17 files changed, 937 insertions, 267 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733ecd17f..70e5a75901aa 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_AK4554
+	select SND_SOC_AK4613 if I2C
 	select SND_SOC_AK4641 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
@@ -79,7 +80,6 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_MAX9877 if I2C
 	select SND_SOC_MC13783 if MFD_MC13XXX
 	select SND_SOC_ML26124 if I2C
-	select SND_SOC_HDMI_CODEC
 	select SND_SOC_PCM1681 if I2C
 	select SND_SOC_PCM1792A if SPI_MASTER
 	select SND_SOC_PCM3008
@@ -319,6 +319,10 @@ config SND_SOC_AK4535
 config SND_SOC_AK4554
 	tristate "AKM AK4554 CODEC"
 
+config SND_SOC_AK4613
+	tristate "AKM AK4613 CODEC"
+	depends on I2C
+
 config SND_SOC_AK4641
 	tristate
 
@@ -442,9 +446,6 @@ config SND_SOC_BT_SCO
 config SND_SOC_DMIC
 	tristate
 
-config SND_SOC_HDMI_CODEC
-       tristate "HDMI stub CODEC"
-
 config SND_SOC_ES8328
 	tristate "Everest Semi ES8328 CODEC"
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077954ae..be1491acb6ae 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o
 snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4554-objs := ak4554.o
+snd-soc-ak4613-objs := ak4613.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
@@ -72,7 +73,6 @@ snd-soc-max98925-objs := max98925.o
 snd-soc-max9850-objs := max9850.o
 snd-soc-mc13783-objs := mc13783.o
 snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
 snd-soc-pcm1681-objs := pcm1681.o
 snd-soc-pcm1792a-codec-objs := pcm1792a.o
 snd-soc-pcm3008-objs := pcm3008.o
@@ -216,6 +216,7 @@ obj-$(CONFIG_SND_SOC_ADS117X)	+= snd-soc-ads117x.o
 obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4554)	+= snd-soc-ak4554.o
+obj-$(CONFIG_SND_SOC_AK4613)	+= snd-soc-ak4613.o
 obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
@@ -264,7 +265,6 @@ obj-$(CONFIG_SND_SOC_MAX98925)	+= snd-soc-max98925.o
 obj-$(CONFIG_SND_SOC_MAX9850)	+= snd-soc-max9850.o
 obj-$(CONFIG_SND_SOC_MC13783)	+= snd-soc-mc13783.o
 obj-$(CONFIG_SND_SOC_ML26124)	+= snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
 obj-$(CONFIG_SND_SOC_PCM1681)	+= snd-soc-pcm1681.o
 obj-$(CONFIG_SND_SOC_PCM1792A)	+= snd-soc-pcm1792a-codec.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 000000000000..07a266460ec3
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c  --  Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * 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/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1	0x00 /* Power Management 1 */
+#define PW_MGMT2	0x01 /* Power Management 2 */
+#define PW_MGMT3	0x02 /* Power Management 3 */
+#define CTRL1		0x03 /* Control 1 */
+#define CTRL2		0x04 /* Control 2 */
+#define DEMP1		0x05 /* De-emphasis1 */
+#define DEMP2		0x06 /* De-emphasis2 */
+#define OFD		0x07 /* Overflow Detect */
+#define ZRD		0x08 /* Zero Detect */
+#define ICTRL		0x09 /* Input Control */
+#define OCTRL		0x0a /* Output Control */
+#define LOUT1		0x0b /* LOUT1 Volume Control */
+#define ROUT1		0x0c /* ROUT1 Volume Control */
+#define LOUT2		0x0d /* LOUT2 Volume Control */
+#define ROUT2		0x0e /* ROUT2 Volume Control */
+#define LOUT3		0x0f /* LOUT3 Volume Control */
+#define ROUT3		0x10 /* ROUT3 Volume Control */
+#define LOUT4		0x11 /* LOUT4 Volume Control */
+#define ROUT4		0x12 /* ROUT4 Volume Control */
+#define LOUT5		0x13 /* LOUT5 Volume Control */
+#define ROUT5		0x14 /* ROUT5 Volume Control */
+#define LOUT6		0x15 /* LOUT6 Volume Control */
+#define ROUT6		0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN		BIT(0)
+#define PMDAC		BIT(1)
+#define PMADC		BIT(2)
+#define PMVR		BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL	0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL	0x3f
+
+/* CTRL1 */
+#define DIF0		BIT(3)
+#define DIF1		BIT(4)
+#define DIF2		BIT(5)
+#define TDM0		BIT(6)
+#define TDM1		BIT(7)
+#define NO_FMT		(0xff)
+#define FMT_MASK	(0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED	(0 << 2)
+#define DFS_DOUBLE_SPEED	(1 << 2)
+#define DFS_QUAD_SPEED		(2 << 2)
+
+struct ak4613_priv {
+	struct mutex lock;
+
+	unsigned int fmt;
+	u8 fmt_ctrl;
+	int cnt;
+};
+
+struct ak4613_formats {
+	unsigned int width;
+	unsigned int fmt;
+};
+
+struct ak4613_interface {
+	struct ak4613_formats capture;
+	struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ *       ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+	SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+			 0, 0xFF, 1, out_tlv),
+	SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+			 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+	{ 0x0,  0x0f }, { 0x1,  0x07 }, { 0x2,  0x3f }, { 0x3,  0x20 },
+	{ 0x4,  0x20 }, { 0x5,  0x55 }, { 0x6,  0x05 }, { 0x7,  0x07 },
+	{ 0x8,  0x0f }, { 0x9,  0x07 }, { 0xa,  0x3f }, { 0xb,  0x00 },
+	{ 0xc,  0x00 }, { 0xd,  0x00 }, { 0xe,  0x00 }, { 0xf,  0x00 },
+	{ 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+	{ 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+	/* capture */				/* playback */
+	[0] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(16, RIGHT_J) },
+	[1] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(20, RIGHT_J) },
+	[2] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(24, RIGHT_J) },
+	[3] = {	AUDIO_IFACE(24, LEFT_J),	AUDIO_IFACE(24, LEFT_J) },
+	[4] = {	AUDIO_IFACE(24, I2S),		AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+	.reg_bits		= 8,
+	.val_bits		= 8,
+	.max_register		= 0x16,
+	.reg_defaults		= ak4613_reg,
+	.num_reg_defaults	= ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+	{ .compatible = "asahi-kasei,ak4613",	.data = &ak4613_regmap_cfg },
+	{},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+	{ "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+	{ }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+	/* Outputs */
+	SND_SOC_DAPM_OUTPUT("LOUT1"),
+	SND_SOC_DAPM_OUTPUT("LOUT2"),
+	SND_SOC_DAPM_OUTPUT("LOUT3"),
+	SND_SOC_DAPM_OUTPUT("LOUT4"),
+	SND_SOC_DAPM_OUTPUT("LOUT5"),
+	SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+	SND_SOC_DAPM_OUTPUT("ROUT1"),
+	SND_SOC_DAPM_OUTPUT("ROUT2"),
+	SND_SOC_DAPM_OUTPUT("ROUT3"),
+	SND_SOC_DAPM_OUTPUT("ROUT4"),
+	SND_SOC_DAPM_OUTPUT("ROUT5"),
+	SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+	/* Inputs */
+	SND_SOC_DAPM_INPUT("LIN1"),
+	SND_SOC_DAPM_INPUT("LIN2"),
+
+	SND_SOC_DAPM_INPUT("RIN1"),
+	SND_SOC_DAPM_INPUT("RIN2"),
+
+	/* DAC */
