Merge tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As the diffstat shows, we've had lots of developments in a wide range
  at this time; the majority of changes are about ASoC, including
  subsystem-wide cleanups, continued SOF / Intel updates and a bunch of
  new drivers (as usual), while there have been some significant (but
  almost invisible) improvements in ALSA core side, too.

  Below are some highlights:

  Core:

   - Faster lookups of control elements with Xarray; normal user won't
     notice, but on the devices with tons of control elements, it can be
     visibly faster

   - Support for input validation for controls; this will harden for
     badly written drivers in general with a slight overhead

   - Deferred async signal handling for working around the potential
     deadlocks

   - Cleanup / refactoring raw MIDI locking code

  ASoC:

   - Restructing of the set_fmt() callbacks for making things clearer in
     situations like CODEC to CODEC links

   - Clean up and modernizing the DAI naming scheme setups

   - Merge of more of the Intel AVS driver stack, including some board
     integrations

   - New version 4 mechanism for communication with SOF DSPs

   - Suppoort for dynamically selecting the PLL to use at runtime on
     i.MX platforms

   - Improvements for CODEC to CODEC support in the generic cards

   - Support for AMD Jadeite and various machines, AMD RPL, Intel
     MetorLake DSPs, Mediatek MT8186 DSPs and MT6366, nVidia Tegra
     MDDRC, OPE and PEQ, NXP TFA9890, Qualcomm SDM845, WCD9335 and
     WAS883x, and Texas Instruments TAS2780

  HD- and USB-audio:

   - Continued improvement for CS35L41 (sub)codec support

   - More quirks for various devices (HP, Lenovo, Dell, Clevo)"

* tag 'sound-6.0-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (778 commits)
  ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
  ALSA: line6: Replace sprintf() with sysfs_emit()
  ALSA: hda: Replace sprintf() with sysfs_emit()
  ALSA: pcm: Replace sprintf() with sysfs_emit()
  ALSA: core: Replace scnprintf() with sysfs_emit()
  ALSA: control-led: Replace sprintf() with sysfs_emit()
  ALSA: aoa: Replace sprintf() with sysfs_emit()
  ALSA: ac97: Replace sprintf() with sysfs_emit()
  ALSA: hda/realtek: Add quirk for Clevo NV45PZ
  ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7
  ALSA: control: Use deferred fasync helper
  ALSA: pcm: Use deferred fasync helper
  ALSA: timer: Use deferred fasync helper
  ALSA: core: Add async signal helpers
  ASoC: q6asm: use kcalloc() instead of kzalloc()
  ACPI: scan: Add CLSA0101 Laptop Support
  ALSA: hda: cs35l41: Support CLSA0101
  ALSA: hda: cs35l41: Use the CS35L41 HDA internal define
  ASoC: dt-bindings: use spi-peripheral-props.yaml
  ASoC: codecs: va-macro: use fsgen as clock
  ...

31 files changed, 323 insertions, 38 deletions

diff --git a/include/dt-bindings/sound/qcom,wcd9335.h b/include/dt-bindings/sound/qcom,wcd9335.h
new file mode 100644
index 000000000000..f5e9f1db091e
--- /dev/null
+++ b/include/dt-bindings/sound/qcom,wcd9335.h
@@ -0,0 +1,15 @@
+/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) */
+
+#ifndef __DT_SOUND_QCOM_WCD9335_H
+#define __DT_SOUND_QCOM_WCD9335_H
+
+#define AIF1_PB                 0
+#define AIF1_CAP                1
+#define AIF2_PB                 2
+#define AIF2_CAP                3
+#define AIF3_PB                 4
+#define AIF3_CAP                5
+#define AIF4_PB                 6
+#define NUM_CODEC_DAIS          7
+
+#endif
diff --git a/include/linux/acpi.h b/include/linux/acpi.h
index f6d4539c3895..7ee10b2848d5 100644
--- a/include/linux/acpi.h
+++ b/include/linux/acpi.h
@@ -764,6 +764,7 @@ static inline u64 acpi_arch_get_root_pointer(void)
 #endif
 
 int acpi_get_local_address(acpi_handle handle, u32 *addr);
+const char *acpi_get_subsystem_id(acpi_handle handle);
 
 #else	/* !CONFIG_ACPI */
 
@@ -1025,6 +1026,11 @@ static inline int acpi_get_local_address(acpi_handle handle, u32 *addr)
 	return -ENODEV;
 }
 
+static inline const char *acpi_get_subsystem_id(acpi_handle handle)
+{
+	return ERR_PTR(-ENODEV);
+}
+
 static inline int acpi_register_wakeup_handler(int wake_irq,
 	bool (*wakeup)(void *context), void *context)
 {
diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h
index 30055706cce2..cad828e21c72 100644
--- a/include/linux/firmware/cirrus/cs_dsp.h
+++ b/include/linux/firmware/cirrus/cs_dsp.h
@@ -11,6 +11,7 @@
 #ifndef __CS_DSP_H
 #define __CS_DSP_H
 
+#include <linux/bits.h>
 #include <linux/device.h>
 #include <linux/firmware.h>
 #include <linux/list.h>
@@ -34,6 +35,7 @@
 #define CS_ADSP2_REGION_ALL (CS_ADSP2_REGION_0 | CS_ADSP2_REGION_1_9)
 
 #define CS_DSP_DATA_WORD_SIZE                3
+#define CS_DSP_DATA_WORD_BITS                (3 * BITS_PER_BYTE)
 
