Merge tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "We had again a busy development cycle with many new drivers as well as
  lots of core improvements / cleanups. Let's go for highlights:

  ALSA core:

   - PCM locking scheme was refactored for reducing a global rwlock

   - PCM suspend is handled in the device type PM ops now; lots of
     explicit calls were reduced by this action

   - Cleanups about PCM buffer preallocation calls

   - Kill NULL device object in memory allocations

   - Lots of procfs API cleanups

  ASoC core:

   - Support for only powering up channels that are actively being used

   - Cleanups / fixes of topology API

  ASoC drivers:

   - MediaTek BTCVSD for a Bluetooth radio chip, which is the first such
     driver we've had upstream!

   - Quite a few improvements to simplify the generic card drivers,
     especially the merge of the SCU cards into the main generic drivers

   - Lots of fixes for probing on Intel systems to follow more standard
     styles

   - A big refresh and cleanup of the Samsung drivers

   - New drivers: Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341
     and CS35L26, Google ChromeOS embedded controllers, Ingenic JZ4725B,
     MediaTek BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328,
     Spreadtrum DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM
     formatters

  ALSA drivers:

   - Improvements of Tegra HD-audio controller driver for supporting new
     chips

   - HD-audio codec quirks for ALC294 S4 resume, ASUS laptop, Chrome
     headset button support and Dell workstations

   - Improved DSD support on USB-audio

   - Quirk for MOTU MicroBook II USB-audio

   - Support for Fireface UCX support and Solid State Logic Duende
     Classic/Mini"

* tag 'sound-5.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (461 commits)
  ALSA: usb-audio: Add quirk for MOTU MicroBook II
  ASoC: stm32: i2s: skip useless write in slave mode
  ASoC: stm32: i2s: fix race condition in irq handler
  ASoC: stm32: i2s: remove useless callback
  ASoC: stm32: i2s: fix dma configuration
  ASoC: stm32: i2s: fix stream count management
  ASoC: stm32: i2s: fix 16 bit format support
  ASoC: stm32: i2s: fix IRQ clearing
  ASoC: qcom: Kconfig: fix dependency for sdm845
  ASoC: Intel: Boards: Add Maxim98373 support
  ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
  ALSA: firewire-motu: fix construction of PCM frame for capture direction
  ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
  ALSA: hda: Extend i915 component bind timeout
  ASoC: wm_adsp: Improve logging messages
  ASoC: wm_adsp: Add support for multiple compressed buffers
  ASoC: wm_adsp: Refactor compress stream initialisation
  ASoC: wm_adsp: Reorder some functions for improved clarity
  ASoC: wm_adsp: Factor out stripping padding from ADSP data
  ASoC: cs35l36: Fix an IS_ERR() vs NULL checking bug
  ...

15 files changed, 218 insertions, 55 deletions

diff --git a/include/linux/acpi.h b/include/linux/acpi.h
index 87715f20b69a..03b4c4f225d0 100644
--- a/include/linux/acpi.h
+++ b/include/linux/acpi.h
@@ -1014,6 +1014,13 @@ struct acpi_gpio_mapping {
 
 /* Ignore IoRestriction field */
 #define ACPI_GPIO_QUIRK_NO_IO_RESTRICTION	BIT(0)
+/*
+ * When ACPI GPIO mapping table is in use the index parameter inside it
+ * refers to the GPIO resource in _CRS method. That index has no
+ * distinction of actual type of the resource. When consumer wants to
+ * get GpioIo type explicitly, this quirk may be used.
+ */
+#define ACPI_GPIO_QUIRK_ONLY_GPIOIO		BIT(1)
 
 	unsigned int quirks;
 };
diff --git a/include/sound/core.h b/include/sound/core.h
index 36a5934cf4b1..e923c23e05dd 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -120,7 +120,6 @@ struct snd_card {
 	struct list_head ctl_files;	/* active control files */
 
 	struct snd_info_entry *proc_root;	/* root for soundcard specific files */
-	struct snd_info_entry *proc_id;	/* the card id */
 	struct proc_dir_entry *proc_root_link;	/* number link to real id */
 
