Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

19 files changed, 1613 insertions, 136 deletions

diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig
index fa6a6d2abffc..fe9d548a6837 100644
--- a/sound/soc/blackfin/Kconfig
+++ b/sound/soc/blackfin/Kconfig
@@ -27,6 +27,19 @@ config SND_SOC_BFIN_EVAL_ADAU1701
 	  board connected to one of the Blackfin evaluation boards like the
 	  BF5XX-STAMP or BF5XX-EZKIT.
 
+config SND_SOC_BFIN_EVAL_ADAV80X
+	tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards"
+	depends on SND_BF5XX_I2S && (SPI_MASTER || I2C)
+	select SND_BF5XX_SOC_I2S
+	select SND_SOC_ADAV80X
+	help
+	  Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or
+	  EVAL-ADAV803 board connected to one of the Blackfin evaluation boards
+	  like the BF5XX-STAMP or BF5XX-EZKIT.
+
+	  Note: This driver assumes that the ADAV80X digital record and playback
+	  interfaces are connected to the first SPORT port on the BF5XX board.
+
 config SND_BF5XX_SOC_AD73311
 	tristate "SoC AD73311 Audio support for Blackfin"
 	depends on SND_BF5XX_I2S
diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile
index f01bff63177d..6018bf52a234 100644
--- a/sound/soc/blackfin/Makefile
+++ b/sound/soc/blackfin/Makefile
@@ -22,6 +22,7 @@ snd-ssm2602-objs := bf5xx-ssm2602.o
 snd-ad73311-objs := bf5xx-ad73311.o
 snd-ad193x-objs := bf5xx-ad193x.o
 snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o
+snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o
 
 obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
@@ -29,3 +30,4 @@ obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
 obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o
 obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o
+obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 4a805a859723..61ddf942fd4d 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream)
 	pr_debug("%s enter\n", __func__);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		diff = sport_curr_offset_tx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	} else {
 		diff = sport_curr_offset_rx(sport);
-		frames = bytes_to_frames(substream->runtime, diff);
 	}
+
+	/*
+	 * TX at least can report one frame beyond the end of the
+	 * buffer if we hit the wraparound case - clamp to within the
+	 * buffer as the ALSA APIs require.
+	 */
+	if (diff == snd_pcm_lib_buffer_bytes(substream))
+		diff = 0;
+
+	frames = bytes_to_frames(substream->runtime, diff);
+
 	return frames;
 }
 
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
new file mode 100644
index 000000000000..8d014d01626e
--- /dev/null
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -0,0 +1,173 @@
+/*
+ * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin
+ * evaluation boards.
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include "../codecs/adav80x.h"
+
+static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = {
+	{ "Line Out", NULL, "VOUTL" },
+	{ "Line Out", NULL, "VOUTR" },
+
+	{ "VINL", NULL, "Line In" },
+	{ "VINR", NULL, "Line In" },
+};
+
+static int bfin_eval_adav80x_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 *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	int 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)
+		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)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL,
+			27000000, params_rate(params) * 256);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1,
+			params_rate(params) * 256, SND_SOC_CLOCK_IN);
+
+	return ret;
+}
+
+static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0,
+	    SND_SOC_CLOCK_OUT);
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0,
+	    SND_SOC_CLOCK_OUT);
+
+	snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0);
+
+	return 0;
+}
+
+static struct snd_soc_ops bfin_eval_adav80x_ops = {
+	.hw_params = bfin_eval_adav80x_hw_params,
+};
+
+static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = {
+	{
+		.name = "adav80x",
+		.stream_name = "ADAV80x HiFi",
+		.cpu_dai_name = "bfin-i2s.0",
+		.codec_dai_name = "adav80x-hifi",
+		.platform_name = "bfin-i2s-pcm-audio",
+		.init = bfin_eval_adav80x_codec_init,
+		.ops = &bfin_eval_adav80x_ops,
+	},
+};
+
+static struct snd_soc_card bfin_eval_adav80x = {
+	.name = "bfin-eval-adav80x",
+	.dai_link = bfin_eval_adav80x_dais,
+	.num_links = ARRAY_SIZE(bfin_eval_adav80x_dais),
+
+	.dapm_widgets		= bfin_eval_adav80x_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets),
+	.dapm_routes		= bfin_eval_adav80x_dapm_routes,
+	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adav80x_dapm_routes),
+};
+
+enum bfin_eval_adav80x_type {
+	BFIN_EVAL_ADAV801,
+	BFIN_EVAL_ADAV803,
+};
+
+static int bfin_eval_adav80x_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &bfin_eval_adav80x;
+	const char *codec_name;
+
+	switch (platform_get_device_id(pdev)->driver_data) {
+	case BFIN_EVAL_ADAV801:
+		codec_name = "spi0.1";
+		break;
+	case BFIN_EVAL_ADAV803:
+		codec_name = "adav803.0-0034";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	bfin_eval_adav80x_dais[0].codec_name = codec_name;
+
+	card->dev = &pdev->dev;
+
+	return snd_soc_register_card(&bfin_eval_adav80x);
+}
+
+static int __devexit bfin_eval_adav80x_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+
+	return 0;
+}
+
+static const struct platform_device_id bfin_eval_adav80x_ids[] = {
+	{ "bfin-eval-adav801", BFIN_EVAL_ADAV801 },
+	{ "bfin-eval-adav803", BFIN_EVAL_ADAV803 },
+	{ },
+};
+MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids);
+
+static struct platform_driver bfin_eval_adav80x_driver = {
+	.driver = {
+		.name = "bfin-eval-adav80x",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+	},
+	.probe = bfin_eval_adav80x_probe,
+	.remove = __devexit_p(bfin_eval_adav80x_remove),
+	.id_table = bfin_eval_adav80x_ids,
+};
+
+static int __init bfin_eval_adav80x_init(void)
+{
+	return platform_driver_register(&bfin_eval_adav80x_driver);
+}
+module_init(bfin_eval_adav80x_init);
+
+static void __exit bfin_eval_adav80x_exit(void)
+{
+	platform_driver_unregister(&bfin_eval_adav80x_driver);
+}
+module_exit(bfin_eval_adav80x_exit);
+
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 2998e659379c..ff43405752a1 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311
+	select SND_SOC_ADAV80X
 	select SND_SOC_ADS117X
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
@@ -137,6 +138,9 @@ config SND_SOC_ADAU1701
 	select SIGMA
 	tristate
 
