Merge tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This is the first big chunk for 3.5 merges of sound stuff.

  There are a few big changes in different areas.  First off, the
  streaming logic of USB-audio endpoints has been largely rewritten for
  the better support of "implicit feedback".  If anything about USB got
  broken, this change has to be checked.

  For HD-audio, the resume procedure was changed; instead of delaying
  the resume of the hardware until the first use, now waking up
  immediately at resume.  This is for buggy BIOS.

  For ASoC, dynamic PCM support and the improved support for digital
  links between off-SoC devices are major framework changes.

  Some highlights are below:

  * HD-audio
   - Avoid accesses of invalid pin-control bits that may stall the codec
   - V-ref setup cleanups
   - Fix the races in power-saving code
   - Fix the races in codec cache hashes and connection lists
   - Split some common codes for BIOS auto-parser to hda_auto_parser.c
   - Changed the PM resume code to wake up immediately for buggy BIOS
   - Creative SoundCore3D support
   - Add Conexant CX20751/2/3/4 codec support

  * ASoC
   - Dynamic PCM support, allowing support for SoCs with internal
     routing through components with tight sequencing and formatting
     constraints within their internal paths or where there are multiple
     components connected with CPU managed DMA controllers inside the
     SoC.
   - Greatly improved support for direct digital links between off-SoC
     devices, providing a much simpler way of connecting things like
     digital basebands to CODECs.
   - Much more fine grained and robust locking, cleaning up some of the
     confusion that crept in with multi-component.
   - CPU support for nVidia Tegra 30 I2S and audio hub controllers and
     ST-Ericsson MSP I2S controolers
   - New CODEC drivers for Cirrus CS42L52, LAPIS Semiconductor ML26124,
     Texas Instruments LM49453.
   - Some regmap changes needed by the Tegra I2S driver.
   - mc13783 audio support.

  * Misc
   - Rewrite with module_pci_driver()
   - Xonar DGX support for snd-oxygen
   - Improvement of packet handling in snd-firewire driver
   - New USB-endpoint streaming logic
   - Enhanced M-audio FTU quirks and relevant cleanups
   - Increment the support of OSS devices to 256
   - snd-aloop accuracy improvement

  There are a few more pending changes for 3.5, but they will be sent
  slightly later as partly depending on the changes of DRM."

Fix up conflicts in regmap (due to duplicate patches, with some further
updates then having already come in from the regmap tree).  Also some
fairly trivial context conflicts in the imx and mcx soc drivers.

* tag 'sound-3.5' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (280 commits)
  ALSA: snd-usb: fix stream info output in /proc
  ALSA: pcm - Add proper state checks to snd_pcm_drain()
  ALSA: sh: Fix up namespace collision in sh_dac_audio.
  ALSA: hda/realtek - Fix unused variable compile warning
  ASoC: sh: fsi: enable chip specific data transfer mode
  ASoC: sh: fsi: call fsi_hw_startup/shutdown from fsi_dai_trigger()
  ASoC: sh: fsi: use same format for IN/OUT
  ASoC: sh: fsi: add fsi_version() and removed meaningless version check
  ASoC: sh: fsi: use register field macro name on IN/OUT_DMAC
  ASoC: tegra: Add machine driver for WM8753 codec
  ALSA: hda - Fix possible races of accesses to connection list array
  ASoC: OMAP: HDMI: Introduce codec
  ARM: mx31_3ds: Add sound support
  ASoC: imx-mc13783 cleanup
  mx31moboard: Add sound support
  ASoC: mc13783 codec cleanups
  ASoC: add imx-mc13783 sound support
  ASoC: Add mc13783 codec
  mfd: mc13xxx: add codec platform data
  ASoC: don't flip master of DT-instantiated DAI links
  ...

