Merge remote branch 'broonie-asoc/for-2.6.37' into for-2.6.37

29 files changed, 2870 insertions, 87 deletions

diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index eabf66af12cd..5d230cee3fa7 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -789,13 +789,14 @@ static struct snd_soc_dai_driver atmel_ssc_dai[NUM_SSC_DEVICES] = {
 
 static __devinit int asoc_ssc_probe(struct platform_device *pdev)
 {
-	return snd_soc_register_dais(&pdev->dev, atmel_ssc_dai,
-			ARRAY_SIZE(atmel_ssc_dai));
+	BUG_ON(pdev->id < 0);
+	BUG_ON(pdev->id >= ARRAY_SIZE(atmel_ssc_dai));
+	return snd_soc_register_dai(&pdev->dev, &atmel_ssc_dai[pdev->id]);
 }
 
 static int __devexit asoc_ssc_remove(struct platform_device *pdev)
 {
-	snd_soc_unregister_dais(&pdev->dev, ARRAY_SIZE(atmel_ssc_dai));
+	snd_soc_unregister_dai(&pdev->dev);
 	return 0;
 }
 
@@ -809,6 +810,56 @@ static struct platform_driver asoc_ssc_driver = {
 	.remove = __devexit_p(asoc_ssc_remove),
 };
 
+/**
+ * atmel_ssc_set_audio - Allocate the specified SSC for audio use.
+ */
+int atmel_ssc_set_audio(int ssc_id)
+{
+	struct ssc_device *ssc;
+	static struct platform_device *dma_pdev;
+	struct platform_device *ssc_pdev;
+	int ret;
+
+	if (ssc_id < 0 || ssc_id >= ARRAY_SIZE(atmel_ssc_dai))
+		return -EINVAL;
+
+	/* Allocate a dummy device for DMA if we don't have one already */
+	if (!dma_pdev) {
+		dma_pdev = platform_device_alloc("atmel-pcm-audio", -1);
+		if (!dma_pdev)
+			return -ENOMEM;
+
+		ret = platform_device_add(dma_pdev);
+		if (ret < 0) {
+			platform_device_put(dma_pdev);
+			dma_pdev = NULL;
+			return ret;
+		}
+	}
+
+	ssc_pdev = platform_device_alloc("atmel-ssc-dai", ssc_id);
+	if (!ssc_pdev) {
+		ssc_free(ssc);
+		return -ENOMEM;
+	}
+
+	/* If we can grab the SSC briefly to parent the DAI device off it */
+	ssc = ssc_request(ssc_id);
+	if (IS_ERR(ssc))
+		pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n",
+			PTR_ERR(ssc));
+	else
+		ssc_pdev->dev.parent = &(ssc->pdev->dev);
+	ssc_free(ssc);
+
+	ret = platform_device_add(ssc_pdev);
+	if (ret < 0)
+		platform_device_put(ssc_pdev);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(atmel_ssc_set_audio);
+
 static int __init snd_atmel_ssc_init(void)
 {
 	return platform_driver_register(&asoc_ssc_driver);
diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h
index 392a46953112..5d4f0f9b4d9a 100644
--- a/sound/soc/atmel/atmel_ssc_dai.h
+++ b/sound/soc/atmel/atmel_ssc_dai.h
@@ -117,4 +117,6 @@ struct atmel_ssc_info {
 	struct atmel_ssc_state ssc_state;
 };
 
+int atmel_ssc_set_audio(int ssc);
+
 #endif /* _AT91_SSC_DAI_H */
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 66a6f1879689..293569dfd0ed 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,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
 		MCLK_RATE, SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		printk(KERN_ERR "Failed to set WM8731 SYSCLK: %d\n", ret);
@@ -183,8 +183,8 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 	.cpu_dai_name = "atmel-ssc-dai.0",
 	.codec_dai_name = "wm8731-hifi",
 	.init = at91sam9g20ek_wm8731_init,
-	.platform_name = "atmel_pcm-audio",
-	.codec_name = "wm8731-codec.0-001a",
+	.platform_name = "atmel-pcm-audio",
+	.codec_name = "wm8731-codec.0-001b",
 	.ops = &at91sam9g20ek_ops,
 };
 
@@ -205,6 +205,12 @@ static int __init at91sam9g20ek_init(void)
 	if (!(machine_is_at91sam9g20ek() || machine_is_at91sam9g20ek_2mmc()))
 		return -ENODEV;
 
+	ret = atmel_ssc_set_audio(0);
+	if (ret != 0) {
+		pr_err("Failed to set SSC 0 for audio: %d\n", ret);
+		return ret;
+	}
+
 	/*
 	 * Codec MCLK is supplied by PCK0 - set it up.
 	 */
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 8780c90107fc..d8dc8225576a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -49,7 +49,7 @@ static int db1200_i2s_startup(struct snd_pcm_substream *substream)
 	int ret;
 
 	/* WM8731 has its own 12MHz crystal */
-	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+	snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL,
 				12000000, SND_SOC_CLOCK_IN);
 