+config SND_SOC_ADAV80X
+	tristate
+
 config SND_SOC_ADS117X
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 51cd3d489589..4957431e23fc 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
 snd-soc-adau1701-objs := adau1701.o
+snd-soc-adav80x-objs := adav80x.o
 snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
@@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X)	+= snd-soc-ad193x.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
 obj-$(CONFIG_SND_SOC_ADAU1701)  += snd-soc-adau1701.o
+obj-$(CONFIG_SND_SOC_ADAV80X)  += snd-soc-adav80x.o
 obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
new file mode 100644
index 000000000000..e30fba62392d
--- /dev/null
+++ b/sound/soc/codecs/adav80x.c
@@ -0,0 +1,951 @@
+/*
+ * ADAV80X Audio Codec driver supporting ADAV801, ADAV803
+ *
+ * Copyright 2011 Analog Devices Inc.
+ * Author: Yi Li <yi.li@analog.com>
+ * Author: Lars-Peter Clausen <lars@metafoo.de>
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/i2c.h>
+#include <linux/spi/spi.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include <sound/soc.h>
+
+#include "adav80x.h"
+
+#define ADAV80X_PLAYBACK_CTRL	0x04
+#define ADAV80X_AUX_IN_CTRL	0x05
+#define ADAV80X_REC_CTRL	0x06
+#define ADAV80X_AUX_OUT_CTRL	0x07
+#define ADAV80X_DPATH_CTRL1	0x62
+#define ADAV80X_DPATH_CTRL2	0x63
+#define ADAV80X_DAC_CTRL1	0x64
+#define ADAV80X_DAC_CTRL2	0x65
+#define ADAV80X_DAC_CTRL3	0x66
+#define ADAV80X_DAC_L_VOL	0x68
+#define ADAV80X_DAC_R_VOL	0x69
+#define ADAV80X_PGA_L_VOL	0x6c
+#define ADAV80X_PGA_R_VOL	0x6d
+#define ADAV80X_ADC_CTRL1	0x6e
+#define ADAV80X_ADC_CTRL2	0x6f
+#define ADAV80X_ADC_L_VOL	0x70
+#define ADAV80X_ADC_R_VOL	0x71
+#define ADAV80X_PLL_CTRL1	0x74
+#define ADAV80X_PLL_CTRL2	0x75
+#define ADAV80X_ICLK_CTRL1	0x76
+#define ADAV80X_ICLK_CTRL2	0x77
+#define ADAV80X_PLL_CLK_SRC	0x78
+#define ADAV80X_PLL_OUTE	0x7a
+
+#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll)	0x00
+#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll)	(0x40 << (pll))
+#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll)	(0x40 << (pll))
+
+#define ADAV80X_ICLK_CTRL1_DAC_SRC(src)		((src) << 5)
+#define ADAV80X_ICLK_CTRL1_ADC_SRC(src)		((src) << 2)
+#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src)	(src)
+#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src)	((src) << 3)
+
+#define ADAV80X_PLL_CTRL1_PLLDIV		0x10
+#define ADAV80X_PLL_CTRL1_PLLPD(pll)		(0x04 << (pll))
+#define ADAV80X_PLL_CTRL1_XTLPD			0x02
+
+#define ADAV80X_PLL_CTRL2_FIELD(pll, x)		((x) << ((pll) * 4))
+
+#define ADAV80X_PLL_CTRL2_FS_48(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x00)
+#define ADAV80X_PLL_CTRL2_FS_32(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x08)
+#define ADAV80X_PLL_CTRL2_FS_44(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c)
+
+#define ADAV80X_PLL_CTRL2_SEL(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x02)
+#define ADAV80X_PLL_CTRL2_DOUB(pll)	ADAV80X_PLL_CTRL2_FIELD((pll), 0x01)
+#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f)
+
+#define ADAV80X_ADC_CTRL1_MODULATOR_MASK	0x80
+#define ADAV80X_ADC_CTRL1_MODULATOR_128FS	0x00
+#define ADAV80X_ADC_CTRL1_MODULATOR_64FS	0x80
+
+#define ADAV80X_DAC_CTRL1_PD			0x80
+
+#define ADAV80X_DAC_CTRL2_DIV1			0x00
+#define ADAV80X_DAC_CTRL2_DIV1_5		0x10
+#define ADAV80X_DAC_CTRL2_DIV2			0x20
+#define ADAV80X_DAC_CTRL2_DIV3			0x30
+#define ADAV80X_DAC_CTRL2_DIV_MASK		0x30
+
+#define ADAV80X_DAC_CTRL2_INTERPOL_256FS	0x00
+#define ADAV80X_DAC_CTRL2_INTERPOL_128FS	0x40
+#define ADAV80X_DAC_CTRL2_INTERPOL_64FS		0x80
+#define ADAV80X_DAC_CTRL2_INTERPOL_MASK		0xc0
+
+#define ADAV80X_DAC_CTRL2_DEEMPH_NONE		0x00
+#define ADAV80X_DAC_CTRL2_DEEMPH_44		0x01
+#define ADAV80X_DAC_CTRL2_DEEMPH_32		0x02
+#define ADAV80X_DAC_CTRL2_DEEMPH_48		0x03
+#define ADAV80X_DAC_CTRL2_DEEMPH_MASK		0x01
+
+#define ADAV80X_CAPTURE_MODE_MASTER		0x20
+#define ADAV80X_CAPTURE_WORD_LEN24		0x00
+#define ADAV80X_CAPTURE_WORD_LEN20		0x04
+#define ADAV80X_CAPTRUE_WORD_LEN18		0x08
+#define ADAV80X_CAPTURE_WORD_LEN16		0x0c
+#define ADAV80X_CAPTURE_WORD_LEN_MASK		0x0c
+
+#define ADAV80X_CAPTURE_MODE_LEFT_J		0x00
+#define ADAV80X_CAPTURE_MODE_I2S		0x01
+#define ADAV80X_CAPTURE_MODE_RIGHT_J		0x03
+#define ADAV80X_CAPTURE_MODE_MASK		0x03
+
+#define ADAV80X_PLAYBACK_MODE_MASTER		0x10
+#define ADAV80X_PLAYBACK_MODE_LEFT_J		0x00
+#define ADAV80X_PLAYBACK_MODE_I2S		0x01
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24	0x04
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20	0x05
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18	0x06
+#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16	0x07
+#define ADAV80X_PLAYBACK_MODE_MASK		0x07
+
+#define ADAV80X_PLL_OUTE_SYSCLKPD(x)		BIT(2 - (x))
+
+static u8 adav80x_default_regs[] = {
+	0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00,
+	0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37,
+	0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b,
+	0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00,
+	0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee,
+	0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f,
+	0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00,
+	0x00, 0x00, 0x00, 0x00, 0x52, 0x00,
+};
+
+struct adav80x {
+	enum snd_soc_control_type control_type;
+
+	enum adav80x_clk_src clk_src;
+	unsigned int sysclk;
+	enum adav80x_pll_src pll_src;
+
+	unsigned int dai_fmt[2];
+	unsigned int rate;
+	bool deemph;
+	bool sysclk_pd[3];
+};
+
+static const char *adav80x_mux_text[] = {
+	"ADC",
+	"Playback",
+	"Aux Playback",
+};
+
+static const unsigned int adav80x_mux_values[] = {
+	0, 2, 3,
+};
+
+#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \
+	SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \
+		ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \
+		adav80x_mux_values)
+
+static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0);
+static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3);
+static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3);
+
+static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum);
+static const struct snd_kcontrol_new adav80x_capture_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum);
+static const struct snd_kcontrol_new adav80x_dac_mux_ctrl =
+	SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum);
+
+#define ADAV80X_MUX(name, ctrl) \
+	SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl)
+
+static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1),
+	SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1),
+
+	SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0),
+	SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0),
+
+	ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl),
+	ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl),
+	ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl),
+
+	SND_SOC_DAPM_INPUT("VINR"),
+	SND_SOC_DAPM_INPUT("VINL"),
+	SND_SOC_DAPM_OUTPUT("VOUTR"),
+	SND_SOC_DAPM_OUTPUT("VOUTL"),
+
+	SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0),
+	SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0),
+};
+
+static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	const char *clk;
+
+	switch (adav80x->clk_src) {
+	case ADAV80X_CLK_PLL1:
+		clk = "PLL1";
+		break;
+	case ADAV80X_CLK_PLL2:
+		clk = "PLL2";
+		break;
+	case ADAV80X_CLK_XTAL:
+		clk = "OSC";
+		break;
+	default:
+		return 0;
+	}
+
+	return strcmp(source->name, clk) == 0;
+}
+
+static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink)
+{
+	struct snd_soc_codec *codec = source->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL;
+}
+
+
+static const struct snd_soc_dapm_route adav80x_dapm_routes[] = {
+	{ "DAC Select", "ADC", "ADC" },
+	{ "DAC Select", "Playback", "AIFIN" },
+	{ "DAC Select", "Aux Playback", "AIFAUXIN" },
+	{ "DAC", NULL,  "DAC Select" },
+
+	{ "Capture Select", "ADC", "ADC" },
+	{ "Capture Select", "Playback", "AIFIN" },
+	{ "Capture Select", "Aux Playback", "AIFAUXIN" },
+	{ "AIFOUT", NULL,  "Capture Select" },
+
+	{ "Aux Capture Select", "ADC", "ADC" },
+	{ "Aux Capture Select", "Playback", "AIFIN" },
+	{ "Aux Capture Select", "Aux Playback", "AIFAUXIN" },
+	{ "AIFAUXOUT", NULL,  "Aux Capture Select" },
+
+	{ "VOUTR",  NULL, "DAC" },
+	{ "VOUTL",  NULL, "DAC" },
+
+	{ "Left PGA", NULL, "VINL" },
+	{ "Right PGA", NULL, "VINR" },
+	{ "ADC", NULL, "Left PGA" },
+	{ "ADC", NULL, "Right PGA" },
+
+	{ "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check },
+	{ "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check },