+	SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+	SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+	SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+	SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+	SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+	SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+	/* ADC */
+	SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+	SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+	{"LOUT1", NULL, "DAC1"},
+	{"LOUT2", NULL, "DAC2"},
+	{"LOUT3", NULL, "DAC3"},
+	{"LOUT4", NULL, "DAC4"},
+	{"LOUT5", NULL, "DAC5"},
+	{"LOUT6", NULL, "DAC6"},
+
+	{"ROUT1", NULL, "DAC1"},
+	{"ROUT2", NULL, "DAC2"},
+	{"ROUT3", NULL, "DAC3"},
+	{"ROUT4", NULL, "DAC4"},
+	{"ROUT5", NULL, "DAC5"},
+	{"ROUT6", NULL, "DAC6"},
+
+	{"DAC1", NULL, "Playback"},
+	{"DAC2", NULL, "Playback"},
+	{"DAC3", NULL, "Playback"},
+	{"DAC4", NULL, "Playback"},
+	{"DAC5", NULL, "Playback"},
+	{"DAC6", NULL, "Playback"},
+
+	{"Capture", NULL, "ADC1"},
+	{"Capture", NULL, "ADC2"},
+
+	{"ADC1", NULL, "LIN1"},
+	{"ADC2", NULL, "LIN2"},
+
+	{"ADC1", NULL, "RIN1"},
+	{"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct device *dev = codec->dev;
+
+	mutex_lock(&priv->lock);
+	priv->cnt--;
+	if (priv->cnt < 0) {
+		dev_err(dev, "unexpected counter error\n");
+		priv->cnt = 0;
+	}
+	if (!priv->cnt)
+		priv->fmt_ctrl = NO_FMT;
+	mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+	switch (fmt) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+	case SND_SOC_DAIFMT_I2S:
+		priv->fmt = fmt;
+
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ak4613_dai_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 ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+	const struct ak4613_formats *fmts;
+	struct device *dev = codec->dev;
+	unsigned int width = params_width(params);
+	unsigned int fmt = priv->fmt;
+	unsigned int rate;
+	int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+	int i, ret;
+	u8 fmt_ctrl, ctrl2;
+
+	rate = params_rate(params);
+	switch (rate) {
+	case 32000:
+	case 44100:
+	case 48000:
+		ctrl2 = DFS_NORMAL_SPEED;
+		break;
+	case 88200:
+	case 96000:
+		ctrl2 = DFS_DOUBLE_SPEED;
+		break;
+	case 176400:
+	case 192000:
+		ctrl2 = DFS_QUAD_SPEED;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/*
+	 * FIXME
+	 *
+	 * It doesn't support TDM at this point
+	 */
+	fmt_ctrl = NO_FMT;
+	for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+		fmts = (is_play) ?	&ak4613_iface[i].playback :
+					&ak4613_iface[i].capture;
+
+		if (fmts->fmt != fmt)
+			continue;
+
+		if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+			if (fmts->width != width)
+				continue;
+		} else {
+			if (fmts->width < width)
+				continue;
+		}
+
+		fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+		break;
+	}
+
+	ret = -EINVAL;
+	if (fmt_ctrl == NO_FMT)
+		goto hw_params_end;
+
+	mutex_lock(&priv->lock);
+	if ((priv->fmt_ctrl == NO_FMT) ||
+	    (priv->fmt_ctrl == fmt_ctrl)) {
+		priv->fmt_ctrl = fmt_ctrl;
+		priv->cnt++;
+		ret = 0;
+	}
+	mutex_unlock(&priv->lock);
+
+	if (ret < 0)
+		goto hw_params_end;
+
+	snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+	snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+	if (ret < 0)
+		dev_warn(dev, "unsupported data width/format combination\n");
+
+	return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	u8 mgmt1 = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		mgmt1 |= RSTN;
+		/* fall through */
+	case SND_SOC_BIAS_PREPARE:
+		mgmt1 |= PMADC | PMDAC;
+		/* fall through */
+	case SND_SOC_BIAS_STANDBY:
+		mgmt1 |= PMVR;
+		/* fall through */
+	case SND_SOC_BIAS_OFF:
+	default:
+		break;
+	}
+
+	snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+	.shutdown	= ak4613_dai_shutdown,
+	.set_fmt	= ak4613_dai_set_fmt,
+	.hw_params	= ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE		(SNDRV_PCM_RATE_32000  |\
+				 SNDRV_PCM_RATE_44100  |\
+				 SNDRV_PCM_RATE_48000  |\
+				 SNDRV_PCM_RATE_64000  |\
+				 SNDRV_PCM_RATE_88200  |\
+				 SNDRV_PCM_RATE_96000  |\
+				 SNDRV_PCM_RATE_176400 |\
+				 SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT	(SNDRV_PCM_FMTBIT_S16_LE |\
+				 SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+	.name = "ak4613-hifi",
+	.playback = {
+		.stream_name	= "Playback",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= AK4613_PCM_RATE,
+		.formats	= AK4613_PCM_FMTBIT,
+	},
+	.capture = {
+		.stream_name	= "Capture",
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= AK4613_PCM_RATE,
+		.formats	= AK4613_PCM_FMTBIT,
+	},
+	.ops = &ak4613_dai_ops,
+	.symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+	struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+	regcache_mark_dirty(regmap);
+	return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+	.resume			= ak4613_resume,
+	.set_bias_level		= ak4613_set_bias_level,
+	.controls		= ak4613_snd_controls,
+	.num_controls		= ARRAY_SIZE(ak4613_snd_controls),
+	.dapm_widgets		= ak4613_dapm_widgets,
+	.num_dapm_widgets	= ARRAY_SIZE(ak4613_dapm_widgets),
+	.dapm_routes		= ak4613_intercon,
+	.num_dapm_routes	= ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+			    const struct i2c_device_id *id)
+{
+	struct device *dev = &i2c->dev;
+	struct device_node *np = dev->of_node;
+	const struct regmap_config *regmap_cfg;
+	struct regmap *regmap;
+	struct ak4613_priv *priv;
+
+	regmap_cfg = NULL;
+	if (np) {
+		const struct of_device_id *of_id;
+
+		of_id = of_match_device(ak4613_of_match, dev);
+		if (of_id)
+			regmap_cfg = of_id->data;
+	} else {
+		regmap_cfg = (const struct regmap_config *)id->driver_data;
+	}
+
+	if (!regmap_cfg)
+		return -EINVAL;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->fmt_ctrl		= NO_FMT;
+	priv->cnt		= 0;
+
+	mutex_init(&priv->lock);
+
+	i2c_set_clientdata(i2c, priv);
+
+	regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+	if (IS_ERR(regmap))
+		return PTR_ERR(regmap);
+
+	return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+				      &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+	snd_soc_unregister_codec(&client->dev);
+	return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+	.driver = {
+		.name = "ak4613-codec",
+		.owner = THIS_MODULE,
+		.of_match_table = ak4613_of_match,
+	},
+	.probe		= ak4613_i2c_probe,
+	.remove		= ak4613_i2c_remove,
+	.id_table	= ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143d0e90..cda27c22812a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
  * AK4648 is tested.
  */
 