 	struct list_head files_list;	/* all files associated to this card */
diff --git a/include/sound/cs35l36.h b/include/sound/cs35l36.h
new file mode 100644
index 000000000000..8f8049d390f0
--- /dev/null
+++ b/include/sound/cs35l36.h
@@ -0,0 +1,43 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * linux/sound/cs35l36.h -- Platform data for CS35L36
+ *
+ * Copyright 2018 Cirrus Logic, Inc.
+ *
+ * Author: James Schulman <james.schulman@cirrus.com>
+ *
+ */
+
+#ifndef __CS35L36_H
+#define __CS35L36_H
+
+struct cs35l36_vpbr_cfg {
+	bool is_present;
+	bool vpbr_en;
+	int vpbr_thld;
+	int vpbr_atk_rate;
+	int vpbr_atk_vol;
+	int vpbr_max_attn;
+	int vpbr_wait;
+	int vpbr_rel_rate;
+	int vpbr_mute_en;
+};
+
+struct cs35l36_platform_data {
+	bool multi_amp_mode;
+	bool dcm_mode;
+	bool amp_pcm_inv;
+	bool imon_pol_inv;
+	bool vmon_pol_inv;
+	int boost_ind;
+	int bst_vctl;
+	int bst_vctl_sel;
+	int bst_ipk;
+	bool extern_boost;
+	int temp_warn_thld;
+	int irq_drv_sel;
+	int irq_gpio_sel;
+	struct cs35l36_vpbr_cfg vpbr_config;
+};
+
+#endif /* __CS35L36_H */
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 2c4cfaa135a6..c679f6116580 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -99,10 +99,6 @@ 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/hda_register.h b/include/sound/hda_register.h
index 2ab39fb52d7a..0fd39295b426 100644
--- a/include/sound/hda_register.h
+++ b/include/sound/hda_register.h
@@ -79,6 +79,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 
 /* stream register offsets from stream base */
 #define AZX_REG_SD_CTL			0x00
+#define AZX_REG_SD_CTL_3B		0x02 /* 3rd byte of SD_CTL register */
 #define AZX_REG_SD_STS			0x03
 #define AZX_REG_SD_LPIB			0x04
 #define AZX_REG_SD_CBL			0x08
@@ -165,6 +166,7 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
 #define SD_INT_COMPLETE		0x04	/* completion interrupt */
 #define SD_INT_MASK		(SD_INT_DESC_ERR|SD_INT_FIFO_ERR|\
 				 SD_INT_COMPLETE)
+#define SD_CTL_STRIPE_MASK	0x3	/* stripe control mask */
 
 /* SD_STS */
 #define SD_STS_FIFO_READY	0x20	/* FIFO ready */
diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h
index 2a8573a00ea6..e36b77531c5c 100644
--- a/include/sound/hda_verbs.h
+++ b/include/sound/hda_verbs.h
@@ -66,6 +66,7 @@ enum {
 #define AC_VERB_GET_CONFIG_DEFAULT		0x0f1c
 /* f20: AFG/MFG */
 #define AC_VERB_GET_SUBSYSTEM_ID		0x0f20
+#define AC_VERB_GET_STRIPE_CONTROL		0x0f24
 #define AC_VERB_GET_CVT_CHAN_COUNT		0x0f2d
 #define AC_VERB_GET_HDMI_DIP_SIZE		0x0f2e
 #define AC_VERB_GET_HDMI_ELDD			0x0f2f
@@ -110,6 +111,7 @@ enum {
 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3	0x71f
 #define AC_VERB_SET_EAPD				0x788
 #define AC_VERB_SET_CODEC_RESET			0x7ff
+#define AC_VERB_SET_STRIPE_CONTROL		0x724
 #define AC_VERB_SET_CVT_CHAN_COUNT		0x72d
 #define AC_VERB_SET_HDMI_DIP_INDEX		0x730
 #define AC_VERB_SET_HDMI_DIP_DATA		0x731
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index b4fa1c775251..45f944d57982 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -539,6 +539,9 @@ void snd_hdac_stream_sync(struct hdac_stream *azx_dev, bool start,
 			  unsigned int streams);
 void snd_hdac_stream_timecounter_init(struct hdac_stream *azx_dev,
 				      unsigned int streams);
+int snd_hdac_get_stream_stripe_ctl(struct hdac_bus *bus,
+				struct snd_pcm_substream *substream);
+
 /*
  * macros for easy use
  */
diff --git a/include/sound/info.h b/include/sound/info.h
index becdf66d2825..97fdda41e076 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -82,7 +82,6 @@ struct snd_info_entry {
 		struct snd_info_entry_ops *ops;
 	} c;
 	struct snd_info_entry *parent;
-	struct snd_card *card;
 	struct module *module;
 	void *private_data;
 	void (*private_free)(struct snd_info_entry *entry);
@@ -160,6 +159,13 @@ static inline void snd_info_set_text_ops(struct snd_info_entry *entry,
 	entry->c.text.read = read;
 }
 