+	{ "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check },
+	{ "PLL1", NULL, "OSC", adav80x_dapm_pll_check },
+	{ "PLL2", NULL, "OSC", adav80x_dapm_pll_check },
+
+	{ "ADC", NULL, "SYSCLK" },
+	{ "DAC", NULL, "SYSCLK" },
+	{ "AIFOUT", NULL, "SYSCLK" },
+	{ "AIFAUXOUT", NULL, "SYSCLK" },
+	{ "AIFIN", NULL, "SYSCLK" },
+	{ "AIFAUXIN", NULL, "SYSCLK" },
+};
+
+static int adav80x_set_deemph(struct snd_soc_codec *codec)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (adav80x->deemph) {
+		switch (adav80x->rate) {
+		case 32000:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_32;
+			break;
+		case 44100:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_44;
+			break;
+		case 48000:
+		case 64000:
+		case 88200:
+		case 96000:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_48;
+			break;
+		default:
+			val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+			break;
+		}
+	} else {
+		val = ADAV80X_DAC_CTRL2_DEEMPH_NONE;
+	}
+
+	return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+		ADAV80X_DAC_CTRL2_DEEMPH_MASK, val);
+}
+
+static int adav80x_put_deemph(struct snd_kcontrol *kcontrol,
+		struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int deemph = ucontrol->value.enumerated.item[0];
+
+	if (deemph > 1)
+		return -EINVAL;
+
+	adav80x->deemph = deemph;
+
+	return adav80x_set_deemph(codec);
+}
+
+static int adav80x_get_deemph(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	ucontrol->value.enumerated.item[0] = adav80x->deemph;
+	return 0;
+};
+
+static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0);
+static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0);
+
+static const struct snd_kcontrol_new adav80x_controls[] = {
+	SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL,
+		ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+	SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL,
+			ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv),
+
+	SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL,
+			ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv),
+
+	SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1),
+
+	SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0),
+
+	SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0,
+			adav80x_get_deemph, adav80x_put_deemph),
+};
+
+static unsigned int adav80x_port_ctrl_regs[2][2] = {
+	{ ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, },
+	{ ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL },
+};
+
+static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int capture = 0x00;
+	unsigned int playback = 0x00;
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		capture |= ADAV80X_CAPTURE_MODE_MASTER;
+		playback |= ADAV80X_PLAYBACK_MODE_MASTER;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		capture |= ADAV80X_CAPTURE_MODE_I2S;
+		playback |= ADAV80X_PLAYBACK_MODE_I2S;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		capture |= ADAV80X_CAPTURE_MODE_LEFT_J;
+		playback |= ADAV80X_PLAYBACK_MODE_LEFT_J;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		capture |= ADAV80X_CAPTURE_MODE_RIGHT_J;
+		playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+		ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER,
+		capture);
+	snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback);
+
+	adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+
+	return 0;
+}
+
+static int adav80x_set_adc_clock(struct snd_soc_codec *codec,
+		unsigned int sample_rate)
+{
+	unsigned int val;
+
+	if (sample_rate <= 48000)
+		val = ADAV80X_ADC_CTRL1_MODULATOR_128FS;
+	else
+		val = ADAV80X_ADC_CTRL1_MODULATOR_64FS;
+
+	snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1,
+		ADAV80X_ADC_CTRL1_MODULATOR_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_set_dac_clock(struct snd_soc_codec *codec,
+		unsigned int sample_rate)
+{
+	unsigned int val;
+
+	if (sample_rate <= 48000)
+		val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS;
+	else
+		val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS;
+
+	snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2,
+		ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK,
+		val);
+
+	return 0;
+}
+
+static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec,
+		struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+	unsigned int val;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAV80X_CAPTURE_WORD_LEN16;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		val = ADAV80X_CAPTRUE_WORD_LEN18;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAV80X_CAPTURE_WORD_LEN20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAV80X_CAPTURE_WORD_LEN24;
+		break;
+	default:
+		break;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0],
+		ADAV80X_CAPTURE_WORD_LEN_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec,
+		struct snd_soc_dai *dai, snd_pcm_format_t format)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+
+	if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J)
+		return 0;
+
+	switch (format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16;
+		break;
+	case SNDRV_PCM_FORMAT_S18_3LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18;
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24;
+		break;
+	default:
+		break;
+	}
+
+	snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1],
+		ADAV80X_PLAYBACK_MODE_MASK, val);
+
+	return 0;
+}
+
+static int adav80x_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;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int rate = params_rate(params);
+
+	if (rate * 256 != adav80x->sysclk)
+		return -EINVAL;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		adav80x_set_playback_pcm_format(codec, dai,
+			params_format(params));
+		adav80x_set_dac_clock(codec, rate);
+	} else {
+		adav80x_set_capture_pcm_format(codec, dai,
+			params_format(params));
+		adav80x_set_adc_clock(codec, rate);
+	}
+	adav80x->rate = rate;
+	adav80x_set_deemph(codec);
+
+	return 0;
+}
+
+static int adav80x_set_sysclk(struct snd_soc_codec *codec,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (dir == SND_SOC_CLOCK_IN) {
+		switch (clk_id) {
+		case ADAV80X_CLK_XIN:
+		case ADAV80X_CLK_XTAL:
+		case ADAV80X_CLK_MCLKI:
+		case ADAV80X_CLK_PLL1:
+		case ADAV80X_CLK_PLL2:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		adav80x->sysclk = freq;
+
+		if (adav80x->clk_src != clk_id) {
+			unsigned int iclk_ctrl1, iclk_ctrl2;
+
+			adav80x->clk_src = clk_id;
+			if (clk_id == ADAV80X_CLK_XTAL)
+				clk_id = ADAV80X_CLK_XIN;
+
+			iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) |
+					ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) |
+					ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id);
+			iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id);
+
+			snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1);
+			snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2);
+
+			snd_soc_dapm_sync(&codec->dapm);
+		}
+	} else {
+		unsigned int mask;
+
+		switch (clk_id) {
+		case ADAV80X_CLK_SYSCLK1:
+		case ADAV80X_CLK_SYSCLK2:
+		case ADAV80X_CLK_SYSCLK3:
+			break;
+		default:
+			return -EINVAL;
+		}
+
+		clk_id -= ADAV80X_CLK_SYSCLK1;
+		mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id);
+
+		if (freq == 0) {
+			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask);
+			adav80x->sysclk_pd[clk_id] = true;
+		} else {
+			snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0);
+			adav80x->sysclk_pd[clk_id] = false;
+		}
+
+		if (adav80x->sysclk_pd[0])
+			snd_soc_dapm_disable_pin(&codec->dapm, "PLL1");
+		else
+			snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+
+		if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2])
+			snd_soc_dapm_disable_pin(&codec->dapm, "PLL2");
+		else
+			snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+		snd_soc_dapm_sync(&codec->dapm);
+	}
+
+	return 0;
+}
+
+static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id,
+		int source, unsigned int freq_in, unsigned int freq_out)
+{
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+	unsigned int pll_ctrl1 = 0;
+	unsigned int pll_ctrl2 = 0;
+	unsigned int pll_src;
+
+	switch (source) {
+	case ADAV80X_PLL_SRC_XTAL:
+	case ADAV80X_PLL_SRC_XIN:
+	case ADAV80X_PLL_SRC_MCLKI:
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (!freq_out)
+		return 0;
+
+	switch (freq_in) {
+	case 27000000:
+		break;
+	case 54000000:
+		if (source == ADAV80X_PLL_SRC_XIN) {
+			pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV;
+			break;
+		}
+	default:
