Merge tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "No dramatic changes are found in this development cycle, but as usual,
  many commits are applied in a wide range of drivers.

  Most of big changes are in ASoC, where a few bits of framework work
  and quite a lot of cleanups and improvements to existing code have
  been done. The rest are usual stuff, a few HD-audio and USB-audio
  quirks and fixes, as well as the drop of kthread usages in the whole
  subsystem.

  Below are some highlights:

  ASoC:
   - support for stereo DAPM controls
   - some initial work on the of-graph sound card
   - regmap conversions of the remaining AC'97 drivers
   - a new version of the topology ABI; this should be backward
     compatible
   - updates / cleanups of rsnd, sunxi, sti, nau8825, samsung, arizona,
     Intel skylake, atom-sst
   - new drivers for Cirrus Logic CS42L42, Qualcomm MSM8916-WCD, and
     Realtek RT5665

  USB-audio:
   - yet another race fix at disconnection
   - tolerated packet size calculation for some Android devices
   - quirks for Axe-Fx II, QuickCam, TEAC 501/503

  HD-audio:
   - improvement of Dell pin fixup mapping
   - quirks for HP Z1 Gen3, Alienware 15 R2 2016 and ALC622 headset mic

  Misc:
   - replace all kthread usages with simple works"

* tag 'sound-4.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (296 commits)
  ALSA: hiface: Fix M2Tech hiFace driver sampling rate change
  ALSA: usb-audio: Eliminate noise at the start of DSD playback.
  ALSA: usb-audio: Add native DSD support for TEAC 501/503 DAC
  ASoC: wm_adsp: wm_adsp_buf_alloc should use kfree in error path
  ASoC: topology: avoid uninitialized kcontrol_type
  ALSA: usb-audio: Add QuickCam Communicate Deluxe/S7500 to volume_control_quirks
  ALSA: usb-audio: add implicit fb quirk for Axe-Fx II
  ASoC: zte: spdif: correct ZX_SPDIF_CLK_RAT define
  ASoC: zte: spdif and i2s drivers are not zx296702 specific
  ASoC: rsnd: setup BRGCKR/BRRA/BRRB when starting
  ASoC: rsnd: enable/disable ADG when suspend/resume timing
  ASoC: rsnd: tidyup ssi->usrcnt counter check in hw_params
  ALSA: cs46xx: add a new line
  ASoC: Intel: update bxt_da7219_max98357a to support quad ch dmic capture
  ASoC: nau8825: disable sinc filter for high THD of ADC
  ALSA: usb-audio: more tolerant packetsize
  ALSA: usb-audio: avoid setting of sample rate multiple times on bus
  ASoC: cs35l34: Simplify the logic to set CS35L34_MCLK_CTL setting
  ALSA: hda - Gate the mic jack on HP Z1 Gen3 AiO
  ALSA: hda: when comparing pin configurations, ignore assoc in addition to seq
  ...

