Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

16 files changed, 128 insertions, 99 deletions

diff --git a/include/dt-bindings/sound/qcom,q6asm.h b/include/dt-bindings/sound/qcom,q6asm.h
index 1eb77d87c2e8..f59d74f14395 100644
--- a/include/dt-bindings/sound/qcom,q6asm.h
+++ b/include/dt-bindings/sound/qcom,q6asm.h
@@ -19,4 +19,8 @@
 #define	MSM_FRONTEND_DAI_MULTIMEDIA15	14
 #define	MSM_FRONTEND_DAI_MULTIMEDIA16	15
 
+#define Q6ASM_DAI_TX_RX	0
+#define Q6ASM_DAI_TX	1
+#define Q6ASM_DAI_RX	2
+
 #endif /* __DT_BINDINGS_Q6_ASM_H__ */
diff --git a/include/sound/control.h b/include/sound/control.h
index aeaed2a05bae..e128cff10dfa 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -188,20 +188,21 @@ int snd_ctl_enum_info(struct snd_ctl_elem_info *info, unsigned int channels,
  */
 struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 						 const unsigned int *tlv);
-int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
-		       unsigned int flags);
-/* optional flags for slave */
-#define SND_CTL_SLAVE_NEED_UPDATE	(1 << 0)
+int _snd_ctl_add_follower(struct snd_kcontrol *master,
+			  struct snd_kcontrol *follower,
+			  unsigned int flags);
+/* optional flags for follower */
+#define SND_CTL_FOLLOWER_NEED_UPDATE	(1 << 0)
 
 /**
- * snd_ctl_add_slave - Add a virtual slave control
+ * snd_ctl_add_follower - Add a virtual follower control
  * @master: vmaster element
- * @slave: slave element to add
+ * @follower: follower element to add
  *
- * Add a virtual slave control to the given master element created via
+ * Add a virtual follower control to the given master element created via
  * snd_ctl_create_virtual_master() beforehand.
  *
- * All slaves must be the same type (returning the same information
+ * All followers must be the same type (returning the same information
  * via info callback).  The function doesn't check it, so it's your
  * responsibility.
  *
@@ -213,18 +214,18 @@ int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
  * Return: Zero if successful or a negative error code.
  */
 static inline int
-snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+snd_ctl_add_follower(struct snd_kcontrol *master, struct snd_kcontrol *follower)
 {
-	return _snd_ctl_add_slave(master, slave, 0);
+	return _snd_ctl_add_follower(master, follower, 0);
 }
 
 /**
- * snd_ctl_add_slave_uncached - Add a virtual slave control
+ * snd_ctl_add_follower_uncached - Add a virtual follower control
  * @master: vmaster element
- * @slave: slave element to add
+ * @follower: follower element to add
  *
- * Add a virtual slave control to the given master.
- * Unlike snd_ctl_add_slave(), the element added via this function
+ * Add a virtual follower control to the given master.
+ * Unlike snd_ctl_add_follower(), the element added via this function
  * is supposed to have volatile values, and get callback is called
  * at each time queried from the master.
  *
@@ -235,10 +236,10 @@ snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
  * Return: Zero if successful or a negative error code.
  */
 static inline int
-snd_ctl_add_slave_uncached(struct snd_kcontrol *master,
-			   struct snd_kcontrol *slave)
+snd_ctl_add_follower_uncached(struct snd_kcontrol *master,
+			      struct snd_kcontrol *follower)
 {
-	return _snd_ctl_add_slave(master, slave, SND_CTL_SLAVE_NEED_UPDATE);
+	return _snd_ctl_add_follower(master, follower, SND_CTL_FOLLOWER_NEED_UPDATE);
 }
 
 int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
@@ -246,11 +247,11 @@ int snd_ctl_add_vmaster_hook(struct snd_kcontrol *kctl,
 			     void *private_data);
 void snd_ctl_sync_vmaster(struct snd_kcontrol *kctl, bool hook_only);
 #define snd_ctl_sync_vmaster_hook(kctl)	snd_ctl_sync_vmaster(kctl, true)
-int snd_ctl_apply_vmaster_slaves(struct snd_kcontrol *kctl,
-				 int (*func)(struct snd_kcontrol *vslave,
-					     struct snd_kcontrol *slave,
-					     void *arg),
-				 void *arg);
+int snd_ctl_apply_vmaster_followers(struct snd_kcontrol *kctl,
+				    int (*func)(struct snd_kcontrol *vfollower,
+						struct snd_kcontrol *follower,
+						void *arg),
+				    void *arg);
 