12 files changed, 496 insertions, 42 deletions

diff --git a/include/linux/mfd/mc13xxx.h b/include/linux/mfd/mc13xxx.h
index 10e038bac8dd..bf070755982e 100644
--- a/include/linux/mfd/mc13xxx.h
+++ b/include/linux/mfd/mc13xxx.h
@@ -170,6 +170,16 @@ struct mc13xxx_ts_platform_data {
 	bool atox;
 };
 
+enum mc13783_ssi_port {
+	MC13783_SSI1_PORT,
+	MC13783_SSI2_PORT,
+};
+
+struct mc13xxx_codec_platform_data {
+	enum mc13783_ssi_port adc_ssi_port;
+	enum mc13783_ssi_port dac_ssi_port;
+};
+
 struct mc13xxx_platform_data {
 #define MC13XXX_USE_TOUCHSCREEN (1 << 0)
 #define MC13XXX_USE_CODEC	(1 << 1)
@@ -181,6 +191,7 @@ struct mc13xxx_platform_data {
 	struct mc13xxx_leds_platform_data *leds;
 	struct mc13xxx_buttons_platform_data *buttons;
 	struct mc13xxx_ts_platform_data touch;
+	struct mc13xxx_codec_platform_data *codec;
 };
 
 #define MC13XXX_ADC_MODE_TS		1
diff --git a/include/sound/asound.h b/include/sound/asound.h
index a2e4ff5ba9e9..0876a1e76aef 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -70,6 +70,20 @@ struct snd_aes_iec958 {
 
 /****************************************************************************
  *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+
+struct snd_cea_861_aud_if {
+	unsigned char db1_ct_cc; /* coding type and channel count */
+	unsigned char db2_sf_ss; /* sample frequency and size */
+	unsigned char db3; /* not used, all zeros */
+	unsigned char db4_ca; /* channel allocation code */
+	unsigned char db5_dminh_lsv; /* downmix inhibit & level-shit values */
+};
+
+/****************************************************************************
+ *                                                                          *
  *      Section for driver hardware dependent interface - /dev/snd/hw?      *
  *                                                                          *
  ****************************************************************************/
diff --git a/include/sound/asoundef.h b/include/sound/asoundef.h
index 20ebf3298eba..bb05c02f89b0 100644
--- a/include/sound/asoundef.h
+++ b/include/sound/asoundef.h
@@ -170,6 +170,47 @@
 #define IEC958_AES5_CON_CGMSA_COPYNOMORE (2<<0)	/* condition not be used */
 #define IEC958_AES5_CON_CGMSA_COPYNEVER	(3<<0)	/* no copying is permitted */
 
