * Copyright (C) Nokia Corporation
  *
- * Author: 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
@@ -495,7 +495,7 @@ static void __exit tpa6130a2_exit(void)
 	i2c_del_driver(&tpa6130a2_i2c_driver);
 }
 
-MODULE_AUTHOR("Peter Ujfalusi");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
 MODULE_DESCRIPTION("TPA6130A2 Headphone amplifier driver");
 MODULE_LICENSE("GPL");
 

sound/soc/codecs/tpa6130a2.h

@@ -3,7 +3,7 @@
  *
  * Copyright (C) Nokia Corporation
  *
- * Author: 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

sound/soc/codecs/twl6040.c

@@ -960,9 +960,9 @@ static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0);
 
 /*
  * AFMGAIN volume control:
- * from 18 to 24 dB in 6 dB steps
+ * from -18 to 24 dB in 6 dB steps
  */
-static DECLARE_TLV_DB_SCALE(afm_amp_tlv, 1800, 600, 0);
+static DECLARE_TLV_DB_SCALE(afm_amp_tlv, -1800, 600, 0);
 
 /*
  * HSGAIN volume control:
@@ -1049,7 +1049,7 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = {
 
 	/* AFM gains */
 	SOC_DOUBLE_TLV("Aux FM Volume",
-		TWL6040_REG_LINEGAIN, 0, 4, 0xF, 0, afm_amp_tlv),
+		TWL6040_REG_LINEGAIN, 0, 3, 7, 0, afm_amp_tlv),
 
 	/* Playback gains */
 	SOC_TWL6040_DOUBLE_TLV("Headset Playback Volume",

sound/soc/codecs/wm1250-ev1.c

@@ -22,7 +22,7 @@ SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0),
 SND_SOC_DAPM_DAC("DAC", "wm1250-ev1 Playback", SND_SOC_NOPM, 0, 0),
 
 SND_SOC_DAPM_INPUT("WM1250 Input"),
-SND_SOC_DAPM_INPUT("WM1250 Output"),
+SND_SOC_DAPM_OUTPUT("WM1250 Output"),
 };
 
 static const struct snd_soc_dapm_route wm1250_ev1_dapm_routes[] = {

sound/soc/codecs/wm8731.c

@@ -198,7 +198,7 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
 {
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
 
-	return wm8731->sysclk_type == WM8731_SYSCLK_MCLK;
+	return wm8731->sysclk_type == WM8731_SYSCLK_XTAL;
 }
 
 static const struct snd_soc_dapm_route wm8731_intercon[] = {

sound/soc/codecs/wm8903.c

@@ -382,7 +382,8 @@ static void wm8903_seq_notifier(struct snd_soc_dapm_context *dapm,
 static int wm8903_class_w_put(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
 	struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec);
 	u16 reg;

sound/soc/codecs/wm8915.c

@@ -19,7 +19,6 @@
 #include <linux/gcd.h>
 #include <linux/gpio.h>
 #include <linux/i2c.h>
-#include <linux/delay.h>
 #include <linux/regulator/consumer.h>
 #include <linux/slab.h>
 #include <linux/workqueue.h>

sound/soc/codecs/wm8958-dsp2.c

@@ -99,7 +99,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
 	len = fw->size - len;
 	while (len) {
 		if (len < 12) {
-			dev_err(codec->dev, "%s short data block of %d\n",
+			dev_err(codec->dev, "%s short data block of %zd\n",
 				name, len);
 			goto err;
 		}
@@ -107,7 +107,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
 		memcpy(&data32, data + 4, sizeof(data32));
 		block_len = be32_to_cpu(data32);
 		if (block_len + 8 > len) {
-			dev_err(codec->dev, "%d byte block longer than file\n",
+			dev_err(codec->dev, "%zd byte block longer than file\n",
 				block_len);
 			goto err;
 		}
@@ -141,7 +141,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name,
 		case WM_FW_BLOCK_I:
 		case WM_FW_BLOCK_A:
 		case WM_FW_BLOCK_C:
-			dev_dbg(codec->dev, "%s: %d bytes of %x@%x\n", name,
+			dev_dbg(codec->dev, "%s: %zd bytes of %x@%x\n", name,
 				block_len, (data32 >> 24) & 0xff,
 				data32 & 0xffffff);
 
@@ -362,6 +362,10 @@ static void wm8958_dsp_apply(struct snd_soc_codec *codec, int path, int start)
 		path, wm8994->dsp_active, start, pwr_reg, reg);
 
 	if (start && ena) {
+		/* If the DSP is already running then noop */
+		if (reg & WM8958_DSP2_ENA)
+			return;
+
 		/* If either AIFnCLK is not yet enabled postpone */
 		if (!(snd_soc_read(codec, WM8994_AIF1_CLOCKING_1)
 		      & WM8994_AIF1CLK_ENA_MASK) &&
@@ -508,6 +512,9 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+	if (wm8994->mbc_ena[mbc] == ucontrol->value.integer.value[0])
+		return 0;
+
 	if (ucontrol->value.integer.value[0] > 1)
 		return -EINVAL;
 
