Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

28 files changed, 1729 insertions, 328 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 45b72561c615..2f45f00e31b0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4641 if I2C
 	select SND_SOC_AK4642 if I2C
 	select SND_SOC_AK4671 if I2C
+	select SND_SOC_AK5386
 	select SND_SOC_ALC5623 if I2C
 	select SND_SOC_ALC5632 if I2C
 	select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC
@@ -63,6 +64,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_STA32X if I2C
 	select SND_SOC_STA529 if I2C
 	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
+	select SND_SOC_TAS5086 if I2C
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC32X4 if I2C
@@ -203,6 +205,9 @@ config SND_SOC_AK4642
 config SND_SOC_AK4671
 	tristate
 
+config SND_SOC_AK5386
+	tristate
+
 config SND_SOC_ALC5623
        tristate
 config SND_SOC_ALC5632
@@ -320,11 +325,14 @@ config SND_SOC_STA529
 config SND_SOC_STAC9766
 	tristate
 
+config SND_SOC_TAS5086
+	tristate
+
 config SND_SOC_TLV320AIC23
 	tristate
 
 config SND_SOC_TLV320AIC26
-	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
+	tristate
 	depends on SPI
 
 config SND_SOC_TLV320AIC32X4
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 6a3b3c3b8b41..b9e41c9a1f4c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-ak4641-objs := ak4641.o
 snd-soc-ak4642-objs := ak4642.o
 snd-soc-ak4671-objs := ak4671.o
+snd-soc-ak5386-objs := ak5386.o
 snd-soc-arizona-objs := arizona.o
 snd-soc-cq93vc-objs := cq93vc.o
 snd-soc-cs42l51-objs := cs42l51.o
@@ -55,6 +56,7 @@ snd-soc-ssm2602-objs := ssm2602.o
 snd-soc-sta32x-objs := sta32x.o
 snd-soc-sta529-objs := sta529.o
 snd-soc-stac9766-objs := stac9766.o
+snd-soc-tas5086-objs := tas5086.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -137,6 +139,7 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_AK4641)	+= snd-soc-ak4641.o
 obj-$(CONFIG_SND_SOC_AK4642)	+= snd-soc-ak4642.o
 obj-$(CONFIG_SND_SOC_AK4671)	+= snd-soc-ak4671.o
+obj-$(CONFIG_SND_SOC_AK5386)	+= snd-soc-ak5386.o
 obj-$(CONFIG_SND_SOC_ALC5623)    += snd-soc-alc5623.o
 obj-$(CONFIG_SND_SOC_ALC5632)	+= snd-soc-alc5632.o
 obj-$(CONFIG_SND_SOC_ARIZONA)	+= snd-soc-arizona.o
@@ -177,6 +180,7 @@ obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
 obj-$(CONFIG_SND_SOC_STA32X)   += snd-soc-sta32x.o
 obj-$(CONFIG_SND_SOC_STA529)   += snd-soc-sta529.o
 obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
+obj-$(CONFIG_SND_SOC_TAS5086)	+= snd-soc-tas5086.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c
index 068b3ae56a17..1aa10ddf3a61 100644
--- a/sound/soc/codecs/adau1373.c
+++ b/sound/soc/codecs/adau1373.c
@@ -133,6 +133,8 @@ struct adau1373 {
 #define ADAU1373_DAI_FORMAT_DSP		0x3
 
 #define ADAU1373_BCLKDIV_SOURCE		BIT(5)
+#define ADAU1373_BCLKDIV_SR_MASK	(0x07 << 2)
+#define ADAU1373_BCLKDIV_BCLK_MASK	0x03
 #define ADAU1373_BCLKDIV_32		0x03
 #define ADAU1373_BCLKDIV_64		0x02
 #define ADAU1373_BCLKDIV_128		0x01
@@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream,
 	adau1373_dai->enable_src = (div != 0);
 
 	snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id),
-		~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64);
+		ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK,
+		(div << 2) | ADAU1373_BCLKDIV_64);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
index 6f6c335a5baa..c7cfdf957e4d 100644
--- a/sound/soc/codecs/ak4104.c
+++ b/sound/soc/codecs/ak4104.c
@@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 			      unsigned int format)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
 	int val = 0;
 	int ret;
 
@@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
 		return -EINVAL;
 
-	ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-				  AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
-				  val);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+				 AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1,
+				 val);
 	if (ret < 0)
 		return ret;
 
@@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 			    struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	int val = 0;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+	int ret, val = 0;
 
 	/* set the IEC958 bits: consumer mode, no copyright bit */
 	val |= IEC958_AES0_CON_NOT_COPYRIGHT;
-	snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val);
+	regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val);
 
 	val = 0;
 