+/****************************************************************************
+ *                                                                          *
+ *        CEA-861 Audio InfoFrame. Used in HDMI and DisplayPort		    *
+ *                                                                          *
+ ****************************************************************************/
+#define CEA861_AUDIO_INFOFRAME_DB1CC		(7<<0) /* mask - channel count */
+#define CEA861_AUDIO_INFOFRAME_DB1CT		(0xf<<4) /* mask - coding type */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM (0<<4) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_IEC60958	(1<<4) /* IEC-60958 L-PCM */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AC3	(2<<4) /* AC-3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG1	(3<<4) /* MPEG1 Layers 1 & 2 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MP3	(4<<4) /* MPEG1 Layer 3 */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MPEG2_MULTICH (5<<4) /* MPEG2 Multichannel */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_AAC	(6<<4) /* AAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS	(7<<4) /* DTS */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ATRAC	(8<<4) /* ATRAC */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_ONEBIT	(9<<4) /* One Bit Audio */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DOLBY_DIG_PLUS (10<<4) /* Dolby Digital + */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DTS_HD	(11<<4) /* DTS-HD */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_MAT	(12<<4) /* MAT (MLP) */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_DST	(13<<4) /* DST */
+#define CEA861_AUDIO_INFOFRAME_DB1CT_WMA_PRO	(14<<4) /* WMA Pro */
+#define CEA861_AUDIO_INFOFRAME_DB2SF		(7<<2) /* mask - sample frequency */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM (0<<2) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_32000	(1<<2) /* 32kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_44100	(2<<2) /* 44.1kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_48000	(3<<2) /* 48kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_88200	(4<<2) /* 88.2kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_96000	(5<<2) /* 96kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_176400	(6<<2) /* 176.4kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SF_192000	(7<<2) /* 192kHz */
+#define CEA861_AUDIO_INFOFRAME_DB2SS		(3<<0) /* mask - sample size */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM (0<<0) /* refer to stream */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_16BIT	(1<<0) /* 16 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_20BIT	(2<<0) /* 20 bits */
+#define CEA861_AUDIO_INFOFRAME_DB2SS_24BIT	(3<<0) /* 24 bits */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH	(1<<7) /* mask - inhibit downmixing */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED (0<<7) /* stereo downmix permitted */
+#define CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED (1<<7) /* stereo downmis prohibited */
+#define CEA861_AUDIO_INFOFRAME_DB5_LSV		(0xf<<3) /* mask - level-shift values */
+
 /*****************************************************************************
  *                                                                           *
  *                            MIDI v1.0 interface                            *
diff --git a/include/sound/cs42l52.h b/include/sound/cs42l52.h
new file mode 100644
index 000000000000..4c68955f7330
--- /dev/null
+++ b/include/sound/cs42l52.h
@@ -0,0 +1,36 @@
+/*
+ * linux/sound/cs42l52.h -- Platform data for CS42L52
+ *
+ * Copyright (c) 2012 Cirrus Logic Inc.
+ *
+ * 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 __CS42L52_H
+#define __CS42L52_H
+
+struct cs42l52_platform_data {
+
+	/* MICBIAS Level. Check datasheet Pg48 */
+	unsigned int micbias_lvl;
+
+	/* MICA mode selection 0=Single 1=Differential */
+	unsigned int mica_cfg;
+
+	/* MICB mode selection 0=Single 1=Differential */
+	unsigned int micb_cfg;
+
+	/* MICA Select 0=MIC1A 1=MIC2A */
+	unsigned int mica_sel;
+
+	/* MICB Select 0=MIC2A 1=MIC2B */
+	unsigned int micb_sel;
+
+	/* Charge Pump Freq. Check datasheet Pg73 */
+	unsigned int chgfreq;
+
+};
+
+#endif /* __CS42L52_H */
diff --git a/include/sound/max98095.h b/include/sound/max98095.h
index 7513a42dd4aa..e87ae67b0a55 100644
--- a/include/sound/max98095.h
+++ b/include/sound/max98095.h
@@ -49,6 +49,18 @@ struct max98095_pdata {
 	 */
 	unsigned int digmic_left_mode:1;
 	unsigned int digmic_right_mode:1;
+
+	/* Pin5 is the mechanical method of sensing jack insertion
+	 * but it is something that might not be supported.
+	 * 0 = PIN5 not supported
+	 * 1 = PIN5 supported
+	 */
+	unsigned int jack_detect_pin5en:1;
+
+	/* Slew amount for jack detection. Calculated as 4 * (delay + 1).
+	 * Default delay is 24 to get a time of 100ms.
+	 */
+	unsigned int jack_detect_delay;
 };
 
 #endif
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
index b457e87fbd08..906010344dd7 100644
--- a/include/sound/sh_fsi.h
+++ b/include/sound/sh_fsi.h
@@ -21,10 +21,11 @@
 /*
  * flags format
  *
- * 0x000000BA
+ * 0x00000CBA
  *
  * A:  inversion
  * B:  format mode
+ * C:  chip specific
  */
 
 /* A: clock inversion */
@@ -39,6 +40,9 @@
 #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 */
 
 /*
  * set_rate return value
@@ -84,16 +88,4 @@ struct sh_fsi_platform_info {
 	struct sh_fsi_port_info port_b;
 };
 
-/*
- * for fsi-ak4642
- */
-struct fsi_ak4642_info {
-	const char *name;
-	const char *card;
-	const char *cpu_dai;
-	const char *codec;
-	const char *platform;
-	int id;
-};
-
 #endif /* __SOUND_FSI_H */
diff --git a/include/sound/simple_card.h b/include/sound/simple_card.h
new file mode 100644
index 000000000000..4b62b8dc6a4f
--- /dev/null
+++ b/include/sound/simple_card.h
@@ -0,0 +1,38 @@
+/*
+ * ASoC simple sound card support
+ *
+ * Copyright (C) 2012 Renesas Solutions Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.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 __SIMPLE_CARD_H
+#define __SIMPLE_CARD_H
+
+#include <sound/soc.h>
+
+struct asoc_simple_dai_init_info {
+	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 */
+
+	/* used in simple-card.c */
+	struct snd_soc_dai_link snd_link;
+	struct snd_soc_card snd_card;
+};
+
+#endif /* __SIMPLE_CARD_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index c429f248cf4e..1f69e0af2941 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -173,6 +173,8 @@ struct snd_soc_dai_ops {
 		struct snd_soc_dai *);
 	int (*trigger)(struct snd_pcm_substream *, int,
 		struct snd_soc_dai *);
+	int (*bespoke_trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
 	/*
 	 * For hardware based FIFO caused delay reporting.
 	 * Optional.
@@ -196,6 +198,7 @@ struct snd_soc_dai_driver {
 	const char *name;
 	unsigned int id;
 	int ac97_control;
+	unsigned int base;
 
