Merge tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "The biggest change in this update is the unification of HD-audio codec
  parsers.  Now the HD-audio codec is parsed in a generic parser code
  which is invoked by each HD-audio codec driver.

  Some background information is found in David Henningsson's blog
  entry:

      http://voices.canonical.com/david.henningsson/2013/01/18/upcoming-changes-to-the-intel-hda-drivers/

  Other than that, some random updates/fixes like USB-audio and a bunch
  of small AoC updates as usual.

  Highlights:

   - Unification of HD-audio parser code (aka generic parser)

   - Support of new Intel HD-audio controller, new IDT codecs

   - Fixes for HD-audio HDMI audio hotplug

   - Haswell HDMI audio fixup

   - Support of Creative CA0132 DSP code

   - A few fixes of HDSP driver

   - USB-audio fix for Roland A-PRO, M-Audio FT C600

   - Support PM for aloop driver (and fixes Oops)

   - Compress API updates for gapless playback support

  For ASoC part:

   - Support for a wider range of hardware in the compressed stream code

   - The ability to mute capture streams as well as playback streams
     while inactive

   - DT support for AK4642, FSI, Samsung I2S and WM8962

   - AC'97 support for Tegra

   - New driver for max98090, replacing the stub which was there

   - A new driver from Dialog

  Note that due to dependencies, DTification of DMA support for Samsung
  platforms (used only by the and I2S driver and SPI) is merged here as
  well."

Fix up trivial conflict in drivers/spi/spi-s3c64xx.c due to removed code
being changed.

* tag 'sound-3.9' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (453 commits)
  ALSA: usb: Fix Processing Unit Descriptor parsers
  ALSA: hda - hdmi: Notify userspace when ELD control changes
  ALSA: hda - hdmi: Protect ELD buffer
  ALSA: hda - hdmi: Refactor hdmi_eld into parsed_hdmi_eld
  ALSA: hda - hdmi: Do not expose eld data when eld is invalid
  ALSA: hda - hdmi: ELD shouldn't be valid after unplug
  ALSA: hda - Fix the silent speaker output on Fujitsu S7020 laptop
  ALSA: hda - add quirks for mute LED on two HP machines
  ALSA: usb/quirks, fix out-of-bounds access
  ASoC: codecs: Add da7213 codec
  ALSA: au88x0 - Define channel map for au88x0
  ALSA: compress: add support for gapless playback
  ALSA: hda - Remove speaker clicks on CX20549
  ALSA: hda - Disable runtime PM for Intel 5 Series/3400
  ALSA: hda - Increase badness for missing multi-io
  ASoC: arizona: Automatically manage input mutes
  ALSA: hda - Fix broken workaround for HDMI/SPDIF conflicts
  ALSA: hda/ca0132 - Add missing \n to debug prints
  ALSA: hda/ca0132 - Fix type of INVALID_CHIP_ADDRESS
  ALSA: hda - update documentation for no-primary-hp fixup
  ...

17 files changed, 207 insertions, 104 deletions

diff --git a/include/linux/mfd/arizona/pdata.h b/include/linux/mfd/arizona/pdata.h
index 8b1d1daaae1..ec3e2a2a6d7 100644
--- a/include/linux/mfd/arizona/pdata.h
+++ b/include/linux/mfd/arizona/pdata.h
@@ -62,6 +62,8 @@
 
 #define ARIZONA_MAX_OUTPUT 6
 
+#define ARIZONA_MAX_AIF 3
+
 #define ARIZONA_HAP_ACT_ERM 0
 #define ARIZONA_HAP_ACT_LRA 2
 
@@ -96,6 +98,13 @@ struct arizona_pdata {
 	/** Pin state for GPIO pins */
 	int gpio_defaults[ARIZONA_MAX_GPIO];
 
+	/**
+	 * Maximum number of channels clocks will be generated for,
+	 * useful for systems where and I2S bus with multiple data
+	 * lines is mastered.
+	 */
+	int max_channels_clocked[ARIZONA_MAX_AIF];
+
 	/** GPIO for mic detection polarity */
 	int micd_pol_gpio;
 