+		return -EINVAL;
+	}
+
+	if (freq_out > 12288000) {
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id);
+		freq_out /= 2;
+	}
+
+	/* freq_out = sample_rate * 256 */
+	switch (freq_out) {
+	case 8192000:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id);
+		break;
+	case 11289600:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id);
+		break;
+	case 12288000:
+		pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV,
+		pll_ctrl1);
+	snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2,
+			ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2);
+
+	if (source != adav80x->pll_src) {
+		if (source == ADAV80X_PLL_SRC_MCLKI)
+			pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id);
+		else
+			pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id);
+
+		snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC,
+				ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src);
+
+		adav80x->pll_src = source;
+
+		snd_soc_dapm_sync(&codec->dapm);
+	}
+
+	return 0;
+}
+
+static int adav80x_set_bias_level(struct snd_soc_codec *codec,
+		enum snd_soc_bias_level level)
+{
+	unsigned int mask = ADAV80X_DAC_CTRL1_PD;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+	case SND_SOC_BIAS_PREPARE:
+		break;
+	case SND_SOC_BIAS_STANDBY:
+		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00);
+		break;
+	case SND_SOC_BIAS_OFF:
+		snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask);
+		break;
+	}
+
+	codec->dapm.bias_level = level;
+	return 0;
+}
+
+/* Enforce the same sample rate on all audio interfaces */
+static int adav80x_dai_startup(struct snd_pcm_substream *substream,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (!codec->active || !adav80x->rate)
+		return 0;
+
+	return snd_pcm_hw_constraint_minmax(substream->runtime,
+			SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+}
+
+static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	if (!codec->active)
+		adav80x->rate = 0;
+}
+
+static const struct snd_soc_dai_ops adav80x_dai_ops = {
+	.set_fmt = adav80x_set_dai_fmt,
+	.hw_params = adav80x_hw_params,
+	.startup = adav80x_dai_startup,
+	.shutdown = adav80x_dai_shutdown,
+};
+
+#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \
+	SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
+
+#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver adav80x_dais[] = {
+	{
+		.name = "adav80x-hifi",
+		.id = 0,
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_PLAYBACK_RATES,
+			.formats = ADAV80X_FORMATS,
+	},
+		.capture = {
+			.stream_name = "HiFi Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_CAPTURE_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.ops = &adav80x_dai_ops,
+	},
+	{
+		.name = "adav80x-aux",
+		.id = 1,
+		.playback = {
+			.stream_name = "Aux Playback",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_PLAYBACK_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.capture = {
+			.stream_name = "Aux Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = ADAV80X_CAPTURE_RATES,
+			.formats = ADAV80X_FORMATS,
+		},
+		.ops = &adav80x_dai_ops,
+	},
+};
+
+static int adav80x_probe(struct snd_soc_codec *codec)
+{
+	int ret;
+	struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec);
+
+	ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type);
+	if (ret) {
+		dev_err(codec->dev, "failed to set cache I/O: %d\n", ret);
+		return ret;
+	}
+
+	/* Force PLLs on for SYSCLK output */
+	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1");
+	snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2");
+
+	/* Power down S/PDIF receiver, since it is currently not supported */
+	snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20);
+	/* Disable DAC zero flag */
+	snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6);
+
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+}
+
+static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state)
+{
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int adav80x_resume(struct snd_soc_codec *codec)
+{
+	adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+	codec->cache_sync = 1;
+	snd_soc_cache_sync(codec);
+
+	return 0;
+}
+
+static int adav80x_remove(struct snd_soc_codec *codec)
+{
+	return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static struct snd_soc_codec_driver adav80x_codec_driver = {
+	.probe = adav80x_probe,
+	.remove = adav80x_remove,
+	.suspend = adav80x_suspend,
+	.resume = adav80x_resume,
+	.set_bias_level = adav80x_set_bias_level,
+
+	.set_pll = adav80x_set_pll,
+	.set_sysclk = adav80x_set_sysclk,
+
+	.reg_word_size = sizeof(u8),
+	.reg_cache_size = ARRAY_SIZE(adav80x_default_regs),
+	.reg_cache_default = adav80x_default_regs,
+
+	.controls = adav80x_controls,
+	.num_controls = ARRAY_SIZE(adav80x_controls),
+	.dapm_widgets = adav80x_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets),
+	.dapm_routes = adav80x_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes),
+};
+
+static int __devinit adav80x_bus_probe(struct device *dev,
+		enum snd_soc_control_type control_type)
+{
+	struct adav80x *adav80x;
+	int ret;
+
+	adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL);
+	if (!adav80x)
+		return -ENOMEM;
+
+	dev_set_drvdata(dev, adav80x);
+	adav80x->control_type = control_type;
+
+	ret = snd_soc_register_codec(dev, &adav80x_codec_driver,
+		adav80x_dais, ARRAY_SIZE(adav80x_dais));
+	if (ret)
+		kfree(adav80x);
+
+	return ret;
+}
+
+static int __devexit adav80x_bus_remove(struct device *dev)
+{
+	snd_soc_unregister_codec(dev);
+	kfree(dev_get_drvdata(dev));
+	return 0;
+}
+
+#if defined(CONFIG_SPI_MASTER)
+static int __devinit adav80x_spi_probe(struct spi_device *spi)
+{
+	return adav80x_bus_probe(&spi->dev, SND_SOC_SPI);
+}
+
+static int __devexit adav80x_spi_remove(struct spi_device *spi)
+{
+	return adav80x_bus_remove(&spi->dev);
+}
+
+static struct spi_driver adav80x_spi_driver = {
+	.driver = {
+		.name	= "adav801",
+		.owner	= THIS_MODULE,
+	},
+	.probe		= adav80x_spi_probe,
+	.remove		= __devexit_p(adav80x_spi_remove),
+};
+#endif
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+static const struct i2c_device_id adav80x_id[] = {
+	{ "adav803", 0 },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, adav80x_id);
+
+static int __devinit adav80x_i2c_probe(struct i2c_client *client,
+		const struct i2c_device_id *id)
+{
+	return adav80x_bus_probe(&client->dev, SND_SOC_I2C);
+}
+
+static int __devexit adav80x_i2c_remove(struct i2c_client *client)
+{
+	return adav80x_bus_remove(&client->dev);
+}
+
+static struct i2c_driver adav80x_i2c_driver = {
+	.driver = {
+		.name = "adav803",
+		.owner = THIS_MODULE,
+	},
+	.probe = adav80x_i2c_probe,
+	.remove = __devexit_p(adav80x_i2c_remove),
+	.id_table = adav80x_id,
+};
+#endif
+
+static int __init adav80x_init(void)
+{
+	int ret = 0;
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&adav80x_i2c_driver);
+	if (ret)
+		return ret;
+#endif
+
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&adav80x_spi_driver);
+#endif
+
+	return ret;
+}
+module_init(adav80x_init);
+
+static void __exit adav80x_exit(void)
+{
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&adav80x_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&adav80x_spi_driver);
+#endif
+}
+module_exit(adav80x_exit);
+
+MODULE_DESCRIPTION("ASoC ADAV80x driver");
+MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
+MODULE_AUTHOR("Yi Li <yi.li@analog.com>>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h
new file mode 100644
index 000000000000..adb0fc76d4e3
--- /dev/null
+++ b/sound/soc/codecs/adav80x.h
@@ -0,0 +1,35 @@
+/*
+ * header file for ADAV80X parts
+ *
+ * Copyright 2011 Analog Devices Inc.
+ *
+ * Licensed under the GPL-2 or later.
+ */
+
+#ifndef _ADAV80X_H
+#define _ADAV80X_H
+
+enum adav80x_pll_src {
+	ADAV80X_PLL_SRC_XIN,
+	ADAV80X_PLL_SRC_XTAL,
+	ADAV80X_PLL_SRC_MCLKI,
+};
+
+enum adav80x_pll {
+	ADAV80X_PLL1 = 0,
+	ADAV80X_PLL2 = 1,
+};
+
+enum adav80x_clk_src {
+	ADAV80X_CLK_XIN = 0,
+	ADAV80X_CLK_MCLKI = 1,
+	ADAV80X_CLK_PLL1 = 2,
+	ADAV80X_CLK_PLL2 = 3,
+	ADAV80X_CLK_XTAL = 6,
+
+	ADAV80X_CLK_SYSCLK1 = 6,
+	ADAV80X_CLK_SYSCLK2 = 7,
+	ADAV80X_CLK_SYSCLK3 = 8,
+};
+
+#endif
diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c
index dc2350e6350b..70a68fd96c46 100644
--- a/sound/soc/codecs/wm8994.c
+++ b/sound/soc/codecs/wm8994.c
@@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src,
 		snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset,
 				    WM8994_FLL1_ENA | WM8994_FLL1_FRAC,
 				    reg);
+
+		msleep(5);
 	}
 