 	/* DAI driver callbacks */
 	int (*probe)(struct snd_soc_dai *dai);
@@ -241,6 +244,7 @@ struct snd_soc_dai {
 
 	struct snd_soc_dapm_widget *playback_widget;
 	struct snd_soc_dapm_widget *capture_widget;
+	struct snd_soc_dapm_context dapm;
 
 	/* DAI DMA data */
 	void *playback_dma_data;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 8da3c2409060..e3833d9f1914 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -141,10 +141,6 @@ struct device;
 {       .id = snd_soc_dapm_mixer, .name = wname, .reg = wreg, .shift = wshift, \
 	.invert = winvert, .kcontrol_news = wcontrols, \
 	.num_kcontrols = wncontrols, .event = wevent, .event_flags = wflags}
-#define SND_SOC_DAPM_MICBIAS_E(wname, wreg, wshift, winvert, wevent, wflags) \
-{	.id = snd_soc_dapm_micbias, .name = wname, .reg = wreg, .shift = wshift, \
-	.invert = winvert, .kcontrol_news = NULL, .num_kcontrols = 0, \
-	.event = wevent, .event_flags = wflags}
 #define SND_SOC_DAPM_SWITCH_E(wname, wreg, wshift, winvert, wcontrols, \
 	wevent, wflags) \
 {	.id = snd_soc_dapm_switch, .name = wname, .reg = wreg, .shift = wshift, \
@@ -324,6 +320,8 @@ struct snd_soc_dapm_path;
 struct snd_soc_dapm_pin;
 struct snd_soc_dapm_route;
 struct snd_soc_dapm_context;
+struct regulator;
+struct snd_soc_dapm_widget_list;
 
 int dapm_reg_event(struct snd_soc_dapm_widget *w,
 		   struct snd_kcontrol *kcontrol, int event);
@@ -359,6 +357,10 @@ int snd_soc_dapm_new_controls(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 				 struct snd_soc_dai *dai);
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
+int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
+			 const struct snd_soc_pcm_stream *params,
+			 struct snd_soc_dapm_widget *source,
+			 struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
@@ -369,8 +371,8 @@ int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm,
 			     const struct snd_soc_dapm_route *route, int num);
 
 /* dapm events */
-int snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
-			      struct snd_soc_dai *dai, int event);
+void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream,
+	int event);
 void snd_soc_dapm_shutdown(struct snd_soc_card *card);
 
 /* external DAPM widget events */
@@ -402,6 +404,10 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
 /* Mostly internal - should not normally be used */
 void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
 
+/* dapm path query */
+int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
+	struct snd_soc_dapm_widget_list **list);
+
 /* dapm widget types */
 enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
@@ -430,6 +436,12 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_aif_out,		/* audio interface output */
 	snd_soc_dapm_siggen,		/* signal generator */
 	snd_soc_dapm_dai,		/* link to DAI structure */
+	snd_soc_dapm_dai_link,		/* link between two DAI structures */
+};
+
+enum snd_soc_dapm_subclass {
+	SND_SOC_DAPM_CLASS_INIT		= 0,
+	SND_SOC_DAPM_CLASS_RUNTIME	= 1,
 };
 
 /*
@@ -482,9 +494,11 @@ struct snd_soc_dapm_widget {
 	struct snd_soc_dapm_context *dapm;
 
 	void *priv;				/* widget specific data */
+	struct regulator *regulator;		/* attached regulator */
+	const struct snd_soc_pcm_stream *params; /* params for dai links */
 