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index f2912abacdf..ff6c74153fa 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -71,6 +71,8 @@ struct snd_compr_runtime {
  * @runtime: pointer to runtime structure
  * @device: device pointer
  * @direction: stream direction, playback/recording
+ * @metadata_set: metadata set flag, true when set
+ * @next_track: has userspace signall next track transistion, true when set
  * @private_data: pointer to DSP private data
  */
 struct snd_compr_stream {
@@ -79,6 +81,8 @@ struct snd_compr_stream {
 	struct snd_compr_runtime *runtime;
 	struct snd_compr *device;
 	enum snd_compr_direction direction;
+	bool metadata_set;
+	bool next_track;
 	void *private_data;
 };
 
@@ -110,6 +114,10 @@ struct snd_compr_ops {
 			struct snd_compr_params *params);
 	int (*get_params)(struct snd_compr_stream *stream,
 			struct snd_codec *params);
+	int (*set_metadata)(struct snd_compr_stream *stream,
+			struct snd_compr_metadata *metadata);
+	int (*get_metadata)(struct snd_compr_stream *stream,
+			struct snd_compr_metadata *metadata);
 	int (*trigger)(struct snd_compr_stream *stream, int cmd);
 	int (*pointer)(struct snd_compr_stream *stream,
 			struct snd_compr_tstamp *tstamp);
diff --git a/include/sound/core.h b/include/sound/core.h
index 93896ad1fcd..7cede2d6aa8 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -394,8 +394,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
 
 #else /* !CONFIG_SND_DEBUG */
 
-#define snd_printd(fmt, args...)	do { } while (0)
-#define _snd_printd(level, fmt, args...) do { } while (0)
+__printf(1, 2)
+static inline void snd_printd(const char *format, ...) {}
+__printf(2, 3)
+static inline void _snd_printd(int level, const char *format, ...) {}
+
 #define snd_BUG()			do { } while (0)
 static inline int __snd_bug_on(int cond)
 {
@@ -416,7 +419,8 @@ static inline int __snd_bug_on(int cond)
 #define snd_printdd(format, args...) \
 	__snd_printk(2, __FILE__, __LINE__, format, ##args)
 #else
-#define snd_printdd(format, args...)	do { } while (0)
+__printf(1, 2)
+static inline void snd_printdd(const char *format, ...) {}
 #endif
 
 
@@ -454,6 +458,7 @@ struct snd_pci_quirk {
 #define SND_PCI_QUIRK_MASK(vend, mask, dev, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev),			\
 			.value = (val), .name = (xname)}
+#define snd_pci_quirk_name(q)	((q)->name)
 #else
 #define SND_PCI_QUIRK(vend,dev,xname,val) \
 	{_SND_PCI_QUIRK_ID(vend, dev), .value = (val)}
@@ -461,6 +466,7 @@ struct snd_pci_quirk {
 	{_SND_PCI_QUIRK_ID_MASK(vend, mask, dev), .value = (val)}
 #define SND_PCI_QUIRK_VENDOR(vend, xname, val)			\
 	{_SND_PCI_QUIRK_ID_MASK(vend, 0, 0), .value = (val)}
+#define snd_pci_quirk_name(q)	""
 #endif
 
 const struct snd_pci_quirk *
diff --git a/include/sound/cs4271.h b/include/sound/cs4271.h
index dd8c48d14ed..70f45355aca 100644
--- a/include/sound/cs4271.h
+++ b/include/sound/cs4271.h
@@ -20,6 +20,21 @@
 struct cs4271_platform_data {
 	int gpio_nreset;	/* GPIO driving Reset pin, if any */
 	bool amutec_eq_bmutec;	/* flag to enable AMUTEC=BMUTEC */
+
+	/*
+	 * The CS4271 requires its LRCLK and MCLK to be stable before its RESET
+	 * line is de-asserted. That also means that clocks cannot be changed
+	 * without putting the chip back into hardware reset, which also requires
+	 * a complete re-initialization of all registers.
+	 *
+	 * One (undocumented) workaround is to assert and de-assert the PDN bit
+	 * in the MODE2 register. This workaround can be enabled with the
+	 * following flag.
+	 *
+	 * Note that this is not needed in case the clocks are stable
+	 * throughout the entire runtime of the codec.
+	 */
+	bool enable_soft_reset;
 };
 