 	wm8994->fll[id].in = freq_in;
diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c
index 91c6b39de50c..a4691321f9b3 100644
--- a/sound/soc/codecs/wm9081.c
+++ b/sound/soc/codecs/wm9081.c
@@ -727,7 +727,7 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0,
 SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0),
 
 SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0),
-SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
+SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0),
 
 SND_SOC_DAPM_OUTPUT("LINEOUT"),
 SND_SOC_DAPM_OUTPUT("SPKN"),
diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c
index 4ebc131b7f36..2d6c88b68a14 100644
--- a/sound/soc/codecs/wm_hubs.c
+++ b/sound/soc/codecs/wm_hubs.c
@@ -133,9 +133,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 		break;
 	case 1:
 		reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
-		reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
+		reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
 			>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-		reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
+		reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
 		break;
 	default:
 		WARN(1, "Unknown DCS readback method\n");
@@ -149,13 +149,13 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 		dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
 			hubs->dcs_codes);
 
-		/* HPOUT1L */
-		offset = reg_l;
+		/* HPOUT1R */
+		offset = reg_r;
 		offset += hubs->dcs_codes;
 		dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
 
-		/* HPOUT1R */
-		offset = reg_r;
+		/* HPOUT1L */
+		offset = reg_l;
 		offset += hubs->dcs_codes;
 		dcs_cfg |= (u8)offset;
 
@@ -167,8 +167,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec)
 				  WM8993_DCS_TRIG_DAC_WR_0 |
 				  WM8993_DCS_TRIG_DAC_WR_1);
 	} else {
-		dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
-		dcs_cfg |= reg_r;
+		dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
+		dcs_cfg |= reg_l;
 	}
 
 	/* Save the callibrated offset if we're in class W mode and
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index c611e45ff71d..54b0e4b7faf7 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -171,6 +171,14 @@ config SND_SOC_SMDK_WM8580_PCM
 	help
 	  Say Y if you want to add support for SoC audio on the SMDK.
 
+config SND_SOC_SMDK_WM8994_PCM
+	tristate "SoC PCM Audio support for WM8994 on SMDK"
+	depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310)
+	select SND_SOC_WM8994
+	select SND_SAMSUNG_PCM
+	help
+	  Say Y if you want to add support for SoC audio on the SMDK
+
 config SND_SOC_SPEYSIDE
 	tristate "Audio support for Wolfson Speyside"
 	depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410
diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile
index e04df65db1fc..9eb3b12eb72f 100644
--- a/sound/soc/samsung/Makefile
+++ b/sound/soc/samsung/Makefile
@@ -35,6 +35,7 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o
 snd-soc-goni-wm8994-objs := goni_wm8994.o
 snd-soc-smdk-spdif-objs := smdk_spdif.o
 snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o
+snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o
 snd-soc-speyside-objs := speyside.o
 snd-soc-speyside-wm8962-objs := speyside_wm8962.o
 