 #define CS_DSP_ACKED_CTL_TIMEOUT_MS          100
 #define CS_DSP_ACKED_CTL_N_QUICKPOLLS        10
@@ -189,7 +191,8 @@ struct cs_dsp {
  * @control_remove:	Called under the pwr_lock when a control is destroyed
  * @pre_run:		Called under the pwr_lock by cs_dsp_run() before the core is started
  * @post_run:		Called under the pwr_lock by cs_dsp_run() after the core is started
- * @post_stop:		Called under the pwr_lock by cs_dsp_stop()
+ * @pre_stop:		Called under the pwr_lock by cs_dsp_stop() before the core is stopped
+ * @post_stop:		Called under the pwr_lock by cs_dsp_stop() after the core is stopped
  * @watchdog_expired:	Called when a watchdog expiry is detected
  *
  * These callbacks give the cs_dsp client an opportunity to respond to events
@@ -200,6 +203,7 @@ struct cs_dsp_client_ops {
 	void (*control_remove)(struct cs_dsp_coeff_ctl *ctl);
 	int (*pre_run)(struct cs_dsp *dsp);
 	int (*post_run)(struct cs_dsp *dsp);
+	void (*pre_stop)(struct cs_dsp *dsp);
 	void (*post_stop)(struct cs_dsp *dsp);
 	void (*watchdog_expired)(struct cs_dsp *dsp);
 };
@@ -250,4 +254,75 @@ struct cs_dsp_alg_region *cs_dsp_find_alg_region(struct cs_dsp *dsp,
 
 const char *cs_dsp_mem_region_name(unsigned int type);
 
+/**
+ * struct cs_dsp_chunk - Describes a buffer holding data formatted for the DSP
+ * @data:	Pointer to underlying buffer memory
+ * @max:	Pointer to end of the buffer memory
+ * @bytes:	Number of bytes read/written into the memory chunk
+ * @cache:	Temporary holding data as it is formatted
+ * @cachebits:	Number of bits of data currently in cache
+ */
+struct cs_dsp_chunk {
+	u8 *data;
+	u8 *max;
+	int bytes;
+
+	u32 cache;
+	int cachebits;
+};
+
+/**
+ * cs_dsp_chunk() - Create a DSP memory chunk
+ * @data: Pointer to the buffer that will be used to store data
+ * @size: Size of the buffer in bytes
+ *
+ * Return: A cs_dsp_chunk structure
+ */
+static inline struct cs_dsp_chunk cs_dsp_chunk(void *data, int size)
+{
+	struct cs_dsp_chunk ch = {
+		.data = data,
+		.max = data + size,
+	};
+
+	return ch;
+}
+
+/**
+ * cs_dsp_chunk_end() - Check if a DSP memory chunk is full
+ * @ch: Pointer to the chunk structure
+ *
+ * Return: True if the whole buffer has been read/written
+ */
+static inline bool cs_dsp_chunk_end(struct cs_dsp_chunk *ch)
+{
+	return ch->data == ch->max;
+}
+
+/**
+ * cs_dsp_chunk_bytes() - Number of bytes written/read from a DSP memory chunk
+ * @ch: Pointer to the chunk structure
+ *
+ * Return: Number of bytes read/written to the buffer
+ */
+static inline int cs_dsp_chunk_bytes(struct cs_dsp_chunk *ch)
+{
+	return ch->bytes;
+}
+
+/**
+ * cs_dsp_chunk_valid_addr() - Check if an address is in a DSP memory chunk
+ * @ch: Pointer to the chunk structure
+ *
+ * Return: True if the given address is within the buffer
+ */
+static inline bool cs_dsp_chunk_valid_addr(struct cs_dsp_chunk *ch, void *addr)
+{
+	return (u8 *)addr >= ch->data && (u8 *)addr < ch->max;
+}
+
+int cs_dsp_chunk_write(struct cs_dsp_chunk *ch, int nbits, u32 val);
+int cs_dsp_chunk_flush(struct cs_dsp_chunk *ch);
+int cs_dsp_chunk_read(struct cs_dsp_chunk *ch, int nbits);
+
 #endif
diff --git a/include/linux/soundwire/sdw_intel.h b/include/linux/soundwire/sdw_intel.h
index 67e0d3e750b5..ec16ae49e6a4 100644
--- a/include/linux/soundwire/sdw_intel.h
+++ b/include/linux/soundwire/sdw_intel.h
@@ -9,6 +9,8 @@
 
 #define SDW_SHIM_BASE			0x2C000
 #define SDW_ALH_BASE			0x2C800
+#define SDW_SHIM_BASE_ACE		0x38000
+#define SDW_ALH_BASE_ACE		0x24000
 #define SDW_LINK_BASE			0x30000
 #define SDW_LINK_SIZE			0x10000
 
@@ -119,6 +121,7 @@ struct sdw_intel_ops {
 			     struct sdw_intel_stream_params_data *params_data);
 	int (*free_stream)(struct device *dev,
 			   struct sdw_intel_stream_free_data *free_data);
+	int (*trigger)(struct snd_soc_dai *dai, int cmd, int stream);
 };
 
 /**
diff --git a/include/sound/control.h b/include/sound/control.h
index 985c51a8fb74..eae443ba79ba 100644
--- a/include/sound/control.h
+++ b/include/sound/control.h
@@ -23,7 +23,7 @@ typedef int (snd_kcontrol_tlv_rw_t)(struct snd_kcontrol *kcontrol,
 				    unsigned int __user *tlv);
 
 /* internal flag for skipping validations */
-#ifdef CONFIG_SND_CTL_VALIDATION
+#ifdef CONFIG_SND_CTL_DEBUG
 #define SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK	(1 << 24)
 #define snd_ctl_skip_validation(info) \
 	((info)->access & SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK)
@@ -109,7 +109,7 @@ struct snd_ctl_file {
 	int preferred_subdevice[SND_CTL_SUBDEV_ITEMS];
 	wait_queue_head_t change_sleep;
 	spinlock_t read_lock;
-	struct fasync_struct *fasync;
+	struct snd_fasync *fasync;
 	int subscribed;			/* read interface is activated */
 	struct list_head events;	/* waiting events for read */
 };
diff --git a/include/sound/core.h b/include/sound/core.h
index 6d4cc49584c6..4365c35d038b 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -14,6 +14,7 @@
 #include <linux/pm.h>			/* pm_message_t */
 #include <linux/stringify.h>
 #include <linux/printk.h>
+#include <linux/xarray.h>
 