+int snd_card_rw_proc_new(struct snd_card *card, const char *name,
+			 void *private_data,
+			 void (*read)(struct snd_info_entry *,
+				      struct snd_info_buffer *),
+			 void (*write)(struct snd_info_entry *entry,
+				       struct snd_info_buffer *buffer));
+
 int snd_info_check_reserved_words(const char *str);
 
 #else
@@ -189,10 +195,38 @@ static inline int snd_card_proc_new(struct snd_card *card, const char *name,
 static inline void snd_info_set_text_ops(struct snd_info_entry *entry __attribute__((unused)),
 					 void *private_data,
 					 void (*read)(struct snd_info_entry *, struct snd_info_buffer *)) {}
+static inline int snd_card_rw_proc_new(struct snd_card *card, const char *name,
+				       void *private_data,
+				       void (*read)(struct snd_info_entry *,
+						    struct snd_info_buffer *),
+				       void (*write)(struct snd_info_entry *entry,
+						     struct snd_info_buffer *buffer))
+{
+	return 0;
+}
 static inline int snd_info_check_reserved_words(const char *str) { return 1; }
 
 #endif
 
+/**
+ * snd_card_ro_proc_new - Create a read-only text proc file entry for the card
+ * @card: the card instance
+ * @name: the file name
+ * @private_data: the arbitrary private data
+ * @read: the read callback
+ *
+ * This proc file entry will be registered via snd_card_register() call, and
+ * it will be removed automatically at the card removal, too.
+ */
+static inline int
+snd_card_ro_proc_new(struct snd_card *card, const char *name,
+		     void *private_data,
+		     void (*read)(struct snd_info_entry *,
+				  struct snd_info_buffer *))
+{
+	return snd_card_rw_proc_new(card, name, private_data, read, NULL);
+}
+
 /*
  * OSS info part
  */
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index af3fa577fa06..1ac0dd82a916 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -37,7 +37,6 @@ struct snd_dma_device {
 };
 
 #define snd_dma_pci_data(pci)	(&(pci)->dev)
-#define snd_dma_isa_data()	NULL
 #define snd_dma_continuous_data(x)	((struct device *)(__force unsigned long)(x))
 
 
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index d6bd3caf6878..465d7d033c4c 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -30,6 +30,7 @@
 #include <linux/mm.h>
 #include <linux/bitops.h>
 #include <linux/pm_qos.h>
+#include <linux/refcount.h>
 
 #define snd_pcm_substream_chip(substream) ((substream)->private_data)
 #define snd_pcm_chip(pcm) ((pcm)->private_data)
@@ -439,7 +440,7 @@ struct snd_pcm_group {		/* keep linked substreams */
 	spinlock_t lock;
 	struct mutex mutex;
 	struct list_head substreams;
-	int count;
+	refcount_t refs;
 };
 