 /*
  * Helper functions for jack-detection controls
diff --git a/include/sound/gus.h b/include/sound/gus.h
index 410939ecf3a5..cd8da68cab92 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -613,4 +613,8 @@ int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
 int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
 		      unsigned int addr, unsigned int size, int rom);
 
+/* gus_timer.c */
+void snd_gf1_timers_init(struct snd_gus_card *gus);
+void snd_gf1_timers_done(struct snd_gus_card *gus);
+
 #endif /* __SOUND_GUS_H */
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index d16a4229209b..0fea49bfc5e8 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -208,7 +208,7 @@ struct hda_codec {
 	struct mutex control_mutex;
 	struct snd_array spdif_out;
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
-	const hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+	const hda_nid_t *follower_dig_outs; /* optional digital out follower widgets */
 	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
 	struct snd_array driver_pins;	/* pin configs set by codec parser */
 	struct snd_array cvt_setups;	/* audio convert setups */
@@ -415,6 +415,8 @@ __printf(2, 3)
 struct hda_pcm *snd_hda_codec_pcm_new(struct hda_codec *codec,
 				      const char *fmt, ...);
 
+void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec);
+
 static inline void snd_hda_codec_pcm_get(struct hda_pcm *pcm)
 {
 	kref_get(&pcm->kref);
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index c1f78d9a6e47..6eed61e6cf8a 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -347,6 +347,9 @@ struct hdac_bus {
 
 	int bdl_pos_adj;		/* BDL position adjustment */
 
+	/* delay time in us for dma stop */
+	unsigned int dma_stop_delay;
+
 	/* locks */
 	spinlock_t reg_lock;
 	struct mutex cmd_mutex;
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 83b17682e01c..7754631a3102 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -2,7 +2,7 @@
 /*
  * hdmi-codec.h - HDMI Codec driver API
  *
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
  *
  * Author: Jyri Sarha <jsarha@ti.com>
  */
@@ -76,7 +76,8 @@ struct hdmi_codec_ops {
 	 * Mute/unmute HDMI audio stream.
 	 * Optional
 	 */
-	int (*digital_mute)(struct device *dev, void *data, bool enable);
+	int (*mute_stream)(struct device *dev, void *data,
+			   bool enable, int direction);
 
 	/*
 	 * Provides EDID-Like-Data from connected HDMI device.
@@ -99,6 +100,9 @@ struct hdmi_codec_ops {
 	int (*hook_plugged_cb)(struct device *dev, void *data,
 			       hdmi_codec_plugged_cb fn,
 			       struct device *codec_dev);
+
+	/* bit field */
+	unsigned int no_capture_mute:1;
 };
 
 /* HDMI codec initalization data */
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index 3b47832b1c1f..5daa937684a4 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -94,7 +94,11 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
 					   size_t offset)
 {
 	struct snd_sg_buf *sgbuf = dmab->private_data;
-	dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
+	dma_addr_t addr;
+
+	if (!sgbuf)
+		return dmab->addr + offset;
+	addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
 	addr &= ~((dma_addr_t)PAGE_SIZE - 1);
 	return addr + offset % PAGE_SIZE;
 }
@@ -106,6 +110,9 @@ static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
 				     size_t offset)
 {
 	struct snd_sg_buf *sgbuf = dmab->private_data;
+
+	if (!sgbuf)
+		return dmab->area + offset;
 	return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
 }
 