@@ -628,6 +635,9 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+	if (wm8994->vss_ena[vss] == ucontrol->value.integer.value[0])
+		return 0;
+
 	if (ucontrol->value.integer.value[0] > 1)
 		return -EINVAL;
 
@@ -689,6 +699,16 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+	if (hpf < 3) {
+		if (wm8994->hpf1_ena[hpf % 3] ==
+		    ucontrol->value.integer.value[0])
+			return 0;
+	} else {
+		if (wm8994->hpf2_ena[hpf % 3] ==
+		    ucontrol->value.integer.value[0])
+			return 0;
+	}
+
 	if (ucontrol->value.integer.value[0] > 1)
 		return -EINVAL;
 
@@ -782,6 +802,9 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol,
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
 
+	if (wm8994->enh_eq_ena[eq] == ucontrol->value.integer.value[0])
+		return 0;
+
 	if (ucontrol->value.integer.value[0] > 1)
 		return -EINVAL;
 

sound/soc/codecs/wm8993.c

@@ -718,7 +718,8 @@ static int clk_sys_event(struct snd_soc_dapm_widget *w,
 static int class_w_put(struct snd_kcontrol *kcontrol,
 		       struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *widget = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
 	struct snd_soc_codec *codec = widget->codec;
 	struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec);
 	int ret;

sound/soc/codecs/wm8994.c

@@ -877,7 +877,8 @@ static const char *hp_mux_text[] = {
 static int wm8994_put_hp_enum(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *w = wlist->widgets[0];
 	struct snd_soc_codec *codec = w->codec;
 	int ret;
 
@@ -1004,7 +1005,8 @@ SOC_DAPM_SINGLE("AIF1.1 Switch", WM8994_DAC2_RIGHT_MIXER_ROUTING,
 static int wm8994_put_class_w(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *w = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *w = wlist->widgets[0];
 	struct snd_soc_codec *codec = w->codec;
 	int ret;
 
@@ -2416,8 +2418,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = {
 static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	struct wm8994 *control = codec->control_data;
 	int i, ret;
 
+	switch (control->type) {
+	case WM8994:
+		snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0);
+		break;
+	case WM8958:
+		snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
+				    WM8958_MICD_ENA, 0);
+		break;
+	}
+
 	for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) {
 		memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i],
 		       sizeof(struct wm8994_fll_config));
@@ -2435,6 +2448,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state)
 static int wm8994_resume(struct snd_soc_codec *codec)
 {
 	struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec);
+	struct wm8994 *control = codec->control_data;
 	int i, ret;
 	unsigned int val, mask;
 
@@ -2473,6 +2487,19 @@ static int wm8994_resume(struct snd_soc_codec *codec)
 				 i + 1, ret);
 	}
 
+	switch (control->type) {
+	case WM8994:
+		if (wm8994->micdet[0].jack || wm8994->micdet[1].jack)
+			snd_soc_update_bits(codec, WM8994_MICBIAS,
+					    WM8994_MICD_ENA, WM8994_MICD_ENA);
+		break;
+	case WM8958:
+		if (wm8994->jack_cb)
+			snd_soc_update_bits(codec, WM8958_MIC_DETECT_1,
+					    WM8958_MICD_ENA, WM8958_MICD_ENA);
+		break;
+	}
+
 	return 0;
 }
 #else

sound/soc/codecs/wm8995.c

@@ -305,11 +305,11 @@ static int check_clk_sys(struct snd_soc_dapm_widget *source,
 static int wm8995_put_class_w(struct snd_kcontrol *kcontrol,
 			      struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_dapm_widget *w;
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *w = wlist->widgets[0];
 	struct snd_soc_codec *codec;
 	int ret;
 
-	w = snd_kcontrol_chip(kcontrol);
 	codec = w->codec;
 	ret = snd_soc_dapm_put_volsw(kcontrol, ucontrol);
 	wm8995_update_class_w(codec);

sound/soc/codecs/wm_hubs.c

@@ -215,23 +215,23 @@ static const struct snd_kcontrol_new analogue_snd_controls[] = {
 SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
 	       inpga_tlv),
 SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
 
 SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
 	       inpga_tlv),
 SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
 
 
 SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
 	       inpga_tlv),
 SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
 
 SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
 	       inpga_tlv),
 SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
-SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 0),
+SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
 
 SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
 	       inmix_sw_tlv),
@@ -787,17 +787,17 @@ static const struct snd_soc_dapm_route analogue_routes[] = {
 static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
 	{ "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" },
 	{ "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" },
-	{ "LINEOUT1 Mixer", "Output Switch", "Left Output Mixer" },
+	{ "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" },
 