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
 #include <linux/slab.h>
@@ -128,11 +130,8 @@
 #define I2S		(3 << 0)
 
 /* MD_CTL2 */
-#define FS0		(1 << 0)
-#define FS1		(1 << 1)
-#define FS2		(1 << 2)
-#define FS3		(1 << 5)
-#define FS_MASK		(FS0 | FS1 | FS2 | FS3)
+#define FSs(val)	(((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val)	((val & 0x3) << 6)
 
 /* MD_CTL3 */
 #define BST1		(1 << 3)
@@ -147,6 +146,7 @@ struct ak4642_drvdata {
 
 struct ak4642_priv {
 	const struct ak4642_drvdata *drvdata;
+	struct clk *mcko;
 };
 
 /*
@@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	return 0;
 }
 
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+			   u32 frequency)
+{
+	u32 fs_list[] = {
+		[0] = 8000,
+		[1] = 12000,
+		[2] = 16000,
+		[3] = 24000,
+		[4] = 7350,
+		[5] = 11025,
+		[6] = 14700,
+		[7] = 22050,
+		[10] = 32000,
+		[11] = 48000,
+		[14] = 29400,
+		[15] = 44100,
+	};
+	u32 ps_list[] = {
+		[0] = 256,
+		[1] = 128,
+		[2] = 64,
+		[3] = 32
+	};
+	int ps, fs;
+
+	for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+		for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+			if (frequency == ps_list[ps] * fs_list[fs]) {
+				snd_soc_write(codec, MD_CTL2,
+					      PSs(ps) | FSs(fs));
+				return 0;
+			}
+		}
+	}
+
+	return 0;
+}
+
 static int ak4642_dai_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;
-	u8 rate;
+	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+	u32 rate = clk_get_rate(priv->mcko);
 