 struct pid;
@@ -470,7 +471,6 @@ struct snd_pcm_substream {
 	struct snd_pcm_group self_group;	/* fake group for non linked substream (with substream lock inside) */
 	struct snd_pcm_group *group;		/* pointer to current group */
 	/* -- assigned files -- */
-	void *file;
 	int ref_count;
 	atomic_t mmap_count;
 	unsigned int f_flags;
@@ -482,15 +482,6 @@ struct snd_pcm_substream {
 #endif
 #ifdef CONFIG_SND_VERBOSE_PROCFS
 	struct snd_info_entry *proc_root;
-	struct snd_info_entry *proc_info_entry;
-	struct snd_info_entry *proc_hw_params_entry;
-	struct snd_info_entry *proc_sw_params_entry;
-	struct snd_info_entry *proc_status_entry;
-	struct snd_info_entry *proc_prealloc_entry;
-	struct snd_info_entry *proc_prealloc_max_entry;
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-	struct snd_info_entry *proc_xrun_injection_entry;
-#endif
 #endif /* CONFIG_SND_VERBOSE_PROCFS */
 	/* misc flags */
 	unsigned int hw_opened: 1;
@@ -512,10 +503,8 @@ struct snd_pcm_str {
 #endif
 #ifdef CONFIG_SND_VERBOSE_PROCFS
 	struct snd_info_entry *proc_root;
-	struct snd_info_entry *proc_info_entry;
 #ifdef CONFIG_SND_PCM_XRUN_DEBUG
 	unsigned int xrun_debug;	/* 0 = disabled, 1 = verbose, 2 = stacktrace */
-	struct snd_info_entry *proc_xrun_debug_entry;
 #endif
 #endif
 	struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */
@@ -538,6 +527,7 @@ struct snd_pcm {
 	void (*private_free) (struct snd_pcm *pcm);
 	bool internal; /* pcm is for internal use only */
 	bool nonatomic; /* whole PCM operations are in non-atomic context */
+	bool no_device_suspend; /* don't invoke device PM suspend */
 #if IS_ENABLED(CONFIG_SND_PCM_OSS)
 	struct snd_pcm_oss oss;
 #endif
@@ -581,13 +571,8 @@ int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status);
 int snd_pcm_drain_done(struct snd_pcm_substream *substream);
 int snd_pcm_stop_xrun(struct snd_pcm_substream *substream);
 #ifdef CONFIG_PM
-int snd_pcm_suspend(struct snd_pcm_substream *substream);
 int snd_pcm_suspend_all(struct snd_pcm *pcm);
 #else
-static inline int snd_pcm_suspend(struct snd_pcm_substream *substream)
-{
-	return 0;
-}
 static inline int snd_pcm_suspend_all(struct snd_pcm *pcm)
 {
 	return 0;
@@ -1200,12 +1185,12 @@ static inline void snd_pcm_gettime(struct snd_pcm_runtime *runtime,
  *  Memory
  */
 
-int snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream);
-int snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm);
-int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
+void snd_pcm_lib_preallocate_free(struct snd_pcm_substream *substream);
+void snd_pcm_lib_preallocate_free_for_all(struct snd_pcm *pcm);
+void snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream,
 				  int type, struct device *data,
 				  size_t size, size_t max);
-int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
+void snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm,
 					  int type, void *data,
 					  size_t size, size_t max);
 int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size);
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 6d69ed2bd7b1..7afe45389972 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -75,7 +75,7 @@ void asoc_simple_card_clk_disable(struct asoc_simple_dai *dai);
 				   &dai_link->codec_dai_name,			\
 				   list_name, cells_name, NULL)
 #define asoc_simple_card_parse_platform(node, dai_link, list_name, cells_name)	\
-	asoc_simple_card_parse_dai(node, dai_link->platform,					\
+	asoc_simple_card_parse_dai(node, dai_link->platforms,			\
 		&dai_link->platform_of_node,					\
 		NULL, list_name, cells_name, NULL)
 int asoc_simple_card_parse_dai(struct device_node *node,
@@ -108,7 +108,7 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
 int asoc_simple_card_init_dai(struct snd_soc_dai *dai,
 			      struct asoc_simple_dai *simple_dai);
 
-int asoc_simple_card_canonicalize_dailink(struct snd_soc_dai_link *dai_link);
+void asoc_simple_card_canonicalize_platform(struct snd_soc_dai_link *dai_link);
 void asoc_simple_card_canonicalize_cpu(struct snd_soc_dai_link *dai_link,
 				      int is_single_links);
 
diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h
index 266e64e3c24c..35b38e41e5b2 100644
--- a/include/sound/soc-acpi.h
+++ b/include/sound/soc-acpi.h
@@ -22,20 +22,37 @@ struct snd_soc_acpi_package_context {
 #define SND_ACPI_I2C_ID_LEN (4 + ACPI_ID_LEN + 3 + 1)
 
 #if IS_ENABLED(CONFIG_ACPI)
+/* acpi match */
+struct snd_soc_acpi_mach *
+snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines);
+
 bool snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN],
 				    struct snd_soc_acpi_package_context *ctx);
+
+/* check all codecs */
+struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg);
+
 #else
+/* acpi match */
+static inline struct snd_soc_acpi_mach *
+snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines)
+{
+	return NULL;
+}
+
 static inline bool
 snd_soc_acpi_find_package_from_hid(const u8 hid[ACPI_ID_LEN],
 				   struct snd_soc_acpi_package_context *ctx)
 {
 	return false;
 }
-#endif
 