 /* number of supported soundcards */
 #ifdef CONFIG_SND_DYNAMIC_MINORS
@@ -103,6 +104,11 @@ struct snd_card {
 	size_t user_ctl_alloc_size;	// current memory allocation by user controls.
 	struct list_head controls;	/* all controls for this card */
 	struct list_head ctl_files;	/* active control files */
+#ifdef CONFIG_SND_CTL_FAST_LOOKUP
+	struct xarray ctl_numids;	/* hash table for numids */
+	struct xarray ctl_hash;		/* hash table for ctl id matching */
+	bool ctl_hash_collision;	/* ctl_hash collision seen? */
+#endif
 
 	struct snd_info_entry *proc_root;	/* root for soundcard specific files */
 	struct proc_dir_entry *proc_root_link;	/* number link to real id */
@@ -501,4 +507,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device,
 }
 #endif
 
+/* async signal helpers */
+struct snd_fasync;
+
+int snd_fasync_helper(int fd, struct file *file, int on,
+		      struct snd_fasync **fasyncp);
+void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll);
+void snd_fasync_free(struct snd_fasync *fasync);
+
 #endif /* __SOUND_CORE_H */
diff --git a/include/sound/cs35l41.h b/include/sound/cs35l41.h
index 8972fa697622..9ac5918269a5 100644
--- a/include/sound/cs35l41.h
+++ b/include/sound/cs35l41.h
@@ -665,6 +665,10 @@
 #define CS35L41_BST_EN_DEFAULT		0x2
 #define CS35L41_AMP_EN_SHIFT		0
 #define CS35L41_AMP_EN_MASK		1
+#define CS35L41_VMON_EN_MASK		0x1000
+#define CS35L41_VMON_EN_SHIFT		12
+#define CS35L41_IMON_EN_MASK		0x2000
+#define CS35L41_IMON_EN_SHIFT		13
 
 #define CS35L41_PDN_DONE_MASK		0x00800000
 #define CS35L41_PDN_DONE_SHIFT		23
@@ -881,6 +885,9 @@ void cs35l41_configure_cs_dsp(struct device *dev, struct regmap *reg, struct cs_
 int cs35l41_set_cspl_mbox_cmd(struct device *dev, struct regmap *regmap,
 			      enum cs35l41_cspl_mbox_cmd cmd);
 int cs35l41_write_fs_errata(struct device *dev, struct regmap *regmap);
+int cs35l41_enter_hibernate(struct device *dev, struct regmap *regmap,
+			    enum cs35l41_boost_type b_type);
+int cs35l41_exit_hibernate(struct device *dev, struct regmap *regmap);
 int cs35l41_init_boost(struct device *dev, struct regmap *regmap,
 		       struct cs35l41_hw_cfg *hw_cfg);
 bool cs35l41_safe_reset(struct regmap *regmap, enum cs35l41_boost_type b_type);
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 38ea046e653c..2df54cf02cb3 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -15,6 +15,8 @@
  * snd_pcm_substream_to_dma_direction - Get dma_transfer_direction for a PCM
  *   substream
  * @substream: PCM substream
+ *
+ * Return: DMA transfer direction
  */
 static inline enum dma_transfer_direction
 snd_pcm_substream_to_dma_direction(const struct snd_pcm_substream *substream)
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index b7be300b6b18..6d3c82c4b6ac 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -231,7 +231,6 @@ struct hda_codec {
 	/* misc flags */
 	unsigned int configured:1; /* codec was configured */
 	unsigned int in_freeing:1; /* being released */
-	unsigned int registered:1; /* codec was registered */
 	unsigned int display_power_control:1; /* needs display power */
 	unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each
 					     * status change
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index 15f15075238d..797bf67a164d 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -93,6 +93,7 @@ struct hdac_device {
 	bool lazy_cache:1;	/* don't wake up for writes */
 	bool caps_overwriting:1; /* caps overwrite being in process */
 	bool cache_coef:1;	/* cache COEF read/write too */
+	unsigned int registered:1; /* codec was registered */
 };
 
 /* device/driver type used for matching */
diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h
index 4fc733c8c570..48ad33aba393 100644
--- a/include/sound/hdmi-codec.h
+++ b/include/sound/hdmi-codec.h
@@ -32,8 +32,8 @@ struct hdmi_codec_daifmt {
 	} fmt;
 	unsigned int bit_clk_inv:1;
 	unsigned int frame_clk_inv:1;
-	unsigned int bit_clk_master:1;
-	unsigned int frame_clk_master:1;
+	unsigned int bit_clk_provider:1;
+	unsigned int frame_clk_provider:1;
 	/* bit_fmt could be standard PCM format or
 	 * IEC958 encoded format. ALSA IEC958 plugin will pass
 	 * IEC958_SUBFRAME format to the underneath driver.
diff --git a/include/sound/madera-pdata.h b/include/sound/madera-pdata.h
index e3060f48f108..58398d80c3de 100644
--- a/include/sound/madera-pdata.h
+++ b/include/sound/madera-pdata.h
@@ -9,7 +9,7 @@
 #ifndef MADERA_CODEC_PDATA_H
 #define MADERA_CODEC_PDATA_H
 