-	switch (params_rate(params)) {
-	case 7350:
-		rate = FS2;
-		break;
-	case 8000:
-		rate = 0;
-		break;
-	case 11025:
-		rate = FS2 | FS0;
-		break;
-	case 12000:
-		rate = FS0;
-		break;
-	case 14700:
-		rate = FS2 | FS1;
-		break;
-	case 16000:
-		rate = FS1;
-		break;
-	case 22050:
-		rate = FS2 | FS1 | FS0;
-		break;
-	case 24000:
-		rate = FS1 | FS0;
-		break;
-	case 29400:
-		rate = FS3 | FS2 | FS1;
-		break;
-	case 32000:
-		rate = FS3 | FS1;
-		break;
-	case 44100:
-		rate = FS3 | FS2 | FS1 | FS0;
-		break;
-	case 48000:
-		rate = FS3 | FS1 | FS0;
-		break;
-	default:
-		return -EINVAL;
-	}
-	snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+	if (!rate)
+		rate = params_rate(params) * 256;
 
-	return 0;
+	return ak4642_set_mcko(codec, rate);
 }
 
 static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec)
 	return 0;
 }
 
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+	struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (priv->mcko)
+		ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+	return 0;
+}
+
 static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+	.probe			= ak4642_probe,
 	.resume			= ak4642_resume,
 	.set_bias_level		= ak4642_set_bias_level,
 	.controls		= ak4642_snd_controls,
@@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = {
 	.extended_frequencies = 1,
 };
 
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+	struct device_node *np = dev->of_node;
+	struct clk *clk;
+	const char *clk_name = np->name;
+	const char *parent_clk_name = NULL;
+	u32 rate;
+
+	if (of_property_read_u32(np, "clock-frequency", &rate))
+		return NULL;
+
+	if (of_property_read_bool(np, "clocks"))
+		parent_clk_name = of_clk_get_parent_name(np, 0);
+
+	of_property_read_string(np, "clock-output-names", &clk_name);
+
+	clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+				      (parent_clk_name) ? 0 : CLK_IS_ROOT,
+				      rate);
+	if (!IS_ERR(clk))
+		of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+	return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
 static const struct of_device_id ak4642_of_match[];
 static int ak4642_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct device_node *np = i2c->dev.of_node;
+	struct device *dev = &i2c->dev;
+	struct device_node *np = dev->of_node;
 	const struct ak4642_drvdata *drvdata = NULL;
 	struct regmap *regmap;
 	struct ak4642_priv *priv;
+	struct clk *mcko = NULL;
 
 	if (np) {
 		const struct of_device_id *of_id;
 
-		of_id = of_match_device(ak4642_of_match, &i2c->dev);
+		mcko = ak4642_of_parse_mcko(dev);
+		if (IS_ERR(mcko))
+			mcko = NULL;
+
+		of_id = of_match_device(ak4642_of_match, dev);
 		if (of_id)
 			drvdata = of_id->data;
 	} else {
@@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 	}
 
 	if (!drvdata) {
-		dev_err(&i2c->dev, "Unknown device type\n");
+		dev_err(dev, "Unknown device type\n");
 		return -EINVAL;
 	}
 
-	priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
 	if (!priv)
 		return -ENOMEM;
 
 	priv->drvdata = drvdata;
+	priv->mcko = mcko;
 
 	i2c_set_clientdata(i2c, priv);
 
@@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
 	if (IS_ERR(regmap))
 		return PTR_ERR(regmap);
 
-	return snd_soc_register_codec(&i2c->dev,
+	return snd_soc_register_codec(dev,
 				      &soc_codec_dev_ak4642, &ak4642_dai, 1);
 }
 
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221ab3d10..ac21b85ff75f 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
 						   0x4f5, 0x0da);
 		}
 		break;
+	default:
+		break;
 	}
 
 	return 0;
@@ -689,6 +691,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
 				    ARIZONA_IN_VU, val);
 }
 
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+	unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+	unsigned int val = snd_soc_read(codec, reg);
+
+	return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
 int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		  int event)
 {
@@ -725,6 +736,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
 		if (reg == 0)
 			arizona_in_set_vu(codec, 0);
+		break;
+	default:
+		break;
 	}
 
 	return 0;
@@ -806,6 +820,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 			break;
 		}
 		break;
+	default:
+		break;
 	}
 
 	return 0;
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a418ff4b..7b68d05a0939 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -294,4 +294,6 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai);
 int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
 			    bool diff);
 