 #endif /* __CS4271_H */
diff --git a/include/sound/da7213.h b/include/sound/da7213.h
new file mode 100644
index 00000000000..673f5c39cbf
--- /dev/null
+++ b/include/sound/da7213.h
@@ -0,0 +1,52 @@
+/*
+ * da7213.h - DA7213 ASoC Codec Driver Platform Data
+ *
+ * Copyright (c) 2013 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _DA7213_PDATA_H
+#define _DA7213_PDATA_H
+
+enum da7213_micbias_voltage {
+	DA7213_MICBIAS_1_6V = 0,
+	DA7213_MICBIAS_2_2V = 1,
+	DA7213_MICBIAS_2_5V = 2,
+	DA7213_MICBIAS_3_0V = 3,
+};
+
+enum da7213_dmic_data_sel {
+	DA7213_DMIC_DATA_LRISE_RFALL = 0,
+	DA7213_DMIC_DATA_LFALL_RRISE = 1,
+};
+
+enum da7213_dmic_samplephase {
+	DA7213_DMIC_SAMPLE_ON_CLKEDGE = 0,
+	DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE = 1,
+};
+
+enum da7213_dmic_clk_rate {
+	DA7213_DMIC_CLK_3_0MHZ = 0,
+	DA7213_DMIC_CLK_1_5MHZ = 1,
+};
+
+struct da7213_platform_data {
+	/* Mic Bias voltage */
+	enum da7213_micbias_voltage micbias1_lvl;
+	enum da7213_micbias_voltage micbias2_lvl;
+
+	/* DMIC config */
+	enum da7213_dmic_data_sel dmic_data_sel;
+	enum da7213_dmic_samplephase dmic_samplephase;
+	enum da7213_dmic_clk_rate dmic_clk_rate;
+
+	/* MCLK squaring config */
+	bool mclk_squaring;
+};
+
+#endif /* _DA7213_PDATA_H */
diff --git a/include/sound/max98090.h b/include/sound/max98090.h
new file mode 100755
index 00000000000..95efb13f847
--- /dev/null
+++ b/include/sound/max98090.h
@@ -0,0 +1,29 @@
+/*
+ * Platform data for MAX98090
+ *
+ * Copyright 2011-2012 Maxim Integrated Products
+ *
+ *  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 __SOUND_MAX98090_PDATA_H__
+#define __SOUND_MAX98090_PDATA_H__
+
+/* codec platform data */
+struct max98090_pdata {
+
+	/* Analog/digital microphone configuration:
+	 * 0 = analog microphone input (normal setting)
+	 * 1 = digital microphone input
+	 */
+	unsigned int digmic_left_mode:1;
+	unsigned int digmic_right_mode:1;
+	unsigned int digmic_3_mode:1;
+	unsigned int digmic_4_mode:1;
+};
+
+#endif
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 844af65af62..cf15b8213df 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -37,7 +37,7 @@ struct snd_dma_device {
 #ifndef snd_dma_pci_data
 #define snd_dma_pci_data(pci)	(&(pci)->dev)
 #define snd_dma_isa_data()	NULL
-#define snd_dma_continuous_data(x)	((struct device *)(unsigned long)(x))
+#define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
 #endif
 
 
diff --git a/include/sound/saif.h b/include/sound/saif.h
deleted file mode 100644
index f22f3e16edf..00000000000
--- a/include/sound/saif.h
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved.
- *
- * This program is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License version 2 as
- * published by the Free Software Foundation.
- */
-
-#ifndef __SOUND_SAIF_H__
-#define __SOUND_SAIF_H__
-
-struct mxs_saif_platform_data {
-	bool master_mode;	/* if true use master mode */
-	int master_id;		/* id of the master if in slave mode */
-};
-#endif
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index cc1c919c643..7a9710b4b79 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -11,82 +11,20 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
-
-#define FSI_PORT_A	0
-#define FSI_PORT_B	1
-
 #include <linux/clk.h>
 #include <sound/soc.h>
 