 	{ "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" },
 	{ "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" },
 };
 
 static const struct snd_soc_dapm_route lineout1_se_routes[] = {
-	{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output Mixer" },
-	{ "LINEOUT1N Mixer", "Right Output Switch", "Left Output Mixer" },
+	{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
+	{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
 
-	{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output Mixer" },
+	{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
 
 	{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
 	{ "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" },
@@ -806,17 +806,17 @@ static const struct snd_soc_dapm_route lineout1_se_routes[] = {
 static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
 	{ "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
 	{ "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
-	{ "LINEOUT2 Mixer", "Output Switch", "Right Output Mixer" },
+	{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
 
 	{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
 	{ "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" },
 };
 
 static const struct snd_soc_dapm_route lineout2_se_routes[] = {
-	{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output Mixer" },
-	{ "LINEOUT2N Mixer", "Right Output Switch", "Left Output Mixer" },
+	{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
+	{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
 
-	{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output Mixer" },
+	{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
 
 	{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
 	{ "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" },
@@ -836,17 +836,21 @@ int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
 	snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME,
 			    WM8993_IN2_VU, WM8993_IN2_VU);
 
+	snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT,
+			    WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
 	snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT,
 			    WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
 
 	snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME,
-			    WM8993_HPOUT1L_ZC, WM8993_HPOUT1L_ZC);
+			    WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC,
+			    WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC);
 	snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME,
 			    WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC,
 			    WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC);
 
 	snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME,
-			    WM8993_MIXOUTL_ZC, WM8993_MIXOUTL_ZC);
+			    WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU,
+			    WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU);
 	snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME,
 			    WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU,
 			    WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU);

sound/soc/davinci/davinci-mcasp.c

@@ -909,6 +909,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_PLAYBACK];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
+	dma_data->sram_size = pdata->sram_size_playback;
 	dma_data->dma_addr = (dma_addr_t) (pdata->tx_dma_offset +
 							mem->start);
 
@@ -925,6 +926,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 	dma_data = &dev->dma_params[SNDRV_PCM_STREAM_CAPTURE];
 	dma_data->asp_chan_q = pdata->asp_chan_q;
 	dma_data->ram_chan_q = pdata->ram_chan_q;
+	dma_data->sram_size = pdata->sram_size_capture;
 	dma_data->dma_addr = (dma_addr_t)(pdata->rx_dma_offset +
 							mem->start);
 

sound/soc/omap/omap-mcbsp.c

@@ -4,7 +4,7 @@
  * Copyright (C) 2008 Nokia Corporation
  *
  * Contact: Jarkko Nikula <jhnikula@gmail.com>
- *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          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
@@ -146,7 +146,7 @@ static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
 	 * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
 	 * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
 	 */
-	if (cpu_is_omap343x() || cpu_is_omap44xx()) {
+	if (cpu_is_omap34xx() || cpu_is_omap44xx()) {
 		/*
 		* Rule for the buffer size. We should not allow
 		* smaller buffer than the FIFO size to avoid underruns
@@ -258,7 +258,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	default:
 		return -EINVAL;
 	}
-	if (cpu_is_omap343x()) {
+	if (cpu_is_omap34xx()) {
 		dma_data->set_threshold = omap_mcbsp_set_threshold;
 		/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
 		if (omap_mcbsp_get_dma_op_mode(bus_id) ==

sound/soc/omap/omap-mcbsp.h

@@ -4,7 +4,7 @@
  * Copyright (C) 2008 Nokia Corporation
  *
  * Contact: Jarkko Nikula <jhnikula@gmail.com>
- *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          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

sound/soc/omap/omap-pcm.c

@@ -4,7 +4,7 @@
  * Copyright (C) 2008 Nokia Corporation
  *
  * Contact: Jarkko Nikula <jhnikula@gmail.com>
- *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          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
@@ -37,7 +37,8 @@ static const struct snd_pcm_hardware omap_pcm_hardware = {
 				  SNDRV_PCM_INFO_MMAP_VALID |
 				  SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_PAUSE |
-				  SNDRV_PCM_INFO_RESUME,
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
 	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
 				  SNDRV_PCM_FMTBIT_S32_LE,
 	.period_bytes_min	= 32,
@@ -195,7 +196,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	if ((cpu_is_omap1510()))
 		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
 			      OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
-	else
+	else if (!substream->runtime->no_period_wakeup)
 		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
 
 	if (!(cpu_class_is_omap1())) {

sound/soc/omap/omap-pcm.h

@@ -4,7 +4,7 @@
  * Copyright (C) 2008 Nokia Corporation
  *
  * Contact: Jarkko Nikula <jhnikula@gmail.com>
- *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ *          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

sound/soc/omap/rx51.c

@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 - 2009 Nokia Corporation
  *
- * Contact: Peter Ujfalusi <peter.ujfalusi@nokia.com>
+ * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
  *          Eduardo Valentin <eduardo.valentin@nokia.com>
  *          Jarkko Nikula <jhnikula@gmail.com>
  *

sound/soc/pxa/Kconfig

@@ -155,6 +155,15 @@ config SND_SOC_RAUMFELD
 	help
 	  Say Y if you want to add support for SoC audio on Raumfeld devices
 