@@ -55,5 +56,6 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o
 obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o
 obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o
 obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o
+obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o
 obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o
 obj-$(CONFIG_SND_SOC_SPEYSIDE_WM8962) += snd-soc-speyside-wm8962.o
diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h
new file mode 100644
index 000000000000..c0e6d9a19efc
--- /dev/null
+++ b/sound/soc/samsung/i2s-regs.h
@@ -0,0 +1,143 @@
+/*
+ * linux/sound/soc/samsung/i2s-regs.h
+ *
+ * Copyright (c) 2011 Samsung Electronics Co., Ltd.
+ *		http://www.samsung.com
+ *
+ * Samsung I2S driver's register header
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __SND_SOC_SAMSUNG_I2S_REGS_H
+#define __SND_SOC_SAMSUNG_I2S_REGS_H
+
+#define I2SCON		0x0
+#define I2SMOD		0x4
+#define I2SFIC		0x8
+#define I2SPSR		0xc
+#define I2STXD		0x10
+#define I2SRXD		0x14
+#define I2SFICS		0x18
+#define I2STXDS		0x1c
+#define I2SAHB		0x20
+#define I2SSTR0		0x24
+#define I2SSIZE		0x28
+#define I2STRNCNT	0x2c
+#define I2SLVL0ADDR	0x30
+#define I2SLVL1ADDR	0x34
+#define I2SLVL2ADDR	0x38
+#define I2SLVL3ADDR	0x3c
+
+#define CON_RSTCLR		(1 << 31)
+#define CON_FRXOFSTATUS		(1 << 26)
+#define CON_FRXORINTEN		(1 << 25)
+#define CON_FTXSURSTAT		(1 << 24)
+#define CON_FTXSURINTEN		(1 << 23)
+#define CON_TXSDMA_PAUSE	(1 << 20)
+#define CON_TXSDMA_ACTIVE	(1 << 18)
+
+#define CON_FTXURSTATUS		(1 << 17)
+#define CON_FTXURINTEN		(1 << 16)
+#define CON_TXFIFO2_EMPTY	(1 << 15)
+#define CON_TXFIFO1_EMPTY	(1 << 14)
+#define CON_TXFIFO2_FULL	(1 << 13)
+#define CON_TXFIFO1_FULL	(1 << 12)
+
+#define CON_LRINDEX		(1 << 11)
+#define CON_TXFIFO_EMPTY	(1 << 10)
+#define CON_RXFIFO_EMPTY	(1 << 9)
+#define CON_TXFIFO_FULL		(1 << 8)
+#define CON_RXFIFO_FULL		(1 << 7)
+#define CON_TXDMA_PAUSE		(1 << 6)
+#define CON_RXDMA_PAUSE		(1 << 5)
+#define CON_TXCH_PAUSE		(1 << 4)
+#define CON_RXCH_PAUSE		(1 << 3)
+#define CON_TXDMA_ACTIVE	(1 << 2)
+#define CON_RXDMA_ACTIVE	(1 << 1)
+#define CON_ACTIVE		(1 << 0)
+
+#define MOD_OPCLK_CDCLK_OUT	(0 << 30)
+#define MOD_OPCLK_CDCLK_IN	(1 << 30)
+#define MOD_OPCLK_BCLK_OUT	(2 << 30)
+#define MOD_OPCLK_PCLK		(3 << 30)
+#define MOD_OPCLK_MASK		(3 << 30)
+#define MOD_TXS_IDMA		(1 << 28) /* Sec_TXFIFO use I-DMA */
+
+#define MOD_BLCS_SHIFT		26
+#define MOD_BLCS_16BIT		(0 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_8BIT		(1 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_24BIT		(2 << MOD_BLCS_SHIFT)
+#define MOD_BLCS_MASK		(3 << MOD_BLCS_SHIFT)
+#define MOD_BLCP_SHIFT		24
+#define MOD_BLCP_16BIT		(0 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_8BIT		(1 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_24BIT		(2 << MOD_BLCP_SHIFT)
+#define MOD_BLCP_MASK		(3 << MOD_BLCP_SHIFT)
+
+#define MOD_C2DD_HHALF		(1 << 21) /* Discard Higher-half */
+#define MOD_C2DD_LHALF		(1 << 20) /* Discard Lower-half */
+#define MOD_C1DD_HHALF		(1 << 19)
+#define MOD_C1DD_LHALF		(1 << 18)
+#define MOD_DC2_EN		(1 << 17)
+#define MOD_DC1_EN		(1 << 16)
+#define MOD_BLC_16BIT		(0 << 13)
+#define MOD_BLC_8BIT		(1 << 13)
+#define MOD_BLC_24BIT		(2 << 13)
+#define MOD_BLC_MASK		(3 << 13)
+
+#define MOD_IMS_SYSMUX		(1 << 10)
+#define MOD_SLAVE		(1 << 11)
+#define MOD_TXONLY		(0 << 8)
+#define MOD_RXONLY		(1 << 8)
+#define MOD_TXRX		(2 << 8)
+#define MOD_MASK		(3 << 8)
+#define MOD_LR_LLOW		(0 << 7)
+#define MOD_LR_RLOW		(1 << 7)
+#define MOD_SDF_IIS		(0 << 5)
+#define MOD_SDF_MSB		(1 << 5)
+#define MOD_SDF_LSB		(2 << 5)
+#define MOD_SDF_MASK		(3 << 5)
+#define MOD_RCLK_256FS		(0 << 3)
+#define MOD_RCLK_512FS		(1 << 3)
+#define MOD_RCLK_384FS		(2 << 3)
+#define MOD_RCLK_768FS		(3 << 3)
+#define MOD_RCLK_MASK		(3 << 3)
+#define MOD_BCLK_32FS		(0 << 1)
+#define MOD_BCLK_48FS		(1 << 1)
+#define MOD_BCLK_16FS		(2 << 1)
+#define MOD_BCLK_24FS		(3 << 1)
+#define MOD_BCLK_MASK		(3 << 1)
+#define MOD_8BIT		(1 << 0)
+
+#define MOD_CDCLKCON		(1 << 12)
+
+#define PSR_PSREN		(1 << 15)
+
+#define FIC_TX2COUNT(x)		(((x) >>  24) & 0xf)
+#define FIC_TX1COUNT(x)		(((x) >>  16) & 0xf)
+
+#define FIC_TXFLUSH		(1 << 15)
+#define FIC_RXFLUSH		(1 << 7)
+
+#define FIC_TXCOUNT(x)		(((x) >>  8) & 0xf)
+#define FIC_RXCOUNT(x)		(((x) >>  0) & 0xf)
+#define FICS_TXCOUNT(x)		(((x) >>  8) & 0x7f)
+
+#define AHB_INTENLVL0		(1 << 24)
+#define AHB_LVL0INT		(1 << 20)
+#define AHB_CLRLVL0INT		(1 << 16)
+#define AHB_DMARLD		(1 << 5)
+#define AHB_INTMASK		(1 << 3)
+#define AHB_DMAEN		(1 << 0)
+#define AHB_LVLINTMASK		(0xf << 20)
+
+#define I2SSIZE_TRNMSK		(0xffff)
+#define I2SSIZE_SHIFT		(16)
+
+#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */
+
+
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index 992a732b5211..1568eea31f41 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -22,109 +22,7 @@
 