 /*
- * flags format
- *
- * 0x00000CBA
- *
- * A:  inversion
- * B:  format mode
- * C:  chip specific
- * D:  clock selecter if master mode
+ * flags
  */
-
-/* A: clock inversion */
-#define SH_FSI_INVERSION_MASK	0x0000000F
-#define SH_FSI_LRM_INV		(1 << 0)
-#define SH_FSI_BRM_INV		(1 << 1)
-#define SH_FSI_LRS_INV		(1 << 2)
-#define SH_FSI_BRS_INV		(1 << 3)
-
-/* B: format mode */
-#define SH_FSI_FMT_MASK		0x000000F0
-#define SH_FSI_FMT_DAI		(0 << 4)
-#define SH_FSI_FMT_SPDIF	(1 << 4)
-
-/* C: chip specific */
-#define SH_FSI_OPTION_MASK	0x00000F00
-#define SH_FSI_ENABLE_STREAM_MODE	(1 << 8) /* for 16bit data */
-
-/* D:  clock selecter if master mode */
-#define SH_FSI_CLK_MASK		0x0000F000
-#define SH_FSI_CLK_EXTERNAL	(0 << 12)
-#define SH_FSI_CLK_CPG		(1 << 12) /* FSIxCK + FSI-DIV */
-
-/*
- * set_rate return value
- *
- * see ACKMD/BPFMD on
- *     ACK_MD (FSI2)
- *     CKG1   (FSI)
- *
- * err		: return value <  0
- * no change	: return value == 0
- * change xMD	: return value >  0
- *
- * 0x-00000AB
- *
- * A:  ACKMD value
- * B:  BPFMD value
- */
-
-#define SH_FSI_ACKMD_MASK	(0xF << 0)
-#define SH_FSI_ACKMD_512	(1 << 0)
-#define SH_FSI_ACKMD_256	(2 << 0)
-#define SH_FSI_ACKMD_128	(3 << 0)
-#define SH_FSI_ACKMD_64		(4 << 0)
-#define SH_FSI_ACKMD_32		(5 << 0)
-
-#define SH_FSI_BPFMD_MASK	(0xF << 4)
-#define SH_FSI_BPFMD_512	(1 << 4)
-#define SH_FSI_BPFMD_256	(2 << 4)
-#define SH_FSI_BPFMD_128	(3 << 4)
-#define SH_FSI_BPFMD_64		(4 << 4)
-#define SH_FSI_BPFMD_32		(5 << 4)
-#define SH_FSI_BPFMD_16		(6 << 4)
+#define SH_FSI_FMT_SPDIF		(1 << 0) /* spdif for HDMI */
+#define SH_FSI_ENABLE_STREAM_MODE	(1 << 1) /* for 16bit data */
+#define SH_FSI_CLK_CPG			(1 << 2) /* FSIxCK + FSI-DIV */
 
 struct sh_fsi_port_info {
 	unsigned long flags;
 	int tx_id;
 	int rx_id;
-	int (*set_rate)(struct device *dev, int rate, int enable);
 };
 
 struct sh_fsi_platform_info {
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
index 4b62b8dc6a4..6c74527d492 100644
--- a/include/sound/simple_card.h
+++ b/include/sound/simple_card.h
@@ -14,21 +14,21 @@
 
 #include <sound/soc.h>
 
-struct asoc_simple_dai_init_info {
+struct asoc_simple_dai {
+	const char *name;
 	unsigned int fmt;
-	unsigned int cpu_daifmt;
-	unsigned int codec_daifmt;
 	unsigned int sysclk;
 };
 
 struct asoc_simple_card_info {
 	const char *name;
 	const char *card;
-	const char *cpu_dai;
 	const char *codec;
 	const char *platform;
-	const char *codec_dai;
-	struct asoc_simple_dai_init_info *init; /* for snd_link.init */
+
+	unsigned int daifmt;
+	struct asoc_simple_dai cpu_dai;
+	struct asoc_simple_dai codec_dai;
 
 	/* used in simple-card.c */
 	struct snd_soc_dai_link snd_link;
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 3953cea0ecf..3d84808952b 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -45,7 +45,7 @@ struct snd_compr_stream;
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(2 << 4) /* clock is gated */
+#define SND_SOC_DAIFMT_GATED		(0 << 4) /* clock is gated */
 
 /*
  * DAI hardware signal inversions.
@@ -53,7 +53,7 @@ struct snd_compr_stream;
  * Specifies whether the DAI can also support inverted clocks for the specified
  * format.
  */
-#define SND_SOC_DAIFMT_NB_NF		(1 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
 #define SND_SOC_DAIFMT_NB_IF		(2 << 8) /* normal BCLK + inv FRM */
 #define SND_SOC_DAIFMT_IB_NF		(3 << 8) /* invert BCLK + nor FRM */
 #define SND_SOC_DAIFMT_IB_IF		(4 << 8) /* invert BCLK + FRM */
@@ -126,7 +126,8 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 
 /* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute,
+			     int direction);
 
 struct snd_soc_dai_ops {
 	/*
@@ -157,6 +158,7 @@ struct snd_soc_dai_ops {
 	 * Called by soc-core to minimise any pops.
 	 */
 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+	int (*mute_stream)(struct snd_soc_dai *dai, int mute, int stream);
 