-/* acpi match */
-struct snd_soc_acpi_mach *
-snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines);
+/* check all codecs */
+static inline struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg)
+{
+	return NULL;
+}
+#endif
 
 /**
  * snd_soc_acpi_mach_params: interface for machine driver configuration
@@ -69,9 +86,6 @@ struct snd_soc_acpi_mach_params {
  *  is not constant since this field may be updated at run-time
  * @sof_fw_filename: Sound Open Firmware file name, if enabled
  * @sof_tplg_filename: Sound Open Firmware topology file name, if enabled
- * @asoc_plat_name: ASoC platform name, used for binding machine drivers
- * if non NULL
- * @new_mach_data: machine driver private data fixup
  */
 /* Descriptor for SST ASoC machine driver */
 struct snd_soc_acpi_mach {
@@ -85,8 +99,6 @@ struct snd_soc_acpi_mach {
 	struct snd_soc_acpi_mach_params mach_params;
 	const char *sof_fw_filename;
 	const char *sof_tplg_filename;
-	const char *asoc_plat_name;
-	struct platform_device * (*new_mach_data)(void *pdata);
 };
 
 #define SND_SOC_ACPI_MAX_CODECS 3
@@ -105,7 +117,4 @@ struct snd_soc_acpi_codecs {
 	u8 codecs[SND_SOC_ACPI_MAX_CODECS][ACPI_ID_LEN];
 };
 
-/* check all codecs */
-struct snd_soc_acpi_mach *snd_soc_acpi_codec_list(void *arg);
-
 #endif
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index bd8163f151cb..c00a0b8ade08 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -214,21 +214,21 @@ struct device;
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
 
 /* stream domain */
-#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wchan, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
-	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
-#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wslot, wreg, wshift, winvert, \
+	.channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
+#define SND_SOC_DAPM_AIF_IN_E(wname, stname, wchan, wreg, wshift, winvert, \
 			      wevent, wflags)				\
 {	.id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
-	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags }
-#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wchan, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
-	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
-#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wslot, wreg, wshift, winvert, \
+	.channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), }
+#define SND_SOC_DAPM_AIF_OUT_E(wname, stname, wchan, wreg, wshift, winvert, \
 			     wevent, wflags)				\
 {	.id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
-	SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
+	.channel = wchan, SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \
 	.event = wevent, .event_flags = wflags }
 #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, \
@@ -407,6 +407,10 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm,
 int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card);
 
+int snd_soc_dapm_update_dai(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai);
+
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
@@ -519,6 +523,9 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_asrc,		/* DSP/CODEC ASRC component */
 	snd_soc_dapm_encoder,		/* FW/SW audio encoder component */
 	snd_soc_dapm_decoder,		/* FW/SW audio decoder component */
+
+	/* Don't edit below this line */
+	SND_SOC_DAPM_TYPE_COUNT
 };
 
 enum snd_soc_dapm_subclass {
@@ -540,6 +547,8 @@ struct snd_soc_dapm_route {
 	/* Note: currently only supported for links where source is a supply */
 	int (*connected)(struct snd_soc_dapm_widget *source,
 			 struct snd_soc_dapm_widget *sink);
+
+	struct snd_soc_dobj dobj;
 };
 
 /* dapm audio path between two widgets */
@@ -625,6 +634,8 @@ struct snd_soc_dapm_widget {
 	int endpoints[2];
 
 	struct clk *clk;
+
+	int channel;
 };
 
 struct snd_soc_dapm_update {
diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h
index fa4b8413d2e2..5223896de26f 100644
--- a/include/sound/soc-topology.h
+++ b/include/sound/soc-topology.h
@@ -38,12 +38,14 @@ struct snd_soc_dapm_route;
 enum snd_soc_dobj_type {
 	SND_SOC_DOBJ_NONE		= 0,	/* object is not dynamic */
 	SND_SOC_DOBJ_MIXER,
-	SND_SOC_DOBJ_ENUM,
 	SND_SOC_DOBJ_BYTES,
-	SND_SOC_DOBJ_PCM,
+	SND_SOC_DOBJ_ENUM,
+	SND_SOC_DOBJ_GRAPH,
+	SND_SOC_DOBJ_WIDGET,
 	SND_SOC_DOBJ_DAI_LINK,
+	SND_SOC_DOBJ_PCM,
 	SND_SOC_DOBJ_CODEC_LINK,
-	SND_SOC_DOBJ_WIDGET,
+	SND_SOC_DOBJ_BACKEND_LINK,
 };
 