15 files changed, 393 insertions, 64 deletions

diff --git a/include/dt-bindings/sound/cs42l42.h b/include/dt-bindings/sound/cs42l42.h
new file mode 100644
index 000000000000..399a123aed58
--- /dev/null
+++ b/include/dt-bindings/sound/cs42l42.h
@@ -0,0 +1,73 @@
+/*
+ * cs42l42.h -- CS42L42 ALSA SoC audio driver DT bindings header
+ *
+ * Copyright 2016 Cirrus Logic, Inc.
+ *
+ * Author: James Schulman <james.schulman@cirrus.com>
+ * Author: Brian Austin <brian.austin@cirrus.com>
+ * Author: Michael White <michael.white@cirrus.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 __DT_CS42L42_H
+#define __DT_CS42L42_H
+
+/* HPOUT Load Capacity */
+#define CS42L42_HPOUT_LOAD_1NF		0
+#define CS42L42_HPOUT_LOAD_10NF		1
+
+/* HPOUT Clamp to GND Overide */
+#define CS42L42_HPOUT_CLAMP_EN		0
+#define CS42L42_HPOUT_CLAMP_DIS		1
+
+/* Tip Sense Inversion */
+#define CS42L42_TS_INV_DIS			0
+#define CS42L42_TS_INV_EN			1
+
+/* Tip Sense Debounce */
+#define CS42L42_TS_DBNCE_0			0
+#define CS42L42_TS_DBNCE_125			1
+#define CS42L42_TS_DBNCE_250			2
+#define CS42L42_TS_DBNCE_500			3
+#define CS42L42_TS_DBNCE_750			4
+#define CS42L42_TS_DBNCE_1000			5
+#define CS42L42_TS_DBNCE_1250			6
+#define CS42L42_TS_DBNCE_1500			7
+
+/* Button Press Software Debounce Times */
+#define CS42L42_BTN_DET_INIT_DBNCE_MIN		0
+#define CS42L42_BTN_DET_INIT_DBNCE_DEFAULT	100
+#define CS42L42_BTN_DET_INIT_DBNCE_MAX		200
+
+#define CS42L42_BTN_DET_EVENT_DBNCE_MIN		0
+#define CS42L42_BTN_DET_EVENT_DBNCE_DEFAULT	10
+#define CS42L42_BTN_DET_EVENT_DBNCE_MAX		20
+
+/* Button Detect Level Sensitivities */
+#define CS42L42_NUM_BIASES		4
+
+#define CS42L42_HS_DET_LEVEL_15		0x0F
+#define CS42L42_HS_DET_LEVEL_8		0x08
+#define CS42L42_HS_DET_LEVEL_4		0x04
+#define CS42L42_HS_DET_LEVEL_1		0x01
+
+#define CS42L42_HS_DET_LEVEL_MIN	0
+#define CS42L42_HS_DET_LEVEL_MAX	0x3F
+
+/* HS Bias Ramp Rate */
+
+#define CS42L42_HSBIAS_RAMP_FAST_RISE_SLOW_FALL		0
+#define CS42L42_HSBIAS_RAMP_FAST			1
+#define CS42L42_HSBIAS_RAMP_SLOW			2
+#define CS42L42_HSBIAS_RAMP_SLOWEST			3
+
+#define CS42L42_HSBIAS_RAMP_TIME0			10
+#define CS42L42_HSBIAS_RAMP_TIME1			40
+#define CS42L42_HSBIAS_RAMP_TIME2			90
+#define CS42L42_HSBIAS_RAMP_TIME3			170
+
+#endif /* __DT_CS42L42_H */
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index cee8c00f3d3e..9924bc9cbc7c 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -155,6 +155,7 @@ struct snd_compr {
 	struct mutex lock;
 	int device;
 #ifdef CONFIG_SND_VERBOSE_PROCFS
+	/* private: */
 	char id[64];
 	struct snd_info_entry *proc_root;
 	struct snd_info_entry *proc_info_entry;
diff --git a/include/sound/core.h b/include/sound/core.h
index 31079ea5e484..f7d8c10c4c45 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -308,8 +308,8 @@ __printf(4, 5)
 void __snd_printk(unsigned int level, const char *file, int line,
 		  const char *format, ...);
 #else
-#define __snd_printk(level, file, line, format, args...) \
-	printk(format, ##args)
+#define __snd_printk(level, file, line, format, ...) \
+	printk(format, ##__VA_ARGS__)
 #endif
 
 /**
@@ -319,8 +319,8 @@ void __snd_printk(unsigned int level, const char *file, int line,
  * Works like printk() but prints the file and the line of the caller
  * when configured with CONFIG_SND_VERBOSE_PRINTK.
  */
-#define snd_printk(fmt, args...) \
-	__snd_printk(0, __FILE__, __LINE__, fmt, ##args)
+#define snd_printk(fmt, ...) \
+	__snd_printk(0, __FILE__, __LINE__, fmt, ##__VA_ARGS__)
 
 #ifdef CONFIG_SND_DEBUG
 /**
@@ -330,10 +330,10 @@ void __snd_printk(unsigned int level, const char *file, int line,
  * Works like snd_printk() for debugging purposes.
  * Ignored when CONFIG_SND_DEBUG is not set.
  */
-#define snd_printd(fmt, args...) \
-	__snd_printk(1, __FILE__, __LINE__, fmt, ##args)
-#define _snd_printd(level, fmt, args...) \
-	__snd_printk(level, __FILE__, __LINE__, fmt, ##args)
+#define snd_printd(fmt, ...) \
+	__snd_printk(1, __FILE__, __LINE__, fmt, ##__VA_ARGS__)
+#define _snd_printd(level, fmt, ...) \
+	__snd_printk(level, __FILE__, __LINE__, fmt, ##__VA_ARGS__)
 