 	/*
 	 * ALSA PCM audio operations - all optional.
diff --git a/include/sound/soc.h b/include/sound/soc.h
index bc56738cb10..a6a059ca387 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -906,8 +906,8 @@ struct snd_soc_dai_link {
 			struct snd_pcm_hw_params *params);
 
 	/* machine stream operations */
-	struct snd_soc_ops *ops;
-	struct snd_soc_compr_ops *compr_ops;
+	const struct snd_soc_ops *ops;
+	const struct snd_soc_compr_ops *compr_ops;
 };
 
 struct snd_soc_codec_conf {
@@ -1171,6 +1171,8 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card,
 			       const char *propname);
 int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 				   const char *propname);
+unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
+				     const char *prefix);
 
 #include <sound/soc-dai.h>
 
diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h
index ffd9bc79310..9407fd00363 100644
--- a/include/sound/tlv320aic3x.h
+++ b/include/sound/tlv320aic3x.h
@@ -46,6 +46,13 @@ enum {
 	AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ	= 15
 };
 
+enum aic3x_micbias_voltage {
+	AIC3X_MICBIAS_OFF = 0,
+	AIC3X_MICBIAS_2_0V = 1,
+	AIC3X_MICBIAS_2_5V = 2,
+	AIC3X_MICBIAS_AVDDV = 3,
+};
+
 struct aic3x_setup_data {
 	unsigned int gpio_func[2];
 };
@@ -53,6 +60,9 @@ struct aic3x_setup_data {
 struct aic3x_pdata {
 	int gpio_reset; /* < 0 if not used */
 	struct aic3x_setup_data *setup;
+
+	/* Selects the micbias voltage */
+	enum aic3x_micbias_voltage micbias_vg;
 };
 
 #endif
diff --git a/include/sound/wm2000.h b/include/sound/wm2000.h
index aa388ca9ec6..4de81f41c90 100644
--- a/include/sound/wm2000.h
+++ b/include/sound/wm2000.h
@@ -15,9 +15,6 @@ struct wm2000_platform_data {
 	/** Filename for system-specific image to download to device. */
 	const char *download_file;
 
-	/** Divide MCLK by 2 for system clock? */
-	unsigned int mclkdiv2:1;
-
 	/** Disable speech clarity enhancement, for use when an
 	 * external algorithm is used. */
 	unsigned int speech_enh_disable:1;
diff --git a/include/sound/wm2200.h b/include/sound/wm2200.h
index 79bf55be7ff..bc7ab1a4b48 100644
--- a/include/sound/wm2200.h
+++ b/include/sound/wm2200.h
@@ -12,6 +12,7 @@
 #define __LINUX_SND_WM2200_H
 
 #define WM2200_GPIO_SET 0x10000
+#define WM2200_MAX_MICBIAS 2
 
 enum wm2200_in_mode {
 	WM2200_IN_SE = 0,
@@ -25,6 +26,24 @@ enum wm2200_dmic_sup {
 	WM2200_DMIC_SUP_MICBIAS2 = 2,
 };
 
+enum wm2200_mbias_lvl {
+	WM2200_MBIAS_LVL_1V5 = 1,
+	WM2200_MBIAS_LVL_1V8 = 2,
+	WM2200_MBIAS_LVL_1V9 = 3,
+	WM2200_MBIAS_LVL_2V0 = 4,
+	WM2200_MBIAS_LVL_2V2 = 5,
+	WM2200_MBIAS_LVL_2V4 = 6,
+	WM2200_MBIAS_LVL_2V5 = 7,
+	WM2200_MBIAS_LVL_2V6 = 8,
+};
+
+struct wm2200_micbias {
+	enum wm2200_mbias_lvl mb_lvl;      /** Regulated voltage */
+	unsigned int discharge:1;          /** Actively discharge */
+	unsigned int fast_start:1;         /** Enable aggressive startup ramp rate */
+	unsigned int bypass:1;             /** Use bypass mode */
+};
+
 struct wm2200_pdata {
 	int reset;      /** GPIO controlling /RESET, if any */
 	int ldo_ena;    /** GPIO controlling LODENA, if any */
@@ -35,7 +54,8 @@ struct wm2200_pdata {
 	enum wm2200_in_mode in_mode[3];
 	enum wm2200_dmic_sup dmic_sup[3];
 