 	/* dapm control */
-	short reg;						/* negative reg = no direct dapm */
+	int reg;				/* negative reg = no direct dapm */
 	unsigned char shift;			/* bits to shift */
 	unsigned int saved_value;		/* widget saved value */
 	unsigned int value;				/* widget current value */
diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h
new file mode 100644
index 000000000000..04598f1efd77
--- /dev/null
+++ b/include/sound/soc-dpcm.h
@@ -0,0 +1,138 @@
+/*
+ * linux/sound/soc-dpcm.h -- ALSA SoC Dynamic PCM Support
+ *
+ * Author:		Liam Girdwood <lrg@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.
+ */
+
+#ifndef __LINUX_SND_SOC_DPCM_H
+#define __LINUX_SND_SOC_DPCM_H
+
+#include <linux/list.h>
+#include <sound/pcm.h>
+
+struct snd_soc_pcm_runtime;
+
+/*
+ * Types of runtime_update to perform. e.g. originated from FE PCM ops
+ * or audio route changes triggered by muxes/mixers.
+ */
+enum snd_soc_dpcm_update {
+	SND_SOC_DPCM_UPDATE_NO	= 0,
+	SND_SOC_DPCM_UPDATE_BE,
+	SND_SOC_DPCM_UPDATE_FE,
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link management states.
+ */
+enum snd_soc_dpcm_link_state {
+	SND_SOC_DPCM_LINK_STATE_NEW	= 0,	/* newly created link */
+	SND_SOC_DPCM_LINK_STATE_FREE,		/* link to be dismantled */
+};
+
+/*
+ * Dynamic PCM Frontend -> Backend link PCM states.
+ */
+enum snd_soc_dpcm_state {
+	SND_SOC_DPCM_STATE_NEW	= 0,
+	SND_SOC_DPCM_STATE_OPEN,
+	SND_SOC_DPCM_STATE_HW_PARAMS,
+	SND_SOC_DPCM_STATE_PREPARE,
+	SND_SOC_DPCM_STATE_START,
+	SND_SOC_DPCM_STATE_STOP,
+	SND_SOC_DPCM_STATE_PAUSED,
+	SND_SOC_DPCM_STATE_SUSPEND,
+	SND_SOC_DPCM_STATE_HW_FREE,
+	SND_SOC_DPCM_STATE_CLOSE,
+};
+
+/*
+ * Dynamic PCM trigger ordering. Triggering flexibility is required as some
+ * DSPs require triggering before/after their CPU platform and DAIs.
+ *
+ * i.e. some clients may want to manually order this call in their PCM
+ * trigger() whilst others will just use the regular core ordering.
+ */
+enum snd_soc_dpcm_trigger {
+	SND_SOC_DPCM_TRIGGER_PRE		= 0,
+	SND_SOC_DPCM_TRIGGER_POST,
+	SND_SOC_DPCM_TRIGGER_BESPOKE,
+};
+
+/*
+ * Dynamic PCM link
+ * This links together a FE and BE DAI at runtime and stores the link
+ * state information and the hw_params configuration.
+ */
+struct snd_soc_dpcm {
+	/* FE and BE DAIs*/
+	struct snd_soc_pcm_runtime *be;
+	struct snd_soc_pcm_runtime *fe;
+
+	/* link state */
+	enum snd_soc_dpcm_link_state state;
+
+	/* list of BE and FE for this DPCM link */
+	struct list_head list_be;
+	struct list_head list_fe;
+
+	/* hw params for this link - may be different for each link */
+	struct snd_pcm_hw_params hw_params;
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_state;
+#endif
+};
+
+/*
+ * Dynamic PCM runtime data.
+ */
+struct snd_soc_dpcm_runtime {
+	struct list_head be_clients;
+	struct list_head fe_clients;
+
+	int users;
+	struct snd_pcm_runtime *runtime;
+	struct snd_pcm_hw_params hw_params;
+
+	/* state and update */
+	enum snd_soc_dpcm_update runtime_update;
+	enum snd_soc_dpcm_state state;
+};
+
+/* can this BE stop and free */
+int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* can this BE perform a hw_params() */
+int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* is the current PCM operation for this FE ? */
+int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream);
+
+/* is the current PCM operation for this BE ? */
+int snd_soc_dpcm_be_can_update(struct snd_soc_pcm_runtime *fe,
+		struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the substream for this BE */
+struct snd_pcm_substream *
+	snd_soc_dpcm_get_substream(struct snd_soc_pcm_runtime *be, int stream);
+
+/* get the BE runtime state */
+enum snd_soc_dpcm_state
+	snd_soc_dpcm_be_get_state(struct snd_soc_pcm_runtime *be, int stream);
+
+/* set the BE runtime state */
+void snd_soc_dpcm_be_set_state(struct snd_soc_pcm_runtime *be, int stream,
+	enum snd_soc_dpcm_state state);
+
+/* internal use only */
+int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute);
+int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd);
+int soc_dpcm_runtime_update(struct snd_soc_dapm_widget *);
+
+#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 2ebf7877c148..c703871f5f65 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -55,6 +55,18 @@
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\
 	.put = snd_soc_put_volsw, \
 	.private_value =  SOC_SINGLE_VALUE(reg, shift, max, invert) }
+#define SOC_SINGLE_SX_TLV(xname, xreg, xshift, xmin, xmax, tlv_array) \
+{       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+	SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p  = (tlv_array),\
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_sx,\
+	.put = snd_soc_put_volsw_sx, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .rreg = xreg, \
+		.shift = xshift, .rshift = xshift, \
+		.max = xmax, .min = xmin} }
 #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \
@@ -85,6 +97,18 @@
 	.get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \
 					    xmax, xinvert) }
+#define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \
+{       .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+	SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p  = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw_sx, \
+	.put = snd_soc_put_volsw_sx, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .rreg = xrreg, \
+		.shift = xshift, .rshift = xshift, \
+		.max = xmax, .min = xmin} }
 #define SOC_DOUBLE_S8_TLV(xname, xreg, xmin, xmax, tlv_array) \
 {	.iface  = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
@@ -171,20 +195,6 @@
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = (unsigned long)&xenum }
 