 	/* codec is bitclock and lrclk master */
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
new file mode 100644
index 000000000000..01d19e9f53f9
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -0,0 +1,1486 @@
+/*
+ * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ * Author: Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/i2c.h>
+#include <linux/platform_device.h>
+#include <linux/mfd/88pm860x.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/tlv.h>
+#include <sound/initval.h>
+#include <sound/jack.h>
+
+#include "88pm860x-codec.h"
+
+#define MAX_NAME_LEN		20
+#define REG_CACHE_SIZE		0x40
+#define REG_CACHE_BASE		0xb0
+
+/* Status Register 1 (0x01) */
+#define REG_STATUS_1		0x01
+#define MIC_STATUS		(1 << 7)
+#define HOOK_STATUS		(1 << 6)
+#define HEADSET_STATUS		(1 << 5)
+
+/* Mic Detection Register (0x37) */
+#define REG_MIC_DET		0x37
+#define CONTINUOUS_POLLING	(3 << 1)
+#define EN_MIC_DET		(1 << 0)
+#define MICDET_MASK		0x07
+
+/* Headset Detection Register (0x38) */
+#define REG_HS_DET		0x38
+#define EN_HS_DET		(1 << 0)
+
+/* Misc2 Register (0x42) */
+#define REG_MISC2		0x42
+#define AUDIO_PLL		(1 << 5)
+#define AUDIO_SECTION_RESET	(1 << 4)
+#define AUDIO_SECTION_ON	(1 << 3)
+
+/* PCM Interface Register 2 (0xb1) */
+#define PCM_INF2_BCLK		(1 << 6)	/* Bit clock polarity */
+#define PCM_INF2_FS		(1 << 5)	/* Frame Sync polarity */
+#define PCM_INF2_MASTER		(1 << 4)	/* Master / Slave */
+#define PCM_INF2_18WL		(1 << 3)	/* 18 / 16 bits */
+#define PCM_GENERAL_I2S		0
+#define PCM_EXACT_I2S		1
+#define PCM_LEFT_I2S		2
+#define PCM_RIGHT_I2S		3
+#define PCM_SHORT_FS		4
+#define PCM_LONG_FS		5
+#define PCM_MODE_MASK		7
+
+/* I2S Interface Register 4 (0xbe) */
+#define I2S_EQU_BYP		(1 << 6)
+
+/* DAC Offset Register (0xcb) */
+#define DAC_MUTE		(1 << 7)
+#define MUTE_LEFT		(1 << 6)
+#define MUTE_RIGHT		(1 << 2)
+
+/* ADC Analog Register 1 (0xd0) */
+#define REG_ADC_ANA_1		0xd0
+#define MIC1BIAS_MASK		0x60
+
+/* Earpiece/Speaker Control Register 2 (0xda) */
+#define REG_EAR2		0xda
+#define RSYNC_CHANGE		(1 << 2)
+
+/* Audio Supplies Register 2 (0xdc) */
+#define REG_SUPPLIES2		0xdc
+#define LDO15_READY		(1 << 4)
+#define LDO15_EN		(1 << 3)
+#define CPUMP_READY		(1 << 2)
+#define CPUMP_EN		(1 << 1)
+#define AUDIO_EN		(1 << 0)
+#define SUPPLY_MASK		(LDO15_EN | CPUMP_EN | AUDIO_EN)
+
+/* Audio Enable Register 1 (0xdd) */
+#define ADC_MOD_RIGHT		(1 << 1)
+#define ADC_MOD_LEFT		(1 << 0)
+
+/* Audio Enable Register 2 (0xde) */
+#define ADC_LEFT		(1 << 5)
+#define ADC_RIGHT		(1 << 4)
+
+/* DAC Enable Register 2 (0xe1) */
+#define DAC_LEFT		(1 << 5)
+#define DAC_RIGHT		(1 << 4)
+#define MODULATOR		(1 << 3)
+
+/* Shorts Register (0xeb) */
+#define REG_SHORTS		0xeb
+#define CLR_SHORT_LO2		(1 << 7)
+#define SHORT_LO2		(1 << 6)
+#define CLR_SHORT_LO1		(1 << 5)
+#define SHORT_LO1		(1 << 4)
+#define CLR_SHORT_HS2		(1 << 3)
+#define SHORT_HS2		(1 << 2)
+#define CLR_SHORT_HS1		(1 << 1)
+#define SHORT_HS1		(1 << 0)
+
+/*
+ * This widget should be just after DAC & PGA in DAPM power-on sequence and
+ * before DAC & PGA in DAPM power-off sequence.
+ */
+#define PM860X_DAPM_OUTPUT(wname, wevent)	\
+{	.id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
+	.shift = 0, .invert = 0, .kcontrols = NULL, \
+	.num_kcontrols = 0, .event = wevent, \
+	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }
+
+struct pm860x_det {
+	struct snd_soc_jack	*hp_jack;
+	struct snd_soc_jack	*mic_jack;
+	int			hp_det;
+	int			mic_det;
+	int			hook_det;
+	int			hs_shrt;
+	int			lo_shrt;
+};
+
+struct pm860x_priv {
+	unsigned int		sysclk;
+	unsigned int		pcmclk;
+	unsigned int		dir;
+	unsigned int		filter;
+	struct snd_soc_codec	*codec;
+	struct i2c_client	*i2c;
+	struct pm860x_chip	*chip;
+	struct pm860x_det	det;
+
+	int			irq[4];
+	unsigned char		name[4][MAX_NAME_LEN];
+	unsigned char		reg_cache[REG_CACHE_SIZE];
+};
+
+/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
+static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);
+
+/* -9dB to 0db in 3dB steps */
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);
+
+/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
+static const unsigned int mic_tlv[] = {
+	TLV_DB_RANGE_HEAD(5),
+	0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
+	1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
+	2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
+	3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
+	4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
+};
+
+/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
+static const unsigned int aux_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
+	3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
+};
+
+/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
+static const unsigned int out_tlv[] = {
+	TLV_DB_RANGE_HEAD(4),
+	0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
+	4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
+	5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
+	6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
+};
+
+static const unsigned int st_tlv[] = {
+	TLV_DB_RANGE_HEAD(8),
+	0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
+	2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
+	4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
+	6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
+	8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
+	10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
+	14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
+	18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
+};
+
+/* Sidetone Gain = M * 2^(-5-N) */
+struct st_gain {
+	unsigned int	db;
+	unsigned int	m;
+	unsigned int	n;
+};
+
+static struct st_gain st_table[] = {
+	{-12041,  1, 15}, {-11439,  1, 14}, {-11087,  3, 15}, {-10837,  1, 13},
+	{-10643,  5, 15}, {-10485,  3, 14}, {-10351,  7, 15}, {-10235,  1, 12},
+	{-10133,  9, 15}, {-10041,  5, 14}, { -9958, 11, 15}, { -9883,  3, 13},
+	{ -9813, 13, 15}, { -9749,  7, 14}, { -9689, 15, 15}, { -9633,  1, 11},
+	{ -9580, 17, 15}, { -9531,  9, 14}, { -9484, 19, 15}, { -9439,  5, 13},
+	{ -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281,  3, 12},
+	{ -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147,  7, 13},
+	{ -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031,  1, 10},
+	{ -8978, 17, 14}, { -8929,  9, 13}, { -8882, 19, 14}, { -8837,  5, 12},
+	{ -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679,  3, 11},
+	{ -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545,  7, 12},
+	{ -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429,  1,  9},
+	{ -8376, 17, 13}, { -8327,  9, 12}, { -8280, 19, 13}, { -8235,  5, 11},
+	{ -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077,  3, 10},
+	{ -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943,  7, 11},
+	{ -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827,  1,  8},
+	{ -7774, 17, 12}, { -7724,  9, 11}, { -7678, 19, 12}, { -7633,  5, 10},
+	{ -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475,  3,  9},
+	{ -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341,  7, 10},
+	{ -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225,  1,  7},
+	{ -7172, 17, 11}, { -7122,  9, 10}, { -7075, 19, 11}, { -7031,  5,  9},
+	{ -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873,  3,  8},
+	{ -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739,  7,  9},
+	{ -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623,  1,  6},
+	{ -6570, 17, 10}, { -6520,  9,  9}, { -6473, 19, 10}, { -6429,  5,  8},
+	{ -6386, 21, 10}, { -6346, 11,  9}, { -6307, 23, 10}, { -6270,  3,  7},
+	{ -6235, 25, 10}, { -6201, 13,  9}, { -6168, 27, 10}, { -6137,  7,  8},
+	{ -6106, 29, 10}, { -6077, 15,  9}, { -6048, 31, 10}, { -6021,  1,  5},
+	{ -5968, 17,  9}, { -5918,  9,  8}, { -5871, 19,  9}, { -5827,  5,  7},
+	{ -5784, 21,  9}, { -5744, 11,  8}, { -5705, 23,  9}, { -5668,  3,  6},
+	{ -5633, 25,  9}, { -5599, 13,  8}, { -5566, 27,  9}, { -5535,  7,  7},
+	{ -5504, 29,  9}, { -5475, 15,  8}, { -5446, 31,  9}, { -5419,  1,  4},
+	{ -5366, 17,  8}, { -5316,  9,  7}, { -5269, 19,  8}, { -5225,  5,  6},
+	{ -5182, 21,  8}, { -5142, 11,  7}, { -5103, 23,  8}, { -5066,  3,  5},
+	{ -5031, 25,  8}, { -4997, 13,  7}, { -4964, 27,  8}, { -4932,  7,  6},
+	{ -4902, 29,  8}, { -4873, 15,  7}, { -4844, 31,  8}, { -4816,  1,  3},
+	{ -4764, 17,  7}, { -4714,  9,  6}, { -4667, 19,  7}, { -4623,  5,  5},
+	{ -4580, 21,  7}, { -4540, 11,  6}, { -4501, 23,  7}, { -4464,  3,  4},
+	{ -4429, 25,  7}, { -4395, 13,  6}, { -4362, 27,  7}, { -4330,  7,  5},
+	{ -4300, 29,  7}, { -4270, 15,  6}, { -4242, 31,  7}, { -4214,  1,  2},
+	{ -4162, 17,  6}, { -4112,  9,  5}, { -4065, 19,  6}, { -4021,  5,  4},
+	{ -3978, 21,  6}, { -3938, 11,  5}, { -3899, 23,  6}, { -3862,  3,  3},
+	{ -3827, 25,  6}, { -3793, 13,  5}, { -3760, 27,  6}, { -3728,  7,  4},
+	{ -3698, 29,  6}, { -3668, 15,  5}, { -3640, 31,  6}, { -3612,  1,  1},
+	{ -3560, 17,  5}, { -3510,  9,  4}, { -3463, 19,  5}, { -3419,  5,  3},
+	{ -3376, 21,  5}, { -3336, 11,  4}, { -3297, 23,  5}, { -3260,  3,  2},
+	{ -3225, 25,  5}, { -3191, 13,  4}, { -3158, 27,  5}, { -3126,  7,  3},
+	{ -3096, 29,  5}, { -3066, 15,  4}, { -3038, 31,  5}, { -3010,  1,  0},
+	{ -2958, 17,  4}, { -2908,  9,  3}, { -2861, 19,  4}, { -2816,  5,  2},
+	{ -2774, 21,  4}, { -2734, 11,  3}, { -2695, 23,  4}, { -2658,  3,  1},
+	{ -2623, 25,  4}, { -2589, 13,  3}, { -2556, 27,  4}, { -2524,  7,  2},
+	{ -2494, 29,  4}, { -2464, 15,  3}, { -2436, 31,  4}, { -2408,  2,  0},
+	{ -2356, 17,  3}, { -2306,  9,  2}, { -2259, 19,  3}, { -2214,  5,  1},
+	{ -2172, 21,  3}, { -2132, 11,  2}, { -2093, 23,  3}, { -2056,  3,  0},
+	{ -2021, 25,  3}, { -1987, 13,  2}, { -1954, 27,  3}, { -1922,  7,  1},
+	{ -1892, 29,  3}, { -1862, 15,  2}, { -1834, 31,  3}, { -1806,  4,  0},
+	{ -1754, 17,  2}, { -1704,  9,  1}, { -1657, 19,  2}, { -1612,  5,  0},
+	{ -1570, 21,  2}, { -1530, 11,  1}, { -1491, 23,  2}, { -1454,  6,  0},
+	{ -1419, 25,  2}, { -1384, 13,  1}, { -1352, 27,  2}, { -1320,  7,  0},
+	{ -1290, 29,  2}, { -1260, 15,  1}, { -1232, 31,  2}, { -1204,  8,  0},
+	{ -1151, 17,  1}, { -1102,  9,  0}, { -1055, 19,  1}, { -1010, 10,  0},
+	{  -968, 21,  1}, {  -928, 11,  0}, {  -889, 23,  1}, {  -852, 12,  0},
+	{  -816, 25,  1}, {  -782, 13,  0}, {  -750, 27,  1}, {  -718, 14,  0},
+	{  -688, 29,  1}, {  -658, 15,  0}, {  -630, 31,  1}, {  -602, 16,  0},
+	{  -549, 17,  0}, {  -500, 18,  0}, {  -453, 19,  0}, {  -408, 20,  0},
+	{  -366, 21,  0}, {  -325, 22,  0}, {  -287, 23,  0}, {  -250, 24,  0},
+	{  -214, 25,  0}, {  -180, 26,  0}, {  -148, 27,  0}, {  -116, 28,  0},
+	{   -86, 29,  0}, {   -56, 30,  0}, {   -28, 31,  0}, {     0,  0,  0},
+};
+
+static int pm860x_volatile(unsigned int reg)
+{
+	BUG_ON(reg >= REG_CACHE_SIZE);
+
+	switch (reg) {
+	case PM860X_AUDIO_SUPPLIES_2:
+		return 1;
+	}
+
+	return 0;
+}
+
+static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
+					  unsigned int reg)
+{
+	unsigned char *cache = codec->reg_cache;
+
+	BUG_ON(reg >= REG_CACHE_SIZE);
+
+	if (pm860x_volatile(reg))
+		return cache[reg];
+
+	reg += REG_CACHE_BASE;
+
+	return pm860x_reg_read(codec->control_data, reg);
+}
+
+static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
+				  unsigned int reg, unsigned int value)
+{
+	unsigned char *cache = codec->reg_cache;
+
+	BUG_ON(reg >= REG_CACHE_SIZE);
+
+	if (!pm860x_volatile(reg))
+		cache[reg] = (unsigned char)value;
+
+	reg += REG_CACHE_BASE;
+
+	return pm860x_reg_write(codec->control_data, reg, value);
+}
+
+static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	int val[2], val2[2], i;
+
+	val[0] = snd_soc_read(codec, reg) & 0x3f;
+	val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
+	val2[0] = snd_soc_read(codec, reg2) & 0x3f;
+	val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;
+
+	for (i = 0; i < ARRAY_SIZE(st_table); i++) {
+		if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
+			ucontrol->value.integer.value[0] = i;
+		if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
+			ucontrol->value.integer.value[1] = i;
+	}
+	return 0;
+}
+
+static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	int err;
+	unsigned int val, val2;
+
+	val = ucontrol->value.integer.value[0];
+	val2 = ucontrol->value.integer.value[1];
+
+	err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
+	if (err < 0)
+		return err;
+	err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
+				  st_table[val].n << 4);
+	if (err < 0)
+		return err;
+
+	err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
+	if (err < 0)
+		return err;
+	err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
+				  st_table[val2].n);
+	return err;
+}
+
+static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max, val, val2;
+	unsigned int mask = (1 << fls(max)) - 1;
+
+	val = snd_soc_read(codec, reg) >> shift;
+	val2 = snd_soc_read(codec, reg2) >> shift;
+	ucontrol->value.integer.value[0] = (max - val) & mask;
+	ucontrol->value.integer.value[1] = (max - val2) & mask;
+
+	return 0;
+}
+
+static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
+	unsigned int shift = mc->shift;
+	int max = mc->max;
+	unsigned int mask = (1 << fls(max)) - 1;
+	int err;
+	unsigned int val, val2, val_mask;
+
+	val_mask = mask << shift;
+	val = ((max - ucontrol->value.integer.value[0]) & mask);
+	val2 = ((max - ucontrol->value.integer.value[1]) & mask);
+
+	val = val << shift;
+	val2 = val2 << shift;
+
+	err = snd_soc_update_bits(codec, reg, val_mask, val);
+	if (err < 0)
+		return err;
+
+	err = snd_soc_update_bits(codec, reg2, val_mask, val2);
+	return err;
+}
+
+/* DAPM Widget Events */
+/*
+ * A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
+ * after updating these registers. Otherwise, these updated registers won't
+ * be effective.
+ */
+static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
+			      struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	/*
+	 * In order to avoid current on the load, mute power-on and power-off
+	 * should be transients.
+	 * Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
+	 * finished.
+	 */
+	snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
+			    struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	unsigned int dac = 0;
+	int data;
+
+	if (!strcmp(w->name, "Left DAC"))
+		dac = DAC_LEFT;
+	if (!strcmp(w->name, "Right DAC"))
+		dac = DAC_RIGHT;
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (dac) {
+			/* Auto mute in power-on sequence. */
+			dac |= MODULATOR;
+			snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
+					    DAC_MUTE, DAC_MUTE);
+			snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+					    RSYNC_CHANGE, RSYNC_CHANGE);
+			/* update dac */
+			snd_soc_update_bits(codec, PM860X_DAC_EN_2,
+					    dac, dac);
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		if (dac) {
+			/* Auto mute in power-off sequence. */
+			snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
+					    DAC_MUTE, DAC_MUTE);
+			snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+					    RSYNC_CHANGE, RSYNC_CHANGE);
+			/* update dac */