 /**
  * snd_BUG - give a BUG warning message and stack trace
@@ -383,8 +383,8 @@ static inline bool snd_printd_ratelimit(void) { return false; }
  * Works like snd_printk() for debugging purposes.
  * Ignored when CONFIG_SND_DEBUG_VERBOSE is not set.
  */
-#define snd_printdd(format, args...) \
-	__snd_printk(2, __FILE__, __LINE__, format, ##args)
+#define snd_printdd(format, ...) \
+	__snd_printk(2, __FILE__, __LINE__, format, ##__VA_ARGS__)
 #else
 __printf(1, 2)
 static inline void snd_printdd(const char *format, ...) {}
diff --git a/include/sound/cs35l34.h b/include/sound/cs35l34.h
new file mode 100644
index 000000000000..9c927cffbe46
--- /dev/null
+++ b/include/sound/cs35l34.h
@@ -0,0 +1,35 @@
+/*
+ * linux/sound/cs35l34.h -- Platform data for CS35l34
+ *
+ * Copyright (c) 2016 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 __CS35L34_H
+#define __CS35L34_H
+
+struct cs35l34_platform_data {
+	/* Set AIF to half drive strength */
+	bool aif_half_drv;
+	/* Digital Soft Ramp Disable */
+	bool digsft_disable;
+	/* Amplifier Invert */
+	bool amp_inv;
+	/* Peak current (mA) */
+	unsigned int boost_peak;
+	/* Boost inductor value (nH) */
+	unsigned int boost_ind;
+	/* Boost Controller Voltage Setting (mV) */
+	unsigned int boost_vtge;
+	/* Gain Change Zero Cross */
+	bool gain_zc_disable;
+	/* SDIN Left/Right Selection */
+	unsigned int i2s_sdinloc;
+	/* TDM Rising Edge */
+	bool tdm_rising_edge;
+};
+
+#endif /* __CS35L34_H */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 67be2445941a..1c8f9e1ef2a5 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -71,7 +71,6 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * @slave_id: Slave requester id for the DMA channel.
  * @filter_data: Custom DMA channel filter data, this will usually be used when
  * requesting the DMA channel.
- * @chan_name: Custom channel name to use when requesting DMA channel.
  * @fifo_size: FIFO size of the DAI controller in bytes
  * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now
  */
@@ -81,7 +80,6 @@ struct snd_dmaengine_dai_dma_data {
 	u32 maxburst;
 	unsigned int slave_id;
 	void *filter_data;
-	const char *chan_name;
 	unsigned int fifo_size;
 	unsigned int flags;
 };
@@ -107,10 +105,6 @@ void snd_dmaengine_pcm_set_config_from_dai_data(
  * playback.
  */
 #define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3)
-/*
- * The PCM streams have custom channel names specified.
- */
-#define SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME BIT(4)
 
 /**
  * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 5bd134651f5e..4f42affe777c 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1688,7 +1688,8 @@ struct snd_emu1010 {
 	unsigned int internal_clock; /* 44100 or 48000 */
 	unsigned int optical_in; /* 0:SPDIF, 1:ADAT */
 	unsigned int optical_out; /* 0:SPDIF, 1:ADAT */
-	struct task_struct *firmware_thread;
+	struct delayed_work firmware_work;
+	u32 last_reg;
 };
 
 struct snd_emu10k1 {
diff --git a/include/sound/rt5514.h b/include/sound/rt5514.h
new file mode 100644
index 000000000000..ef18494769ee
--- /dev/null
+++ b/include/sound/rt5514.h
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/rt5514.h -- Platform data for RT5514
+ *
+ * Copyright 2016 Realtek Semiconductor Corp.
+ * Author: Oder Chiou <oder_chiou@realtek.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_RT5514_H
+#define __LINUX_SND_RT5514_H
+
+struct rt5514_platform_data {
+	unsigned int dmic_init_delay;
+};
+
+#endif
+
diff --git a/include/sound/rt5665.h b/include/sound/rt5665.h
new file mode 100755
index 000000000000..963229e71dc7
--- /dev/null
+++ b/include/sound/rt5665.h
@@ -0,0 +1,47 @@
+/*
+ * linux/sound/rt5665.h -- Platform data for RT5665
+ *
+ * Copyright 2016 Realtek Microelectronics
+ *
+ * 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_RT5665_H
+#define __LINUX_SND_RT5665_H
+
+enum rt5665_dmic1_data_pin {
+	RT5665_DMIC1_NULL,
+	RT5665_DMIC1_DATA_GPIO4,
+	RT5665_DMIC1_DATA_IN2N,
+};
+
+enum rt5665_dmic2_data_pin {
+	RT5665_DMIC2_NULL,
+	RT5665_DMIC2_DATA_GPIO5,
+	RT5665_DMIC2_DATA_IN2P,
+};
+
+enum rt5665_jd_src {
+	RT5665_JD_NULL,
+	RT5665_JD1,
+};
+
+struct rt5665_platform_data {
+	bool in1_diff;
+	bool in2_diff;
+	bool in3_diff;
+	bool in4_diff;
+
+	int ldo1_en; /* GPIO for LDO1_EN */
+
+	enum rt5665_dmic1_data_pin dmic1_data_pin;
+	enum rt5665_dmic2_data_pin dmic2_data_pin;
+	enum rt5665_jd_src jd_src;
+
+	unsigned int sar_hs_type;
+};
+
+#endif
+
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index fd6412551145..64e90ca9ad32 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -1,5 +1,5 @@
 /*
- * simple_card_core.h
+ * simple_card_utils.h
  *
  * Copyright (c) 2016 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
  *
@@ -7,8 +7,8 @@
  * it under the terms of the GNU General Public License version 2 as
  * published by the Free Software Foundation.
  */
-#ifndef __SIMPLE_CARD_CORE_H
-#define __SIMPLE_CARD_CORE_H
+#ifndef __SIMPLE_CARD_UTILS_H
+#define __SIMPLE_CARD_UTILS_H
 