+config SND_PXA2XX_SOC_HX4700
+	tristate "SoC Audio support for HP iPAQ hx4700"
+	depends on SND_PXA2XX_SOC && MACH_H4700
+	select SND_PXA2XX_SOC_I2S
+	select SND_SOC_AK4641
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  HP iPAQ hx4700.
+
 config SND_PXA2XX_SOC_MAGICIAN
 	tristate "SoC Audio support for HTC Magician"
 	depends on SND_PXA2XX_SOC && MACH_MAGICIAN

sound/soc/pxa/Makefile

@@ -22,6 +22,7 @@ snd-soc-palm27x-objs := palm27x.o
 snd-soc-saarb-objs := saarb.o
 snd-soc-tavorevb3-objs := tavorevb3.o
 snd-soc-zylonite-objs := zylonite.o
+snd-soc-hx4700-objs := hx4700.o
 snd-soc-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
 snd-soc-z2-objs := z2.o
@@ -37,6 +38,7 @@ obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_HX4700) += snd-soc-hx4700.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MAGICIAN) += snd-soc-magician.o
 obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_PXA2XX_SOC_Z2) += snd-soc-z2.o

sound/soc/pxa/hx4700.c

@@ -0,0 +1,255 @@
+/*
+ * SoC audio for HP iPAQ hx4700
+ *
+ * Copyright (c) 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 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/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/delay.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <mach/hx4700.h>
+#include <asm/mach-types.h>
+#include "pxa2xx-i2s.h"
+
+#include "../codecs/ak4641.h"
+
+static struct snd_soc_jack hs_jack;
+
+/* Headphones jack detection DAPM pin */
+static struct snd_soc_jack_pin hs_jack_pin[] = {
+	{
+		.pin	= "Headphone Jack",
+		.mask	= SND_JACK_HEADPHONE,
+	},
+	{
+		.pin	= "Speaker",
+		/* disable speaker when hp jack is inserted */
+		.mask   = SND_JACK_HEADPHONE,
+		.invert	= 1,
+	},
+};
+
+/* Headphones jack detection GPIO */
+static struct snd_soc_jack_gpio hs_jack_gpio = {
+	.gpio		= GPIO75_HX4700_EARPHONE_nDET,
+	.invert		= true,
+	.name		= "hp-gpio",
+	.report		= SND_JACK_HEADPHONE,
+	.debounce_time	= 200,
+};
+
+/*
+ * iPAQ hx4700 uses I2S for capture and playback.
+ */
+static int hx4700_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+			SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the I2S system clock as output */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
+			SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* inform codec driver about clock freq *
+	 * (PXA I2S always uses divider 256)    */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 256 * params_rate(params),
+			SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops hx4700_ops = {
+	.hw_params = hx4700_hw_params,
+};
+
+static int hx4700_spk_power(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(GPIO107_HX4700_SPK_nSD, !!SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int hx4700_hp_power(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *k, int event)
+{
+	gpio_set_value(GPIO92_HX4700_HP_DRIVER, !!SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* hx4700 machine dapm widgets */
+static const struct snd_soc_dapm_widget hx4700_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", hx4700_hp_power),
+	SND_SOC_DAPM_SPK("Speaker", hx4700_spk_power),
+	SND_SOC_DAPM_MIC("Built-in Microphone", NULL),
+};
+
+/* hx4700 machine audio_map */
+static const struct snd_soc_dapm_route hx4700_audio_map[] = {
+
+	/* Headphone connected to LOUT, ROUT */
+	{"Headphone Jack", NULL, "LOUT"},
+	{"Headphone Jack", NULL, "ROUT"},
+
+	/* Speaker connected to MOUT2 */
+	{"Speaker", NULL, "MOUT2"},
+
+	/* Microphone connected to MICIN */
+	{"MICIN", NULL, "Built-in Microphone"},
+	{"AIN", NULL, "MICOUT"},