+			data = snd_soc_read(codec, PM860X_DAC_EN_2);
+			data &= ~dac;
+			if (!(data & (DAC_LEFT | DAC_RIGHT)))
+				data &= ~MODULATOR;
+			snd_soc_write(codec, PM860X_DAC_EN_2, data);
+		}
+		break;
+	}
+	return 0;
+}
+
+static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
+
+static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
+
+static const struct soc_enum pm860x_hs1_opamp_enum =
+	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_hs2_opamp_enum =
+	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_hs1_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_hs2_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_lo1_opamp_enum =
+	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_lo2_opamp_enum =
+	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
+
+static const struct soc_enum pm860x_lo1_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_lo2_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_spk_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_ear_pa_enum =
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
+
+static const struct soc_enum pm860x_spk_ear_opamp_enum =
+	SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
+
+static const struct snd_kcontrol_new pm860x_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
+			PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
+	SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
+			aux_tlv),
+	SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
+			mic_tlv),
+	SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
+			mic_tlv),
+	SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
+			     PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
+			     0, snd_soc_get_volsw_2r_st,
+			     snd_soc_put_volsw_2r_st, st_tlv),
+	SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
+			0, 7, 0, out_tlv),
+	SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
+			 PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
+	SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
+			 PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
+	SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
+			     PM860X_HIFIL_GAIN_LEFT,
+			     PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
+			     snd_soc_get_volsw_2r_out,
+			     snd_soc_put_volsw_2r_out, dpga_tlv),
+	SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
+			     PM860X_HIFIR_GAIN_LEFT,
+			     PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
+			     snd_soc_get_volsw_2r_out,
+			     snd_soc_put_volsw_2r_out, dpga_tlv),
+	SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
+			     PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
+			     snd_soc_get_volsw_2r_out,
+			     snd_soc_put_volsw_2r_out, dpga_tlv),
+	SOC_ENUM("Headset1 Operational Amplifier Current",
+		 pm860x_hs1_opamp_enum),
+	SOC_ENUM("Headset2 Operational Amplifier Current",
+		 pm860x_hs2_opamp_enum),
+	SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
+	SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
+	SOC_ENUM("Lineout1 Operational Amplifier Current",
+		 pm860x_lo1_opamp_enum),
+	SOC_ENUM("Lineout2 Operational Amplifier Current",
+		 pm860x_lo2_opamp_enum),
+	SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
+	SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
+	SOC_ENUM("Speaker Operational Amplifier Current",
+		 pm860x_spk_ear_opamp_enum),
+	SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
+	SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
+};
+
+/*
+ * DAPM Controls
+ */
+
+/* PCM Switch / PCM Interface */
+static const struct snd_kcontrol_new pcm_switch_controls =
+	SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
+
+/* AUX1 Switch */
+static const struct snd_kcontrol_new aux1_switch_controls =
+	SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
+
+/* AUX2 Switch */
+static const struct snd_kcontrol_new aux2_switch_controls =
+	SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);
+
+/* Left Ex. PA Switch */
+static const struct snd_kcontrol_new lepa_switch_controls =
+	SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);
+
+/* Right Ex. PA Switch */
+static const struct snd_kcontrol_new repa_switch_controls =
+	SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
+
+/* PCM Mux / Mux7 */
+static const char *aif1_text[] = {
+	"PCM L", "PCM R",
+};
+
+static const struct soc_enum aif1_enum =
+	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
+
+static const struct snd_kcontrol_new aif1_mux =
+	SOC_DAPM_ENUM("PCM Mux", aif1_enum);
+
+/* I2S Mux / Mux9 */
+static const char *i2s_din_text[] = {
+	"DIN", "DIN1",
+};
+
+static const struct soc_enum i2s_din_enum =
+	SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
+
+static const struct snd_kcontrol_new i2s_din_mux =
+	SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
+
+/* I2S Mic Mux / Mux8 */
+static const char *i2s_mic_text[] = {
+	"Ex PA", "ADC",
+};
+
+static const struct soc_enum i2s_mic_enum =
+	SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
+
+static const struct snd_kcontrol_new i2s_mic_mux =
+	SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
+
+/* ADCL Mux / Mux2 */
+static const char *adcl_text[] = {
+	"ADCR", "ADCL",
+};
+
+static const struct soc_enum adcl_enum =
+	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
+
+static const struct snd_kcontrol_new adcl_mux =
+	SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
+
+/* ADCR Mux / Mux3 */
+static const char *adcr_text[] = {
+	"ADCL", "ADCR",
+};
+
+static const struct soc_enum adcr_enum =
+	SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
+
+static const struct snd_kcontrol_new adcr_mux =
+	SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
+
+/* ADCR EC Mux / Mux6 */
+static const char *adcr_ec_text[] = {
+	"ADCR", "EC",
+};
+
+static const struct soc_enum adcr_ec_enum =
+	SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
+
+static const struct snd_kcontrol_new adcr_ec_mux =
+	SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
+
+/* EC Mux / Mux4 */
+static const char *ec_text[] = {
+	"Left", "Right", "Left + Right",
+};
+
+static const struct soc_enum ec_enum =
+	SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
+
+static const struct snd_kcontrol_new ec_mux =
+	SOC_DAPM_ENUM("EC Mux", ec_enum);
+
+static const char *dac_text[] = {
+	"No input", "Right", "Left", "No input",
+};
+
+/* DAC Headset 1 Mux / Mux10 */
+static const struct soc_enum dac_hs1_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_hs1_mux =
+	SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
+
+/* DAC Headset 2 Mux / Mux11 */
+static const struct soc_enum dac_hs2_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_hs2_mux =
+	SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
+
+/* DAC Lineout 1 Mux / Mux12 */
+static const struct soc_enum dac_lo1_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_lo1_mux =
+	SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
+
+/* DAC Lineout 2 Mux / Mux13 */
+static const struct soc_enum dac_lo2_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_lo2_mux =
+	SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
+
+/* DAC Spearker Earphone Mux / Mux14 */
+static const struct soc_enum dac_spk_ear_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
+
+static const struct snd_kcontrol_new dac_spk_ear_mux =
+	SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
+
+/* Headset 1 Mux / Mux15 */
+static const char *in_text[] = {
+	"Digital", "Analog",
+};
+
+static const struct soc_enum hs1_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
+
+static const struct snd_kcontrol_new hs1_mux =
+	SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
+
+/* Headset 2 Mux / Mux16 */
+static const struct soc_enum hs2_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
+
+static const struct snd_kcontrol_new hs2_mux =
+	SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
+
+/* Lineout 1 Mux / Mux17 */
+static const struct soc_enum lo1_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
+
+static const struct snd_kcontrol_new lo1_mux =
+	SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
+
+/* Lineout 2 Mux / Mux18 */
+static const struct soc_enum lo2_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
+
+static const struct snd_kcontrol_new lo2_mux =
+	SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
+
+/* Speaker Earpiece Demux */
+static const char *spk_text[] = {
+	"Earpiece", "Speaker",
+};
+
+static const struct soc_enum spk_enum =
+	SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
+
+static const struct snd_kcontrol_new spk_demux =
+	SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
+
+/* MIC Mux / Mux1 */
+static const char *mic_text[] = {
+	"Mic 1", "Mic 2",
+};
+
+static const struct soc_enum mic_enum =
+	SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
+
+static const struct snd_kcontrol_new mic_mux =
+	SOC_DAPM_ENUM("MIC Mux", mic_enum);
+
+static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
+	SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
+			    PM860X_ADC_EN_2, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
+			     PM860X_PCM_IFACE_3, 1, 1),
+
+
+	SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
+			    PM860X_DAC_EN_2, 0, 0),
+	SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
+			    PM860X_DAC_EN_2, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
+			     PM860X_I2S_IFACE_3, 5, 1),
+	SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
+	SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
+	SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
+	SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
+	SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
+	SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
+			    &lepa_switch_controls),
+	SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
+			    &repa_switch_controls),
+
+	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
+			 0, 1, 1, 0),
+	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
+			 1, 1, 1, 0),
+	SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
+	SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),
+
+	SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
+			    &aux1_switch_controls),
+	SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
+			    &aux2_switch_controls),
+
+	SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
+	SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
+	SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
+	SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),
+
+	SND_SOC_DAPM_INPUT("AUX1"),
+	SND_SOC_DAPM_INPUT("AUX2"),
+	SND_SOC_DAPM_INPUT("MIC1P"),
+	SND_SOC_DAPM_INPUT("MIC1N"),
+	SND_SOC_DAPM_INPUT("MIC2P"),
+	SND_SOC_DAPM_INPUT("MIC2N"),
+	SND_SOC_DAPM_INPUT("MIC3P"),
+	SND_SOC_DAPM_INPUT("MIC3N"),
+
+	SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
+			   pm860x_dac_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
+			   pm860x_dac_event,
+			   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
+
+	SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
+	SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
+	SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
+	SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
+	SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
+	SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
+	SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
+	SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
+	SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
+	SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
+	SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
+			 &spk_demux),
+
+
+	SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HS1"),
+	SND_SOC_DAPM_OUTPUT("HS2"),
+	SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("LINEOUT1"),
+	SND_SOC_DAPM_OUTPUT("LINEOUT2"),
+	SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("EARP"),
+	SND_SOC_DAPM_OUTPUT("EARN"),
+	SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("LSP"),
+	SND_SOC_DAPM_OUTPUT("LSN"),
+	SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
+			 0, SUPPLY_MASK, SUPPLY_MASK, 0),
+
+	PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* supply */
+	{"Left DAC", NULL, "VCODEC"},
+	{"Right DAC", NULL, "VCODEC"},
+	{"Left ADC", NULL, "VCODEC"},
+	{"Right ADC", NULL, "VCODEC"},
+	{"Left ADC", NULL, "Left ADC MOD"},
+	{"Right ADC", NULL, "Right ADC MOD"},
+
+	/* PCM/AIF1 Inputs */
+	{"PCM SDO", NULL, "ADC Left Mux"},
+	{"PCM SDO", NULL, "ADCR EC Mux"},
+
+	/* PCM/AFI2 Outputs */
+	{"Lofi PGA", NULL, "PCM SDI"},
+	{"Lofi PGA", NULL, "Sidetone PGA"},
+	{"Left DAC", NULL, "Lofi PGA"},
+	{"Right DAC", NULL, "Lofi PGA"},
+
+	/* I2S/AIF2 Inputs */
+	{"MIC Mux", "Mic 1", "MIC1P"},
+	{"MIC Mux", "Mic 1", "MIC1N"},
+	{"MIC Mux", "Mic 2", "MIC2P"},
+	{"MIC Mux", "Mic 2", "MIC2N"},
+	{"MIC1 Volume", NULL, "MIC Mux"},
+	{"MIC3 Volume", NULL, "MIC3P"},
+	{"MIC3 Volume", NULL, "MIC3N"},
+	{"Left ADC", NULL, "MIC1 Volume"},
+	{"Right ADC", NULL, "MIC3 Volume"},
+	{"ADC Left Mux", "ADCR", "Right ADC"},
+	{"ADC Left Mux", "ADCL", "Left ADC"},
+	{"ADC Right Mux", "ADCL", "Left ADC"},
+	{"ADC Right Mux", "ADCR", "Right ADC"},
+	{"Left EPA", "Switch", "Left DAC"},
+	{"Right EPA", "Switch", "Right DAC"},
+	{"EC Mux", "Left", "Left DAC"},
+	{"EC Mux", "Right", "Right DAC"},
+	{"EC Mux", "Left + Right", "Left DAC"},
+	{"EC Mux", "Left + Right", "Right DAC"},
+	{"ADCR EC Mux", "ADCR", "ADC Right Mux"},
+	{"ADCR EC Mux", "EC", "EC Mux"},
+	{"I2S Mic Mux", "Ex PA", "Left EPA"},
+	{"I2S Mic Mux", "Ex PA", "Right EPA"},
+	{"I2S Mic Mux", "ADC", "ADC Left Mux"},
+	{"I2S Mic Mux", "ADC", "ADCR EC Mux"},
+	{"I2S DOUT", NULL, "I2S Mic Mux"},
+
+	/* I2S/AIF2 Outputs */
+	{"I2S DIN Mux", "DIN", "I2S DIN"},
+	{"I2S DIN Mux", "DIN1", "I2S DIN1"},
+	{"Left DAC", NULL, "I2S DIN Mux"},
+	{"Right DAC", NULL, "I2S DIN Mux"},
+	{"DAC HS1 Mux", "Left", "Left DAC"},
+	{"DAC HS1 Mux", "Right", "Right DAC"},
+	{"DAC HS2 Mux", "Left", "Left DAC"},
+	{"DAC HS2 Mux", "Right", "Right DAC"},
+	{"DAC LO1 Mux", "Left", "Left DAC"},
+	{"DAC LO1 Mux", "Right", "Right DAC"},
+	{"DAC LO2 Mux", "Left", "Left DAC"},
+	{"DAC LO2 Mux", "Right", "Right DAC"},
+	{"Headset1 Mux", "Digital", "DAC HS1 Mux"},
+	{"Headset2 Mux", "Digital", "DAC HS2 Mux"},
+	{"Lineout1 Mux", "Digital", "DAC LO1 Mux"},
+	{"Lineout2 Mux", "Digital", "DAC LO2 Mux"},
+	{"Headset1 PGA", NULL, "Headset1 Mux"},
+	{"Headset2 PGA", NULL, "Headset2 Mux"},
+	{"Lineout1 PGA", NULL, "Lineout1 Mux"},
+	{"Lineout2 PGA", NULL, "Lineout2 Mux"},
+	{"DAC SP Mux", "Left", "Left DAC"},
+	{"DAC SP Mux", "Right", "Right DAC"},
+	{"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"},
+	{"Speaker PGA", NULL, "Speaker Earpiece Demux"},
+	{"Earpiece PGA", NULL, "Speaker Earpiece Demux"},
+
+	{"RSYNC", NULL, "Headset1 PGA"},
+	{"RSYNC", NULL, "Headset2 PGA"},
+	{"RSYNC", NULL, "Lineout1 PGA"},
+	{"RSYNC", NULL, "Lineout2 PGA"},
+	{"RSYNC", NULL, "Speaker PGA"},
+	{"RSYNC", NULL, "Speaker PGA"},
+	{"RSYNC", NULL, "Earpiece PGA"},
+	{"RSYNC", NULL, "Earpiece PGA"},
+
+	{"HS1", NULL, "RSYNC"},