 #include <sound/soc.h>
 
@@ -68,4 +68,4 @@ void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
 
 int asoc_simple_card_clean_reference(struct snd_soc_card *card);
 
-#endif /* __SIMPLE_CARD_CORE_H */
+#endif /* __SIMPLE_CARD_UTILS_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 964b7de1a1cc..200e1f04c166 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -15,6 +15,7 @@
 
 
 #include <linux/list.h>
+#include <sound/asoc.h>
 
 struct snd_pcm_substream;
 struct snd_soc_dapm_widget;
@@ -26,13 +27,13 @@ struct snd_compr_stream;
  * Describes the physical PCM data formating and clocking. Add new formats
  * to the end.
  */
-#define SND_SOC_DAIFMT_I2S		1 /* I2S mode */
-#define SND_SOC_DAIFMT_RIGHT_J		2 /* Right Justified mode */
-#define SND_SOC_DAIFMT_LEFT_J		3 /* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A		4 /* L data MSB after FRM LRC */
-#define SND_SOC_DAIFMT_DSP_B		5 /* L data MSB during FRM LRC */
-#define SND_SOC_DAIFMT_AC97		6 /* AC97 */
-#define SND_SOC_DAIFMT_PDM		7 /* Pulse density modulation */
+#define SND_SOC_DAIFMT_I2S		SND_SOC_DAI_FORMAT_I2S
+#define SND_SOC_DAIFMT_RIGHT_J		SND_SOC_DAI_FORMAT_RIGHT_J
+#define SND_SOC_DAIFMT_LEFT_J		SND_SOC_DAI_FORMAT_LEFT_J
+#define SND_SOC_DAIFMT_DSP_A		SND_SOC_DAI_FORMAT_DSP_A
+#define SND_SOC_DAIFMT_DSP_B		SND_SOC_DAI_FORMAT_DSP_B
+#define SND_SOC_DAIFMT_AC97		SND_SOC_DAI_FORMAT_AC97
+#define SND_SOC_DAIFMT_PDM		SND_SOC_DAI_FORMAT_PDM
 
 /* left and right justified also known as MSB and LSB respectively */
 #define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
@@ -207,6 +208,30 @@ struct snd_soc_dai_ops {
 		struct snd_soc_dai *);
 };
 
+struct snd_soc_cdai_ops {
+	/*
+	 * for compress ops
+	 */
+	int (*startup)(struct snd_compr_stream *,
+			struct snd_soc_dai *);
+	int (*shutdown)(struct snd_compr_stream *,
+			struct snd_soc_dai *);
+	int (*set_params)(struct snd_compr_stream *,
+			struct snd_compr_params *, struct snd_soc_dai *);
+	int (*get_params)(struct snd_compr_stream *,
+			struct snd_codec *, struct snd_soc_dai *);
+	int (*set_metadata)(struct snd_compr_stream *,
+			struct snd_compr_metadata *, struct snd_soc_dai *);
+	int (*get_metadata)(struct snd_compr_stream *,
+			struct snd_compr_metadata *, struct snd_soc_dai *);
+	int (*trigger)(struct snd_compr_stream *, int,
+			struct snd_soc_dai *);
+	int (*pointer)(struct snd_compr_stream *,
+			struct snd_compr_tstamp *, struct snd_soc_dai *);
+	int (*ack)(struct snd_compr_stream *, size_t,
+			struct snd_soc_dai *);
+};
+
 /*
  * Digital Audio Interface Driver.
  *
@@ -236,6 +261,7 @@ struct snd_soc_dai_driver {
 
 	/* ops */
 	const struct snd_soc_dai_ops *ops;
+	const struct snd_soc_cdai_ops *cops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
@@ -268,8 +294,9 @@ struct snd_soc_dai {
 	unsigned int symmetric_rates:1;
 	unsigned int symmetric_channels:1;
 	unsigned int symmetric_samplebits:1;
+	unsigned int probed:1;
+
 	unsigned int active;
-	unsigned char probed:1;
 
 	struct snd_soc_dapm_widget *playback_widget;
 	struct snd_soc_dapm_widget *capture_widget;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index f60d755f7ac6..a466f4bdc835 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -272,6 +272,16 @@ struct device;
 
 
 /* dapm kcontrol types */
+#define SOC_DAPM_DOUBLE(xname, reg, lshift, rshift, max, invert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+	.private_value = SOC_DOUBLE_VALUE(reg, lshift, rshift, max, invert, 0) }
+#define SOC_DAPM_DOUBLE_R(xname, lreg, rreg, shift, max, invert) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw, \
+	.private_value = SOC_DOUBLE_R_VALUE(lreg, rreg, shift, max, invert) }
 #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_volsw, \
@@ -615,6 +625,10 @@ struct snd_soc_dapm_update {
 	int reg;
 	int mask;
 	int val;
+	int reg2;
+	int mask2;
+	int val2;
+	bool has_second_set;
 };
 
 struct snd_soc_dapm_wcache {
diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h
index b897b9d63161..f9cc7b9271ac 100644
--- a/include/sound/soc-topology.h
+++ b/include/sound/soc-topology.h
@@ -53,7 +53,7 @@ struct snd_soc_dobj_control {
 