-#include <linux/kernel.h>
+#include <linux/types.h>
 
 #define MADERA_MAX_INPUT		6
 #define MADERA_MAX_MUXED_CHANNELS	4
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index 6b99310b5b88..8c48a5bce88c 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -399,7 +399,7 @@ struct snd_pcm_runtime {
 	snd_pcm_uframes_t twake; 	/* do transfer (!poll) wakeup if non-zero */
 	wait_queue_head_t sleep;	/* poll sleep */
 	wait_queue_head_t tsleep;	/* transfer sleep */
-	struct fasync_struct *fasync;
+	struct snd_fasync *fasync;
 	bool stop_operating;		/* sync_stop will be called */
 	struct mutex buffer_mutex;	/* protect for buffer changes */
 	atomic_t buffer_accessing;	/* >0: in r/w operation, <0: blocked */
@@ -607,7 +607,7 @@ snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size)
  * snd_pcm_stream_linked - Check whether the substream is linked with others
  * @substream: substream to check
  *
- * Returns true if the given substream is being linked with others.
+ * Return: true if the given substream is being linked with others
  */
 static inline int snd_pcm_stream_linked(struct snd_pcm_substream *substream)
 {
@@ -673,7 +673,7 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream,
  * snd_pcm_running - Check whether the substream is in a running state
  * @substream: substream to check
  *
- * Returns true if the given substream is in the state RUNNING, or in the
+ * Return: true if the given substream is in the state RUNNING, or in the
  * state DRAINING for playback.
  */
 static inline int snd_pcm_running(struct snd_pcm_substream *substream)
@@ -687,6 +687,8 @@ static inline int snd_pcm_running(struct snd_pcm_substream *substream)
  * bytes_to_samples - Unit conversion of the size from bytes to samples
  * @runtime: PCM runtime instance
  * @size: size in bytes
+ *
+ * Return: the size in samples
  */
 static inline ssize_t bytes_to_samples(struct snd_pcm_runtime *runtime, ssize_t size)
 {
@@ -697,6 +699,8 @@ static inline ssize_t bytes_to_samples(struct snd_pcm_runtime *runtime, ssize_t
  * bytes_to_frames - Unit conversion of the size from bytes to frames
  * @runtime: PCM runtime instance
  * @size: size in bytes
+ *
+ * Return: the size in frames
  */
 static inline snd_pcm_sframes_t bytes_to_frames(struct snd_pcm_runtime *runtime, ssize_t size)
 {
@@ -707,6 +711,8 @@ static inline snd_pcm_sframes_t bytes_to_frames(struct snd_pcm_runtime *runtime,
  * samples_to_bytes - Unit conversion of the size from samples to bytes
  * @runtime: PCM runtime instance
  * @size: size in samples
+ *
+ * Return: the byte size
  */
 static inline ssize_t samples_to_bytes(struct snd_pcm_runtime *runtime, ssize_t size)
 {
@@ -717,6 +723,8 @@ static inline ssize_t samples_to_bytes(struct snd_pcm_runtime *runtime, ssize_t
  * frames_to_bytes - Unit conversion of the size from frames to bytes
  * @runtime: PCM runtime instance
  * @size: size in frames
+ *
+ * Return: the byte size
  */
 static inline ssize_t frames_to_bytes(struct snd_pcm_runtime *runtime, snd_pcm_sframes_t size)
 {
@@ -727,6 +735,8 @@ static inline ssize_t frames_to_bytes(struct snd_pcm_runtime *runtime, snd_pcm_s
  * frame_aligned - Check whether the byte size is aligned to frames
  * @runtime: PCM runtime instance
  * @bytes: size in bytes
+ *
+ * Return: true if aligned, or false if not
  */
 static inline int frame_aligned(struct snd_pcm_runtime *runtime, ssize_t bytes)
 {
@@ -736,6 +746,8 @@ static inline int frame_aligned(struct snd_pcm_runtime *runtime, ssize_t bytes)
 /**
  * snd_pcm_lib_buffer_bytes - Get the buffer size of the current PCM in bytes
  * @substream: PCM substream
+ *
+ * Return: buffer byte size
  */
 static inline size_t snd_pcm_lib_buffer_bytes(struct snd_pcm_substream *substream)
 {
@@ -746,6 +758,8 @@ static inline size_t snd_pcm_lib_buffer_bytes(struct snd_pcm_substream *substrea
 /**
  * snd_pcm_lib_period_bytes - Get the period size of the current PCM in bytes
  * @substream: PCM substream
+ *
+ * Return: period byte size
  */
 static inline size_t snd_pcm_lib_period_bytes(struct snd_pcm_substream *substream)
 {
@@ -758,6 +772,8 @@ static inline size_t snd_pcm_lib_period_bytes(struct snd_pcm_substream *substrea
  * @runtime: PCM runtime instance
  *
  * Result is between 0 ... (boundary - 1)
+ *
+ * Return: available frame size
  */
 static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *runtime)
 {
@@ -774,6 +790,8 @@ static inline snd_pcm_uframes_t snd_pcm_playback_avail(struct snd_pcm_runtime *r
  * @runtime: PCM runtime instance
  *
  * Result is between 0 ... (boundary - 1)
+ *
+ * Return: available frame size
  */
 static inline snd_pcm_uframes_t snd_pcm_capture_avail(struct snd_pcm_runtime *runtime)
 {
@@ -786,6 +804,8 @@ static inline snd_pcm_uframes_t snd_pcm_capture_avail(struct snd_pcm_runtime *ru
 /**
  * snd_pcm_playback_hw_avail - Get the queued space for playback
  * @runtime: PCM runtime instance
+ *
+ * Return: available frame size
  */
 static inline snd_pcm_sframes_t snd_pcm_playback_hw_avail(struct snd_pcm_runtime *runtime)
 {
@@ -795,6 +815,8 @@ static inline snd_pcm_sframes_t snd_pcm_playback_hw_avail(struct snd_pcm_runtime
 /**
  * snd_pcm_capture_hw_avail - Get the free space for capture
  * @runtime: PCM runtime instance
+ *
+ * Return: available frame size
  */
 static inline snd_pcm_sframes_t snd_pcm_capture_hw_avail(struct snd_pcm_runtime *runtime)
 {
@@ -934,6 +956,8 @@ static inline const struct snd_interval *hw_param_interval_c(const struct snd_pc
 /**
  * params_channels - Get the number of channels from the hw params
  * @p: hw params
+ *
+ * Return: the number of channels
  */
 static inline unsigned int params_channels(const struct snd_pcm_hw_params *p)
 {
@@ -943,6 +967,8 @@ static inline unsigned int params_channels(const struct snd_pcm_hw_params *p)
 /**
  * params_rate - Get the sample rate from the hw params
  * @p: hw params
+ *
+ * Return: the sample rate
  */
 static inline unsigned int params_rate(const struct snd_pcm_hw_params *p)
 {
@@ -952,6 +978,8 @@ static inline unsigned int params_rate(const struct snd_pcm_hw_params *p)
 /**
  * params_period_size - Get the period size (in frames) from the hw params
  * @p: hw params
+ *
+ * Return: the period size in frames
  */
 static inline unsigned int params_period_size(const struct snd_pcm_hw_params *p)
 {
@@ -961,6 +989,8 @@ static inline unsigned int params_period_size(const struct snd_pcm_hw_params *p)
 /**
  * params_periods - Get the number of periods from the hw params
  * @p: hw params
+ *
+ * Return: the number of periods
  */
 static inline unsigned int params_periods(const struct snd_pcm_hw_params *p)
 {
@@ -970,6 +1000,8 @@ static inline unsigned int params_periods(const struct snd_pcm_hw_params *p)
 /**
  * params_buffer_size - Get the buffer size (in frames) from the hw params
  * @p: hw params
+ *
+ * Return: the buffer size in frames
  */
 static inline unsigned int params_buffer_size(const struct snd_pcm_hw_params *p)
 {
@@ -979,6 +1011,8 @@ static inline unsigned int params_buffer_size(const struct snd_pcm_hw_params *p)
 /**
  * params_buffer_bytes - Get the buffer size (in bytes) from the hw params
  * @p: hw params
+ *
+ * Return: the buffer size in bytes
  */
 static inline unsigned int params_buffer_bytes(const struct snd_pcm_hw_params *p)
 {
@@ -1241,6 +1275,8 @@ int snd_pcm_set_managed_buffer_all(struct snd_pcm *pcm, int type,
  * only the given sized buffer and doesn't allow re-allocation nor dynamic
  * allocation of a larger buffer unlike the standard one.
  * The function may return -ENOMEM error, hence the caller must check it.
+ *
+ * Return: zero if successful, or a negative error code
  */
 static inline int __must_check
 snd_pcm_set_fixed_buffer(struct snd_pcm_substream *substream, int type,
@@ -1259,6 +1295,8 @@ snd_pcm_set_fixed_buffer(struct snd_pcm_substream *substream, int type,
  * Apply the set up of the fixed buffer via snd_pcm_set_fixed_buffer() for
  * all substream.  If any of allocation fails, it returns -ENOMEM, hence the
  * caller must check the return value.
+ *
+ * Return: zero if successful, or a negative error code
  */
 static inline int __must_check
 snd_pcm_set_fixed_buffer_all(struct snd_pcm *pcm, int type,
@@ -1315,6 +1353,8 @@ static inline int snd_pcm_lib_alloc_vmalloc_32_buffer
  * snd_pcm_sgbuf_get_addr - Get the DMA address at the corresponding offset
  * @substream: PCM substream
  * @ofs: byte offset
+ *
+ * Return: DMA address
  */
 static inline dma_addr_t
 snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
@@ -1328,6 +1368,8 @@ snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
  * @substream: PCM substream
  * @ofs: byte offset
  * @size: byte size to examine
+ *
+ * Return: chunk size
  */
 static inline unsigned int
 snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
@@ -1393,6 +1435,20 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max)
 const char *snd_pcm_format_name(snd_pcm_format_t format);
 