+	{"HS2", NULL, "RSYNC"},
+	{"LINEOUT1", NULL, "RSYNC"},
+	{"LINEOUT2", NULL, "RSYNC"},
+	{"LSP", NULL, "RSYNC"},
+	{"LSN", NULL, "RSYNC"},
+	{"EARP", NULL, "RSYNC"},
+	{"EARN", NULL, "RSYNC"},
+};
+
+/*
+ * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
+ * These bits can also be used to mute.
+ */
+static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
+
+	if (mute)
+		data = mask;
+	snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data);
+	snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
+			    RSYNC_CHANGE, RSYNC_CHANGE);
+	return 0;
+}
+
+static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned char inf = 0, mask = 0;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		inf &= ~PCM_INF2_18WL;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		inf |= PCM_INF2_18WL;
+		break;
+	default:
+		return -EINVAL;
+	}
+	mask |= PCM_INF2_18WL;
+	snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
+
+	/* sample rate */
+	switch (params_rate(params)) {
+	case 8000:
+		inf = 0;
+		break;
+	case 16000:
+		inf = 3;
+		break;
+	case 32000:
+		inf = 6;
+		break;
+	case 48000:
+		inf = 8;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf);
+
+	return 0;
+}
+
+static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				  unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	unsigned char inf = 0, mask = 0;
+	int ret = -EINVAL;
+
+	mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+	case SND_SOC_DAIFMT_CBM_CFS:
+		if (pm860x->dir == PM860X_CLK_DIR_OUT) {
+			inf |= PCM_INF2_MASTER;
+			ret = 0;
+		}
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (pm860x->dir == PM860X_CLK_DIR_IN) {
+			inf &= ~PCM_INF2_MASTER;
+			ret = 0;
+		}
+		break;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		inf |= PCM_EXACT_I2S;
+		ret = 0;
+		break;
+	}
+	mask |= PCM_MODE_MASK;
+	if (ret)
+		return ret;
+	snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
+	return 0;
+}
+
+static int pm860x_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 pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	if (dir == PM860X_CLK_DIR_OUT)
+		pm860x->dir = PM860X_CLK_DIR_OUT;
+	else {
+		pm860x->dir = PM860X_CLK_DIR_IN;
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream,
+				struct snd_pcm_hw_params *params,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned char inf;
+
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		inf = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		inf = PCM_INF2_18WL;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf);
+
+	/* sample rate */
+	switch (params_rate(params)) {
+	case 8000:
+		inf = 0;
+		break;
+	case 11025:
+		inf = 1;
+		break;
+	case 16000:
+		inf = 3;
+		break;
+	case 22050:
+		inf = 4;
+		break;
+	case 32000:
+		inf = 6;
+		break;
+	case 44100:
+		inf = 7;
+		break;
+	case 48000:
+		inf = 8;
+		break;
+	default:
+		return -EINVAL;
+	}
+	snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf);
+
+	return 0;
+}
+
+static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
+				  unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	unsigned char inf = 0, mask = 0;
+
+	mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		if (pm860x->dir == PM860X_CLK_DIR_OUT)
+			inf |= PCM_INF2_MASTER;
+		else
+			return -EINVAL;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		if (pm860x->dir == PM860X_CLK_DIR_IN)
+			inf &= ~PCM_INF2_MASTER;
+		else
+			return -EINVAL;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		inf |= PCM_EXACT_I2S;
+		break;
+	default:
+		return -EINVAL;
+	}
+	mask |= PCM_MODE_MASK;
+	snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf);
+	return 0;
+}
+
+static int pm860x_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	int data;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Enable Audio PLL & Audio section */
+			data = AUDIO_PLL | AUDIO_SECTION_RESET
+				| AUDIO_SECTION_ON;
+			pm860x_reg_write(codec->control_data, REG_MISC2, data);
+		}
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
+		pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
+		break;
+	}
+	codec->bias_level = level;
+	return 0;
+}
+
+static struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
+	.digital_mute	= pm860x_digital_mute,
+	.hw_params	= pm860x_pcm_hw_params,
+	.set_fmt	= pm860x_pcm_set_dai_fmt,
+	.set_sysclk	= pm860x_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
+	.digital_mute	= pm860x_digital_mute,
+	.hw_params	= pm860x_i2s_hw_params,
+	.set_fmt	= pm860x_i2s_set_dai_fmt,
+	.set_sysclk	= pm860x_set_dai_sysclk,
+};
+
+#define PM860X_RATES	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |	\
+			 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_dai_driver pm860x_dai[] = {
+	{
+		/* DAI PCM */
+		.name	= "88pm860x-pcm",
+		.id	= 1,
+		.playback = {
+			.stream_name	= "PCM Playback",
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= PM860X_RATES,
+			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
+					  SNDRV_PCM_FORMAT_S18_3LE,
+		},
+		.capture = {
+			.stream_name	= "PCM Capture",
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= PM860X_RATES,
+			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
+					  SNDRV_PCM_FORMAT_S18_3LE,
+		},
+		.ops	= &pm860x_pcm_dai_ops,
+	}, {
+		/* DAI I2S */
+		.name	= "88pm860x-i2s",
+		.id	= 2,
+		.playback = {
+			.stream_name	= "I2S Playback",
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= SNDRV_PCM_RATE_8000_48000,
+			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
+					  SNDRV_PCM_FORMAT_S18_3LE,
+		},
+		.capture = {
+			.stream_name	= "I2S Capture",
+			.channels_min	= 2,
+			.channels_max	= 2,
+			.rates		= SNDRV_PCM_RATE_8000_48000,
+			.formats	= SNDRV_PCM_FORMAT_S16_LE | \
+					  SNDRV_PCM_FORMAT_S18_3LE,
+		},
+		.ops	= &pm860x_i2s_dai_ops,
+	},
+};
+
+static irqreturn_t pm860x_codec_handler(int irq, void *data)
+{
+	struct pm860x_priv *pm860x = data;
+	int status, shrt, report = 0, mic_report = 0;
+	int mask;
+
+	status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
+	shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
+	mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
+		| pm860x->det.hp_det;
+
+	if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
+		&& (status & HEADSET_STATUS))
+		report |= SND_JACK_HEADPHONE;
+
+	if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
+		&& (status & MIC_STATUS))
+		mic_report |= SND_JACK_MICROPHONE;
+
+	if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
+		report |= pm860x->det.hs_shrt;
+
+	if (pm860x->det.hook_det && (status & HOOK_STATUS))
+		report |= pm860x->det.hook_det;
+
+	if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
+		report |= pm860x->det.lo_shrt;
+
+	if (report)
+		snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
+	if (mic_report)
+		snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
+				    SND_JACK_MICROPHONE);
+
+	dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
+		report, mask);
+	dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
+	return IRQ_HANDLED;
+}
+
+int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
+			  struct snd_soc_jack *jack,
+			  int det, int hook, int hs_shrt, int lo_shrt)
+{
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	int data;
+
+	pm860x->det.hp_jack = jack;
+	pm860x->det.hp_det = det;
+	pm860x->det.hook_det = hook;
+	pm860x->det.hs_shrt = hs_shrt;
+	pm860x->det.lo_shrt = lo_shrt;
+
+	if (det & SND_JACK_HEADPHONE)
+		pm860x_set_bits(codec->control_data, REG_HS_DET,
+				EN_HS_DET, EN_HS_DET);
+	/* headset short detect */
+	if (hs_shrt) {
+		data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
+		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+	}
+	/* Lineout short detect */
+	if (lo_shrt) {
+		data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
+		pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
+	}
+
+	/* sync status */
+	pm860x_codec_handler(0, pm860x);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);
+
+int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
+			   struct snd_soc_jack *jack, int det)
+{
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+
+	pm860x->det.mic_jack = jack;
+	pm860x->det.mic_det = det;
+
+	if (det & SND_JACK_MICROPHONE)
+		pm860x_set_bits(codec->control_data, REG_MIC_DET,
+				MICDET_MASK, MICDET_MASK);
+
+	/* sync status */
+	pm860x_codec_handler(0, pm860x);
+	return 0;
+}
+EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
+
+static int pm860x_probe(struct snd_soc_codec *codec)
+{
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	int i, ret;
+
+	pm860x->codec = codec;
+
+	codec->control_data = pm860x->i2c;
+
+	for (i = 0; i < 4; i++) {
+		ret = request_threaded_irq(pm860x->irq[i], NULL,
+					   pm860x_codec_handler, IRQF_ONESHOT,
+					   pm860x->name[i], pm860x);
+		if (ret < 0) {
+			dev_err(codec->dev, "Failed to request IRQ!\n");
+			goto out_irq;
+		}
+	}
+
+	pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
+			       REG_CACHE_SIZE, codec->reg_cache);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to fill register cache: %d\n",
+			ret);
+		goto out_codec;
+	}
+
+	snd_soc_add_controls(codec, pm860x_snd_controls,
+			     ARRAY_SIZE(pm860x_snd_controls));
+	snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
+				  ARRAY_SIZE(pm860x_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	return 0;
+
+out_codec:
+	i = 3;
+out_irq:
+	for (; i >= 0; i--)
+		free_irq(pm860x->irq[i], pm860x);
+	return -EINVAL;
+}
+
+static int pm860x_remove(struct snd_soc_codec *codec)
+{
+	struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
+	int i;
+
+	for (i = 3; i >= 0; i--)
+		free_irq(pm860x->irq[i], pm860x);
+	pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
+	.probe		= pm860x_probe,
+	.remove		= pm860x_remove,
+	.read		= pm860x_read_reg_cache,
+	.write		= pm860x_write_reg_cache,
+	.reg_cache_size	= REG_CACHE_SIZE,
+	.reg_word_size	= sizeof(u8),
+	.set_bias_level	= pm860x_set_bias_level,
+};
+
+static int __devinit pm860x_codec_probe(struct platform_device *pdev)
+{
+	struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent);
+	struct pm860x_priv *pm860x;
+	struct resource *res;
+	int i, ret;
+
+	pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL);
+	if (pm860x == NULL)
+		return -ENOMEM;
+
+	pm860x->chip = chip;
+	pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
+			: chip->companion;
+	platform_set_drvdata(pdev, pm860x);
+
+	for (i = 0; i < 4; i++) {
+		res = platform_get_resource(pdev, IORESOURCE_IRQ, i);
+		if (!res) {
+			dev_err(&pdev->dev, "Failed to get IRQ resources\n");
+			goto out;
+		}
+		pm860x->irq[i] = res->start + chip->irq_base;
+		strncpy(pm860x->name[i], res->name, MAX_NAME_LEN);
+	}
+
+	ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x,
+				     pm860x_dai, ARRAY_SIZE(pm860x_dai));
+	if (ret) {
+		dev_err(&pdev->dev, "Failed to register codec\n");
+		goto out;
+	}
+	return ret;
+
+out:
+	platform_set_drvdata(pdev, NULL);
+	kfree(pm860x);
+	return -EINVAL;
+}
+
+static int __devexit pm860x_codec_remove(struct platform_device *pdev)
+{
+	struct pm860x_priv *pm860x = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_codec(&pdev->dev);
+	platform_set_drvdata(pdev, NULL);
+	kfree(pm860x);
+	return 0;
+}
+
+static struct platform_driver pm860x_codec_driver = {
+	.driver	= {
+		.name	= "88pm860x-codec",
+		.owner	= THIS_MODULE,
+	},
+	.probe	= pm860x_codec_probe,
+	.remove	= __devexit_p(pm860x_codec_remove),
+};
+
+static __init int pm860x_init(void)
+{
+	return platform_driver_register(&pm860x_codec_driver);
+}
+module_init(pm860x_init);
+
+static __exit void pm860x_exit(void)
+{
+	platform_driver_unregister(&pm860x_codec_driver);
+}
+module_exit(pm860x_exit);
+
+MODULE_DESCRIPTION("ASoC 88PM860x driver");
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:88pm860x-codec");
+
diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h
new file mode 100644
index 000000000000..3364ba4a3607
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.h
@@ -0,0 +1,97 @@
+/*
+ * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
+ *
+ * Copyright 2010 Marvell International Ltd.
+ *	Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __88PM860X_H
+#define __88PM860X_H
+
+/* The offset of these registers are 0xb0 */
+#define PM860X_PCM_IFACE_1		0x00
+#define PM860X_PCM_IFACE_2		0x01
+#define PM860X_PCM_IFACE_3		0x02
+#define PM860X_PCM_RATE			0x03
+#define PM860X_EC_PATH			0x04
+#define PM860X_SIDETONE_L_GAIN		0x05
+#define PM860X_SIDETONE_R_GAIN		0x06
+#define PM860X_SIDETONE_SHIFT		0x07
+#define PM860X_ADC_OFFSET_1		0x08
+#define PM860X_ADC_OFFSET_2		0x09
+#define PM860X_DMIC_DELAY		0x0a
+
+#define PM860X_I2S_IFACE_1		0x0b
+#define PM860X_I2S_IFACE_2		0x0c
+#define PM860X_I2S_IFACE_3		0x0d
+#define PM860X_I2S_IFACE_4		0x0e
+#define PM860X_EQUALIZER_N0_1		0x0f
+#define PM860X_EQUALIZER_N0_2		0x10
+#define PM860X_EQUALIZER_N1_1		0x11
+#define PM860X_EQUALIZER_N1_2		0x12
+#define PM860X_EQUALIZER_D1_1		0x13
+#define PM860X_EQUALIZER_D1_2		0x14
+#define PM860X_LOFI_GAIN_LEFT		0x15
+#define PM860X_LOFI_GAIN_RIGHT		0x16
+#define PM860X_HIFIL_GAIN_LEFT		0x17
+#define PM860X_HIFIL_GAIN_RIGHT		0x18
+#define PM860X_HIFIR_GAIN_LEFT		0x19
+#define PM860X_HIFIR_GAIN_RIGHT		0x1a
+#define PM860X_DAC_OFFSET		0x1b
+#define PM860X_OFFSET_LEFT_1		0x1c
+#define PM860X_OFFSET_LEFT_2		0x1d
+#define PM860X_OFFSET_RIGHT_1		0x1e
+#define PM860X_OFFSET_RIGHT_2		0x1f
+#define PM860X_ADC_ANA_1		0x20
+#define PM860X_ADC_ANA_2		0x21
+#define PM860X_ADC_ANA_3		0x22
+#define PM860X_ADC_ANA_4		0x23
+#define PM860X_ANA_TO_ANA		0x24
+#define PM860X_HS1_CTRL			0x25
+#define PM860X_HS2_CTRL			0x26
+#define PM860X_LO1_CTRL			0x27
+#define PM860X_LO2_CTRL			0x28
+#define PM860X_EAR_CTRL_1		0x29
+#define PM860X_EAR_CTRL_2		0x2a
+#define PM860X_AUDIO_SUPPLIES_1		0x2b
+#define PM860X_AUDIO_SUPPLIES_2		0x2c
+#define PM860X_ADC_EN_1			0x2d
+#define PM860X_ADC_EN_2			0x2e
+#define PM860X_DAC_EN_1			0x2f
+#define PM860X_DAC_EN_2			0x31
+#define PM860X_AUDIO_CAL_1		0x32
+#define PM860X_AUDIO_CAL_2		0x33
+#define PM860X_AUDIO_CAL_3		0x34
+#define PM860X_AUDIO_CAL_4		0x35
+#define PM860X_AUDIO_CAL_5		0x36
+#define PM860X_ANA_INPUT_SEL_1		0x37
+#define PM860X_ANA_INPUT_SEL_2		0x38
+
+#define PM860X_PCM_IFACE_4		0x39
+#define PM860X_I2S_IFACE_5		0x3a
+
+#define PM860X_SHORTS			0x3b
+#define PM860X_PLL_ADJ_1		0x3c
+#define PM860X_PLL_ADJ_2		0x3d
+
+/* bits definition */
+#define PM860X_CLK_DIR_IN		0
+#define PM860X_CLK_DIR_OUT		1
+
+#define PM860X_DET_HEADSET		(1 << 0)
+#define PM860X_DET_MIC			(1 << 1)
+#define PM860X_DET_HOOK			(1 << 2)
+#define PM860X_SHORT_HEADSET		(1 << 3)
+#define PM860X_SHORT_LINEOUT		(1 << 4)
+#define PM860X_DET_MASK			0x1F
+
+extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+				 int, int, int, int);
+extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
+				  int);
+
+#endif	/* __88PM860X_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bfdd92b78fb6..a3cfc184ee50 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_88PM860X if MFD_88PM860X
 	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1836 if SPI_MASTER
@@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS
 