 /* dynamic widget object */
 struct snd_soc_dobj_widget {
-	unsigned int kcontrol_enum:1;	/* this widget is an enum kcontrol */
+	unsigned int kcontrol_type;	/* kcontrol type: mixer, enum, bytes */
 };
 
 /* generic dynamic object - all dynamic objects belong to this struct */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4f1c784e44f6..2b502f6cc6d0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -782,6 +782,8 @@ struct snd_soc_component_driver {
 
 	int (*probe)(struct snd_soc_component *);
 	void (*remove)(struct snd_soc_component *);
+	int (*suspend)(struct snd_soc_component *);
+	int (*resume)(struct snd_soc_component *);
 
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
@@ -807,9 +809,11 @@ struct snd_soc_component {
 
 	unsigned int ignore_pmdown_time:1; /* pmdown_time is ignored at stop */
 	unsigned int registered_as_component:1;
+	unsigned int auxiliary:1; /* for auxiliary component of the card */
+	unsigned int suspended:1; /* is in suspend PM state */
 
 	struct list_head list;
-	struct list_head list_aux; /* for auxiliary component of the card */
+	struct list_head card_list;
 
 	struct snd_soc_dai_driver *dai_drv;
 	int num_dai;
@@ -852,6 +856,8 @@ struct snd_soc_component {
 
 	int (*probe)(struct snd_soc_component *);
 	void (*remove)(struct snd_soc_component *);
+	int (*suspend)(struct snd_soc_component *);
+	int (*resume)(struct snd_soc_component *);
 
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);
@@ -868,11 +874,9 @@ struct snd_soc_codec {
 	const struct snd_soc_codec_driver *driver;
 
 	struct list_head list;
-	struct list_head card_list;
 
 	/* runtime */
 	unsigned int cache_bypass:1; /* Suppress access to the cache */
-	unsigned int suspended:1; /* Codec is in suspend PM state */
 	unsigned int cache_init:1; /* codec cache has been initialized */
 
 	/* codec IO */
@@ -1025,13 +1029,13 @@ struct snd_soc_dai_link {
 	const struct snd_soc_ops *ops;
 	const struct snd_soc_compr_ops *compr_ops;
 
-	/* For unidirectional dai links */
-	bool playback_only;
-	bool capture_only;
-
 	/* Mark this pcm with non atomic ops */
 	bool nonatomic;
 
+	/* For unidirectional dai links */
+	unsigned int playback_only:1;
+	unsigned int capture_only:1;
+
 	/* Keep DAI active over suspend */
 	unsigned int ignore_suspend:1;
 
@@ -1148,7 +1152,6 @@ struct snd_soc_card {
 	 */
 	struct snd_soc_aux_dev *aux_dev;
 	int num_aux_devs;
-	struct list_head aux_comp_list;
 
 	const struct snd_kcontrol_new *controls;
 	int num_controls;
@@ -1170,7 +1173,7 @@ struct snd_soc_card {
 	struct work_struct deferred_resume_work;
 
 	/* lists of probed devices belonging to this card */
-	struct list_head codec_dev_list;
+	struct list_head component_dev_list;
 
 	struct list_head widgets;
 	struct list_head paths;
@@ -1203,14 +1206,11 @@ struct snd_soc_pcm_runtime {
 	enum snd_soc_pcm_subclass pcm_subclass;
 	struct snd_pcm_ops ops;
 
-	unsigned int dev_registered:1;
-
 	/* Dynamic PCM BE runtime data */
 	struct snd_soc_dpcm_runtime dpcm[2];
 	int fe_compr;
 
 	long pmdown_time;
-	unsigned char pop_wait:1;
 
 	/* runtime devices */
 	struct snd_pcm *pcm;
@@ -1219,7 +1219,6 @@ struct snd_soc_pcm_runtime {
 	struct snd_soc_platform *platform;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai;
-	struct snd_soc_component *component; /* Only valid for AUX dev rtds */
 
 	struct snd_soc_dai **codec_dais;
 	unsigned int num_codecs;
@@ -1232,6 +1231,10 @@ struct snd_soc_pcm_runtime {
 
 	unsigned int num; /* 0-based and monotonic increasing */
 	struct list_head list; /* rtd list of the soc card */
+
+	/* bit field */
+	unsigned int dev_registered:1;
+	unsigned int pop_wait:1;
 };
 