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
 #endif
diff --git a/sound/soc/codecs/hdmi.c b/sound/soc/codecs/hdmi.c
deleted file mode 100644
index bd42ad34e004..000000000000
--- a/sound/soc/codecs/hdmi.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * ALSA SoC codec driver for HDMI audio codecs.
- * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
- * Author: Ricardo Neri <ricardo.neri@ti.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.
- *
- * This program is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with this program; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
- * 02110-1301 USA
- *
- */
-#include <linux/module.h>
-#include <sound/soc.h>
-#include <linux/of.h>
-#include <linux/of_device.h>
-
-#define DRV_NAME "hdmi-audio-codec"
-
-static const struct snd_soc_dapm_widget hdmi_widgets[] = {
-	SND_SOC_DAPM_INPUT("RX"),
-	SND_SOC_DAPM_OUTPUT("TX"),
-};
-
-static const struct snd_soc_dapm_route hdmi_routes[] = {
-	{ "Capture", NULL, "RX" },
-	{ "TX", NULL, "Playback" },
-};
-
-static struct snd_soc_dai_driver hdmi_codec_dai = {
-	.name = "hdmi-hifi",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 2,
-		.channels_max = 8,
-		.rates = SNDRV_PCM_RATE_32000 |
-			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
-			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
-			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE,
-		.sig_bits = 24,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = SNDRV_PCM_RATE_32000 |
-			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |
-			SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 |
-			SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE |
-			SNDRV_PCM_FMTBIT_S24_LE,
-	},
-
-};
-
-#ifdef CONFIG_OF
-static const struct of_device_id hdmi_audio_codec_ids[] = {
-	{ .compatible = "linux,hdmi-audio", },
-	{ }
-};
-MODULE_DEVICE_TABLE(of, hdmi_audio_codec_ids);
-#endif
-
-static struct snd_soc_codec_driver hdmi_codec = {
-	.dapm_widgets = hdmi_widgets,
-	.num_dapm_widgets = ARRAY_SIZE(hdmi_widgets),
-	.dapm_routes = hdmi_routes,
-	.num_dapm_routes = ARRAY_SIZE(hdmi_routes),
-	.ignore_pmdown_time = true,
-};
-
-static int hdmi_codec_probe(struct platform_device *pdev)
-{
-	return snd_soc_register_codec(&pdev->dev, &hdmi_codec,
-			&hdmi_codec_dai, 1);
-}
-
-static int hdmi_codec_remove(struct platform_device *pdev)
-{
-	snd_soc_unregister_codec(&pdev->dev);
-	return 0;
-}
-
-static struct platform_driver hdmi_codec_driver = {
-	.driver		= {
-		.name	= DRV_NAME,
-		.of_match_table = of_match_ptr(hdmi_audio_codec_ids),
-	},
-
-	.probe		= hdmi_codec_probe,
-	.remove		= hdmi_codec_remove,
-};
-
-module_platform_driver(hdmi_codec_driver);
-
-MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
-MODULE_DESCRIPTION("ASoC generic HDMI codec driver");
-MODULE_LICENSE("GPL");
-MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c
index 3c2f0f8d6266..f823eb502367 100644
--- a/sound/soc/codecs/rt298.c
+++ b/sound/soc/codecs/rt298.c
@@ -50,24 +50,24 @@ struct rt298_priv {
 };
 
 static struct reg_default rt298_index_def[] = {
-	{ 0x01, 0xaaaa },
-	{ 0x02, 0x8aaa },
+	{ 0x01, 0xa5a8 },
+	{ 0x02, 0x8e95 },
 	{ 0x03, 0x0002 },
-	{ 0x04, 0xaf01 },
-	{ 0x08, 0x000d },
-	{ 0x09, 0xd810 },
-	{ 0x0a, 0x0120 },
+	{ 0x04, 0xaf67 },
+	{ 0x08, 0x200f },
+	{ 0x09, 0xd010 },
+	{ 0x0a, 0x0100 },
 	{ 0x0b, 0x0000 },
 	{ 0x0d, 0x2800 },
-	{ 0x0f, 0x0000 },
-	{ 0x19, 0x0a17 },
+	{ 0x0f, 0x0022 },
+	{ 0x19, 0x0217 },
 	{ 0x20, 0x0020 },
 	{ 0x33, 0x0208 },
 	{ 0x46, 0x0300 },
-	{ 0x49, 0x0004 },
-	{ 0x4f, 0x50e9 },
-	{ 0x50, 0x2000 },
-	{ 0x63, 0x2902 },
+	{ 0x49, 0x4004 },
+	{ 0x4f, 0x50c9 },
+	{ 0x50, 0x3000 },
+	{ 0x63, 0x1b02 },
 	{ 0x67, 0x1111 },
 	{ 0x68, 0x1016 },
 	{ 0x69, 0x273f },
@@ -1214,7 +1214,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c,
 	mdelay(10);
 
 	if (!rt298->pdata.gpio2_en)
-		regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x4000);
+		regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0x40);
 	else
 		regmap_write(rt298->regmap, RT298_SET_DMIC2_DEFAULT, 0);
 
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 4972bf3efa91..080cc1ce3963 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -519,11 +519,11 @@ static const struct snd_kcontrol_new rt5645_snd_controls[] = {
 		RT5645_L_VOL_SFT + 1, RT5645_R_VOL_SFT + 1, 63, 0, adc_vol_tlv),
 
 	/* ADC Boost Volume Control */
-	SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1,
+	SOC_DOUBLE_TLV("ADC Boost Capture Volume", RT5645_ADC_BST_VOL1,
 		RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0,
 		adc_bst_tlv),
-	SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1,
-		RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0,
+	SOC_DOUBLE_TLV("Mono ADC Boost Capture Volume", RT5645_ADC_BST_VOL2,
+		RT5645_MONO_ADC_L_BST_SFT, RT5645_MONO_ADC_R_BST_SFT, 3, 0,
 		adc_bst_tlv),
 
 	/* I2S2 function select */
@@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = {
 static const struct snd_kcontrol_new rt5645_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5645_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_R_SFT, 1, 1),
 };
 
@@ -1381,7 +1381,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
 				regmap_write(rt5645->regmap, RT5645_PR_BASE +
 					RT5645_MAMP_INT_REG2, 0xfc00);
 				snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
-				mdelay(5);
+				msleep(40);
 				rt5645->hp_on = true;
 			} else {
 				/* depop parameters */
@@ -2829,13 +2829,15 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
 		}
-
-		snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001);
+		if (rt5645->pdata.jd_invert)
+			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+				RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV);
 	} else { /* jack out */
 		rt5645->jack_type = 0;
 