           If unsure select "N".
 
+config SND_SOC_88PM860X
+	tristate
+
 config SND_SOC_WM_HUBS
 	tristate
 	default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9c3c39fd99ad..b9c43582c5bd 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,3 +1,4 @@
+snd-soc-88pm860x-objs := 88pm860x-codec.o
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1836-objs := ad1836.o
 snd-soc-ad193x-objs := ad193x.o
@@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
 snd-soc-wm2000-objs := wm2000.o
 snd-soc-wm9090-objs := wm9090.o
 
+obj-$(CONFIG_SND_SOC_88PM860X)	+= snd-soc-88pm860x.o
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1836)	+= snd-soc-ad1836.o
 obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c
index cf4323dbf9c4..e8d27c8f9ba3 100644
--- a/sound/soc/codecs/cx20442.c
+++ b/sound/soc/codecs/cx20442.c
@@ -318,7 +318,7 @@ EXPORT_SYMBOL_GPL(v253_ops);
  */
 
 static struct snd_soc_dai_driver cx20442_dai = {
-	.name = "cx20442-hifi",
+	.name = "cx20442-voice",
 	.playback = {
 		.stream_name = "Playback",
 		.channels_min = 1,
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 0b80e242a66d..efae8b53fd64 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -12,11 +12,11 @@
  *
  * Notes:
  *  The AIC3X is a driver for a low power stereo audio
- *  codecs aic31, aic32, aic33.
+ *  codecs aic31, aic32, aic33, aic3007.
  *
  *  It supports full aic33 codec functionality.
- *  The compatibility with aic32, aic31 is as follows:
- *        aic32        |        aic31
+ *  The compatibility with aic32, aic31 and aic3007 is as follows:
+ *    aic32/aic3007    |        aic31
  *  ---------------------------------------
  *   MONO_LOUT -> N/A  |  MONO_LOUT -> N/A
  *                     |  IN1L -> LINE1L
@@ -70,6 +70,10 @@ struct aic3x_priv {
 	unsigned int sysclk;
 	int master;
 	int gpio_reset;
+#define AIC3X_MODEL_3X 0
+#define AIC3X_MODEL_33 1
+#define AIC3X_MODEL_3007 2
+	u16 model;
 };
 
 /*
@@ -361,6 +365,14 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
+/*
+ * Class-D amplifier gain. From 0 to 18 dB in 6 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0);
+
+static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl =
+	SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv);
+
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
 SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -589,6 +601,15 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
 	SND_SOC_DAPM_INPUT("LINE2R"),
 };
 
+static const struct snd_soc_dapm_widget aic3007_dapm_widgets[] = {
+	/* Class-D outputs */
+	SND_SOC_DAPM_PGA("Left Class-D Out", CLASSD_CTRL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Right Class-D Out", CLASSD_CTRL, 2, 0, NULL, 0),
+
+	SND_SOC_DAPM_OUTPUT("SPOP"),
+	SND_SOC_DAPM_OUTPUT("SPOM"),
+};
+
 static const struct snd_soc_dapm_route intercon[] = {
 	/* Left Output */
 	{"Left DAC Mux", "DAC_L1", "Left DAC"},
@@ -759,14 +780,30 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"GPIO1 dmic modclk", NULL, "DMic Rate 32"},
 };
 
+static const struct snd_soc_dapm_route intercon_3007[] = {
+	/* Class-D outputs */
+	{"Left Class-D Out", NULL, "Left Line Out"},
+	{"Right Class-D Out", NULL, "Left Line Out"},
+	{"SPOP", NULL, "Left Class-D Out"},
+	{"SPOM", NULL, "Right Class-D Out"},
+};
+
 static int aic3x_add_widgets(struct snd_soc_codec *codec)
 {
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
+
 	snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
 				  ARRAY_SIZE(aic3x_dapm_widgets));
 
 	/* set up audio path interconnects */
 	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
 
+	if (aic3x->model == AIC3X_MODEL_3007) {
+		snd_soc_dapm_new_controls(codec, aic3007_dapm_widgets,
+			ARRAY_SIZE(aic3007_dapm_widgets));
+		snd_soc_dapm_add_routes(codec, intercon_3007, ARRAY_SIZE(intercon_3007));
+	}
+
 	return 0;
 }
 
@@ -1117,6 +1154,7 @@ static struct snd_soc_dai_driver aic3x_dai = {
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
 	.ops = &aic3x_dai_ops,
+	.symmetric_rates = 1,
 };
 
 static int aic3x_suspend(struct snd_soc_codec *codec, pm_message_t state)
@@ -1150,6 +1188,7 @@ static int aic3x_resume(struct snd_soc_codec *codec)
  */
 static int aic3x_init(struct snd_soc_codec *codec)
 {
+	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int reg;
 
 	aic3x_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT);
@@ -1218,6 +1257,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	aic3x_write(codec, LINE2L_2_MONOLOPM_VOL, DEFAULT_VOL);
 	aic3x_write(codec, LINE2R_2_MONOLOPM_VOL, DEFAULT_VOL);
 
+	if (aic3x->model == AIC3X_MODEL_3007) {
+		/* Class-D speaker driver init; datasheet p. 46 */
+		aic3x_write(codec, AIC3X_PAGE_SELECT, 0x0D);
+		aic3x_write(codec, 0xD, 0x0D);
+		aic3x_write(codec, 0x8, 0x5C);
+		aic3x_write(codec, 0x8, 0x5D);
+		aic3x_write(codec, 0x8, 0x5C);
+		aic3x_write(codec, AIC3X_PAGE_SELECT, 0x00);
+		aic3x_write(codec, CLASSD_CTRL, 0);
+	}
+
 	/* off, with power on */
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1243,6 +1293,8 @@ static int aic3x_probe(struct snd_soc_codec *codec)
 
 	snd_soc_add_controls(codec, aic3x_snd_controls,
 			     ARRAY_SIZE(aic3x_snd_controls));
+	if (aic3x->model == AIC3X_MODEL_3007)
+		snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1);
 
 	aic3x_add_widgets(codec);
 
@@ -1274,6 +1326,14 @@ static struct snd_soc_codec_driver soc_codec_dev_aic3x = {
  * 0x18, 0x19, 0x1A, 0x1B
  */
 
+static const struct i2c_device_id aic3x_i2c_id[] = {
+	[AIC3X_MODEL_3X] = { "tlv320aic3x", 0 },
+	[AIC3X_MODEL_33] = { "tlv320aic33", 0 },
+	[AIC3X_MODEL_3007] = { "tlv320aic3007", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
+
 /*
  * If the i2c layer weren't so broken, we could pass this kind of data
  * around
@@ -1285,6 +1345,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 	struct aic3x_setup_data *setup = pdata->setup;
 	struct aic3x_priv *aic3x;
 	int ret, i;
+	const struct i2c_device_id *tbl;
 
 	aic3x = kzalloc(sizeof(struct aic3x_priv), GFP_KERNEL);
 	if (aic3x == NULL) {
@@ -1305,6 +1366,12 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 		gpio_direction_output(aic3x->gpio_reset, 0);
 	}
 
+	for (tbl = aic3x_i2c_id; tbl->name[0]; tbl++) {
+		if (!strcmp(tbl->name, id->name))
+			break;
+	}
+	aic3x->model = tbl - aic3x_i2c_id;
+
 	for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++)
 		aic3x->supplies[i].supply = aic3x_supply_names[i];
 
@@ -1359,13 +1426,6 @@ static int aic3x_i2c_remove(struct i2c_client *client)
 	return 0;
 }
 
-static const struct i2c_device_id aic3x_i2c_id[] = {
-	{ "tlv320aic3x", 0 },
-	{ "tlv320aic33", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id);
-
 /* machine i2c codec control layer */
 static struct i2c_driver aic3x_i2c_driver = {
 	.driver = {
diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h
index f6e3d9b42daf..98e44395b662 100644
--- a/sound/soc/codecs/tlv320aic3x.h
+++ b/sound/soc/codecs/tlv320aic3x.h
@@ -111,6 +111,8 @@
 #define DACL1_2_MONOLOPM_VOL		75
 #define DACR1_2_MONOLOPM_VOL		78
 #define MONOLOPM_CTRL			79
+/* Class-D speaker driver on tlv320aic3007 */
+#define CLASSD_CTRL			73
 /* Line Output Plus/Minus control registers */
 #define LINE2L_2_LLOPM_VOL		80
 #define LINE2L_2_RLOPM_VOL		87
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index 19844fc8cb1d..56f540838745 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -46,6 +46,7 @@ struct wm8731_priv {
 	struct regulator_bulk_data supplies[WM8731_NUM_SUPPLIES];
 	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
+	int sysclk_type;
 };
 
 
@@ -110,6 +111,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls =
 SOC_DAPM_ENUM("Input Select", wm8731_enum[0]);
 
 static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = {
+SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0),
 SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1,
 	&wm8731_output_mixer_controls[0],
 	ARRAY_SIZE(wm8731_output_mixer_controls)),
@@ -127,7 +129,18 @@ SND_SOC_DAPM_INPUT("RLINEIN"),
 SND_SOC_DAPM_INPUT("LLINEIN"),
 };
 
+static int wm8731_check_osc(struct snd_soc_dapm_widget *source,
+			    struct snd_soc_dapm_widget *sink)
+{
+	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(source->codec);
+
+	return wm8731->sysclk_type == WM8731_SYSCLK_MCLK;
+}
+
 static const struct snd_soc_dapm_route intercon[] = {
+	{"DAC", NULL, "OSC", wm8731_check_osc},
+	{"ADC", NULL, "OSC", wm8731_check_osc},
+
 	/* output mixer */
 	{"Output Mixer", "Line Bypass Switch", "Line Input"},
 	{"Output Mixer", "HiFi Playback Switch", "DAC"},
@@ -285,6 +298,15 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	struct snd_soc_codec *codec = codec_dai->codec;
 	struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec);
 
+	switch (clk_id) {
+	case WM8731_SYSCLK_XTAL:
+	case WM8731_SYSCLK_MCLK:
+		wm8731->sysclk_type = clk_id;
+		break;
+	default:
+		return -EINVAL;
+	}
+
 	switch (freq) {
 	case 11289600:
 	case 12000000:
@@ -292,9 +314,14 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	case 16934400:
 	case 18432000:
 		wm8731->sysclk = freq;
-		return 0;
+		break;
+	default:
+		return -EINVAL;
 	}
-	return -EINVAL;
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
 }
 
 
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 73a70e206ba9..e9c0c76ab73b 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -31,7 +31,9 @@
 
 #define WM8731_CACHEREGNUM 	10
 
-#define WM8731_SYSCLK	0
+#define WM8731_SYSCLK_XTAL 1
+#define WM8731_SYSCLK_MCLK 2
+
 #define WM8731_DAI		0
 
 #endif
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 981868700388..d754d34d68a6 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,24 +1,36 @@
 config SND_MPC52xx_DMA
 	tristate
 
-# ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
-# for the SSI and the Elo DMA controller.  You will still need to select
-# a platform driver and a codec driver.
-config SND_SOC_MPC8610
+# ASoC platform support for the Freescale PowerPC SOCs that have an SSI and
+# an Elo DMA controller, such as the MPC8610 and P1022.  You will still need to
+# select a platform driver and a codec driver.
+config SND_SOC_POWERPC_SSI
 	tristate
-	depends on MPC8610
+	depends on FSL_SOC
 
 config SND_SOC_MPC8610_HPCD
 	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
 	# I2C is necessary for the CS4270 driver
 	depends on MPC8610_HPCD && I2C
-	select SND_SOC_MPC8610
+	select SND_SOC_POWERPC_SSI
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
 	default y if MPC8610_HPCD
 	help
 	  Say Y if you want to enable audio on the Freescale MPC8610 HPCD.
 