-	int micbias_cfg[2];  /** Register value to configure MICBIAS */
+	/** MICBIAS configurations */
+	struct wm2200_micbias micbias[WM2200_MAX_MICBIAS];
 };
 
 #endif
diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h
index ac90037894d..d2314be4f0c 100644
--- a/include/uapi/linux/usb/audio.h
+++ b/include/uapi/linux/usb/audio.h
@@ -384,14 +384,16 @@ static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_de
 						   int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return desc->baSourceID[desc->bNrInPins + control_size];
+	return *(uac_processing_unit_bmControls(desc, protocol)
+			+ control_size);
 }
 
 static inline __u8 *uac_processing_unit_specific(struct uac_processing_unit_descriptor *desc,
 						 int protocol)
 {
 	__u8 control_size = uac_processing_unit_bControlSize(desc, protocol);
-	return &desc->baSourceID[desc->bNrInPins + control_size + 1];
+	return uac_processing_unit_bmControls(desc, protocol)
+			+ control_size + 1;
 }
 
 /* 4.5.2 Class-Specific AS Interface Descriptor */
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 05341a43fed..d630163b9a2 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -30,7 +30,7 @@
 #include <sound/compress_params.h>
 
 
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 0)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 1, 1)
 /**
  * struct snd_compressed_buffer: compressed buffer
  * @fragment_size: size of buffer fragment in bytes
@@ -122,6 +122,27 @@ struct snd_compr_codec_caps {
 };
 
 /**
+ * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the
+ * end of the track
+ * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the
+ * beginning of the track
+ */
+enum {
+	SNDRV_COMPRESS_ENCODER_PADDING = 1,
+	SNDRV_COMPRESS_ENCODER_DELAY = 2,
+};
+
+/**
+ * struct snd_compr_metadata: compressed stream metadata
+ * @key: key id
+ * @value: key value
+ */
+struct snd_compr_metadata {
+	 __u32 key;
+	 __u32 value[8];
+};
+
+/**
  * compress path ioctl definitions
  * SNDRV_COMPRESS_GET_CAPS: Query capability of DSP
  * SNDRV_COMPRESS_GET_CODEC_CAPS: Query capability of a codec
@@ -145,6 +166,10 @@ struct snd_compr_codec_caps {
 						struct snd_compr_codec_caps)
 #define SNDRV_COMPRESS_SET_PARAMS	_IOW('C', 0x12, struct snd_compr_params)
 #define SNDRV_COMPRESS_GET_PARAMS	_IOR('C', 0x13, struct snd_codec)
+#define SNDRV_COMPRESS_SET_METADATA	_IOW('C', 0x14,\
+						 struct snd_compr_metadata)
+#define SNDRV_COMPRESS_GET_METADATA	_IOWR('C', 0x15,\
+						 struct snd_compr_metadata)
 #define SNDRV_COMPRESS_TSTAMP		_IOR('C', 0x20, struct snd_compr_tstamp)
 #define SNDRV_COMPRESS_AVAIL		_IOR('C', 0x21, struct snd_compr_avail)
 #define SNDRV_COMPRESS_PAUSE		_IO('C', 0x30)
@@ -152,10 +177,14 @@ struct snd_compr_codec_caps {
 #define SNDRV_COMPRESS_START		_IO('C', 0x32)
 #define SNDRV_COMPRESS_STOP		_IO('C', 0x33)
 #define SNDRV_COMPRESS_DRAIN		_IO('C', 0x34)
+#define SNDRV_COMPRESS_NEXT_TRACK	_IO('C', 0x35)
+#define SNDRV_COMPRESS_PARTIAL_DRAIN	_IO('C', 0x36)
 /*
  * TODO
  * 1. add mmap support
  *
  */
 #define SND_COMPR_TRIGGER_DRAIN 7 /*FIXME move this to pcm.h */
+#define SND_COMPR_TRIGGER_NEXT_TRACK 8
+#define SND_COMPR_TRIGGER_PARTIAL_DRAIN 9
 #endif