+		regmap_update_bits(rt5645->regmap, RT5645_HP_VOL,
+			RT5645_L_MUTE | RT5645_R_MUTE,
+			RT5645_L_MUTE | RT5645_R_MUTE);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2,
 			RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
@@ -2848,6 +2850,9 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_disable_pin(dapm, "LDO2");
 		snd_soc_dapm_disable_pin(dapm, "Mic Det Power");
 		snd_soc_dapm_sync(dapm);
+		if (rt5645->pdata.jd_invert)
+			regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2,
+				RT5645_JD_1_1_MASK, RT5645_JD_1_1_NOR);
 	}
 
 	return rt5645->jack_type;
@@ -2880,8 +2885,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec,
 		rt5645->en_button_func = true;
 		regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
 				RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
-		regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
-				RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
 		regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
 				RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
 	}
@@ -3205,9 +3208,42 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
 		},
 	},
+	{
+		.ident = "Google Ultima",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
+		},
+	},
+	{ }
+};
+
+static struct rt5645_platform_data buddy_platform_data = {
+	.dmic1_data_pin = RT5645_DMIC_DATA_GPIO5,
+	.dmic2_data_pin = RT5645_DMIC_DATA_IN2P,
+	.jd_mode = 3,
+	.jd_invert = true,
+};
+
+static int buddy_quirk_cb(const struct dmi_system_id *id)
+{
+	rt5645_pdata = &buddy_platform_data;
+
+	return 1;
+}
+
+static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = {
+	{
+		.ident = "Chrome Buddy",
+		.callback = buddy_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"),
+		},
+	},
 	{ }
 };
 
+
 static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev)
 {
 	rt5645->pdata.in2_diff = device_property_read_bool(dev,
@@ -3240,7 +3276,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c,
 
 	if (pdata)
 		rt5645->pdata = *pdata;
-	else if (dmi_check_system(dmi_platform_intel_braswell))
+	else if (dmi_check_system(dmi_platform_intel_braswell) ||
+			dmi_check_system(dmi_platform_intel_broadwell))
 		rt5645->pdata = *rt5645_pdata;
 	else
 		rt5645_parse_dt(rt5645, &i2c->dev);
diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h
index 0e4cfc6ac649..61bc8ab77646 100644
--- a/sound/soc/codecs/rt5645.h
+++ b/sound/soc/codecs/rt5645.h
@@ -39,8 +39,8 @@
 #define RT5645_STO1_ADC_DIG_VOL			0x1c
 #define RT5645_MONO_ADC_DIG_VOL			0x1d
 #define RT5645_ADC_BST_VOL1			0x1e
-/* Mixer - D-D */
 #define RT5645_ADC_BST_VOL2			0x20
+/* Mixer - D-D */
 #define RT5645_STO1_ADC_MIXER			0x27
 #define RT5645_MONO_ADC_MIXER			0x28
 #define RT5645_AD_DA_MIXER			0x29
@@ -315,12 +315,14 @@
 #define RT5645_STO1_ADC_R_BST_SFT		12
 #define RT5645_STO1_ADC_COMP_MASK		(0x3 << 10)
 #define RT5645_STO1_ADC_COMP_SFT		10
-#define RT5645_STO2_ADC_L_BST_MASK		(0x3 << 8)
-#define RT5645_STO2_ADC_L_BST_SFT		8
-#define RT5645_STO2_ADC_R_BST_MASK		(0x3 << 6)
-#define RT5645_STO2_ADC_R_BST_SFT		6
-#define RT5645_STO2_ADC_COMP_MASK		(0x3 << 4)
-#define RT5645_STO2_ADC_COMP_SFT		4
+
+/* ADC Boost Volume Control (0x20) */
+#define RT5645_MONO_ADC_L_BST_MASK		(0x3 << 14)
+#define RT5645_MONO_ADC_L_BST_SFT		14
+#define RT5645_MONO_ADC_R_BST_MASK		(0x3 << 12)
+#define RT5645_MONO_ADC_R_BST_SFT		12
+#define RT5645_MONO_ADC_COMP_MASK		(0x3 << 10)
+#define RT5645_MONO_ADC_COMP_SFT		10
 
 /* Stereo2 ADC Mixer Control (0x26) */
 #define RT5645_STO2_ADC_SRC_MASK		(0x1 << 15)
@@ -777,8 +779,6 @@
 #define RT5645_PWR_CLS_D_R_BIT			9
 #define RT5645_PWR_CLS_D_L			(0x1 << 8)
 #define RT5645_PWR_CLS_D_L_BIT			8
-#define RT5645_PWR_ADC_R			(0x1 << 1)
-#define RT5645_PWR_ADC_R_BIT			1
 #define RT5645_PWR_DAC_L2			(0x1 << 7)
 #define RT5645_PWR_DAC_L2_BIT			7
 #define RT5645_PWR_DAC_R2			(0x1 << 6)
@@ -1626,6 +1626,10 @@
 #define RT5645_OT_P_NOR				(0x0 << 10)
 #define RT5645_OT_P_INV				(0x1 << 10)
 #define RT5645_IRQ_JD_1_1_EN			(0x1 << 9)
+#define RT5645_JD_1_1_MASK			(0x1 << 7)
+#define RT5645_JD_1_1_SFT			7
+#define RT5645_JD_1_1_NOR			(0x0 << 7)
+#define RT5645_JD_1_1_INV			(0x1 << 7)
 