+config SND_SOC_P1022_DS
+	tristate "ALSA SoC support for the Freescale P1022 DS board"
+	# I2C is necessary for the WM8776 driver
+	depends on P1022_DS && I2C
+	select SND_SOC_POWERPC_SSI
+	select SND_SOC_WM8776
+	default y if P1022_DS
+	help
+	  Say Y if you want to enable audio on the Freescale P1022 DS board.
+	  This will also include the Wolfson Microelectronics WM8776 codec
+	  driver.
+
 config SND_SOC_MPC5200_I2S
 	tristate "Freescale MPC5200 PSC in I2S mode driver"
 	depends on PPC_MPC52xx && PPC_BESTCOMM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 7e472a53fcd3..b4a38c0ac58c 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -2,10 +2,14 @@
 snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
 obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
 
-# MPC8610 Platform Support
+# P1022 DS Machine Support
+snd-soc-p1022-ds-objs := p1022_ds.o
+obj-$(CONFIG_SND_SOC_P1022_DS) += snd-soc-p1022-ds.o
+
+# Freescale PowerPC SSI/DMA Platform Support
 snd-soc-fsl-ssi-objs := fsl_ssi.o
 snd-soc-fsl-dma-objs := fsl_dma.o
-obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
+obj-$(CONFIG_SND_SOC_POWERPC_SSI) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
 # MPC5200 Platform Support
 obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index dfe1cb94a70f..4cf98c03af22 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -23,6 +23,7 @@
 #include <linux/gfp.h>
 #include <linux/of_platform.h>
 #include <linux/list.h>
+#include <linux/slab.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -305,21 +306,29 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->coherent_dma_mask)
 		card->dev->coherent_dma_mask = fsl_dma_dmamask;
 
-	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
-		fsl_dma_hardware.buffer_bytes_max,
-		&pcm->streams[0].substream->dma_buffer);
-	if (ret) {
-		dev_err(card->dev, "can't allocate playback dma buffer\n");
-		return ret;
+	/* Some codecs have separate DAIs for playback and capture, so we
+	 * should allocate a DMA buffer only for the streams that are valid.
+	 */
+
+	if (dai->driver->playback.channels_min) {
+		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
+			fsl_dma_hardware.buffer_bytes_max,
+			&pcm->streams[0].substream->dma_buffer);
+		if (ret) {
+			dev_err(card->dev, "can't alloc playback dma buffer\n");
+			return ret;
+		}
 	}
 
-	ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
-		fsl_dma_hardware.buffer_bytes_max,
-		&pcm->streams[1].substream->dma_buffer);
-	if (ret) {
-		snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
-		dev_err(card->dev, "can't allocate capture dma buffer\n");
-		return ret;
+	if (dai->driver->capture.channels_min) {
+		ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev,
+			fsl_dma_hardware.buffer_bytes_max,
+			&pcm->streams[1].substream->dma_buffer);
+		if (ret) {
+			snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer);
+			dev_err(card->dev, "can't alloc capture dma buffer\n");
+			return ret;
+		}
 	}
 
 	return 0;
@@ -887,11 +896,11 @@ static struct snd_pcm_ops fsl_dma_ops = {
 	.pointer	= fsl_dma_pointer,
 };
 
-static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
+static int __devinit fsl_soc_dma_probe(struct platform_device *pdev,
 				       const struct of_device_id *match)
  {
 	struct dma_object *dma;
-	struct device_node *np = of_dev->dev.of_node;
+	struct device_node *np = pdev->dev.of_node;
 	struct device_node *ssi_np;
 	struct resource res;
 	const uint32_t *iprop;
@@ -900,13 +909,13 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	/* Find the SSI node that points to us. */
 	ssi_np = find_ssi_node(np);
 	if (!ssi_np) {
-		dev_err(&of_dev->dev, "cannot find parent SSI node\n");
+		dev_err(&pdev->dev, "cannot find parent SSI node\n");
 		return -ENODEV;
 	}
 
 	ret = of_address_to_resource(ssi_np, 0, &res);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not determine resources for %s\n",
+		dev_err(&pdev->dev, "could not determine resources for %s\n",
 			ssi_np->full_name);
 		of_node_put(ssi_np);
 		return ret;
@@ -914,7 +923,7 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 
 	dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
 	if (!dma) {
-		dev_err(&of_dev->dev, "could not allocate dma object\n");
+		dev_err(&pdev->dev, "could not allocate dma object\n");
 		of_node_put(ssi_np);
 		return -ENOMEM;
 	}
@@ -937,9 +946,9 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 
 	of_node_put(ssi_np);
 
-	ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
+	ret = snd_soc_register_platform(&pdev->dev, &dma->dai);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not register platform\n");
+		dev_err(&pdev->dev, "could not register platform\n");
 		kfree(dma);
 		return ret;
 	}
@@ -947,16 +956,16 @@ static int __devinit fsl_soc_dma_probe(struct of_device *of_dev,
 	dma->channel = of_iomap(np, 0);
 	dma->irq = irq_of_parse_and_map(np, 0);
 
-	dev_set_drvdata(&of_dev->dev, dma);
+	dev_set_drvdata(&pdev->dev, dma);
 
 	return 0;
 }
 
-static int __devexit fsl_soc_dma_remove(struct of_device *of_dev)
+static int __devexit fsl_soc_dma_remove(struct platform_device *pdev)
 {
-	struct dma_object *dma = dev_get_drvdata(&of_dev->dev);
+	struct dma_object *dma = dev_get_drvdata(&pdev->dev);
 
-	snd_soc_unregister_platform(&of_dev->dev);
+	snd_soc_unregister_platform(&pdev->dev);
 	iounmap(dma->channel);
 	irq_dispose_mapping(dma->irq);
 	kfree(dma);
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index d1c855ade8fb..4cc167a7aeb8 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -624,13 +624,13 @@ static void make_lowercase(char *s)
 	}
 }
 
-static int __devinit fsl_ssi_probe(struct of_device *of_dev,
+static int __devinit fsl_ssi_probe(struct platform_device *pdev,
 				   const struct of_device_id *match)
 {
 	struct fsl_ssi_private *ssi_private;
 	int ret = 0;
 	struct device_attribute *dev_attr = NULL;
-	struct device_node *np = of_dev->dev.of_node;
+	struct device_node *np = pdev->dev.of_node;
 	const char *p, *sprop;
 	const uint32_t *iprop;
 	struct resource res;
@@ -645,14 +645,14 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 
 	/* Check for a codec-handle property. */
 	if (!of_get_property(np, "codec-handle", NULL)) {
-		dev_err(&of_dev->dev, "missing codec-handle property\n");
+		dev_err(&pdev->dev, "missing codec-handle property\n");
 		return -ENODEV;
 	}
 
 	/* We only support the SSI in "I2S Slave" mode */
 	sprop = of_get_property(np, "fsl,mode", NULL);
 	if (!sprop || strcmp(sprop, "i2s-slave")) {
-		dev_notice(&of_dev->dev, "mode %s is unsupported\n", sprop);
+		dev_notice(&pdev->dev, "mode %s is unsupported\n", sprop);
 		return -ENODEV;
 	}
 
@@ -661,7 +661,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	ssi_private = kzalloc(sizeof(struct fsl_ssi_private) + strlen(p),
 			      GFP_KERNEL);
 	if (!ssi_private) {
-		dev_err(&of_dev->dev, "could not allocate DAI object\n");
+		dev_err(&pdev->dev, "could not allocate DAI object\n");
 		return -ENOMEM;
 	}
 
@@ -675,7 +675,7 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	/* Get the addresses and IRQ */
 	ret = of_address_to_resource(np, 0, &res);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not determine device resources\n");
+		dev_err(&pdev->dev, "could not determine device resources\n");
 		kfree(ssi_private);
 		return ret;
 	}
@@ -703,19 +703,19 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	dev_attr->attr.mode = S_IRUGO;
 	dev_attr->show = fsl_sysfs_ssi_show;
 
-	ret = device_create_file(&of_dev->dev, dev_attr);
+	ret = device_create_file(&pdev->dev, dev_attr);
 	if (ret) {
-		dev_err(&of_dev->dev, "could not create sysfs %s file\n",
+		dev_err(&pdev->dev, "could not create sysfs %s file\n",
 			ssi_private->dev_attr.attr.name);
 		goto error;
 	}
 
 	/* Register with ASoC */
-	dev_set_drvdata(&of_dev->dev, ssi_private);
+	dev_set_drvdata(&pdev->dev, ssi_private);
 
-	ret = snd_soc_register_dai(&of_dev->dev, &ssi_private->cpu_dai_drv);
+	ret = snd_soc_register_dai(&pdev->dev, &ssi_private->cpu_dai_drv);
 	if (ret) {
-		dev_err(&of_dev->dev, "failed to register DAI: %d\n", ret);
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		goto error;
 	}
 
@@ -733,20 +733,20 @@ static int __devinit fsl_ssi_probe(struct of_device *of_dev,
 	make_lowercase(name);
 
 	ssi_private->pdev =
-		platform_device_register_data(&of_dev->dev, name, 0, NULL, 0);
+		platform_device_register_data(&pdev->dev, name, 0, NULL, 0);
 	if (IS_ERR(ssi_private->pdev)) {
 		ret = PTR_ERR(ssi_private->pdev);
-		dev_err(&of_dev->dev, "failed to register platform: %d\n", ret);
+		dev_err(&pdev->dev, "failed to register platform: %d\n", ret);
 		goto error;
 	}
 
 	return 0;
 
 error:
-	snd_soc_unregister_dai(&of_dev->dev);
-	dev_set_drvdata(&of_dev->dev, NULL);
+	snd_soc_unregister_dai(&pdev->dev);
+	dev_set_drvdata(&pdev->dev, NULL);
 	if (dev_attr)
-		device_remove_file(&of_dev->dev, dev_attr);
+		device_remove_file(&pdev->dev, dev_attr);
 	irq_dispose_mapping(ssi_private->irq);
 	iounmap(ssi_private->ssi);
 	kfree(ssi_private);
@@ -754,16 +754,16 @@ error:
 	return ret;
 }
 
-static int fsl_ssi_remove(struct of_device *of_dev)
+static int fsl_ssi_remove(struct platform_device *pdev)
 {
-	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&of_dev->dev);
+	struct fsl_ssi_private *ssi_private = dev_get_drvdata(&pdev->dev);
 
 	platform_device_unregister(ssi_private->pdev);
-	snd_soc_unregister_dai(&of_dev->dev);
-	device_remove_file(&of_dev->dev, &ssi_private->dev_attr);
+	snd_soc_unregister_dai(&pdev->dev);
+	device_remove_file(&pdev->dev, &ssi_private->dev_attr);
 
 	kfree(ssi_private);
-	dev_set_drvdata(&of_dev->dev, NULL);
+	dev_set_drvdata(&pdev->dev, NULL);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 38339c158ed9..0d7dcf1e4863 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -13,6 +13,7 @@
 #include <linux/module.h>
 #include <linux/interrupt.h>
 #include <linux/of_device.h>
+#include <linux/slab.h>
 #include <sound/soc.h>
 #include <asm/fsl_guts.h>
 
@@ -323,9 +324,10 @@ static int get_dma_channel(struct device_node *ssi_np,
 static int mpc8610_hpcd_probe(struct platform_device *pdev)
 {
 	struct device *dev = pdev->dev.parent;
-	/* of_dev is the OF device for the SSI node that probed us */
-	struct of_device *of_dev = container_of(dev, struct of_device, dev);
-	struct device_node *np = of_dev->dev.of_node;
+	/* ssi_pdev is the platform device for the SSI node that probed us */
+	struct platform_device *ssi_pdev =
+		container_of(dev, struct platform_device, dev);
+	struct device_node *np = ssi_pdev->dev.of_node;
 	struct device_node *codec_np = NULL;
 	struct platform_device *sound_device = NULL;
 	struct mpc8610_hpcd_data *machine_data;
@@ -348,7 +350,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev)
 	if (!machine_data)
 		return -ENOMEM;
 
-	machine_data->dai[0].cpu_dai_name = dev_name(&of_dev->dev);
+	machine_data->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
 	machine_data->dai[0].ops = &mpc8610_hpcd_ops;
 