+};
+
+/*
+ * Logic for a ak4641 as connected on a HP iPAQ hx4700
+ */
+static int hx4700_ak4641_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int err;
+
+	/* NC codec pins */
+	/* FIXME: is anything connected here? */
+	snd_soc_dapm_nc_pin(dapm, "MOUT1");
+	snd_soc_dapm_nc_pin(dapm, "MICEXT");
+	snd_soc_dapm_nc_pin(dapm, "AUX");
+
+	/* Jack detection API stuff */
+	err = snd_soc_jack_new(codec, "Headphone Jack",
+				SND_JACK_HEADPHONE, &hs_jack);
+	if (err)
+		return err;
+
+	err = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pin),
+					hs_jack_pin);
+	if (err)
+		return err;
+
+	err = snd_soc_jack_add_gpios(&hs_jack, 1, &hs_jack_gpio);
+
+	return err;
+}
+
+/* hx4700 digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link hx4700_dai = {
+	.name = "ak4641",
+	.stream_name = "AK4641",
+	.cpu_dai_name = "pxa2xx-i2s",
+	.codec_dai_name = "ak4641-hifi",
+	.platform_name = "pxa-pcm-audio",
+	.codec_name = "ak4641.0-0012",
+	.init = hx4700_ak4641_init,
+	.ops = &hx4700_ops,
+};
+
+/* hx4700 audio machine driver */
+static struct snd_soc_card snd_soc_card_hx4700 = {
+	.name			= "iPAQ hx4700",
+	.dai_link		= &hx4700_dai,
+	.num_links		= 1,
+	.dapm_widgets		= hx4700_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(hx4700_dapm_widgets),
+	.dapm_routes		= hx4700_audio_map,
+	.num_dapm_routes	= ARRAY_SIZE(hx4700_audio_map),
+};
+
+static struct gpio hx4700_audio_gpios[] = {
+	{ GPIO107_HX4700_SPK_nSD, GPIOF_OUT_INIT_HIGH, "SPK_POWER" },
+	{ GPIO92_HX4700_HP_DRIVER, GPIOF_OUT_INIT_LOW, "EP_POWER" },
+};
+
+static int __devinit hx4700_audio_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (!machine_is_h4700())
+		return -ENODEV;
+
+	ret = gpio_request_array(hx4700_audio_gpios,
+				ARRAY_SIZE(hx4700_audio_gpios));
+	if (ret)
+		return ret;
+
+	snd_soc_card_hx4700.dev = &pdev->dev;
+	ret = snd_soc_register_card(&snd_soc_card_hx4700);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static int __devexit hx4700_audio_remove(struct platform_device *pdev)
+{
+	snd_soc_jack_free_gpios(&hs_jack, 1, &hs_jack_gpio);
+	snd_soc_unregister_card(&snd_soc_card_hx4700);
+
+	gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0);
+	gpio_set_value(GPIO107_HX4700_SPK_nSD, 0);
+
+	gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios));
+	return 0;
+}
+
+static struct platform_driver hx4700_audio_driver = {
+	.driver	= {
+		.name = "hx4700-audio",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe	= hx4700_audio_probe,
+	.remove	= __devexit_p(hx4700_audio_remove),
+};
+
+static int __init hx4700_modinit(void)
+{
+	return platform_driver_register(&hx4700_audio_driver);
+}
+module_init(hx4700_modinit);
+
+static void __exit hx4700_modexit(void)
+{
+	platform_driver_unregister(&hx4700_audio_driver);
+}
+
+module_exit(hx4700_modexit);
+
+MODULE_AUTHOR("Philipp Zabel");
+MODULE_DESCRIPTION("ALSA SoC iPAQ hx4700");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:hx4700-audio");

sound/soc/pxa/raumfeld.c

@@ -151,13 +151,13 @@ static struct snd_soc_ops raumfeld_cs4270_ops = {
 	.hw_params = raumfeld_cs4270_hw_params,
 };
 
-static int raumfeld_line_suspend(struct snd_soc_card *card)
+static int raumfeld_analog_suspend(struct snd_soc_card *card)
 {
 	raumfeld_enable_audio(false);
 	return 0;
 }
 
-static int raumfeld_line_resume(struct snd_soc_card *card)
+static int raumfeld_analog_resume(struct snd_soc_card *card)
 {
 	raumfeld_enable_audio(true);
 	return 0;
@@ -225,32 +225,53 @@ static struct snd_soc_ops raumfeld_ak4104_ops = {
 	.hw_params = raumfeld_ak4104_hw_params,
 };
 