diff --git a/include/sound/omap-hdmi-audio.h b/include/sound/omap-hdmi-audio.h
index 16c007b651f4..e5f82044a404 100644
--- a/include/sound/omap-hdmi-audio.h
+++ b/include/sound/omap-hdmi-audio.h
@@ -2,7 +2,7 @@
 /*
  * hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
  *
- * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
  *
  * Author: Jyri Sarha <jsarha@ti.com>
  */
diff --git a/include/sound/rt5670.h b/include/sound/rt5670.h
deleted file mode 100644
index 02e1d7778354..000000000000
--- a/include/sound/rt5670.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * linux/sound/rt5670.h -- Platform data for RT5670
- *
- * Copyright 2014 Realtek Microelectronics
- */
-
-#ifndef __LINUX_SND_RT5670_H
-#define __LINUX_SND_RT5670_H
-
-struct rt5670_platform_data {
-	int jd_mode;
-	bool in2_diff;
-	bool dev_gpio;
-	bool gpio1_is_ext_spk_en;
-
-	bool dmic_en;
-	unsigned int dmic1_data_pin;
-	/* 0 = GPIO6; 1 = IN2P; 3 = GPIO7*/
-	unsigned int dmic2_data_pin;
-	/* 0 = GPIO8; 1 = IN3N; */
-	unsigned int dmic3_data_pin;
-	/* 0 = GPIO9; 1 = GPIO10; 2 = GPIO5*/
-};
-
-#endif
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index bbdd1542d6f1..86a1e956991e 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -12,9 +12,9 @@
 #include <sound/soc.h>
 
 #define asoc_simple_init_hp(card, sjack, prefix) \
-	asoc_simple_init_jack(card, sjack, 1, prefix)
+	asoc_simple_init_jack(card, sjack, 1, prefix, NULL)
 #define asoc_simple_init_mic(card, sjack, prefix) \
-	asoc_simple_init_jack(card, sjack, 0, prefix)
+	asoc_simple_init_jack(card, sjack, 0, prefix, NULL)
 
 struct asoc_simple_dai {
 	const char *name;
@@ -131,7 +131,7 @@ int asoc_simple_parse_pin_switches(struct snd_soc_card *card,
 
 int asoc_simple_init_jack(struct snd_soc_card *card,
 			       struct asoc_simple_jack *sjack,
-			       int is_hp, char *prefix);
+			       int is_hp, char *prefix, char *pin);
 int asoc_simple_init_priv(struct asoc_simple_priv *priv,
 			       struct link_info *li);
 
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 5663891148e3..089ea9441fd1 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -2,7 +2,8 @@
  *
  * soc-component.h
  *
- * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ * Copyright (C) 2019 Renesas Electronics Corp.
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
  */
 #ifndef __SOC_COMPONENT_H
 #define __SOC_COMPONENT_H
@@ -324,10 +325,12 @@ static inline int snd_soc_component_cache_sync(
 	return regcache_sync(component->regmap);
 }
 
+void snd_soc_component_set_aux(struct snd_soc_component *component,
+			       struct snd_soc_aux_dev *aux);
+int snd_soc_component_init(struct snd_soc_component *component);
+
 /* component IO */
-int snd_soc_component_read(struct snd_soc_component *component,
-			   unsigned int reg, unsigned int *val);
-unsigned int snd_soc_component_read32(struct snd_soc_component *component,
+unsigned int snd_soc_component_read(struct snd_soc_component *component,
 				      unsigned int reg);
 int snd_soc_component_write(struct snd_soc_component *component,
 			    unsigned int reg, unsigned int val);
@@ -359,6 +362,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
 int snd_soc_component_set_bias_level(struct snd_soc_component *component,
 				     enum snd_soc_bias_level level);
 
+void snd_soc_component_setup_regmap(struct snd_soc_component *component);
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
 				   struct regmap *regmap);
@@ -421,16 +425,6 @@ int snd_soc_component_open(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream);
 int snd_soc_component_close(struct snd_soc_component *component,
 			    struct snd_pcm_substream *substream);
-int snd_soc_component_prepare(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream);
-int snd_soc_component_hw_params(struct snd_soc_component *component,
-				struct snd_pcm_substream *substream,
-				struct snd_pcm_hw_params *params);
-int snd_soc_component_hw_free(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream);
-int snd_soc_component_trigger(struct snd_soc_component *component,
-			      struct snd_pcm_substream *substream,
-			      int cmd);
 void snd_soc_component_suspend(struct snd_soc_component *component);
 void snd_soc_component_resume(struct snd_soc_component *component);
 int snd_soc_component_is_suspended(struct snd_soc_component *component);
@@ -455,5 +449,13 @@ int snd_soc_pcm_component_mmap(struct snd_pcm_substream *substream,
 			       struct vm_area_struct *vma);
 int snd_soc_pcm_component_new(struct snd_soc_pcm_runtime *rtd);
 void snd_soc_pcm_component_free(struct snd_soc_pcm_runtime *rtd);
+int snd_soc_pcm_component_prepare(struct snd_pcm_substream *substream);
+int snd_soc_pcm_component_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_component **last);
+void snd_soc_pcm_component_hw_free(struct snd_pcm_substream *substream,
+				   struct snd_soc_component *last);
+int snd_soc_pcm_component_trigger(struct snd_pcm_substream *substream,
+				  int cmd);
 