 /* IRQ Control 2 (0xbe) */
 #define RT5645_IRQ_MB1_OC_MASK			(0x1 << 15)
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index bfda25ef0dd4..f540f82b1f27 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec)
 			sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT);
 
 	snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL,
-			SGTL5000_BIAS_R_MASK,
-			sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT);
+			SGTL5000_BIAS_VOLT_MASK,
+			sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT);
 	/*
 	 * disable DAP
 	 * TODO:
@@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 			else {
 				sgtl5000->micbias_voltage = 0;
 				dev_err(&client->dev,
-					"Unsuitable MicBias resistor\n");
+					"Unsuitable MicBias voltage\n");
 			}
 		} else {
 			sgtl5000->micbias_voltage = 0;
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index e3a0bca28bcf..cc1d3981fa4b 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -549,7 +549,7 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
 /*
  * DAC digital volumes. From -7 to 24 dB in 1 dB steps
  */
-static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 0);
+static DECLARE_TLV_DB_SCALE(dac_tlv, -700, 100, 0);
 
 static const char * const tas2552_din_source_select[] = {
 	"Muted",
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index 1a82b19b2644..a564759845f9 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -80,6 +80,7 @@ struct aic3x_priv {
 	unsigned int sysclk;
 	unsigned int dai_fmt;
 	unsigned int tdm_delay;
+	unsigned int slot_width;
 	struct list_head list;
 	int master;
 	int gpio_reset;
@@ -1025,10 +1026,14 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 	u8 data, j, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
 	u16 d, pll_d = 1;
 	int clk;
+	int width = aic3x->slot_width;
+
+	if (!width)
+		width = params_width(params);
 
 	/* select data word length */
 	data = snd_soc_read(codec, AIC3X_ASD_INTF_CTRLB) & (~(0x3 << 4));
-	switch (params_width(params)) {
+	switch (width) {
 	case 16:
 		break;
 	case 20:
@@ -1170,12 +1175,16 @@ static int aic3x_prepare(struct snd_pcm_substream *substream,
 	struct snd_soc_codec *codec = dai->codec;
 	struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec);
 	int delay = 0;
+	int width = aic3x->slot_width;
+
+	if (!width)
+		width = substream->runtime->sample_bits;
 
 	/* TDM slot selection only valid in DSP_A/_B mode */
 	if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_A)
-		delay += (aic3x->tdm_delay + 1);
+		delay += (aic3x->tdm_delay*width + 1);
 	else if (aic3x->dai_fmt == SND_SOC_DAIFMT_DSP_B)
-		delay += aic3x->tdm_delay;
+		delay += aic3x->tdm_delay*width;
 
 	/* Configure data delay */
 	snd_soc_write(codec, AIC3X_ASD_INTF_CTRLC, delay);
@@ -1296,7 +1305,20 @@ static int aic3x_set_dai_tdm_slot(struct snd_soc_dai *codec_dai,
 		return -EINVAL;
 	}
 
-	aic3x->tdm_delay = lsb * slot_width;
+	switch (slot_width) {
+	case 16:
+	case 20:
+	case 24:
+	case 32:
+		break;
+	default:
+		dev_err(codec->dev, "Unsupported slot width %d\n", slot_width);
+		return -EINVAL;
+	}
+
+
+	aic3x->tdm_delay = lsb;
+	aic3x->slot_width = slot_width;
 
 	/* DOUT in high-impedance on inactive bit clocks */
 	snd_soc_update_bits(codec, AIC3X_ASD_INTF_CTRLA,
@@ -1509,14 +1531,17 @@ static int aic3x_init(struct snd_soc_codec *codec)
 	snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL);
 	snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL);
 
-	/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
-	/* Line2 Line Out default volume, disconnect from Output Mixer */
-	snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
-	snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+	/* On tlv320aic3104, these registers are reserved and must not be written */
+	if (aic3x->model != AIC3X_MODEL_3104) {
+		/* Line2 to HP Bypass default volume, disconnect from Output Mixer */
+		snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL);
+		/* Line2 Line Out default volume, disconnect from Output Mixer */
+		snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL);
+		snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL);
+	}
 
 	switch (aic3x->model) {
 	case AIC3X_MODEL_3X:
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f2c6ad4b8fde..581ec1502228 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -577,7 +577,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
 	unsigned long flags;
 	int ret;
-	const struct firmware *fw;
 	struct spi_message m;
 	struct spi_transfer t;
 	struct dfw_pllrec pll_rec;
@@ -623,14 +622,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	wm0010->state = WM0010_OUT_OF_RESET;
 	spin_unlock_irqrestore(&wm0010->irq_lock, flags);
 
-	/* First the bootloader */
-	ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
-			ret);
-		goto abort;
-	}
-
 	if (!wait_for_completion_timeout(&wm0010->boot_completion,
 					 msecs_to_jiffies(20)))
 		dev_err(codec->dev, "Failed to get interrupt from DSP\n");
@@ -673,7 +664,7 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!img_swap)
-			goto abort;
+			goto abort_out;
 
 		/* We need to re-order for 0010 */
 		byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -688,16 +679,16 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		spi_message_add_tail(&t, &m);
 
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "First PLL write failed: %d\n", ret);
-			goto abort;
+			goto abort_swap;
 		}
 
 		/* Use a second send of the message to get the return status */
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
-			goto abort;
+			goto abort_swap;
 		}
 
 		p = (u32 *)out;
@@ -730,6 +721,10 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 	return 0;
 
+abort_swap:
+	kfree(img_swap);
+abort_out:
+	kfree(out);
 abort:
 	/* Put the chip back into reset */
 	wm0010_halt(codec);
diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c
index 9756578fc752..c04c0bc6f58a 100644
--- a/sound/soc/codecs/wm5110.c
+++ b/sound/soc/codecs/wm5110.c
@@ -38,6 +38,12 @@
 struct wm5110_priv {
 	struct arizona_priv core;
 	struct arizona_fll fll[2];
+
+	unsigned int in_value;
+	int in_pre_pending;
+	int in_post_pending;
+
+	unsigned int in_pga_cache[6];
 };
 
 static const struct wm_adsp_region wm5110_dsp1_regions[] = {
@@ -428,6 +434,127 @@ err:
 	return ret;
 }
 