 /* mixer control */
@@ -1541,11 +1544,10 @@ static inline void *snd_soc_platform_get_drvdata(struct snd_soc_platform *platfo
 
 static inline void snd_soc_initialize_card_lists(struct snd_soc_card *card)
 {
-	INIT_LIST_HEAD(&card->codec_dev_list);
 	INIT_LIST_HEAD(&card->widgets);
 	INIT_LIST_HEAD(&card->paths);
 	INIT_LIST_HEAD(&card->dapm_list);
-	INIT_LIST_HEAD(&card->aux_comp_list);
+	INIT_LIST_HEAD(&card->component_dev_list);
 }
 
 static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc)
@@ -1642,25 +1644,43 @@ static inline struct snd_soc_platform *snd_soc_kcontrol_platform(
 int snd_soc_util_init(void);
 void snd_soc_util_exit(void);
 
-int snd_soc_of_parse_card_name(struct snd_soc_card *card,
-			       const char *propname);
-int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card,
-					  const char *propname);
+#define snd_soc_of_parse_card_name(card, propname) \
+	snd_soc_of_parse_card_name_from_node(card, NULL, propname)
+int snd_soc_of_parse_card_name_from_node(struct snd_soc_card *card,
+					 struct device_node *np,
+					 const char *propname);
+#define snd_soc_of_parse_audio_simple_widgets(card, propname)\
+	snd_soc_of_parse_audio_simple_widgets_from_node(card, NULL, propname)
+int snd_soc_of_parse_audio_simple_widgets_from_node(struct snd_soc_card *card,
+						    struct device_node *np,
+						    const char *propname);
+
 int snd_soc_of_parse_tdm_slot(struct device_node *np,
 			      unsigned int *tx_mask,
 			      unsigned int *rx_mask,
 			      unsigned int *slots,
 			      unsigned int *slot_width);
-void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card,
+#define snd_soc_of_parse_audio_prefix(card, codec_conf, of_node, propname) \
+	snd_soc_of_parse_audio_prefix_from_node(card, NULL, codec_conf, \
+						of_node, propname)
+void snd_soc_of_parse_audio_prefix_from_node(struct snd_soc_card *card,
+				   struct device_node *np,
 				   struct snd_soc_codec_conf *codec_conf,
 				   struct device_node *of_node,
 				   const char *propname);
-int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
-				   const char *propname);
+
+#define snd_soc_of_parse_audio_routing(card, propname) \
+	snd_soc_of_parse_audio_routing_from_node(card, NULL, propname)
+int snd_soc_of_parse_audio_routing_from_node(struct snd_soc_card *card,
+					     struct device_node *np,
+					     const char *propname);
+
 unsigned int snd_soc_of_parse_daifmt(struct device_node *np,
 				     const char *prefix,
 				     struct device_node **bitclkmaster,
 				     struct device_node **framemaster);
+int snd_soc_get_dai_name(struct of_phandle_args *args,
+			 const char **dai_name);
 int snd_soc_of_get_dai_name(struct device_node *of_node,
 			    const char **dai_name);
 int snd_soc_of_get_dai_link_codecs(struct device *dev,
@@ -1671,6 +1691,9 @@ int snd_soc_add_dai_link(struct snd_soc_card *card,
 				struct snd_soc_dai_link *dai_link);
 void snd_soc_remove_dai_link(struct snd_soc_card *card,
 			     struct snd_soc_dai_link *dai_link);
+struct snd_soc_dai_link *snd_soc_find_dai_link(struct snd_soc_card *card,
+					       int id, const char *name,
+					       const char *stream_name);
 
 int snd_soc_register_dai(struct snd_soc_component *component,
 	struct snd_soc_dai_driver *dai_drv);
@@ -1697,4 +1720,24 @@ static inline void snd_soc_dapm_mutex_unlock(struct snd_soc_dapm_context *dapm)
 	mutex_unlock(&dapm->card->dapm_mutex);
 }
 
+int snd_soc_component_enable_pin(struct snd_soc_component *component,
+				 const char *pin);
+int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component,
+					  const char *pin);
+int snd_soc_component_disable_pin(struct snd_soc_component *component,
+				  const char *pin);
+int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component,
+					   const char *pin);
+int snd_soc_component_nc_pin(struct snd_soc_component *component,
+			     const char *pin);
+int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component,
+				      const char *pin);
+int snd_soc_component_get_pin_status(struct snd_soc_component *component,
+				     const char *pin);
+int snd_soc_component_force_enable_pin(struct snd_soc_component *component,
+				       const char *pin);
+int snd_soc_component_force_enable_pin_unlocked(
+					struct snd_soc_component *component,
+					const char *pin);
+
 #endif
diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h
index 819d895edfdc..6702533c8bd8 100644
--- a/include/uapi/sound/asoc.h
+++ b/include/uapi/sound/asoc.h
@@ -33,6 +33,11 @@
  */
 #define SND_SOC_TPLG_STREAM_CONFIG_MAX  8
 