 /**
+ * snd_pcm_direction_name - Get a string naming the direction of a stream
+ * @direction: Stream's direction, one of SNDRV_PCM_STREAM_XXX
+ *
+ * Returns a string naming the direction of the stream.
+ */
+static inline const char *snd_pcm_direction_name(int direction)
+{
+	if (direction == SNDRV_PCM_STREAM_PLAYBACK)
+		return "Playback";
+	else
+		return "Capture";
+}
+
+/**
  * snd_pcm_stream_str - Get a string naming the direction of a stream
  * @substream: the pcm substream instance
  *
@@ -1400,10 +1456,7 @@ const char *snd_pcm_format_name(snd_pcm_format_t format);
  */
 static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream)
 {
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		return "Playback";
-	else
-		return "Capture";
+	return snd_pcm_direction_name(substream->stream);
 }
 
 /*
@@ -1430,6 +1483,8 @@ struct snd_pcm_chmap {
  * snd_pcm_chmap_substream - get the PCM substream assigned to the given chmap info
  * @info: chmap information
  * @idx: the substream number index
+ *
+ * Return: the matched PCM substream, or NULL if not found
  */
 static inline struct snd_pcm_substream *
 snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx)
@@ -1460,6 +1515,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
 /**
  * pcm_format_to_bits - Strong-typed conversion of pcm_format to bitwise
  * @pcm_format: PCM format
+ *
+ * Return: 64bit mask corresponding to the given PCM format
  */
 static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format)
 {
diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h
index 7a08ed2acd60..e1f59b2940af 100644
--- a/include/sound/rawmidi.h
+++ b/include/sound/rawmidi.h
@@ -63,7 +63,6 @@ struct snd_rawmidi_runtime {
 	size_t xruns;		/* over/underruns counter */
 	int buffer_ref;		/* buffer reference count */
 	/* misc */
-	spinlock_t lock;
 	wait_queue_head_t sleep;
 	/* event handler (new bytes, input only) */
 	void (*event)(struct snd_rawmidi_substream *substream);
@@ -85,6 +84,7 @@ struct snd_rawmidi_substream {
 	unsigned int clock_type;	/* clock source to use for input framing */
 	int use_count;			/* use counter (for output) */
 	size_t bytes;
+	spinlock_t lock;
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_str *pstr;
 	char name[32];
@@ -156,10 +156,6 @@ int snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
 int snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream, int count);
 int snd_rawmidi_transmit(struct snd_rawmidi_substream *substream,
 			 unsigned char *buffer, int count);
-int __snd_rawmidi_transmit_peek(struct snd_rawmidi_substream *substream,
-			      unsigned char *buffer, int count);
-int __snd_rawmidi_transmit_ack(struct snd_rawmidi_substream *substream,
-			       int count);
 int snd_rawmidi_proceed(struct snd_rawmidi_substream *substream);
 