-#define SOC_DOUBLE_R_SX_TLV(xname, xreg_left, xreg_right, xshift,\
-		xmin, xmax, tlv_array) \
-{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
-	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
-		  SNDRV_CTL_ELEM_ACCESS_READWRITE, \
-	.tlv.p = (tlv_array), \
-	.info = snd_soc_info_volsw_2r_sx, \
-	.get = snd_soc_get_volsw_2r_sx, \
-	.put = snd_soc_put_volsw_2r_sx, \
-	.private_value = (unsigned long)&(struct soc_mixer_control) \
-		{.reg = xreg_left, \
-		 .rreg = xreg_right, .shift = xshift, \
-		 .min = xmin, .max = xmax} }
-
 #define SND_SOC_BYTES(xname, xbase, xregs)		      \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,   \
 	.info = snd_soc_bytes_info, .get = snd_soc_bytes_get, \
@@ -200,6 +210,19 @@
 		{.base = xbase, .num_regs = xregs,	      \
 		 .mask = xmask }) }
 
+#define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \
+		xmin, xmax, xinvert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.info = snd_soc_info_xr_sx, .get = snd_soc_get_xr_sx, \
+	.put = snd_soc_put_xr_sx, \
+	.private_value = (unsigned long)&(struct soc_mreg_control) \
+		{.regbase = xregbase, .regcount = xregcount, .nbits = xnbits, \
+		.invert = xinvert, .min = xmin, .max = xmax} }
+
+#define SOC_SINGLE_STROBE(xname, xreg, xshift, xinvert) \
+	SOC_SINGLE_EXT(xname, xreg, xshift, 1, xinvert, \
+		snd_soc_get_strobe, snd_soc_put_strobe)
+
 /*
  * Simplified versions of above macros, declaring a struct and calculating
  * ARRAY_SIZE internally
@@ -264,6 +287,7 @@ struct snd_soc_jack_zone;
 struct snd_soc_jack_pin;
 struct snd_soc_cache_ops;
 #include <sound/soc-dapm.h>
+#include <sound/soc-dpcm.h>
 
 #ifdef CONFIG_GPIOLIB
 struct snd_soc_jack_gpio;
@@ -288,6 +312,11 @@ enum snd_soc_pcm_subclass {
 	SND_SOC_PCM_CLASS_BE	= 1,
 };
 
+enum snd_soc_card_subclass {
+	SND_SOC_CARD_CLASS_INIT		= 0,
+	SND_SOC_CARD_CLASS_RUNTIME	= 1,
+};
+
 int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id,
 			     int source, unsigned int freq, int dir);
 int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source,
@@ -333,6 +362,11 @@ int snd_soc_platform_write(struct snd_soc_platform *platform,
 					unsigned int reg, unsigned int val);
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
 
+struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
+		const char *dai_link, int stream);
+struct snd_soc_pcm_runtime *snd_soc_get_pcm_runtime(struct snd_soc_card *card,
+		const char *dai_link);
+
 /* Utility functions to get clock rates from various things */
 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots);
 int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params);
@@ -343,6 +377,9 @@ int snd_soc_params_to_bclk(struct snd_pcm_hw_params *parms);
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
 	const struct snd_pcm_hardware *hw);
 
+int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
+		int cmd, struct snd_soc_platform *platform);
+
 /* Jack reporting */
 int snd_soc_jack_new(struct snd_soc_codec *codec, const char *id, int type,
 		     struct snd_soc_jack *jack);
@@ -413,6 +450,10 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 #define snd_soc_get_volsw_2r snd_soc_get_volsw
 #define snd_soc_put_volsw_2r snd_soc_put_volsw
+int snd_soc_get_volsw_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_info_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_info *uinfo);
 int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol,
@@ -421,19 +462,22 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_limit_volume(struct snd_soc_codec *codec,
 	const char *name, int max);
-int snd_soc_info_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_info *uinfo);
-int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
-int snd_soc_put_volsw_2r_sx(struct snd_kcontrol *kcontrol,
-	struct snd_ctl_elem_value *ucontrol);
 int snd_soc_bytes_info(struct snd_kcontrol *kcontrol,
 		       struct snd_ctl_elem_info *uinfo);
 int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
 		      struct snd_ctl_elem_value *ucontrol);
 int snd_soc_bytes_put(struct snd_kcontrol *kcontrol,
 		      struct snd_ctl_elem_value *ucontrol);
-
+int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo);
+int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_xr_sx(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_get_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
+int snd_soc_put_strobe(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol);
 
 /**
  * struct snd_soc_reg_access - Describes whether a given register is
@@ -513,6 +557,7 @@ struct snd_soc_jack_gpio {
 #endif
 
 struct snd_soc_jack {
+	struct mutex mutex;
 	struct snd_jack *jack;
 	struct snd_soc_codec *codec;
 	struct list_head pins;
@@ -711,6 +756,7 @@ struct snd_soc_platform_driver {
 	/* platform IO - used for platform DAPM */
 	unsigned int (*read)(struct snd_soc_platform *, unsigned int);
 	int (*write)(struct snd_soc_platform *, unsigned int, unsigned int);
+	int (*bespoke_trigger)(struct snd_pcm_substream *, int);
 };
 
 struct snd_soc_platform {
@@ -746,21 +792,36 @@ struct snd_soc_dai_link {
 	const char *cpu_dai_name;
 	const struct device_node *cpu_dai_of_node;
 	const char *codec_dai_name;
+	int be_id;	/* optional ID for machine driver BE identification */
+
+	const struct snd_soc_pcm_stream *params;
 
 	unsigned int dai_fmt;           /* format to set on init */
 
+	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
+
 	/* Keep DAI active over suspend */
 	unsigned int ignore_suspend:1;
 
 	/* Symmetry requirements */
 	unsigned int symmetric_rates:1;
 
+	/* Do not create a PCM for this DAI link (Backend link) */
+	unsigned int no_pcm:1;
+
+	/* This DAI link can route to other DAI links at runtime (Frontend)*/
+	unsigned int dynamic:1;
+
 	/* pmdown_time is ignored at stop */
 	unsigned int ignore_pmdown_time:1;
 
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_pcm_runtime *rtd);
 
+	/* optional hw_params re-writing for BE and FE sync */
+	int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd,
+			struct snd_pcm_hw_params *params);
+
 	/* machine stream operations */
 	struct snd_soc_ops *ops;
 };
@@ -800,6 +861,7 @@ struct snd_soc_card {
 
 	struct list_head list;
 	struct mutex mutex;
+	struct mutex dapm_mutex;
 
 	bool instantiated;
 
@@ -889,9 +951,11 @@ struct snd_soc_pcm_runtime {
 	enum snd_soc_pcm_subclass pcm_subclass;
 	struct snd_pcm_ops ops;
 