+/*
+ * Maximum number of physical link's hardware configs
+ */
+#define SND_SOC_TPLG_HW_CONFIG_MAX	8
+
 /* individual kcontrol info types - can be mixed with other types */
 #define SND_SOC_TPLG_CTL_VOLSW		1
 #define SND_SOC_TPLG_CTL_VOLSW_SX	2
@@ -77,7 +82,8 @@
 #define SND_SOC_TPLG_NUM_TEXTS		16
 
 /* ABI version */
-#define SND_SOC_TPLG_ABI_VERSION	0x5
+#define SND_SOC_TPLG_ABI_VERSION	0x5	/* current version */
+#define SND_SOC_TPLG_ABI_VERSION_MIN	0x4	/* oldest version supported */
 
 /* Max size of TLV data */
 #define SND_SOC_TPLG_TLV_SIZE		32
@@ -99,8 +105,8 @@
 #define SND_SOC_TPLG_TYPE_CODEC_LINK	9
 #define SND_SOC_TPLG_TYPE_BACKEND_LINK	10
 #define SND_SOC_TPLG_TYPE_PDATA		11
-#define SND_SOC_TPLG_TYPE_BE_DAI	12
-#define SND_SOC_TPLG_TYPE_MAX		SND_SOC_TPLG_TYPE_BE_DAI
+#define SND_SOC_TPLG_TYPE_DAI		12
+#define SND_SOC_TPLG_TYPE_MAX		SND_SOC_TPLG_TYPE_DAI
 
 /* vendor block IDs - please add new vendor types to end */
 #define SND_SOC_TPLG_TYPE_VENDOR_FW	1000
@@ -119,11 +125,32 @@
 #define SND_SOC_TPLG_TUPLE_TYPE_WORD	4
 #define SND_SOC_TPLG_TUPLE_TYPE_SHORT	5
 
-/* BE DAI flags */
+/* DAI flags */
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_RATES         (1 << 0)
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_CHANNELS      (1 << 1)
 #define SND_SOC_TPLG_DAI_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
 
+/* DAI physical PCM data formats.
+ * Add new formats to the end of the list.
+ */
+#define SND_SOC_DAI_FORMAT_I2S          1 /* I2S mode */
+#define SND_SOC_DAI_FORMAT_RIGHT_J      2 /* Right Justified mode */
+#define SND_SOC_DAI_FORMAT_LEFT_J       3 /* Left Justified mode */
+#define SND_SOC_DAI_FORMAT_DSP_A        4 /* L data MSB after FRM LRC */
+#define SND_SOC_DAI_FORMAT_DSP_B        5 /* L data MSB during FRM LRC */
+#define SND_SOC_DAI_FORMAT_AC97         6 /* AC97 */
+#define SND_SOC_DAI_FORMAT_PDM          7 /* Pulse density modulation */
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAI_FORMAT_MSB          SND_SOC_DAI_FORMAT_LEFT_J
+#define SND_SOC_DAI_FORMAT_LSB          SND_SOC_DAI_FORMAT_RIGHT_J
+
+/* DAI link flags */
+#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_RATES         (1 << 0)
+#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_CHANNELS      (1 << 1)
+#define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS    (1 << 2)
+#define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP            (1 << 3)
+
 /*
  * Block Header.
  * This header precedes all object and object arrays below.
@@ -267,6 +294,35 @@ struct snd_soc_tplg_stream {
 	__le32 channels;	/* channels */
 } __attribute__((packed));
 
+
+/*
+ * Describes a physical link's runtime supported hardware config,
+ * i.e. hardware audio formats.
+ */
+struct snd_soc_tplg_hw_config {
+	__le32 size;            /* in bytes of this structure */
+	__le32 id;		/* unique ID - - used to match */
+	__le32 fmt;		/* SND_SOC_DAI_FORMAT_ format value */
+	__u8 clock_gated;	/* 1 if clock can be gated to save power */
+	__u8 invert_bclk;	/* 1 for inverted BCLK, 0 for normal */
+	__u8 invert_fsync;	/* 1 for inverted frame clock, 0 for normal */
+	__u8 bclk_master;	/* 1 for master of BCLK, 0 for slave */
+	__u8 fsync_master;	/* 1 for master of FSYNC, 0 for slave */
+	__u8 mclk_direction;    /* 0 for input, 1 for output */
+	__le16 reserved;	/* for 32bit alignment */
+	__le32 mclk_rate;	/* MCLK or SYSCLK freqency in Hz */
+	__le32 bclk_rate;	/* BCLK freqency in Hz */
+	__le32 fsync_rate;	/* frame clock in Hz */
+	__le32 tdm_slots;	/* number of TDM slots in use */
+	__le32 tdm_slot_width;	/* width in bits for each slot */
+	__le32 tx_slots;	/* bit mask for active Tx slots */
+	__le32 rx_slots;	/* bit mask for active Rx slots */
+	__le32 tx_channels;	/* number of Tx channels */
+	__le32 tx_chanmap[SND_SOC_TPLG_MAX_CHAN]; /* array of slot number */
+	__le32 rx_channels;	/* number of Rx channels */
+	__le32 rx_chanmap[SND_SOC_TPLG_MAX_CHAN]; /* array of slot number */
+} __attribute__((packed));
+
 /*
  * Manifest. List totals for each payload type. Not used in parsing, but will
  * be passed to the component driver before any other objects in order for any
@@ -286,7 +342,7 @@ struct snd_soc_tplg_manifest {
 	__le32 graph_elems;	/* number of graph elements */
 	__le32 pcm_elems;	/* number of PCM elements */
 	__le32 dai_link_elems;	/* number of DAI link elements */
-	__le32 be_dai_elems;	/* number of BE DAI elements */
+	__le32 dai_elems;	/* number of physical DAI elements */
 	__le32 reserved[20];	/* reserved for new ABI element types */
 	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
@@ -434,13 +490,16 @@ struct snd_soc_tplg_pcm {
 	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* for DAI link */
 	__le32 num_streams;	/* number of streams */
 	struct snd_soc_tplg_stream_caps caps[2]; /* playback and capture for DAI */
+	__le32 flag_mask;       /* bitmask of flags to configure */
+	__le32 flags;           /* SND_SOC_TPLG_LNK_FLGBIT_* flag value */
+	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
 