 /* main midi functions */
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 8faa649f712b..ab55f40896e0 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -51,7 +51,6 @@ struct prop_nums {
 	int cpus;
 	int codecs;
 	int platforms;
-	int c2c;
 };
 
 struct asoc_simple_priv {
@@ -64,7 +63,6 @@ struct asoc_simple_priv {
 		struct snd_soc_dai_link_component *platforms;
 		struct asoc_simple_data adata;
 		struct snd_soc_codec_conf *codec_conf;
-		struct snd_soc_pcm_stream *c2c_conf;
 		struct prop_nums num;
 		unsigned int mclk_fs;
 	} *dai_props;
@@ -75,7 +73,6 @@ struct asoc_simple_priv {
 	struct snd_soc_dai_link_component *dlcs;
 	struct snd_soc_dai_link_component dummy;
 	struct snd_soc_codec_conf *codec_conf;
-	struct snd_soc_pcm_stream *c2c_conf;
 	struct gpio_desc *pa_gpio;
 	const struct snd_soc_ops *ops;
 	unsigned int dpcm_selectable:1;
@@ -173,7 +170,7 @@ void asoc_simple_canonicalize_platform(struct snd_soc_dai_link_component *platfo
 void asoc_simple_canonicalize_cpu(struct snd_soc_dai_link_component *cpus,
 				  int is_single_links);
 
-int asoc_simple_clean_reference(struct snd_soc_card *card);
+void asoc_simple_clean_reference(struct snd_soc_card *card);
 
 void asoc_simple_convert_fixup(struct asoc_simple_data *data,
 				      struct snd_pcm_hw_params *params);
diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h
index 59551b1f22f3..bc7fd46ec2bc 100644
--- a/include/sound/soc-acpi-intel-match.h
+++ b/include/sound/soc-acpi-intel-match.h
@@ -30,6 +30,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[];
 
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cfl_sdw_machines[];
@@ -37,6 +38,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[];
 extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[];
+extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[];
 
 /*
  * generic table used for HDA codec-based platforms, possibly with
diff --git a/include/sound/soc-card.h b/include/sound/soc-card.h
index df08573bd80c..9d31a5c0db33 100644
--- a/include/sound/soc-card.h
+++ b/include/sound/soc-card.h
@@ -29,6 +29,7 @@ int snd_soc_card_resume_post(struct snd_soc_card *card);
 
 int snd_soc_card_probe(struct snd_soc_card *card);
 int snd_soc_card_late_probe(struct snd_soc_card *card);
+void snd_soc_card_fixup_controls(struct snd_soc_card *card);
 int snd_soc_card_remove(struct snd_soc_card *card);
 
 int snd_soc_card_set_bias_level(struct snd_soc_card *card,
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 5a764c3099d3..c26ffb033777 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -179,7 +179,7 @@ struct snd_soc_component_driver {
 	 * analogue).
 	 */
 	unsigned int endianness:1;
-	unsigned int non_legacy_dai_naming:1;
+	unsigned int legacy_dai_naming:1;
 
 	/* this component uses topology and ignore machine driver FEs */
 	const char *ignore_machine;
@@ -348,11 +348,6 @@ static inline int snd_soc_component_cache_sync(
 	return regcache_sync(component->regmap);
 }
 
-static inline int snd_soc_component_is_codec(struct snd_soc_component *component)
-{
-	return component->driver->non_legacy_dai_naming;
-}
-
 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);
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index bbd821d2df9c..ea7509672086 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -124,6 +124,12 @@ struct snd_compr_stream;
 #define SND_SOC_DAIFMT_CBM_CFS		SND_SOC_DAIFMT_CBP_CFC
 #define SND_SOC_DAIFMT_CBS_CFS		SND_SOC_DAIFMT_CBC_CFC
 
+/* when passed to set_fmt directly indicate if the device is provider or consumer */
+#define SND_SOC_DAIFMT_BP_FP		SND_SOC_DAIFMT_CBP_CFP
+#define SND_SOC_DAIFMT_BC_FP		SND_SOC_DAIFMT_CBC_CFP
+#define SND_SOC_DAIFMT_BP_FC		SND_SOC_DAIFMT_CBP_CFC
+#define SND_SOC_DAIFMT_BC_FC		SND_SOC_DAIFMT_CBC_CFC
+
 /* Describes the possible PCM format */
 #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT	48
 #define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK	(0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT)
diff --git a/include/sound/soc.h b/include/sound/soc.h
index b276dcb5d4e8..aad24a1d3276 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -136,6 +136,18 @@
 	.put = snd_soc_put_volsw, \
 	.private_value = SOC_DOUBLE_VALUE(reg, shift_left, shift_right, \
 					  max, invert, 0) }
+#define SOC_DOUBLE_SX_TLV(xname, xreg, shift_left, shift_right, 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_sx, \
+	.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 = shift_left, .rshift = shift_right, \
+		.max = xmax, .min = xmin} }
 #define SOC_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert, tlv_array) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\
 	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
@@ -414,7 +426,7 @@ enum snd_soc_pcm_subclass {
 };
 
 int snd_soc_register_card(struct snd_soc_card *card);
-int snd_soc_unregister_card(struct snd_soc_card *card);
+void snd_soc_unregister_card(struct snd_soc_card *card);
 int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card);
 #ifdef CONFIG_PM_SLEEP
 int snd_soc_suspend(struct device *dev);
@@ -914,6 +926,7 @@ struct snd_soc_card {
 
 	int (*probe)(struct snd_soc_card *card);
 	int (*late_probe)(struct snd_soc_card *card);
+	void (*fixup_controls)(struct snd_soc_card *card);
 	int (*remove)(struct snd_soc_card *card);
 