@@ -132,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
-	return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val);
+	ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val);
+	if (ret < 0)
+		return ret;
+
+	/* enable transmitter */
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				 AK4104_TX_TXE, AK4104_TX_TXE);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static int ak4104_hw_free(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+	/* disable transmitter */
+	return regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				  AK4104_TX_TXE, 0);
 }
 
 static const struct snd_soc_dai_ops ak4101_dai_ops = {
 	.hw_params = ak4104_hw_params,
+	.hw_free = ak4104_hw_free,
 	.set_fmt = ak4104_set_dai_fmt,
 };
 
@@ -160,20 +184,17 @@ static int ak4104_probe(struct snd_soc_codec *codec)
 	int ret;
 
 	codec->control_data = ak4104->regmap;
-	ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP);
-	if (ret != 0)
-		return ret;
 
 	/* set power-up and non-reset bits */
-	ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-				  AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
-				  AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+				 AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN,
+				 AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
 	if (ret < 0)
 		return ret;
 
 	/* enable transmitter */
-	ret = snd_soc_update_bits(codec, AK4104_REG_TX,
-				  AK4104_TX_TXE, AK4104_TX_TXE);
+	ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX,
+				 AK4104_TX_TXE, AK4104_TX_TXE);
 	if (ret < 0)
 		return ret;
 
@@ -182,8 +203,10 @@ static int ak4104_probe(struct snd_soc_codec *codec)
 
 static int ak4104_remove(struct snd_soc_codec *codec)
 {
-	snd_soc_update_bits(codec, AK4104_REG_CONTROL1,
-			    AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
+	struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1,
+			   AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c
new file mode 100644
index 000000000000..1f303983ae02
--- /dev/null
+++ b/sound/soc/codecs/ak5386.c
@@ -0,0 +1,152 @@
+/*
+ * ALSA SoC driver for
+ *    Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC
+ *
+ * (c) 2013 Daniel Mack <zonque@gmail.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/of.h>
+#include <linux/of_gpio.h>
+#include <linux/of_device.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+
+struct ak5386_priv {
+	int reset_gpio;
+};
+
+static struct snd_soc_codec_driver soc_codec_ak5386;
+
+static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+
+	format &= SND_SOC_DAIFMT_FORMAT_MASK;
+	if (format != SND_SOC_DAIFMT_LEFT_J &&
+	    format != SND_SOC_DAIFMT_I2S) {
+		dev_err(codec->dev, "Invalid DAI format\n");
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static int ak5386_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 ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	/*
+	 * From the datasheet:
+	 *
+	 * All external clocks (MCLK, SCLK and LRCK) must be present unless
+	 * PDN pin = “L”. If these clocks are not provided, the AK5386 may
+	 * draw excess current due to its use of internal dynamically
+	 * refreshed logic. If the external clocks are not present, place
+	 * the AK5386 in power-down mode (PDN pin = “L”).
+	 */
+
+	if (gpio_is_valid(priv->reset_gpio))
+		gpio_set_value(priv->reset_gpio, 1);
+
+	return 0;
+}
+
+static int ak5386_hw_free(struct snd_pcm_substream *substream,
+			  struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+	if (gpio_is_valid(priv->reset_gpio))
+		gpio_set_value(priv->reset_gpio, 0);
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops ak5386_dai_ops = {
+	.set_fmt	= ak5386_set_dai_fmt,
+	.hw_params	= ak5386_hw_params,
+	.hw_free	= ak5386_hw_free,
+};
+
+static struct snd_soc_dai_driver ak5386_dai = {
+	.name		= "ak5386-hifi",
+	.capture	= {
+		.stream_name	= "Capture",
+		.channels_min	= 1,
+		.channels_max	= 2,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
+		.formats	= SNDRV_PCM_FMTBIT_S8     |
+				  SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S24_LE |
+				  SNDRV_PCM_FMTBIT_S24_3LE,
+	},
+	.ops	= &ak5386_dai_ops,
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id ak5386_dt_ids[] = {
+	{ .compatible = "asahi-kasei,ak5386", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, ak5386_dt_ids);
+#endif
+
+static int ak5386_probe(struct platform_device *pdev)
+{
+	struct device *dev = &pdev->dev;
+	struct ak5386_priv *priv;
+
+	priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->reset_gpio = -EINVAL;
+	dev_set_drvdata(dev, priv);
+
+	if (of_match_device(of_match_ptr(ak5386_dt_ids), dev))
+		priv->reset_gpio = of_get_named_gpio(dev->of_node,
+						      "reset-gpio", 0);
+
+	if (gpio_is_valid(priv->reset_gpio))
+		if (devm_gpio_request_one(dev, priv->reset_gpio,
+					  GPIOF_OUT_INIT_LOW,
+					  "AK5386 Reset"))
+			priv->reset_gpio = -EINVAL;
+
+	return snd_soc_register_codec(dev, &soc_codec_ak5386,
+				      &ak5386_dai, 1);
+}
+
+static int ak5386_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static struct platform_driver ak5386_driver = {
+	.probe		= ak5386_probe,
+	.remove		= ak5386_remove,
+	.driver		= {
+		.name	= "ak5386",
+		.owner	= THIS_MODULE,
+		.of_match_table = of_match_ptr(ak5386_dt_ids),
+	},
+};
+
+module_platform_driver(ak5386_driver);
+
+MODULE_DESCRIPTION("ASoC driver for AK5386 ADC");
+MODULE_AUTHOR("Daniel Mack <zonque@gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index e7d34711412c..389f23253831 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -10,6 +10,7 @@
  * published by the Free Software Foundation.
  */
 