 	/* Determine the codec name, it will be used as the codec DAI name */
diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c
new file mode 100644
index 000000000000..f8176e8e1adf
--- /dev/null
+++ b/sound/soc/fsl/p1022_ds.c
@@ -0,0 +1,590 @@
+/**
+ * Freescale P1022DS ALSA SoC Machine driver
+ *
+ * Author: Timur Tabi <timur@freescale.com>
+ *
+ * Copyright 2010 Freescale Semiconductor, Inc.
+ *
+ * This file is licensed under the terms of the GNU General Public License
+ * version 2.  This program is licensed "as is" without any warranty of any
+ * kind, whether express or implied.
+ */
+
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/of_device.h>
+#include <linux/slab.h>
+#include <sound/soc.h>
+#include <asm/fsl_guts.h>
+
+#include "fsl_dma.h"
+#include "fsl_ssi.h"
+
+/* P1022-specific PMUXCR and DMUXCR bit definitions */
+
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_MASK	0x0001c000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI	0x00010000
+#define CCSR_GUTS_PMUXCR_UART0_I2C1_SSI		0x00018000
+
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK	0x00000c00
+#define CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI	0x00000000
+
+#define CCSR_GUTS_DMUXCR_PAD	1	/* DMA controller/channel set to pad */
+#define CCSR_GUTS_DMUXCR_SSI	2	/* DMA controller/channel set to SSI */
+
+/*
+ * Set the DMACR register in the GUTS
+ *
+ * The DMACR register determines the source of initiated transfers for each
+ * channel on each DMA controller.  Rather than have a bunch of repetitive
+ * macros for the bit patterns, we just have a function that calculates
+ * them.
+ *
+ * guts: Pointer to GUTS structure
+ * co: The DMA controller (0 or 1)
+ * ch: The channel on the DMA controller (0, 1, 2, or 3)
+ * device: The device to set as the target (CCSR_GUTS_DMUXCR_xxx)
+ */
+static inline void guts_set_dmuxcr(struct ccsr_guts_85xx __iomem *guts,
+	unsigned int co, unsigned int ch, unsigned int device)
+{
+	unsigned int shift = 16 + (8 * (1 - co) + 2 * (3 - ch));
+
+	clrsetbits_be32(&guts->dmuxcr, 3 << shift, device << shift);
+}
+
+/* There's only one global utilities register */
+static phys_addr_t guts_phys;
+
+#define DAI_NAME_SIZE	32
+
+/**
+ * machine_data: machine-specific ASoC device data
+ *
+ * This structure contains data for a single sound platform device on an
+ * P1022 DS.  Some of the data is taken from the device tree.
+ */
+struct machine_data {
+	struct snd_soc_dai_link dai[2];
+	struct snd_soc_card card;
+	unsigned int dai_format;
+	unsigned int codec_clk_direction;
+	unsigned int cpu_clk_direction;
+	unsigned int clk_frequency;
+	unsigned int ssi_id;		/* 0 = SSI1, 1 = SSI2, etc */
+	unsigned int dma_id[2];		/* 0 = DMA1, 1 = DMA2, etc */
+	unsigned int dma_channel_id[2]; /* 0 = ch 0, 1 = ch 1, etc*/
+	char codec_name[DAI_NAME_SIZE];
+	char platform_name[2][DAI_NAME_SIZE]; /* One for each DMA channel */
+};
+
+/**
+ * p1022_ds_machine_probe: initialize the board
+ *
+ * This function is used to initialize the board-specific hardware.
+ *
+ * Here we program the DMACR and PMUXCR registers.
+ */
+static int p1022_ds_machine_probe(struct platform_device *sound_device)
+{
+	struct snd_soc_card *card = platform_get_drvdata(sound_device);
+	struct machine_data *mdata =
+		container_of(card, struct machine_data, card);
+	struct ccsr_guts_85xx __iomem *guts;
+
+	guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx));
+	if (!guts) {
+		dev_err(card->dev, "could not map global utilities\n");
+		return -ENOMEM;
+	}
+
+	/* Enable SSI Tx signal */
+	clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK,
+			CCSR_GUTS_PMUXCR_UART0_I2C1_UART0_SSI);
+
+	/* Enable SSI Rx signal */
+	clrsetbits_be32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK,
+			CCSR_GUTS_PMUXCR_SSI_DMA_TDM_SSI);
+
+	/* Enable DMA Channel for SSI */
+	guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0],
+			CCSR_GUTS_DMUXCR_SSI);
+
+	guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1],
+			CCSR_GUTS_DMUXCR_SSI);
+
+	iounmap(guts);
+
+	return 0;
+}
+
+/**
+ * p1022_ds_startup: program the board with various hardware parameters
+ *
+ * This function takes board-specific information, like clock frequencies
+ * and serial data formats, and passes that information to the codec and
+ * transport drivers.
+ */
+static int p1022_ds_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct machine_data *mdata =
+		container_of(rtd->card, struct machine_data, card);
+	struct device *dev = rtd->card->dev;
+	int ret = 0;
+
+	/* Tell the codec driver what the serial protocol is. */
+	ret = snd_soc_dai_set_fmt(rtd->codec_dai, mdata->dai_format);
+	if (ret < 0) {
+		dev_err(dev, "could not set codec driver audio format\n");
+		return ret;
+	}
+
+	/*
+	 * Tell the codec driver what the MCLK frequency is, and whether it's
+	 * a slave or master.
+	 */
+	ret = snd_soc_dai_set_sysclk(rtd->codec_dai, 0, mdata->clk_frequency,
+				     mdata->codec_clk_direction);
+	if (ret < 0) {
+		dev_err(dev, "could not set codec driver clock params\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+/**
+ * p1022_ds_machine_remove: Remove the sound device
+ *
+ * This function is called to remove the sound device for one SSI.  We
+ * de-program the DMACR and PMUXCR register.
+ */
+static int p1022_ds_machine_remove(struct platform_device *sound_device)
+{
+	struct snd_soc_card *card = platform_get_drvdata(sound_device);
+	struct machine_data *mdata =
+		container_of(card, struct machine_data, card);
+	struct ccsr_guts_85xx __iomem *guts;
+
+	guts = ioremap(guts_phys, sizeof(struct ccsr_guts_85xx));
+	if (!guts) {
+		dev_err(card->dev, "could not map global utilities\n");
+		return -ENOMEM;
+	}
+
+	/* Restore the signal routing */
+	clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_UART0_I2C1_MASK);
+	clrbits32(&guts->pmuxcr, CCSR_GUTS_PMUXCR_SSI_DMA_TDM_MASK);
+	guts_set_dmuxcr(guts, mdata->dma_id[0], mdata->dma_channel_id[0], 0);
+	guts_set_dmuxcr(guts, mdata->dma_id[1], mdata->dma_channel_id[1], 0);
+
+	iounmap(guts);
+
+	return 0;
+}
+
+/**
+ * p1022_ds_ops: ASoC machine driver operations
+ */
+static struct snd_soc_ops p1022_ds_ops = {
+	.startup = p1022_ds_startup,
+};
+
+/**
+ * get_node_by_phandle_name - get a node by its phandle name
+ *
+ * This function takes a node, the name of a property in that node, and a
+ * compatible string.  Assuming the property is a phandle to another node,
+ * it returns that node, (optionally) if that node is compatible.
+ *
+ * If the property is not a phandle, or the node it points to is not compatible
+ * with the specific string, then NULL is returned.
+ */
+static struct device_node *get_node_by_phandle_name(struct device_node *np,
+	const char *name, const char *compatible)
+{
+	np = of_parse_phandle(np, name, 0);
+	if (!np)
+		return NULL;
+
+	if (!of_device_is_compatible(np, compatible)) {
+		of_node_put(np);
+		return NULL;
+	}
+
+	return np;
+}
+
+/**
+ * get_parent_cell_index -- return the cell-index of the parent of a node
+ *
+ * Return the value of the cell-index property of the parent of the given
+ * node.  This is used for DMA channel nodes that need to know the DMA ID
+ * of the controller they are on.
+ */
+static int get_parent_cell_index(struct device_node *np)
+{
+	struct device_node *parent = of_get_parent(np);
+	const u32 *iprop;
+	int ret = -1;
+
+	if (!parent)
+		return -1;
+
+	iprop = of_get_property(parent, "cell-index", NULL);
+	if (iprop)
+		ret = *iprop;
+
+	of_node_put(parent);
+
+	return ret;
+}
+
+/**
+ * codec_node_dev_name - determine the dev_name for a codec node
+ *
+ * This function determines the dev_name for an I2C node.  This is the name
+ * that would be returned by dev_name() if this device_node were part of a
+ * 'struct device'  It's ugly and hackish, but it works.
+ *
+ * The dev_name for such devices include the bus number and I2C address. For
+ * example, "cs4270-codec.0-004f".
+ */
+static int codec_node_dev_name(struct device_node *np, char *buf, size_t len)
+{
+	const u32 *iprop;
+	int bus, addr;
+	char temp[DAI_NAME_SIZE];
+
+	of_modalias_node(np, temp, DAI_NAME_SIZE);
+
+	iprop = of_get_property(np, "reg", NULL);
+	if (!iprop)
+		return -EINVAL;
+
+	addr = *iprop;
+
+	bus = get_parent_cell_index(np);
+	if (bus < 0)
+		return bus;
+
+	snprintf(buf, len, "%s-codec.%u-%04x", temp, bus, addr);
+
+	return 0;
+}
+
+static int get_dma_channel(struct device_node *ssi_np,
+			   const char *compatible,
+			   struct snd_soc_dai_link *dai,
+			   unsigned int *dma_channel_id,
+			   unsigned int *dma_id)
+{
+	struct resource res;
+	struct device_node *dma_channel_np;
+	const u32 *iprop;
+	int ret;
+
+	dma_channel_np = get_node_by_phandle_name(ssi_np, compatible,
+						  "fsl,ssi-dma-channel");
+	if (!dma_channel_np)
+		return -EINVAL;
+
+	/* Determine the dev_name for the device_node.  This code mimics the
+	 * behavior of of_device_make_bus_id(). We need this because ASoC uses
+	 * the dev_name() of the device to match the platform (DMA) device with
+	 * the CPU (SSI) device.  It's all ugly and hackish, but it works (for
+	 * now).
+	 *
+	 * dai->platform name should already point to an allocated buffer.
+	 */
+	ret = of_address_to_resource(dma_channel_np, 0, &res);
+	if (ret)
+		return ret;
+	snprintf((char *)dai->platform_name, DAI_NAME_SIZE, "%llx.%s",
+		 (unsigned long long) res.start, dma_channel_np->name);
+
+	iprop = of_get_property(dma_channel_np, "cell-index", NULL);
+	if (!iprop) {
+		of_node_put(dma_channel_np);
+		return -EINVAL;
+	}
+
+	*dma_channel_id = *iprop;
+	*dma_id = get_parent_cell_index(dma_channel_np);
+	of_node_put(dma_channel_np);
+
+	return 0;
+}
+
+/**
+ * p1022_ds_probe: platform probe function for the machine driver
+ *
+ * Although this is a machine driver, the SSI node is the "master" node with
+ * respect to audio hardware connections.  Therefore, we create a new ASoC
+ * device for each new SSI node that has a codec attached.
+ */
+static int p1022_ds_probe(struct platform_device *pdev)
+{
+	struct device *dev = pdev->dev.parent;
+	/* ssi_pdev is the platform device for the SSI node that probed us */
+	struct platform_device *ssi_pdev =
+		container_of(dev, struct platform_device, dev);
+	struct device_node *np = ssi_pdev->dev.of_node;
+	struct device_node *codec_np = NULL;
+	struct platform_device *sound_device = NULL;
+	struct machine_data *mdata;
+	int ret = -ENODEV;
+	const char *sprop;
+	const u32 *iprop;
+
+	/* Find the codec node for this SSI. */
+	codec_np = of_parse_phandle(np, "codec-handle", 0);
+	if (!codec_np) {
+		dev_err(dev, "could not find codec node\n");
+		return -EINVAL;
+	}
+
+	mdata = kzalloc(sizeof(struct machine_data), GFP_KERNEL);
+	if (!mdata)
+		return -ENOMEM;
+
+	mdata->dai[0].cpu_dai_name = dev_name(&ssi_pdev->dev);
+	mdata->dai[0].ops = &p1022_ds_ops;
+
+	/* Determine the codec name, it will be used as the codec DAI name */
+	ret = codec_node_dev_name(codec_np, mdata->codec_name, DAI_NAME_SIZE);
+	if (ret) {
+		dev_err(&pdev->dev, "invalid codec node %s\n",
+			codec_np->full_name);
+		ret = -EINVAL;
+		goto error;
+	}
+	mdata->dai[0].codec_name = mdata->codec_name;
+
+	/* We register two DAIs per SSI, one for playback and the other for
+	 * capture.  We support codecs that have separate DAIs for both playback
+	 * and capture.
+	 */
+	memcpy(&mdata->dai[1], &mdata->dai[0], sizeof(struct snd_soc_dai_link));
+
+	/* The DAI names from the codec (snd_soc_dai_driver.name) */
+	mdata->dai[0].codec_dai_name = "wm8776-hifi-playback";
+	mdata->dai[1].codec_dai_name = "wm8776-hifi-capture";
+
+	/* Get the device ID */
+	iprop = of_get_property(np, "cell-index", NULL);
+	if (!iprop) {
+		dev_err(&pdev->dev, "cell-index property not found\n");
+		ret = -EINVAL;
+		goto error;
+	}
+	mdata->ssi_id = *iprop;
+
+	/* Get the serial format and clock direction. */
+	sprop = of_get_property(np, "fsl,mode", NULL);
+	if (!sprop) {
+		dev_err(&pdev->dev, "fsl,mode property not found\n");
+		ret = -EINVAL;
+		goto error;
+	}
+
+	if (strcasecmp(sprop, "i2s-slave") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_I2S;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+
+		/* In i2s-slave mode, the codec has its own clock source, so we
+		 * need to get the frequency from the device tree and pass it to
+		 * the codec driver.
+		 */
+		iprop = of_get_property(codec_np, "clock-frequency", NULL);
+		if (!iprop || !*iprop) {
+			dev_err(&pdev->dev, "codec bus-frequency "
+				"property is missing or invalid\n");
+			ret = -EINVAL;
+			goto error;
+		}
+		mdata->clk_frequency = *iprop;
+	} else if (strcasecmp(sprop, "i2s-master") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_I2S;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+	} else if (strcasecmp(sprop, "lj-slave") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+	} else if (strcasecmp(sprop, "lj-master") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_LEFT_J;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+	} else if (strcasecmp(sprop, "rj-slave") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+	} else if (strcasecmp(sprop, "rj-master") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_RIGHT_J;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+	} else if (strcasecmp(sprop, "ac97-slave") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_AC97;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_OUT;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_IN;
+	} else if (strcasecmp(sprop, "ac97-master") == 0) {
+		mdata->dai_format = SND_SOC_DAIFMT_AC97;
+		mdata->codec_clk_direction = SND_SOC_CLOCK_IN;
+		mdata->cpu_clk_direction = SND_SOC_CLOCK_OUT;
+	} else {
+		dev_err(&pdev->dev,
+			"unrecognized fsl,mode property '%s'\n", sprop);
+		ret = -EINVAL;
+		goto error;
+	}
+
+	if (!mdata->clk_frequency) {
+		dev_err(&pdev->dev, "unknown clock frequency\n");
+		ret = -EINVAL;
+		goto error;
+	}
+
+	/* Find the playback DMA channel to use. */
+	mdata->dai[0].platform_name = mdata->platform_name[0];
+	ret = get_dma_channel(np, "fsl,playback-dma", &mdata->dai[0],
+			      &mdata->dma_channel_id[0],
+			      &mdata->dma_id[0]);
+	if (ret) {
+		dev_err(&pdev->dev, "missing/invalid playback DMA phandle\n");
+		goto error;
+	}
+
+	/* Find the capture DMA channel to use. */
+	mdata->dai[1].platform_name = mdata->platform_name[1];
+	ret = get_dma_channel(np, "fsl,capture-dma", &mdata->dai[1],
+			      &mdata->dma_channel_id[1],
+			      &mdata->dma_id[1]);
+	if (ret) {
+		dev_err(&pdev->dev, "missing/invalid capture DMA phandle\n");
+		goto error;
+	}
+
+	/* Initialize our DAI data structure.  */
+	mdata->dai[0].stream_name = "playback";
+	mdata->dai[1].stream_name = "capture";
+	mdata->dai[0].name = mdata->dai[0].stream_name;
+	mdata->dai[1].name = mdata->dai[1].stream_name;
+
+	mdata->card.probe = p1022_ds_machine_probe;
+	mdata->card.remove = p1022_ds_machine_remove;
+	mdata->card.name = pdev->name; /* The platform driver name */
+	mdata->card.num_links = 2;
+	mdata->card.dai_link = mdata->dai;
+
+	/* Allocate a new audio platform device structure */
+	sound_device = platform_device_alloc("soc-audio", -1);
+	if (!sound_device) {
+		dev_err(&pdev->dev, "platform device alloc failed\n");
+		ret = -ENOMEM;
+		goto error;
+	}
+
+	/* Associate the card data with the sound device */
+	platform_set_drvdata(sound_device, &mdata->card);
+
+	/* Register with ASoC */
+	ret = platform_device_add(sound_device);
+	if (ret) {
+		dev_err(&pdev->dev, "platform device add failed\n");
+		goto error;
+	}
+
+	of_node_put(codec_np);
+
+	return 0;
+
+error:
+	of_node_put(codec_np);
+
+	if (sound_device)
+		platform_device_unregister(sound_device);
+
+	kfree(mdata);
+
+	return ret;
+}
+
+/**
+ * p1022_ds_remove: remove the platform device
+ *
+ * This function is called when the platform device is removed.
+ */
+static int __devexit p1022_ds_remove(struct platform_device *pdev)
+{
+	struct platform_device *sound_device = dev_get_drvdata(&pdev->dev);
+	struct snd_soc_card *card = platform_get_drvdata(sound_device);
+	struct machine_data *mdata =
+		container_of(card, struct machine_data, card);
+
+	platform_device_unregister(sound_device);
+
+	kfree(mdata);
+	sound_device->dev.platform_data = NULL;
+
+	dev_set_drvdata(&pdev->dev, NULL);
+
+	return 0;
+}
+
+static struct platform_driver p1022_ds_driver = {
+	.probe = p1022_ds_probe,
+	.remove = __devexit_p(p1022_ds_remove),
+	.driver = {
+		/* The name must match the 'model' property in the device tree,
+		 * in lowercase letters, but only the part after that last
+		 * comma.  This is because some model properties have a "fsl,"
+		 * prefix.
+		 */
+		.name = "snd-soc-p1022",
+		.owner = THIS_MODULE,
+	},
+};
+
+/**
+ * p1022_ds_init: machine driver initialization.
+ *
+ * This function is called when this module is loaded.
+ */
+static int __init p1022_ds_init(void)
+{
+	struct device_node *guts_np;
+	struct resource res;
+
+	pr_info("Freescale P1022 DS ALSA SoC machine driver\n");
+
+	/* Get the physical address of the global utilities registers */
+	guts_np = of_find_compatible_node(NULL, NULL, "fsl,p1022-guts");
+	if (of_address_to_resource(guts_np, 0, &res)) {
+		pr_err("p1022-ds: missing/invalid global utilities node\n");
+		return -EINVAL;
+	}
+	guts_phys = res.start;
+	of_node_put(guts_np);
+
+	return platform_driver_register(&p1022_ds_driver);
+}
+
+/**
+ * p1022_ds_exit: machine driver exit
+ *
+ * This function is called when this driver is unloaded.
+ */
+static void __exit p1022_ds_exit(void)
+{
+	platform_driver_unregister(&p1022_ds_driver);
+}
+
+module_init(p1022_ds_init);
+module_exit(p1022_ds_exit);
+
+MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
+MODULE_DESCRIPTION("Freescale P1022 DS ALSA SoC machine driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 9d88efa06e3c..438146addbb8 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -584,7 +584,7 @@ static struct snd_soc_dai_link ams_delta_dai_link = {
 	.name = "CX20442",
 	.stream_name = "CX20442",
 	.cpu_dai_name ="omap-mcbsp-dai.0",
-	.codec_dai_name = "cx20442-hifi",
+	.codec_dai_name = "cx20442-voice",
 	.init = ams_delta_cx20442_init,
 	.platform_name = "omap-pcm-audio",
 	.codec_name = "cx20442-codec",
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index e30c8325f35e..37f191bbfdd9 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -117,6 +117,24 @@ config SND_PXA2XX_SOC_PALM27X
 	  Say Y if you want to add support for SoC audio on
 	  Palm T|X, T5, E2 or LifeDrive handheld computer.
 