 
 /*
- * Describes the BE or CC link runtime supported configs or params
+ * Describes the physical link runtime supported configs or params
  *
- * File block representation for BE/CC link config :-
+ * File block representation for physical link config :-
  * +-----------------------------------+-----+
  * | struct snd_soc_tplg_hdr           |  1  |
  * +-----------------------------------+-----+
@@ -450,21 +509,30 @@ struct snd_soc_tplg_pcm {
 struct snd_soc_tplg_link_config {
 	__le32 size;            /* in bytes of this structure */
 	__le32 id;              /* unique ID - used to match */
+	char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
+	char stream_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* stream name - used to match */
 	struct snd_soc_tplg_stream stream[SND_SOC_TPLG_STREAM_CONFIG_MAX]; /* supported configs playback and captrure */
 	__le32 num_streams;     /* number of streams */
+	struct snd_soc_tplg_hw_config hw_config[SND_SOC_TPLG_HW_CONFIG_MAX]; /* hw configs */
+	__le32 num_hw_configs;         /* number of hw configs */
+	__le32 default_hw_config_id;   /* default hw config ID for init */
+	__le32 flag_mask;       /* bitmask of flags to configure */
+	__le32 flags;           /* SND_SOC_TPLG_LNK_FLGBIT_* flag value */
+	struct snd_soc_tplg_private priv;
 } __attribute__((packed));
 
 /*
- * Describes SW/FW specific features of BE DAI.
+ * Describes SW/FW specific features of physical DAI.
+ * It can be used to configure backend DAIs for DPCM.
  *
- * File block representation for BE DAI :-
+ * File block representation for physical DAI :-
  * +-----------------------------------+-----+
  * | struct snd_soc_tplg_hdr           |  1  |
  * +-----------------------------------+-----+
- * | struct snd_soc_tplg_be_dai        |  N  |
+ * | struct snd_soc_tplg_dai           |  N  |
  * +-----------------------------------+-----+
  */
-struct snd_soc_tplg_be_dai {
+struct snd_soc_tplg_dai {
 	__le32 size;            /* in bytes of this structure */
 	char dai_name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* name - used to match */
 	__le32 dai_id;          /* unique ID - used to match */
diff --git a/include/uapi/sound/snd_sst_tokens.h b/include/uapi/sound/snd_sst_tokens.h
index 1ee2e943d66a..93392bedcc58 100644
--- a/include/uapi/sound/snd_sst_tokens.h
+++ b/include/uapi/sound/snd_sst_tokens.h
@@ -157,6 +157,10 @@
  *
  * %SKL_TKN_STR_LIB_NAME:       Specifies the library name
  *
+ * %SKL_TKN_U32_PMODE:		Specifies the power mode for pipe
+ *
+ * %SKL_TKL_U32_D0I3_CAPS:	Specifies the D0i3 capability for module
+ *
  * module_id and loadable flags dont have tokens as these values will be
  * read from the DSP FW manifest
  */
@@ -208,7 +212,9 @@ enum SKL_TKNS {
 	SKL_TKN_U32_PROC_DOMAIN,
 	SKL_TKN_U32_LIB_COUNT,
 	SKL_TKN_STR_LIB_NAME,
-	SKL_TKN_MAX = SKL_TKN_STR_LIB_NAME,
+	SKL_TKN_U32_PMODE,
+	SKL_TKL_U32_D0I3_CAPS,
+	SKL_TKN_MAX = SKL_TKL_U32_D0I3_CAPS,
 };
 
 #endif