+#include <linux/delay.h>
 #include <linux/gcd.h>
 #include <linux/module.h>
 #include <linux/pm_runtime.h>
@@ -65,6 +66,163 @@
 #define arizona_aif_dbg(_dai, fmt, ...) \
 	dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__)
 
+static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *kcontrol,
+			  int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct arizona *arizona = dev_get_drvdata(codec->dev->parent);
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	bool manual_ena = false;
+	int val;
+
+	switch (arizona->type) {
+	case WM5102:
+		switch (arizona->rev) {
+		case 0:
+			break;
+		default:
+			manual_ena = true;
+			break;
+		}
+	default:
+		break;
+	}
+
+	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		if (!priv->spk_ena && manual_ena) {
+			snd_soc_write(codec, 0x4f5, 0x25a);
+			priv->spk_ena_pending = true;
+		}
+		break;
+	case SND_SOC_DAPM_POST_PMU:
+		val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3);
+		if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+			dev_crit(arizona->dev,
+				 "Speaker not enabled due to temperature\n");
+			return -EBUSY;
+		}
+
+		snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
+				    1 << w->shift, 1 << w->shift);
+
+		if (priv->spk_ena_pending) {
+			msleep(75);
+			snd_soc_write(codec, 0x4f5, 0xda);
+			priv->spk_ena_pending = false;
+			priv->spk_ena++;
+		}
+		break;
+	case SND_SOC_DAPM_PRE_PMD:
+		if (manual_ena) {
+			priv->spk_ena--;
+			if (!priv->spk_ena)
+				snd_soc_write(codec, 0x4f5, 0x25a);
+		}
+
+		snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1,
+				    1 << w->shift, 0);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		if (manual_ena) {
+			if (!priv->spk_ena)
+				snd_soc_write(codec, 0x4f5, 0x0da);
+		}
+		break;
+	}
+
+	return 0;
+}
+
+static irqreturn_t arizona_thermal_warn(int irq, void *data)
+{
+	struct arizona *arizona = data;
+	unsigned int val;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3,
+			  &val);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to read thermal status: %d\n",
+			ret);
+	} else if (val & ARIZONA_SPK_SHUTDOWN_WARN_STS) {
+		dev_crit(arizona->dev, "Thermal warning\n");
+	}
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t arizona_thermal_shutdown(int irq, void *data)
+{
+	struct arizona *arizona = data;
+	unsigned int val;
+	int ret;
+
+	ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3,
+			  &val);
+	if (ret != 0) {
+		dev_err(arizona->dev, "Failed to read thermal status: %d\n",
+			ret);
+	} else if (val & ARIZONA_SPK_SHUTDOWN_STS) {
+		dev_crit(arizona->dev, "Thermal shutdown\n");
+		ret = regmap_update_bits(arizona->regmap,
+					 ARIZONA_OUTPUT_ENABLES_1,
+					 ARIZONA_OUT4L_ENA |
+					 ARIZONA_OUT4R_ENA, 0);
+		if (ret != 0)
+			dev_crit(arizona->dev,
+				 "Failed to disable speaker outputs: %d\n",
+				 ret);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static const struct snd_soc_dapm_widget arizona_spkl =
+	SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM,
+			   ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev,
+			   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU);
+
+static const struct snd_soc_dapm_widget arizona_spkr =
+	SND_SOC_DAPM_PGA_E("OUT4R", SND_SOC_NOPM,
+			   ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev,
+			   SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU);
+
+int arizona_init_spk(struct snd_soc_codec *codec)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	struct arizona *arizona = priv->arizona;
+	int ret;
+
+	ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1);
+	if (ret != 0)
+		return ret;
+
+	ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1);
+	if (ret != 0)
+		return ret;
+
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN,
+				  "Thermal warning", arizona_thermal_warn,
+				  arizona);
+	if (ret != 0)
+		dev_err(arizona->dev,
+			"Failed to get thermal warning IRQ: %d\n",
+			ret);
+
+	ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN,
+				  "Thermal shutdown", arizona_thermal_shutdown,
+				  arizona);
+	if (ret != 0)
+		dev_err(arizona->dev,
+			"Failed to get thermal shutdown IRQ: %d\n",
+			ret);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(arizona_init_spk);
+
 const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
 	"None",
 	"Tone Generator 1",
@@ -274,6 +432,33 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values);
 const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
 EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
 