+static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct snd_soc_card *card = dapm->card;
+	int ret;
+
+	/*
+	 * PGA Volume is also used as part of the enable sequence, so
+	 * usage of it should be avoided whilst that is running.
+	 */
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_get_volsw_range(kcontrol, ucontrol);
+
+	mutex_unlock(&card->dapm_mutex);
+
+	return ret;
+}
+
+static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+	struct snd_soc_card *card = dapm->card;
+	int ret;
+
+	/*
+	 * PGA Volume is also used as part of the enable sequence, so
+	 * usage of it should be avoided whilst that is running.
+	 */
+	mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME);
+
+	ret = snd_soc_put_volsw_range(kcontrol, ucontrol);
+
+	mutex_unlock(&card->dapm_mutex);
+
+	return ret;
+}
+
+static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w,
+			       struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	unsigned int reg, mask;
+	struct reg_sequence analog_seq[] = {
+		{ 0x80, 0x3 },
+		{ 0x35d, 0 },
+		{ 0x80, 0x0 },
+	};
+
+	reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4);
+	mask = ARIZONA_IN1L_PGA_VOL_MASK;
+
+	switch (event) {
+	case SND_SOC_DAPM_WILL_PMU:
+		wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2);
+		wm5110->in_pre_pending++;
+		wm5110->in_post_pending++;
+		return 0;
+	case SND_SOC_DAPM_PRE_PMU:
+		wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg);
+
+		snd_soc_update_bits(codec, reg, mask,
+				    0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT);
+
+		wm5110->in_pre_pending--;
+		if (wm5110->in_pre_pending == 0) {
+			analog_seq[1].def = wm5110->in_value;
+			regmap_multi_reg_write_bypassed(arizona->regmap,
+							analog_seq,
+							ARRAY_SIZE(analog_seq));
+
+			msleep(55);
+
+			wm5110->in_value = 0;
+		}
+
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		snd_soc_update_bits(codec, reg, mask,
+				    wm5110->in_pga_cache[w->shift]);
+
+		wm5110->in_post_pending--;
+		if (wm5110->in_post_pending == 0)
+			regmap_multi_reg_write_bypassed(arizona->regmap,
+							analog_seq,
+							ARRAY_SIZE(analog_seq));
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int wm5110_in_ev(struct snd_soc_dapm_widget *w,
+			struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+
+	switch (arizona->rev) {
+	case 0 ... 4:
+		if (arizona_input_analog(codec, w->shift))
+			wm5110_in_analog_ev(w, kcontrol, event);
+
+		break;
+	default:
+		break;
+	}
+
+	return arizona_in_ev(w, kcontrol, event);
+}
+
 static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0);
 static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0);
 static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0);
@@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]),
 SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]),
 SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]),
 
-SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
-		     ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
-		     ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
-		     ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
-		     ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
-		     ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
-SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
-		     ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL,
+			 ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL,
+			 ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL,
+			 ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL,
+			 ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL,
+			 ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
+SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL,
+			 ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0,
+			 wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv),
 
 SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum),
 
@@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"),
 SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"),
 
 SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT,
-		   0, NULL, 0, arizona_in_ev,
+		   0, NULL, 0, wm5110_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
-		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU),
+		   SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU |
+		   SND_SOC_DAPM_WILL_PMU),
 SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT,
 		   0, NULL, 0, arizona_in_ev,
 		   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD |
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index e3b7d0c57411..dbd88408861a 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -211,28 +211,38 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 	return wm8960_set_deemph(codec);
 }
 
-static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
-static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
-static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
+static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
+static const unsigned int micboost_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
+	2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
+};
 
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
-		 0, 63, 0, adc_tlv),
+		 0, 63, 0, inpga_tlv),
 SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 	6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
 	7, 1, 0),
 
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
-	       WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
-	       WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
-	       WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
-	       WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
+		WM8960_RINPATH, 4, 3, 0, micboost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
+		WM8960_LINPATH, 4, 3, 0, micboost_tlv),
 
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b4eb975da981..39ebd7bf4f53 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2944,7 +2944,8 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 				   WM8962_DAC_MUTE, val);
 }
 
-#define WM8962_RATES SNDRV_PCM_RATE_8000_96000
+#define WM8962_RATES (SNDRV_PCM_RATE_8000_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
@@ -3759,7 +3760,7 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
 	ret = snd_soc_register_codec(&i2c->dev,
 				     &soc_codec_dev_wm8962, &wm8962_dai, 1);
 	if (ret < 0)
-		goto err_enable;
+		goto err_pm_runtime;
 
 	regcache_cache_only(wm8962->regmap, true);
 
@@ -3768,6 +3769,8 @@ static int wm8962_i2c_probe(struct i2c_client *i2c,
 
 	return 0;
 
+err_pm_runtime:
+	pm_runtime_disable(&i2c->dev);
 err_enable:
 	regulator_bulk_disable(ARRAY_SIZE(wm8962->supplies), wm8962->supplies);
 err:
@@ -3777,6 +3780,7 @@ err:
 static int wm8962_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_codec(&client->dev);
+	pm_runtime_disable(&client->dev);
 	return 0;
 }
 
@@ -3804,6 +3808,8 @@ static int wm8962_runtime_resume(struct device *dev)
 
 	wm8962_reset(wm8962);
 
+	regcache_mark_dirty(wm8962->regmap);
+
 	/* SYSCLK defaults to on; make sure it is off so we can safely
 	 * write to registers if the device is declocked.
 	 */