 /* dynamic control object */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e665f111b0d2..eb7db605955b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -802,6 +802,9 @@ struct snd_soc_component_driver {
 	int probe_order;
 	int remove_order;
 
+	/* signal if the module handling the component cannot be removed */
+	unsigned int ignore_module_refcount:1;
+
 	/* bits */
 	unsigned int idle_bias_on:1;
 	unsigned int suspend_bias_off:1;
@@ -891,6 +894,18 @@ struct snd_soc_dai_link {
 	/* config - must be set by machine driver */
 	const char *name;			/* Codec name */
 	const char *stream_name;		/* Stream name */
+
+	/*
+	 *	cpu_name
+	 *	cpu_of_node
+	 *	cpu_dai_name
+	 *
+	 * These are legacy style, and will be replaced to
+	 * modern style (= snd_soc_dai_link_component) in the future,
+	 * but, not yet supported so far.
+	 * If modern style was supported for CPU, all driver will switch
+	 * to use it, and, legacy style code will be removed from ALSA SoC.
+	 */
 	/*
 	 * You MAY specify the link's CPU-side device, either by device name,
 	 * or by DT/OF node, but not both. If this information is omitted,
@@ -906,6 +921,19 @@ struct snd_soc_dai_link {
 	 * only, which only works well when that device exposes a single DAI.
 	 */
 	const char *cpu_dai_name;
+
+	/*
+	 *	codec_name
+	 *	codec_of_node
+	 *	codec_dai_name
+	 *
+	 * These are legacy style, it will be converted to modern style
+	 * (= snd_soc_dai_link_component) automatically in soc-core
+	 * if driver is using legacy style.
+	 * Driver shouldn't use both legacy and modern style in the same time.
+	 * If modern style was supported for CPU, all driver will switch
+	 * to use it, and, legacy style code will be removed from ALSA SoC.
+	 */
 	/*
 	 * You MUST specify the link's codec, either by device name, or by
 	 * DT/OF node, but not both.
@@ -919,13 +947,25 @@ struct snd_soc_dai_link {
 	unsigned int num_codecs;
 
 	/*
+	 *	platform_name
+	 *	platform_of_node
+	 *
+	 * These are legacy style, it will be converted to modern style
+	 * (= snd_soc_dai_link_component) automatically in soc-core
+	 * if driver is using legacy style.
+	 * Driver shouldn't use both legacy and modern style in the same time.
+	 * If modern style was supported for CPU, all driver will switch
+	 * to use it, and, legacy style code will be removed from ALSA SoC.
+	 */
+	/*
 	 * You MAY specify the link's platform/PCM/DMA driver, either by
 	 * device name, or by DT/OF node, but not both. Some forms of link
 	 * do not need a platform.
 	 */
 	const char *platform_name;
 	struct device_node *platform_of_node;
-	struct snd_soc_dai_link_component *platform;
+	struct snd_soc_dai_link_component *platforms;
+	unsigned int num_platforms;
 
 	int id;	/* optional ID for machine driver link identification */
 
@@ -1543,6 +1583,37 @@ struct snd_soc_dai *snd_soc_card_get_codec_dai(struct snd_soc_card *card,
 	return NULL;
 }
 
+static inline
+int snd_soc_fixup_dai_links_platform_name(struct snd_soc_card *card,
+					  const char *platform_name)
+{
+	struct snd_soc_dai_link *dai_link;
+	const char *name;
+	int i;
+
+	if (!platform_name) /* nothing to do */
+		return 0;
+
+	/* set platform name for each dailink */
+	for_each_card_prelinks(card, i, dai_link) {
+		name = devm_kstrdup(card->dev, platform_name, GFP_KERNEL);
+		if (!name)
+			return -ENOMEM;
+
+		if (dai_link->platforms)
+			/* only single platform is supported for now */
+			dai_link->platforms->name = name;
+		else
+			/*
+			 * legacy mode, this case will be removed when all
+			 * derivers are switched to modern style dai_link.
+			 */
+			dai_link->platform_name = name;
+	}
+
+	return 0;
+}
+
 #ifdef CONFIG_DEBUG_FS
 extern struct dentry *snd_soc_debugfs_root;
 #endif