-static struct snd_soc_dai_link raumfeld_dai[] = {
+#define DAI_LINK_CS4270		\
+{							\
+	.name		= "CS4270",			\
+	.stream_name	= "CS4270",			\
+	.cpu_dai_name	= "pxa-ssp-dai.0",		\
+	.platform_name	= "pxa-pcm-audio",		\
+	.codec_dai_name	= "cs4270-hifi",		\
+	.codec_name	= "cs4270-codec.0-0048",	\
+	.ops		= &raumfeld_cs4270_ops,		\
+}
+
+#define DAI_LINK_AK4104		\
+{							\
+	.name		= "ak4104",			\
+	.stream_name	= "Playback",			\
+	.cpu_dai_name	= "pxa-ssp-dai.1",		\
+	.codec_dai_name	= "ak4104-hifi",		\
+	.platform_name	= "pxa-pcm-audio",		\
+	.ops		= &raumfeld_ak4104_ops,		\
+	.codec_name	= "spi0.0",			\
+}
+
+static struct snd_soc_dai_link snd_soc_raumfeld_connector_dai[] =
 {
-	.name		= "ak4104",
-	.stream_name	= "Playback",
-	.cpu_dai_name	= "pxa-ssp-dai.1",
-	.codec_dai_name	= "ak4104-hifi",
-	.platform_name	= "pxa-pcm-audio",
-	.ops		= &raumfeld_ak4104_ops,
-	.codec_name	= "ak4104-codec.0",
-},
+	DAI_LINK_CS4270,
+	DAI_LINK_AK4104,
+};
+
+static struct snd_soc_dai_link snd_soc_raumfeld_speaker_dai[] =
 {
-	.name		= "CS4270",
-	.stream_name	= "CS4270",
-	.cpu_dai_name	= "pxa-ssp-dai.0",
-	.platform_name	= "pxa-pcm-audio",
-	.codec_dai_name	= "cs4270-hifi",
-	.codec_name	= "cs4270-codec.0-0048",
-	.ops		= &raumfeld_cs4270_ops,
-},};
-
-static struct snd_soc_card snd_soc_raumfeld = {
-	.name		= "Raumfeld",
-	.dai_link	= raumfeld_dai,
-	.suspend_post	= raumfeld_line_suspend,
-	.resume_pre	= raumfeld_line_resume,
-	.num_links	= ARRAY_SIZE(raumfeld_dai),
+	DAI_LINK_CS4270,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_connector = {
+	.name		= "Raumfeld Connector",
+	.dai_link	= snd_soc_raumfeld_connector_dai,
+	.num_links	= ARRAY_SIZE(snd_soc_raumfeld_connector_dai),
+	.suspend_post	= raumfeld_analog_suspend,
+	.resume_pre	= raumfeld_analog_resume,
+};
+
+static struct snd_soc_card snd_soc_raumfeld_speaker = {
+	.name		= "Raumfeld Speaker",
+	.dai_link	= snd_soc_raumfeld_speaker_dai,
+	.num_links	= ARRAY_SIZE(snd_soc_raumfeld_speaker_dai),
+	.suspend_post	= raumfeld_analog_suspend,
+	.resume_pre	= raumfeld_analog_resume,
 };
 
 static struct platform_device *raumfeld_audio_device;
@@ -271,22 +292,25 @@ static int __init raumfeld_audio_init(void)
 
 	set_max9485_clk(MAX9485_MCLK_FREQ_122880);
 
-	/* Register LINE and SPDIF */
+	/* Register analog device */
 	raumfeld_audio_device = platform_device_alloc("soc-audio", 0);
 	if (!raumfeld_audio_device)
 		return -ENOMEM;
 
-	platform_set_drvdata(raumfeld_audio_device,
-			     &snd_soc_raumfeld);
-	ret = platform_device_add(raumfeld_audio_device);
-
-	/* no S/PDIF on Speakers */
 	if (machine_is_raumfeld_speaker())
+		platform_set_drvdata(raumfeld_audio_device,
+				     &snd_soc_raumfeld_speaker);
+
+	if (machine_is_raumfeld_connector())
+		platform_set_drvdata(raumfeld_audio_device,
+				     &snd_soc_raumfeld_connector);
+
+	ret = platform_device_add(raumfeld_audio_device);
+	if (ret < 0)
 		return ret;
 
 	raumfeld_enable_audio(true);
-
-	return ret;
+	return 0;
 }
 
 static void __exit raumfeld_audio_exit(void)

sound/soc/soc-cache.c

@@ -20,25 +20,15 @@
 
 #include <trace/events/asoc.h>
 
-#if defined(CONFIG_SPI_MASTER)
-static int do_spi_write(void *control_data, const void *msg,
-			int len)
+#ifdef CONFIG_SPI_MASTER
+static int do_spi_write(void *control, const char *data, int len)
 {
-	struct spi_device *spi = control_data;
-	struct spi_transfer t;
-	struct spi_message m;
-
-	if (len <= 0)
-		return 0;
-
-	spi_message_init(&m);
-	memset(&t, 0, sizeof t);
-
-	t.tx_buf = msg;
-	t.len = len;
+	struct spi_device *spi = control;
+	int ret;
 