+const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
+	"SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
+};
+EXPORT_SYMBOL_GPL(arizona_rate_text);
+
+const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = {
+	0, 1, 2, 8,
+};
+EXPORT_SYMBOL_GPL(arizona_rate_val);
+
+
+const struct soc_enum arizona_isrc_fsl[] = {
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2,
+			      ARIZONA_ISRC1_FSL_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_2,
+			      ARIZONA_ISRC2_FSL_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+	SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_2,
+			      ARIZONA_ISRC3_FSL_SHIFT, 0xf,
+			      ARIZONA_RATE_ENUM_SIZE,
+			      arizona_rate_text, arizona_rate_val),
+};
+EXPORT_SYMBOL_GPL(arizona_isrc_fsl);
+
 static const char *arizona_vol_ramp_text[] = {
 	"0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB",
 	"15ms/6dB", "30ms/6dB",
@@ -332,9 +517,27 @@ const struct soc_enum arizona_ng_hold =
 			4, arizona_ng_hold_text);
 EXPORT_SYMBOL_GPL(arizona_ng_hold);
 
+static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
+{
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec);
+	unsigned int val;
+	int i;
+
+	if (ena)
+		val = ARIZONA_IN_VU;
+	else
+		val = 0;
+
+	for (i = 0; i < priv->num_inputs; i++)
+		snd_soc_update_bits(codec,
+				    ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 4),
+				    ARIZONA_IN_VU, val);
+}
+
 int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		  int event)
 {
+	struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec);
 	unsigned int reg;
 
 	if (w->shift % 2)
@@ -343,13 +546,29 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
 		reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8);
 
 	switch (event) {
+	case SND_SOC_DAPM_PRE_PMU:
+		priv->in_pending++;
+		break;
 	case SND_SOC_DAPM_POST_PMU:
 		snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0);
+
+		/* If this is the last input pending then allow VU */
+		priv->in_pending--;
+		if (priv->in_pending == 0) {
+			msleep(1);
+			arizona_in_set_vu(w->codec, 1);
+		}
 		break;
 	case SND_SOC_DAPM_PRE_PMD:
-		snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE,
-				    ARIZONA_IN1L_MUTE);
+		snd_soc_update_bits(w->codec, reg,
+				    ARIZONA_IN1L_MUTE | ARIZONA_IN_VU,
+				    ARIZONA_IN1L_MUTE | ARIZONA_IN_VU);
 		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Disable volume updates if no inputs are enabled */
+		reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES);
+		if (reg == 0)
+			arizona_in_set_vu(w->codec, 0);
 	}
 
 	return 0;
@@ -360,6 +579,24 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol,
 		   int event)
 {
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		switch (w->shift) {
+		case ARIZONA_OUT1L_ENA_SHIFT:
+		case ARIZONA_OUT1R_ENA_SHIFT:
+		case ARIZONA_OUT2L_ENA_SHIFT:
+		case ARIZONA_OUT2R_ENA_SHIFT:
+		case ARIZONA_OUT3L_ENA_SHIFT:
+		case ARIZONA_OUT3R_ENA_SHIFT:
+			msleep(17);
+			break;
+
+		default:
+			break;
+		}
+		break;
+	}
+
 	return 0;
 }
 EXPORT_SYMBOL_GPL(arizona_out_ev);
@@ -502,27 +739,27 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 		break;
 	case 11289600:
 	case 12288000:
-		val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_12MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 22579200:
 	case 24576000:
-		val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_24MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 45158400:
 	case 49152000:
-		val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_49MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 67737600:
 	case 73728000:
-		val |= 4 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_73MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 90316800:
 	case 98304000:
-		val |= 5 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_98MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 135475200:
 	case 147456000:
-		val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT;
+		val |= ARIZONA_CLK_147MHZ << ARIZONA_SYSCLK_FREQ_SHIFT;
 		break;
 	case 0:
 		dev_dbg(arizona->dev, "%s cleared\n", name);
@@ -816,7 +1053,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 	struct arizona *arizona = priv->arizona;
 	int base = dai->driver->base;
 	const int *rates;
-	int i, ret;
+	int i, ret, val;
 	int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1];
 	int bclk, lrclk, wl, frame, bclk_target;
 