 	/* the pre and post PM functions are used to do any PM work before and
diff --git a/include/sound/sof.h b/include/sound/sof.h
index 1a82a0db5e7f..367dccfea7ad 100644
--- a/include/sound/sof.h
+++ b/include/sound/sof.h
@@ -138,6 +138,7 @@ struct sof_dev_desc {
 
 	struct snd_sof_dsp_ops *ops;
 	int (*ops_init)(struct snd_sof_dev *sdev);
+	void (*ops_free)(struct snd_sof_dev *sdev);
 };
 
 int sof_dai_get_mclk(struct snd_soc_pcm_runtime *rtd);
diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h
index 90d09dbdd709..92f45c180b7c 100644
--- a/include/sound/sof/dai-amd.h
+++ b/include/sound/sof/dai-amd.h
@@ -18,4 +18,11 @@ struct sof_ipc_dai_acp_params {
 	uint32_t fsync_rate;    /* FSYNC frequency in Hz */
 	uint32_t tdm_slots;
 } __packed;
+
+/* ACPDMIC Configuration Request - SOF_IPC_DAI_AMD_CONFIG */
+struct sof_ipc_dai_acpdmic_params {
+	uint32_t pdm_rate;
+	uint32_t pdm_ch;
+} __packed;
+
 #endif
diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h
index 7a266f41983c..5b93b7292f5e 100644
--- a/include/sound/sof/dai-intel.h
+++ b/include/sound/sof/dai-intel.h
@@ -52,6 +52,8 @@
 #define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_ES               BIT(6)
 /* bclk early start */
 #define SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_ES               BIT(7)
+/* mclk always on */
+#define SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_AON		BIT(8)
 
 /* DMIC max. four controllers for eight microphone channels */
 #define SOF_DAI_INTEL_DMIC_NUM_CTRL			4
diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h
index a818a0f0a226..21d98f31a9ca 100644
--- a/include/sound/sof/dai.h
+++ b/include/sound/sof/dai.h
@@ -111,7 +111,7 @@ struct sof_ipc_dai_config {
 		struct sof_ipc_dai_sai_params sai;
 		struct sof_ipc_dai_acp_params acpbt;
 		struct sof_ipc_dai_acp_params acpsp;
-		struct sof_ipc_dai_acp_params acpdmic;
+		struct sof_ipc_dai_acpdmic_params acpdmic;
 		struct sof_ipc_dai_mtk_afe_params afe;
 	};
 } __packed;
diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h
index b8b8e5b5e3e1..a795deacc2ea 100644
--- a/include/sound/sof/ipc4/header.h
+++ b/include/sound/sof/ipc4/header.h
@@ -385,6 +385,14 @@ struct sof_ipc4_fw_version {
 	uint16_t build;
 } __packed;
 
+/* Payload data for SOF_IPC4_MOD_SET_DX */
+struct sof_ipc4_dx_state_info {
+	/* core(s) to apply the change */
+	uint32_t core_mask;
+	/* core state: 0: put core_id to D3; 1: put core_id to D0 */
+	uint32_t dx_mask;
+} __packed __aligned(4);
+
 /* Reply messages */
 
 /*
diff --git a/include/sound/sof/stream.h b/include/sound/sof/stream.h
index 1db3bbc3e65d..9377113f13e4 100644
--- a/include/sound/sof/stream.h
+++ b/include/sound/sof/stream.h
@@ -86,9 +86,11 @@ struct sof_ipc_stream_params {
 	uint32_t host_period_bytes;
 	uint16_t no_stream_position; /**< 1 means don't send stream position */
 	uint8_t cont_update_posn; /**< 1 means continuous update stream position */
-
-	uint8_t reserved[5];
+	uint8_t reserved0;
+	int16_t ext_data_length; /**< 0, means no extended data */
+	uint8_t reserved[2];
 	uint16_t chmap[SOF_IPC_MAX_CHANNELS];	/**< channel map - SOF_CHMAP_ */
+	uint8_t ext_data[]; /**< extended data */
 } __packed;
 
 /* PCM params info - SOF_IPC_STREAM_PCM_PARAMS */
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index 9555f31c8425..3aef123dbd7f 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -123,7 +123,7 @@ struct snd_compr_codec_caps {
 } __attribute__((packed, aligned(4)));
 
 /**
- * enum sndrv_compress_encoder
+ * enum sndrv_compress_encoder - encoder metadata key
  * @SNDRV_COMPRESS_ENCODER_PADDING: no of samples appended by the encoder at the
  * end of the track
  * @SNDRV_COMPRESS_ENCODER_DELAY: no of samples inserted by the encoder at the
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index 79b14389ae41..726361716919 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -250,7 +250,7 @@ struct snd_enc_wma {
 
 
 /**
- * struct snd_enc_vorbis
+ * struct snd_enc_vorbis - Vorbis encoder parameters
  * @quality: Sets encoding quality to n, between -1 (low) and 10 (high).
  * In the default mode of operation, the quality level is 3.
  * Normal quality range is 0 - 10.
@@ -279,7 +279,7 @@ struct snd_enc_vorbis {
 