-	spi_message_add_tail(&t, &m);
-	spi_sync(spi, &m);
+	ret = spi_write(spi, data, len);
+	if (ret < 0)
+		return ret;
 
 	return len;
 }
@@ -101,28 +91,12 @@ static unsigned int snd_soc_4_12_read(struct snd_soc_codec *codec,
 static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
-	u8 data[2];
-
-	data[0] = (reg << 4) | ((value >> 8) & 0x000f);
-	data[1] = value & 0x00ff;
-
-	return do_hw_write(codec, reg, value, data, 2);
-}
-
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_4_12_spi_write(void *control_data, const char *data,
-				  int len)
-{
-	u8 msg[2];
+	u16 data;
 
-	msg[0] = data[1];
-	msg[1] = data[0];
+	data = cpu_to_be16((reg << 12) | (value & 0xffffff));
 
-	return do_spi_write(control_data, msg, len);
+	return do_hw_write(codec, reg, value, &data, 2);
 }
-#else
-#define snd_soc_4_12_spi_write NULL
-#endif
 
 static unsigned int snd_soc_7_9_read(struct snd_soc_codec *codec,
 				     unsigned int reg)
@@ -140,21 +114,6 @@ static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg,
 	return do_hw_write(codec, reg, value, &data, 2);
 }
 
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_7_9_spi_write(void *control_data, const char *data,
-				 int len)
-{
-	u8 msg[2];
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	return do_spi_write(control_data, msg, len);
-}
-#else
-#define snd_soc_7_9_spi_write NULL
-#endif
-
 static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg,
 			     unsigned int value)
 {
@@ -173,21 +132,6 @@ static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec,
 	return do_hw_read(codec, reg);
 }
 
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_8_8_spi_write(void *control_data, const char *data,
-				 int len)
-{
-	u8 msg[2];
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-
-	return do_spi_write(control_data, msg, len);
-}
-#else
-#define snd_soc_8_8_spi_write NULL
-#endif
-
 static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg,
 			      unsigned int value)
 {
@@ -206,22 +150,6 @@ static unsigned int snd_soc_8_16_read(struct snd_soc_codec *codec,
 	return do_hw_read(codec, reg);
 }
 
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_8_16_spi_write(void *control_data, const char *data,
-				  int len)
-{
-	u8 msg[3];
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-	msg[2] = data[2];
-
-	return do_spi_write(control_data, msg, len);
-}
-#else
-#define snd_soc_8_16_spi_write NULL
-#endif
-
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 static unsigned int do_i2c_read(struct snd_soc_codec *codec,
 				void *reg, int reglen,
@@ -318,27 +246,10 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	memcpy(data, &rval, sizeof(rval));
 	data[2] = value;
-	reg &= 0xff;
 
 	return do_hw_write(codec, reg, value, data, 3);
 }
 
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_16_8_spi_write(void *control_data, const char *data,
-				  int len)
-{
-	u8 msg[3];
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-	msg[2] = data[2];
-
-	return do_spi_write(control_data, msg, len);
-}
-#else
-#define snd_soc_16_8_spi_write NULL
-#endif
-
 #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
 static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec,
 					   unsigned int r)
@@ -373,23 +284,6 @@ static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg,
 	return do_hw_write(codec, reg, value, data, sizeof(data));
 }
 
-#if defined(CONFIG_SPI_MASTER)
-static int snd_soc_16_16_spi_write(void *control_data, const char *data,
-				   int len)
-{
-	u8 msg[4];
-
-	msg[0] = data[0];
-	msg[1] = data[1];
-	msg[2] = data[2];
-	msg[3] = data[3];
-
-	return do_spi_write(control_data, msg, len);
-}
-#else
-#define snd_soc_16_16_spi_write NULL
-#endif
-
 /* Primitive bulk write support for soc-cache.  The data pointed to by
  * `data' needs to already be in the form the hardware expects
  * including any leading register specific data.  Any data written
@@ -419,7 +313,7 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r
 #endif
 #if defined(CONFIG_SPI_MASTER)
 	case SND_SOC_SPI:
-		ret = do_spi_write(codec->control_data, data, len);
+		ret = spi_write(codec->control_data, data, len);
 		break;
 #endif
 	default:
@@ -438,43 +332,36 @@ static struct {
 	int addr_bits;
 	int data_bits;
 	int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int);
-	int (*spi_write)(void *, const char *, int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int);
 } io_types[] = {
 	{
 		.addr_bits = 4, .data_bits = 12,
 		.write = snd_soc_4_12_write, .read = snd_soc_4_12_read,
-		.spi_write = snd_soc_4_12_spi_write,
 	},
 	{
 		.addr_bits = 7, .data_bits = 9,
 		.write = snd_soc_7_9_write, .read = snd_soc_7_9_read,
-		.spi_write = snd_soc_7_9_spi_write,
 	},
 	{
 		.addr_bits = 8, .data_bits = 8,
 		.write = snd_soc_8_8_write, .read = snd_soc_8_8_read,
 		.i2c_read = snd_soc_8_8_read_i2c,
-		.spi_write = snd_soc_8_8_spi_write,
 	},
 	{
 		.addr_bits = 8, .data_bits = 16,
 		.write = snd_soc_8_16_write, .read = snd_soc_8_16_read,
 		.i2c_read = snd_soc_8_16_read_i2c,
-		.spi_write = snd_soc_8_16_spi_write,
 	},
 	{
 		.addr_bits = 16, .data_bits = 8,
 		.write = snd_soc_16_8_write, .read = snd_soc_16_8_read,
 		.i2c_read = snd_soc_16_8_read_i2c,
-		.spi_write = snd_soc_16_8_spi_write,
 	},
 	{
 		.addr_bits = 16, .data_bits = 16,
 		.write = snd_soc_16_16_write, .read = snd_soc_16_16_read,
 		.i2c_read = snd_soc_16_16_read_i2c,
-		.spi_write = snd_soc_16_16_spi_write,
 	},
 };
 