@@ -832,6 +1069,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream,
 		bclk_target *= chan_limit;
 	}
 
+	/* Force stereo for I2S mode */
+	val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT);
+	if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) {
+		arizona_aif_dbg(dai, "Forcing stereo mode\n");
+		bclk_target *= 2;
+	}
+
 	for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) {
 		if (rates[i] >= bclk_target &&
 		    rates[i] % params_rate(params) == 0) {
@@ -988,6 +1232,16 @@ static struct {
 	{ 1000000, 13500000, 0,  1 },
 };
 
+static struct {
+	unsigned int min;
+	unsigned int max;
+	u16 gain;
+} fll_gains[] = {
+	{       0,   256000, 0 },
+	{  256000,  1000000, 2 },
+	{ 1000000, 13500000, 4 },
+};
+
 struct arizona_fll_cfg {
 	int n;
 	int theta;
@@ -995,6 +1249,7 @@ struct arizona_fll_cfg {
 	int refdiv;
 	int outdiv;
 	int fratio;
+	int gain;
 };
 
 static int arizona_calc_fll(struct arizona_fll *fll,
@@ -1054,6 +1309,18 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 		return -EINVAL;
 	}
 
+	for (i = 0; i < ARRAY_SIZE(fll_gains); i++) {
+		if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) {
+			cfg->gain = fll_gains[i].gain;
+			break;
+		}
+	}
+	if (i == ARRAY_SIZE(fll_gains)) {
+		arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n",
+				Fref);
+		return -EINVAL;
+	}
+
 	cfg->n = target / (ratio * Fref);
 
 	if (target % (ratio * Fref)) {
@@ -1081,13 +1348,15 @@ static int arizona_calc_fll(struct arizona_fll *fll,
 			cfg->n, cfg->theta, cfg->lambda);
 	arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n",
 			cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv);
+	arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain);
 
 	return 0;
 
 }
 
 static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
-			      struct arizona_fll_cfg *cfg, int source)
+			      struct arizona_fll_cfg *cfg, int source,
+			      bool sync)
 {
 	regmap_update_bits(arizona->regmap, base + 3,
 			   ARIZONA_FLL1_THETA_MASK, cfg->theta);
@@ -1102,87 +1371,84 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base,
 			   cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT |
 			   source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT);
 
+	if (sync)
+		regmap_update_bits(arizona->regmap, base + 0x7,
+				   ARIZONA_FLL1_GAIN_MASK,
+				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+	else
+		regmap_update_bits(arizona->regmap, base + 0x9,
+				   ARIZONA_FLL1_GAIN_MASK,
+				   cfg->gain << ARIZONA_FLL1_GAIN_SHIFT);
+
 	regmap_update_bits(arizona->regmap, base + 2,
 			   ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK,
 			   ARIZONA_FLL1_CTRL_UPD | cfg->n);
 }
 
-int arizona_set_fll(struct arizona_fll *fll, int source,
-		    unsigned int Fref, unsigned int Fout)
+static bool arizona_is_enabled_fll(struct arizona_fll *fll)
 {
 	struct arizona *arizona = fll->arizona;
-	struct arizona_fll_cfg cfg, sync;
-	unsigned int reg, val;
-	int syncsrc;
-	bool ena;
+	unsigned int reg;
 	int ret;
 
-	if (fll->fref == Fref && fll->fout == Fout)
-		return 0;
-
 	ret = regmap_read(arizona->regmap, fll->base + 1, &reg);
 	if (ret != 0) {
 		arizona_fll_err(fll, "Failed to read current state: %d\n",
 				ret);
 		return ret;
 	}
-	ena = reg & ARIZONA_FLL1_ENA;
 
-	if (Fout) {
-		/* Do we have a 32kHz reference? */
-		regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val);
-		switch (val & ARIZONA_CLK_32K_SRC_MASK) {
-		case ARIZONA_CLK_SRC_MCLK1:
-		case ARIZONA_CLK_SRC_MCLK2:
-			syncsrc = val & ARIZONA_CLK_32K_SRC_MASK;
-			break;
-		default:
-			syncsrc = -1;
-		}
+	return reg & ARIZONA_FLL1_ENA;
+}
 
-		if (source == sync