+config SND_SOC_SAARB
+	tristate "SoC Audio support for Marvell Saarb"
+	depends on SND_PXA2XX_SOC && MACH_SAARB
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
+config SND_SOC_TAVOREVB3
+	tristate "SoC Audio support for Marvell Tavor EVB3"
+	depends on SND_PXA2XX_SOC && MACH_TAVOREVB3
+	select SND_PXA_SOC_SSP
+	select SND_SOC_88PM860X
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  Marvell Saarb reference platform.
+
 config SND_SOC_ZYLONITE
 	tristate "SoC Audio support for Marvell Zylonite"
 	depends on SND_PXA2XX_SOC && MACH_ZYLONITE
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index caa03d8f4789..07660165ec8d 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -19,6 +19,8 @@ snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 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-magician-objs := magician.o
 snd-soc-mioa701-objs := mioa701_wm9713.o
@@ -38,6 +40,8 @@ obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
 obj-$(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
+obj-$(CONFIG_SND_SOC_SAARB) += snd-soc-saarb.o
+obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
 obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o
 obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 555689cf6727..97e9423615c9 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -149,7 +149,7 @@ static int corgi_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
index f614607b2055..c82cedb602fd 100644
--- a/sound/soc/pxa/e740_wm9705.c
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -198,6 +198,9 @@ free_mic_amp_gpio:
 static void __exit e740_exit(void)
 {
 	platform_device_unregister(e740_snd_device);
+	gpio_free(GPIO_E740_WM9705_nAVDD2);
+	gpio_free(GPIO_E740_AMP_ON);
+	gpio_free(GPIO_E740_MIC_ON);
 }
 
 module_init(e740_init);
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index add0e1c25bc8..fa752f6ec37d 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -128,7 +128,7 @@ static int poodle_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 
 	/* set the codec system clock for DAC and ADC */
-	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK_XTAL, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 9c2bafa112ad..ac51c6d25c42 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -24,7 +24,6 @@
 #include <mach/dma.h>
 #include <mach/audio.h>
 
-#include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
 static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97)
diff --git a/sound/soc/pxa/saarb.c b/sound/soc/pxa/saarb.c
new file mode 100644
index 000000000000..d63cb474b4e1
--- /dev/null
+++ b/sound/soc/pxa/saarb.c
@@ -0,0 +1,200 @@
+/*
+ * saarb.c -- SoC audio for saarb
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *saarb_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* saarb machine dapm widgets */
+static const struct snd_soc_dapm_widget saarb_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* saarb machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int saarb_i2s_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 width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+
+	return ret;
+}
+
+static struct snd_soc_ops saarb_i2s_ops = {
+	.hw_params	= saarb_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link saarb_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= saarb_pm860x_init,
+		.ops		= &saarb_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_saarb = {
+	.name = "Saarb",
+	.dai_link = saarb_dai,
+	.num_links = ARRAY_SIZE(saarb_dai),
+};
+
+static int saarb_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, saarb_dapm_widgets,
+				  ARRAY_SIZE(saarb_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init saarb_init(void)
+{
+	int ret;
+
+	if (!machine_is_saarb())
+		return -ENODEV;
+	saarb_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!saarb_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(saarb_snd_device, &snd_soc_card_saarb);
+
+	ret = platform_device_add(saarb_snd_device);
+	if (ret)
+		platform_device_put(saarb_snd_device);
+
+	return ret;
+}
+
+static void __exit saarb_exit(void)
+{
+	platform_device_unregister(saarb_snd_device);
+}
+
+module_init(saarb_init);
+module_exit(saarb_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Saarb");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/tavorevb3.c b/sound/soc/pxa/tavorevb3.c
new file mode 100644
index 000000000000..248c283fc4df
--- /dev/null
+++ b/sound/soc/pxa/tavorevb3.c
@@ -0,0 +1,200 @@
+/*
+ * tavorevb3.c -- SoC audio for Tavor EVB3
+ *
+ * Copyright (C) 2010 Marvell International Ltd.
+ * 	Haojian Zhuang <haojian.zhuang@marvell.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/device.h>
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/88pm860x-codec.h"
+#include "pxa-ssp.h"
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd);
+
+static struct platform_device *evb3_snd_device;
+
+static struct snd_soc_jack hs_jack, mic_jack;
+
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{ .pin = "Headset Stereophone",	.mask = SND_JACK_HEADPHONE, },
+};
+
+static struct snd_soc_jack_pin mic_jack_pins[] = {
+	{ .pin = "Headset Mic 2",	.mask = SND_JACK_MICROPHONE, },
+};
+
+/* tavorevb3 machine dapm widgets */
+static const struct snd_soc_dapm_widget evb3_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 1", NULL),
+	SND_SOC_DAPM_LINE("Lineout Out 2", NULL),
+	SND_SOC_DAPM_SPK("Ext Speaker", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 1", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic 2", NULL),
+	SND_SOC_DAPM_MIC("Ext Mic 3", NULL),
+};
+
+/* tavorevb3 machine audio map */
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headset Stereophone", NULL, "HS1"},
+	{"Headset Stereophone", NULL, "HS2"},
+
+	{"Ext Speaker", NULL, "LSP"},
+	{"Ext Speaker", NULL, "LSN"},
+
+	{"Lineout Out 1", NULL, "LINEOUT1"},
+	{"Lineout Out 2", NULL, "LINEOUT2"},
+
+	{"MIC1P", NULL, "Mic1 Bias"},
+	{"MIC1N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Ext Mic 1"},
+
+	{"MIC2P", NULL, "Mic1 Bias"},
+	{"MIC2N", NULL, "Mic1 Bias"},
+	{"Mic1 Bias", NULL, "Headset Mic 2"},
+
+	{"MIC3P", NULL, "Mic3 Bias"},
+	{"MIC3N", NULL, "Mic3 Bias"},
+	{"Mic3 Bias", NULL, "Ext Mic 3"},
+};
+
+static int evb3_i2s_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 width = snd_pcm_format_physical_width(params_format(params));
+	int ret;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_NET_PLL, 0,
+				     PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 0, PM860X_CLK_DIR_OUT);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 3, 3, 2, width);
+	return ret;
+}
+
+static struct snd_soc_ops evb3_i2s_ops = {
+	.hw_params	= evb3_i2s_hw_params,
+};
+
+static struct snd_soc_dai_link evb3_dai[] = {
+	{
+		.name		= "88PM860x I2S",
+		.stream_name	= "I2S Audio",
+		.cpu_dai_name	= "pxa-ssp-dai.1",
+		.codec_dai_name	= "88pm860x-i2s",
+		.platform_name	= "pxa-pcm-audio",
+		.codec_name	= "88pm860x-codec",
+		.init		= evb3_pm860x_init,
+		.ops		= &evb3_i2s_ops,
+	},
+};
+
+static struct snd_soc_card snd_soc_card_evb3 = {
+	.name = "Tavor EVB3",
+	.dai_link = evb3_dai,
+	.num_links = ARRAY_SIZE(evb3_dai),
+};
+
+static int evb3_pm860x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	int ret;
+
+	snd_soc_dapm_new_controls(codec, evb3_dapm_widgets,
+				  ARRAY_SIZE(evb3_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Speaker");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 1");
+	snd_soc_dapm_enable_pin(codec, "Ext Mic 3");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic 2");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE
+			| SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2,
+			&hs_jack);
+	snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+			      hs_jack_pins);
+	snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE,
+			 &mic_jack);
+	snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
+			      mic_jack_pins);
+
+	/* headphone, microphone detection & headset short detection */
+	pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE,
+			      SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2);
+	pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE);
+	return 0;
+}
+
+static int __init tavorevb3_init(void)
+{
+	int ret;
+
+	if (!machine_is_tavorevb3())
+		return -ENODEV;
+	evb3_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!evb3_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(evb3_snd_device, &snd_soc_card_evb3);
+
+	ret = platform_device_add(evb3_snd_device);
+	if (ret)
+		platform_device_put(evb3_snd_device);
+
+	return ret;
+}
+
+static void __exit tavorevb3_exit(void)
+{
+	platform_device_unregister(evb3_snd_device);
+}
+
+module_init(tavorevb3_init);
+module_exit(tavorevb3_exit);
+
+MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
+MODULE_DESCRIPTION("ALSA SoC 88PM860x Tavor EVB3");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 7093c1787128..65352c7d4b7f 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -3122,10 +3122,12 @@ int snd_soc_register_codec(struct device *dev,
 		fixup_codec_formats(&dai_drv[i].capture);
 	}
 
-	/* register DAIs */
-	ret = snd_soc_register_dais(dev, dai_drv, num_dai);
-	if (ret < 0)
+	/* register any DAIs */
+	if (num_dai) {
+		ret = snd_soc_register_dais(dev, dai_drv, num_dai);
+		if (ret < 0)
 			goto error;
+	}
 
 	mutex_lock(&client_mutex);
 	list_add(&codec->list, &codec_list);
@@ -3164,8 +3166,9 @@ void snd_soc_unregister_codec(struct device *dev)
 	return;
 
 found:
-	for (i = 0; i < codec->num_dai; i++)
-		snd_soc_unregister_dai(dev);
+	if (codec->num_dai)
+		for (i = 0; i < codec->num_dai; i++)
+			snd_soc_unregister_dai(dev);
 
 	mutex_lock(&client_mutex);
 	list_del(&codec->list);