-	unsigned int complete:1;
 	unsigned int dev_registered:1;
 
+	/* Dynamic PCM BE runtime data */
+	struct snd_soc_dpcm_runtime dpcm[2];
+
 	long pmdown_time;
 
 	/* runtime devices */
@@ -902,6 +966,10 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_dai *cpu_dai;
 
 	struct delayed_work delayed_work;
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_dpcm_root;
+	struct dentry *debugfs_dpcm_state;
+#endif
 };
 
 /* mixer control */
@@ -916,6 +984,12 @@ struct soc_bytes {
 	u32 mask;
 };
 
+/* multi register control */
+struct soc_mreg_control {
+	long min, max;
+	unsigned int regbase, regcount, nbits, invert;
+};
+
 /* enumerated kcontrol */
 struct soc_enum {
 	unsigned short reg;
diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h
index ab26f8aa3c78..5fc2dcdd21cd 100644
--- a/include/trace/events/asoc.h
+++ b/include/trace/events/asoc.h
@@ -7,6 +7,8 @@
 #include <linux/ktime.h>
 #include <linux/tracepoint.h>
 
+#define DAPM_DIRECT "(direct)"
+
 struct snd_soc_jack;
 struct snd_soc_codec;
 struct snd_soc_platform;
@@ -241,6 +243,84 @@ TRACE_EVENT(snd_soc_dapm_walk_done,
 		  (int)__entry->path_checks, (int)__entry->neighbour_checks)
 );
 
+TRACE_EVENT(snd_soc_dapm_output_path,
+
+	TP_PROTO(struct snd_soc_dapm_widget *widget,
+		struct snd_soc_dapm_path *path),
+
+	TP_ARGS(widget, path),
+
+	TP_STRUCT__entry(
+		__string(	wname,	widget->name		)
+		__string(	pname,	path->name ? path->name : DAPM_DIRECT)
+		__string(	psname,	path->sink->name	)
+		__field(	int,	path_sink		)
+		__field(	int,	path_connect		)
+	),
+
+	TP_fast_assign(
+		__assign_str(wname, widget->name);
+		__assign_str(pname, path->name ? path->name : DAPM_DIRECT);
+		__assign_str(psname, path->sink->name);
+		__entry->path_connect = path->connect;
+		__entry->path_sink = (long)path->sink;
+	),
+
+	TP_printk("%c%s -> %s -> %s\n",
+		(int) __entry->path_sink &&
+		(int) __entry->path_connect ? '*' : ' ',
+		__get_str(wname), __get_str(pname), __get_str(psname))
+);
+
+TRACE_EVENT(snd_soc_dapm_input_path,
+
+	TP_PROTO(struct snd_soc_dapm_widget *widget,
+		struct snd_soc_dapm_path *path),
+
+	TP_ARGS(widget, path),
+
+	TP_STRUCT__entry(
+		__string(	wname,	widget->name		)
+		__string(	pname,	path->name ? path->name : DAPM_DIRECT)
+		__string(	psname,	path->source->name	)
+		__field(	int,	path_source		)
+		__field(	int,	path_connect		)
+	),
+
+	TP_fast_assign(
+		__assign_str(wname, widget->name);
+		__assign_str(pname, path->name ? path->name : DAPM_DIRECT);
+		__assign_str(psname, path->source->name);
+		__entry->path_connect = path->connect;
+		__entry->path_source = (long)path->source;
+	),
+
+	TP_printk("%c%s <- %s <- %s\n",
+		(int) __entry->path_source &&
+		(int) __entry->path_connect ? '*' : ' ',
+		__get_str(wname), __get_str(pname), __get_str(psname))
+);
+
+TRACE_EVENT(snd_soc_dapm_connected,
+
+	TP_PROTO(int paths, int stream),
+
+	TP_ARGS(paths, stream),
+
+	TP_STRUCT__entry(
+		__field(	int,	paths		)
+		__field(	int,	stream		)
+	),
+
+	TP_fast_assign(
+		__entry->paths = paths;
+		__entry->stream = stream;
+	),
+
+	TP_printk("%s: found %d paths\n",
+		__entry->stream ? "capture" : "playback", __entry->paths)
+);
+
 TRACE_EVENT(snd_soc_jack_irq,
 
 	TP_PROTO(const char *name),