@@ -535,8 +422,9 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
 		break;
 
 	case SND_SOC_SPI:
-		if (io_types[i].spi_write)
-			codec->hw_write = io_types[i].spi_write;
+#ifdef CONFIG_SPI_MASTER
+		codec->hw_write = do_spi_write;
+#endif
 
 		codec->control_data = container_of(codec->dev,
 						   struct spi_device,

sound/soc/soc-core.c

@@ -242,7 +242,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 		const char __user *user_buf, size_t count, loff_t *ppos)
 {
 	char buf[32];
-	int buf_size;
+	size_t buf_size;
 	char *start = buf;
 	unsigned long reg, value;
 	int step = 1;
@@ -1307,10 +1307,6 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num)
 	/* no, then find CPU DAI from registered DAIs*/
 	list_for_each_entry(cpu_dai, &dai_list, list) {
 		if (!strcmp(cpu_dai->name, dai_link->cpu_dai_name)) {
-
-			if (!try_module_get(cpu_dai->dev->driver->owner))
-				return -ENODEV;
-
 			rtd->cpu_dai = cpu_dai;
 			goto find_codec;
 		}
@@ -1623,11 +1619,15 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num)
 
 	/* probe the cpu_dai */
 	if (!cpu_dai->probed) {
+		if (!try_module_get(cpu_dai->dev->driver->owner))
+			return -ENODEV;
+
 		if (cpu_dai->driver->probe) {
 			ret = cpu_dai->driver->probe(cpu_dai);
 			if (ret < 0) {
 				printk(KERN_ERR "asoc: failed to probe CPU DAI %s\n",
 						cpu_dai->name);
+				module_put(cpu_dai->dev->driver->owner);
 				return ret;
 			}
 		}
@@ -1927,9 +1927,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card)
 					card->num_dapm_routes);
 
 	snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname),
-		 "%s",  card->name);
-	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
 		 "%s", card->name);
+	snprintf(card->snd_card->longname, sizeof(card->snd_card->longname),
+		 "%s", card->long_name ? card->long_name : card->name);
+	snprintf(card->snd_card->driver, sizeof(card->snd_card->driver),
+		 "%s", card->driver_name ? card->driver_name : card->name);
 
 	if (card->late_probe) {
 		ret = card->late_probe(card);

sound/soc/soc-dapm.c

@@ -325,6 +325,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm,
 }
 
 static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
+	struct snd_soc_dapm_widget *kcontrolw,
 	const struct snd_kcontrol_new *kcontrol_new,
 	struct snd_kcontrol **kcontrol)
 {
@@ -334,6 +335,8 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm,
 	*kcontrol = NULL;
 
 	list_for_each_entry(w, &dapm->card->widgets, list) {
+		if (w == kcontrolw || w->dapm != kcontrolw->dapm)
+			continue;
 		for (i = 0; i < w->num_kcontrols; i++) {
 			if (&w->kcontrol_news[i] == kcontrol_new) {
 				if (w->kcontrols)
@@ -468,7 +471,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm,
 		return -EINVAL;
 	}
 
-	shared = dapm_is_shared_kcontrol(dapm, &w->kcontrol_news[0],
+	shared = dapm_is_shared_kcontrol(dapm, w, &w->kcontrol_news[0],
 					 &kcontrol);
 	if (kcontrol) {
 		wlist = kcontrol->private_data;
@@ -1110,7 +1113,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event)
 	trace_snd_soc_dapm_start(card);
 
 	list_for_each_entry(d, &card->dapm_list, list)
-		if (d->n_widgets)
+		if (d->n_widgets || d->codec == NULL)
 			d->dev_power = 0;
 
 	/* Check which widgets we need to power and store them in

sound/soc/tegra/tegra_i2s.c

@@ -114,7 +114,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s)
 		debugfs_remove(i2s->debug);
 }
 #else
-static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s)
+static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id)
 {
 }