 
 /**
- * struct snd_enc_real
+ * struct snd_enc_real - RealAudio encoder parameters
  * @quant_bits: number of coupling quantization bits in the stream
  * @start_region: coupling start region in the stream
  * @num_regions: number of regions value
@@ -294,7 +294,7 @@ struct snd_enc_real {
 } __attribute__((packed, aligned(4)));
 
 /**
- * struct snd_enc_flac
+ * struct snd_enc_flac - FLAC encoder parameters
  * @num: serial number, valid only for OGG formats
  *	needs to be set by application
  * @gain: Add replay gain tags
diff --git a/include/uapi/sound/sof/abi.h b/include/uapi/sound/sof/abi.h
index 0e7dccdc25fd..3566630ca965 100644
--- a/include/uapi/sound/sof/abi.h
+++ b/include/uapi/sound/sof/abi.h
@@ -24,9 +24,11 @@
 #ifndef __INCLUDE_UAPI_SOUND_SOF_ABI_H__
 #define __INCLUDE_UAPI_SOUND_SOF_ABI_H__
 
+#include <linux/types.h>
+
 /* SOF ABI version major, minor and patch numbers */
 #define SOF_ABI_MAJOR 3
-#define SOF_ABI_MINOR 21
+#define SOF_ABI_MINOR 23
 #define SOF_ABI_PATCH 0
 
 /* SOF ABI version number. Format within 32bit word is MMmmmppp */
diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h
index dbf137516522..e9bba93a5399 100644
--- a/include/uapi/sound/sof/header.h
+++ b/include/uapi/sound/sof/header.h
@@ -26,4 +26,34 @@ struct sof_abi_hdr {
 	__u32 data[];		/**< Component data - opaque to core */
 }  __packed;
 
+#define SOF_MANIFEST_DATA_TYPE_NHLT 1
+
+/**
+ * struct sof_manifest_tlv - SOF manifest TLV data
+ * @type: type of data
+ * @size: data size (not including the size of this struct)
+ * @data: payload data
+ */
+struct sof_manifest_tlv {
+	__le32 type;
+	__le32 size;
+	__u8 data[];
+};
+
+/**
+ * struct sof_manifest - SOF topology manifest
+ * @abi_major: Major ABI version
+ * @abi_minor: Minor ABI version
+ * @abi_patch: ABI patch
+ * @count: count of tlv items
+ * @items: consecutive variable size tlv items
+ */
+struct sof_manifest {
+	__le16 abi_major;
+	__le16 abi_minor;
+	__le16 abi_patch;
+	__le16 count;
+	struct sof_manifest_tlv items[];
+};
+
 #endif
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index b72fa385bebf..5caf75cadaf8 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -52,11 +52,17 @@
 #define SOF_TKN_SCHED_FRAMES			204
 #define SOF_TKN_SCHED_TIME_DOMAIN		205
 #define SOF_TKN_SCHED_DYNAMIC_PIPELINE		206
+#define SOF_TKN_SCHED_LP_MODE			207
+#define SOF_TKN_SCHED_MEM_USAGE			208
 
 /* volume */
 #define SOF_TKN_VOLUME_RAMP_STEP_TYPE		250
 #define SOF_TKN_VOLUME_RAMP_STEP_MS		251
 
+#define SOF_TKN_GAIN_RAMP_TYPE			260
+#define SOF_TKN_GAIN_RAMP_DURATION		261
+#define SOF_TKN_GAIN_VAL			262
+
 /* SRC */
 #define SOF_TKN_SRC_RATE_IN			300
 #define SOF_TKN_SRC_RATE_OUT			301
@@ -79,6 +85,9 @@
  */
 #define SOF_TKN_COMP_CORE_ID			404
 #define SOF_TKN_COMP_UUID                       405
+#define SOF_TKN_COMP_CPC			406
+#define SOF_TKN_COMP_IS_PAGES			409
+#define SOF_TKN_COMP_NUM_AUDIO_FORMATS		410
 
 /* SSP */
 #define SOF_TKN_INTEL_SSP_CLKS_CONTROL		500
@@ -145,4 +154,39 @@
 #define SOF_TKN_MEDIATEK_AFE_CH			1601
 #define SOF_TKN_MEDIATEK_AFE_FORMAT		1602
 
+/* MIXER */
+#define SOF_TKN_MIXER_TYPE			1700
+
+/* ACPDMIC */
+#define SOF_TKN_AMD_ACPDMIC_RATE		1800
+#define SOF_TKN_AMD_ACPDMIC_CH			1801
+
+/* CAVS AUDIO FORMAT */
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_RATE	1900
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_BIT_DEPTH	1901
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_VALID_BIT	1902
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_CHANNELS	1903
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_CH_MAP	1904
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_CH_CFG	1905
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_INTERLEAVING_STYLE	1906
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_FMT_CFG	1907
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IN_SAMPLE_TYPE	1908
+/* intentional token numbering discontinuity, reserved for future use */
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_RATE	1930
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_BIT_DEPTH	1931
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_VALID_BIT	1932
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_CHANNELS	1933
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_CH_MAP	1934
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_CH_CFG	1935
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_INTERLEAVING_STYLE	1936
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_FMT_CFG	1937
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OUT_SAMPLE_TYPE	1938
+/* intentional token numbering discontinuity, reserved for future use */
+#define SOF_TKN_CAVS_AUDIO_FORMAT_IBS		1970
+#define SOF_TKN_CAVS_AUDIO_FORMAT_OBS		1971
+#define SOF_TKN_CAVS_AUDIO_FORMAT_DMA_BUFFER_SIZE	1972
+
+/* COPIER */
+#define SOF_TKN_INTEL_COPIER_NODE_TYPE		1980
+
 #endif