 #include "dma.h"
 #include "i2s.h"
-
-#define I2SCON		0x0
-#define I2SMOD		0x4
-#define I2SFIC		0x8
-#define I2SPSR		0xc
-#define I2STXD		0x10
-#define I2SRXD		0x14
-#define I2SFICS		0x18
-#define I2STXDS		0x1c
-
-#define CON_RSTCLR		(1 << 31)
-#define CON_FRXOFSTATUS		(1 << 26)
-#define CON_FRXORINTEN		(1 << 25)
-#define CON_FTXSURSTAT		(1 << 24)
-#define CON_FTXSURINTEN		(1 << 23)
-#define CON_TXSDMA_PAUSE	(1 << 20)
-#define CON_TXSDMA_ACTIVE	(1 << 18)
-
-#define CON_FTXURSTATUS		(1 << 17)
-#define CON_FTXURINTEN		(1 << 16)
-#define CON_TXFIFO2_EMPTY	(1 << 15)
-#define CON_TXFIFO1_EMPTY	(1 << 14)
-#define CON_TXFIFO2_FULL	(1 << 13)
-#define CON_TXFIFO1_FULL	(1 << 12)
-
-#define CON_LRINDEX		(1 << 11)
-#define CON_TXFIFO_EMPTY	(1 << 10)
-#define CON_RXFIFO_EMPTY	(1 << 9)
-#define CON_TXFIFO_FULL		(1 << 8)
-#define CON_RXFIFO_FULL		(1 << 7)
-#define CON_TXDMA_PAUSE		(1 << 6)
-#define CON_RXDMA_PAUSE		(1 << 5)
-#define CON_TXCH_PAUSE		(1 << 4)
-#define CON_RXCH_PAUSE		(1 << 3)
-#define CON_TXDMA_ACTIVE	(1 << 2)
-#define CON_RXDMA_ACTIVE	(1 << 1)
-#define CON_ACTIVE		(1 << 0)
-
-#define MOD_OPCLK_CDCLK_OUT	(0 << 30)
-#define MOD_OPCLK_CDCLK_IN	(1 << 30)
-#define MOD_OPCLK_BCLK_OUT	(2 << 30)
-#define MOD_OPCLK_PCLK		(3 << 30)
-#define MOD_OPCLK_MASK		(3 << 30)
-#define MOD_TXS_IDMA		(1 << 28) /* Sec_TXFIFO use I-DMA */
-
-#define MOD_BLCS_SHIFT	26
-#define MOD_BLCS_16BIT	(0 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_8BIT	(1 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_24BIT	(2 << MOD_BLCS_SHIFT)
-#define MOD_BLCS_MASK	(3 << MOD_BLCS_SHIFT)
-#define MOD_BLCP_SHIFT	24
-#define MOD_BLCP_16BIT	(0 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_8BIT	(1 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_24BIT	(2 << MOD_BLCP_SHIFT)
-#define MOD_BLCP_MASK	(3 << MOD_BLCP_SHIFT)
-
-#define MOD_C2DD_HHALF		(1 << 21) /* Discard Higher-half */
-#define MOD_C2DD_LHALF		(1 << 20) /* Discard Lower-half */
-#define MOD_C1DD_HHALF		(1 << 19)
-#define MOD_C1DD_LHALF		(1 << 18)
-#define MOD_DC2_EN		(1 << 17)
-#define MOD_DC1_EN		(1 << 16)
-#define MOD_BLC_16BIT		(0 << 13)
-#define MOD_BLC_8BIT		(1 << 13)
-#define MOD_BLC_24BIT		(2 << 13)
-#define MOD_BLC_MASK		(3 << 13)
-
-#define MOD_IMS_SYSMUX		(1 << 10)
-#define MOD_SLAVE		(1 << 11)
-#define MOD_TXONLY		(0 << 8)
-#define MOD_RXONLY		(1 << 8)
-#define MOD_TXRX		(2 << 8)
-#define MOD_MASK		(3 << 8)
-#define MOD_LR_LLOW		(0 << 7)
-#define MOD_LR_RLOW		(1 << 7)
-#define MOD_SDF_IIS		(0 << 5)
-#define MOD_SDF_MSB		(1 << 5)
-#define MOD_SDF_LSB		(2 << 5)
-#define MOD_SDF_MASK		(3 << 5)
-#define MOD_RCLK_256FS		(0 << 3)
-#define MOD_RCLK_512FS		(1 << 3)
-#define MOD_RCLK_384FS		(2 << 3)
-#define MOD_RCLK_768FS		(3 << 3)
-#define MOD_RCLK_MASK		(3 << 3)
-#define MOD_BCLK_32FS		(0 << 1)
-#define MOD_BCLK_48FS		(1 << 1)
-#define MOD_BCLK_16FS		(2 << 1)
-#define MOD_BCLK_24FS		(3 << 1)
-#define MOD_BCLK_MASK		(3 << 1)
-#define MOD_8BIT		(1 << 0)
-
-#define MOD_CDCLKCON		(1 << 12)
-
-#define PSR_PSREN		(1 << 15)
-
-#define FIC_TX2COUNT(x)		(((x) >>  24) & 0xf)
-#define FIC_TX1COUNT(x)		(((x) >>  16) & 0xf)
-
-#define FIC_TXFLUSH		(1 << 15)
-#define FIC_RXFLUSH		(1 << 7)
-#define FIC_TXCOUNT(x)		(((x) >>  8) & 0xf)
-#define FIC_RXCOUNT(x)		(((x) >>  0) & 0xf)
-#define FICS_TXCOUNT(x)		(((x) >>  8) & 0x7f)
+#include "i2s-regs.h"
 
 #define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t)
 
diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c
new file mode 100644
index 000000000000..5f2111685480
--- /dev/null
+++ b/sound/soc/samsung/smdk_wm8994pcm.c
@@ -0,0 +1,176 @@
+/*
+ *  sound/soc/samsung/smdk_wm8994pcm.c
+ *
+ *  Copyright (c) 2011 Samsung Electronics Co., Ltd
+ *		http://www.samsung.com
+ *
+ *  This program is free software; you can redistribute  it and/or  modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "../codecs/wm8994.h"
+#include "dma.h"
+#include "pcm.h"
+
+/*
+ * Board Settings:
+ *  o '1' means 'ON'
+ *  o '0' means 'OFF'
+ *  o 'X' means 'Don't care'
+ *
+ * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111
+ */
+
+/*
+ * Configure audio route as :-
+ * $ amixer sset 'DAC1' on,on
+ * $ amixer sset 'Right Headphone Mux' 'DAC'
+ * $ amixer sset 'Left Headphone Mux' 'DAC'
+ * $ amixer sset 'DAC1R Mixer AIF1.1' on
+ * $ amixer sset 'DAC1L Mixer AIF1.1' on
+ * $ amixer sset 'IN2L' on
+ * $ amixer sset 'IN2L PGA IN2LN' on
+ * $ amixer sset 'MIXINL IN2L' on
+ * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on
+ * $ amixer sset 'IN2R' on
+ * $ amixer sset 'IN2R PGA IN2RN' on
+ * $ amixer sset 'MIXINR IN2R' on
+ * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on
+ */
+
+/* SMDK has a 16.9344MHZ crystal attached to WM8994 */
+#define SMDK_WM8994_FREQ 16934400
+
+static int smdk_wm8994_pcm_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;
+	unsigned long mclk_freq;
+	int rfs, ret;
+
+	switch(params_rate(params)) {
+	case 8000:
+		rfs = 512;
+		break;
+	default:
+		dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n",
+		__func__, __LINE__, params_rate(params));
+		return -EINVAL;
+	}
+
+	mclk_freq = params_rate(params) * rfs;
+
+	/* Set the codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B
+				| SND_SOC_DAIFMT_IB_NF
+				| SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* Set the cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B
+				| SND_SOC_DAIFMT_IB_NF
+				| SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1,
+					mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1,
+					SMDK_WM8994_FREQ, mclk_freq);
+	if (ret < 0)
+		return ret;
+
+	/* Set PCM source clock on CPU */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX,
+					mclk_freq, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	/* Set SCLK_DIV for making bclk */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops smdk_wm8994_pcm_ops = {
+	.hw_params = smdk_wm8994_pcm_hw_params,
+};
+
+static struct snd_soc_dai_link smdk_dai[] = {
+	{
+		.name = "WM8994 PAIF PCM",
+		.stream_name = "Primary PCM",
+		.cpu_dai_name = "samsung-pcm.0",
+		.codec_dai_name = "wm8994-aif1",
+		.platform_name = "samsung-audio",
+		.codec_name = "wm8994-codec",
+		.ops = &smdk_wm8994_pcm_ops,
+	},
+};
+
+static struct snd_soc_card smdk_pcm = {
+	.name = "SMDK-PCM",
+	.dai_link = smdk_dai,
+	.num_links = 1,
+};
+
+static int __devinit snd_smdk_probe(struct platform_device *pdev)
+{
+	int ret = 0;
+
+	smdk_pcm.dev = &pdev->dev;
+	ret = snd_soc_register_card(&smdk_pcm);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int __devexit snd_smdk_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_card(&smdk_pcm);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver snd_smdk_driver = {
+	.driver = {
+		.owner = THIS_MODULE,
+		.name = "samsung-smdk-pcm",
+	},
+	.probe = snd_smdk_probe,
+	.remove = __devexit_p(snd_smdk_remove),
+};
+
+static int __init smdk_audio_init(void)
+{
+	return platform_driver_register(&snd_smdk_driver);
+}
+
+module_init(smdk_audio_init);
+
+static void __exit smdk_audio_exit(void)
+{
+	platform_driver_unregister(&snd_smdk_driver);
+}
+
+module_exit(smdk_audio_exit);
+
+MODULE_AUTHOR("Sangbeom Kim, <sbkim73@samsung.com>");
+MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c
index c0ba0bfd7f57..8ac42bf82090 100644
--- a/sound/soc/samsung/speyside_wm8962.c
+++ b/sound/soc/samsung/speyside_wm8962.c
@@ -30,14 +30,16 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card,
 						  WM8962_FLL_MCLK, 32768,
 						  44100 * 256);
 			if (ret < 0)
-				pr_err("Failed to start FLL\n");
+				pr_err("Failed to start FLL: %d\n", ret);
 