 #endif /* __SOC_COMPONENT_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 71e178c89793..776a60529e70 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -39,7 +39,7 @@ struct snd_compr_stream;
 /*
  * DAI Clock gating.
  *
- * DAI bit clocks can be be gated (disabled) when the DAI is not
+ * DAI bit clocks can be gated (disabled) when the DAI is not
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(1 << 4) /* continuous clock */
@@ -76,12 +76,12 @@ struct snd_compr_stream;
  *
  * This is wrt the codec, the inverse is true for the interface
  * i.e. if the codec is clk and FRM master then the interface is
- * clk and frame slave.
+ * clk and frame secondary.
  */
 #define SND_SOC_DAIFMT_CBM_CFM		(1 << 12) /* codec clk & FRM master */
-#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk slave & FRM master */
-#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM slave */
+#define SND_SOC_DAIFMT_CBS_CFM		(2 << 12) /* codec clk secondary & FRM master */
+#define SND_SOC_DAIFMT_CBM_CFS		(3 << 12) /* codec clk master & frame secondary */
+#define SND_SOC_DAIFMT_CBS_CFS		(4 << 12) /* codec clk & FRM secondary */
 
 #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
@@ -247,7 +247,6 @@ struct snd_soc_dai_ops {
 	 * DAI digital mute - optional.
 	 * 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);
 
 	/*
@@ -281,6 +280,9 @@ struct snd_soc_dai_ops {
 	 */
 	snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
 		struct snd_soc_dai *);
+
+	/* bit field */
+	unsigned int no_capture_mute:1;
 };
 
 struct snd_soc_cdai_ops {
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index cc3dcb815282..c3039e97929a 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -16,6 +16,8 @@
 #include <sound/asoc.h>
 
 struct device;
+struct snd_soc_pcm_runtime;
+struct soc_enum;
 
 /* widget has no PM register bit */
 #define SND_SOC_NOPM	-1
@@ -376,6 +378,24 @@ struct snd_soc_dapm_widget_list;
 struct snd_soc_dapm_update;
 enum snd_soc_dapm_direction;
 
+/*
+ * Bias levels
+ *
+ * @ON:      Bias is fully on for audio playback and capture operations.
+ * @PREPARE: Prepare for audio operations. Called before DAPM switching for
+ *           stream start and stop operations.
+ * @STANDBY: Low power standby state when no playback/capture operations are
+ *           in progress. NOTE: The transition time between STANDBY and ON
+ *           should be as fast as possible and no longer than 10ms.
+ * @OFF:     Power Off. No restrictions on transition times.
+ */
+enum snd_soc_bias_level {
+	SND_SOC_BIAS_OFF = 0,
+	SND_SOC_BIAS_STANDBY = 1,
+	SND_SOC_BIAS_PREPARE = 2,
+	SND_SOC_BIAS_ON = 3,
+};
+
 int dapm_regulator_event(struct snd_soc_dapm_widget *w,
 			 struct snd_kcontrol *kcontrol, int event);
 int dapm_clock_event(struct snd_soc_dapm_widget *w,
diff --git a/include/sound/soc-link.h b/include/sound/soc-link.h
index 3dd6e33e94ec..337ac5666757 100644
--- a/include/sound/soc-link.h
+++ b/include/sound/soc-link.h
@@ -9,6 +9,7 @@
 #define __SOC_LINK_H
 
 int snd_soc_link_init(struct snd_soc_pcm_runtime *rtd);
+void snd_soc_link_exit(struct snd_soc_pcm_runtime *rtd);
 int snd_soc_link_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
 				    struct snd_pcm_hw_params *params);
 