 			ret = snd_soc_dai_set_sysclk(codec_dai,
 						     WM8962_SYSCLK_FLL,
 						     44100 * 256,
 						     SND_SOC_CLOCK_IN);
-			if (ret < 0)
+			if (ret < 0) {
+				pr_err("Failed to set SYSCLK: %d\n");
 				return ret;
+			}
 		}
 		break;
 
@@ -59,13 +61,15 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card,
 	case SND_SOC_BIAS_STANDBY:
 		ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
 					     32768, SND_SOC_CLOCK_IN);
-		if (ret < 0)
+		if (ret < 0) {
+			pr_err("Failed to switch away from FLL: %d\n", ret);
 			return ret;
+		}
 
 		ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
 					  0, 0, 0);
 		if (ret < 0) {
-			pr_err("Failed to stop FLL\n");
+			pr_err("Failed to stop FLL: %d\n", ret);
 			return ret;
 		}
 		break;
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 32bc50387f61..d08abf4e3eb3 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -986,6 +986,39 @@ err_probe:
 	return ret;
 }
 
+static int soc_probe_platform(struct snd_soc_card *card,
+			   struct snd_soc_platform *platform)
+{
+	int ret = 0;
+	const struct snd_soc_platform_driver *driver = platform->driver;
+
+	platform->card = card;
+
+	if (!try_module_get(platform->dev->driver->owner))
+		return -ENODEV;
+
+	if (driver->probe) {
+		ret = driver->probe(platform);
+		if (ret < 0) {
+			dev_err(platform->dev,
+				"asoc: failed to probe platform %s: %d\n",
+				platform->name, ret);
+			goto err_probe;
+		}
+	}
+
+	/* mark platform as probed and add to card platform list */
+	platform->probed = 1;
+	list_add(&platform->card_list, &card->platform_dev_list);
+
+	return 0;
+
+err_probe:
+	module_put(platform->dev->driver->owner);
+
+	return ret;
+}
+
 static void rtd_release(struct device *dev) {}
 
 static int soc_post_component_init(struct snd_soc_card *card,
@@ -1109,21 +1142,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order)
 	/* probe the platform */
 	if (!platform->probed &&
 			platform->driver->probe_order == order) {
-		if (!try_module_get(platform->dev->driver->owner))
-			return -ENODEV;
-
-		if (platform->driver->probe) {
-			ret = platform->driver->probe(platform);
-			if (ret < 0) {
-				printk(KERN_ERR "asoc: failed to probe platform %s\n",
-						platform->name);
-				module_put(platform->dev->driver->owner);
-				return ret;
-			}
-		}
-		/* mark platform as probed and add to card platform list */
-		platform->probed = 1;
-		list_add(&platform->card_list, &card->platform_dev_list);
+		ret = soc_probe_platform(card, platform);
+		if (ret < 0)
+			return ret;
 	}
 
 	/* probe the CODEC DAI */
@@ -1619,6 +1640,36 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec,
 }
 EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register);
 
+int snd_soc_platform_read(struct snd_soc_platform *platform,
+					unsigned int reg)
+{
+	unsigned int ret;
+
+	if (!platform->driver->read) {
+		dev_err(platform->dev, "platform has no read back\n");
+		return -1;
+	}
+
+	ret = platform->driver->read(platform, reg);
+	dev_dbg(platform->dev, "read %x => %x\n", reg, ret);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_read);
+
+int snd_soc_platform_write(struct snd_soc_platform *platform,
+					 unsigned int reg, unsigned int val)
+{
+	if (!platform->driver->write) {
+		dev_err(platform->dev, "platform has no write back\n");
+		return -1;
+	}
+
+	dev_dbg(platform->dev, "write %x = %x\n", reg, val);
+	return platform->driver->write(platform, reg, val);
+}
+EXPORT_SYMBOL_GPL(snd_soc_platform_write);
+
 /**
  * snd_soc_new_ac97_codec - initailise AC97 device
  * @codec: audio codec
diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c
index 6b817e20548c..f36b9969cfec 100644
--- a/sound/soc/tegra/tegra_i2s.c
+++ b/sound/soc/tegra/tegra_i2s.c
@@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (i2sclock % (2 * srate))
 		reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE;
 
+	if (!i2s->clk_refs)
+		clk_enable(i2s->clk_i2s);
+
 	tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg);
 
 	tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR,
 		TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS |
 		TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS);
 
+	if (!i2s->clk_refs)
+		clk_disable(i2s->clk_i2s);
+
 	return 0;
 }
 
@@ -348,7 +354,6 @@ struct snd_soc_dai_driver tegra_i2s_dai[] = {
 static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 {
 	struct tegra_i2s * i2s;
-	char clk_name[12]; /* tegra-i2s.0 */
 	struct resource *mem, *memregion, *dmareq;
 	int ret;
 
@@ -383,8 +388,7 @@ static __devinit int tegra_i2s_platform_probe(struct platform_device *pdev)
 	}
 	dev_set_drvdata(&pdev->dev, i2s);
 
-	snprintf(clk_name, sizeof(clk_name), DRV_NAME ".%d", pdev->id);
-	i2s->clk_i2s = clk_get_sys(clk_name, NULL);
+	i2s->clk_i2s = clk_get(&pdev->dev, NULL);
 	if (IS_ERR(i2s->clk_i2s)) {
 		dev_err(&pdev->dev, "Can't retrieve i2s clock\n");
 		ret = PTR_ERR(i2s->clk_i2s);