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3ce7f0f5aa92..5e3919ffb00c 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -368,24 +368,6 @@
 #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \
 	const struct soc_enum name = SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(xtexts), xtexts)
 
-/*
- * Bias levels
- *
- * @ON:      Bias is fully on for audio playback and capture operations.
- * @PREPARE: Prepare for audio operations. Called before DAPM switching for
- *           stream start and stop operations.
- * @STANDBY: Low power standby state when no playback/capture operations are
- *           in progress. NOTE: The transition time between STANDBY and ON
- *           should be as fast as possible and no longer than 10ms.
- * @OFF:     Power Off. No restrictions on transition times.
- */
-enum snd_soc_bias_level {
-	SND_SOC_BIAS_OFF = 0,
-	SND_SOC_BIAS_STANDBY = 1,
-	SND_SOC_BIAS_PREPARE = 2,
-	SND_SOC_BIAS_ON = 3,
-};
-
 struct device_node;
 struct snd_jack;
 struct snd_soc_card;
@@ -432,11 +414,12 @@ static inline int snd_soc_resume(struct device *dev)
 }
 #endif
 int snd_soc_poweroff(struct device *dev);
-int snd_soc_add_component(struct device *dev,
-		struct snd_soc_component *component,
-		const struct snd_soc_component_driver *component_driver,
-		struct snd_soc_dai_driver *dai_drv,
-		int num_dai);
+int snd_soc_component_initialize(struct snd_soc_component *component,
+				 const struct snd_soc_component_driver *driver,
+				 struct device *dev);
+int snd_soc_add_component(struct snd_soc_component *component,
+			  struct snd_soc_dai_driver *dai_drv,
+			  int num_dai);
 int snd_soc_register_component(struct device *dev,
 			 const struct snd_soc_component_driver *component_driver,
 			 struct snd_soc_dai_driver *dai_drv, int num_dai);
@@ -801,6 +784,9 @@ struct snd_soc_dai_link {
 	/* codec/machine specific init - e.g. add machine controls */
 	int (*init)(struct snd_soc_pcm_runtime *rtd);
 
+	/* codec/machine specific exit - dual of init() */
+	void (*exit)(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);
@@ -1183,6 +1169,8 @@ struct snd_soc_pcm_runtime {
 /* see soc_new_pcm_runtime()  */
 #define asoc_rtd_to_cpu(rtd, n)   (rtd)->dais[n]
 #define asoc_rtd_to_codec(rtd, n) (rtd)->dais[n + (rtd)->num_cpus]
+#define asoc_substream_to_rtd(substream) \
+	(struct snd_soc_pcm_runtime *)snd_pcm_substream_chip(substream)
 
 #define for_each_rtd_components(rtd, i, component)			\
 	for ((i) = 0, component = NULL;					\
diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
index d22e84805025..275fd5b201ce 100644
--- a/include/sound/wm8960.h
+++ b/include/sound/wm8960.h
@@ -16,6 +16,23 @@ struct wm8960_data {
 	bool capless;  /* Headphone outputs configured in capless mode */
 
 	bool shared_lrclk;  /* DAC and ADC LRCLKs are wired together */
+
+	/*
+	 * Setup for headphone detection
+	 *
+	 * hp_cfg[0]: HPSEL[1:0] of R48 (Additional Control 4)
+	 * hp_cfg[1]: {HPSWEN:HPSWPOL} of R24 (Additional Control 2).
+	 * hp_cfg[2]: {TOCLKSEL:TOEN} of R23 (Additional Control 1).
+	 */
+	u32 hp_cfg[3];
+
+	/*
+	 * Setup for gpio configuration
+	 *
+	 * gpio_cfg[0]: ALRCGPIO of R9 (Audio interface)
+	 * gpio_cfg[1]: {GPIOPOL:GPIOSEL[2:0]} of R48 (Additional Control 4).
+	 */
+	u32 gpio_cfg[2];
 };
 
 #endif