Merge tag 'sound-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "It's been relatively calm in this cycle from the feature POV, but
  there were lots of cleanup works in the wide-range of code for
  converting with the auto-cleanup macros like guard().

  The mostly user-visible changes are the support of a couple of new
  compress-offload API extensions, and the support of new ASoC codec /
  platform drivers as well as USB-audio quirks.

  Here we go with some highlights:

  Core:
   - Compress-offload API extension for 64bit timestamp support
   - Compress-offload API extension for OPUS codec support
   - Workaround for PCM locking issue with PREEMPT_RT and softirq
   - KCSAN warning fix for ALSA sequencer core

  ASoC:
   - Continued cleanup works for ASoC core APIs
   - Lots of cleanups and conversions of DT bindings
   - Substantial maintainance work on the Intel AVS drivers
   - Support for Qualcomm Glymur and PM4125, Realtek RT1321, Shanghai
     FourSemi FS2104/5S, Texas Instruments PCM1754 and TAS2783A
   - Remove support for TI WL1273 for old Nokia systems

  USB-audio:
   - Support for Tascam US-144mkII, Presonus S1824c support
   - More flexible quirk option handling
   - Fix for USB MIDI timer bug triggered by fuzzer

  Others:
   - A large series of cleanups with guard() & co macros over (non-ASoC)
     sound drivers (PCI, ISA, HD-audio, USB-audio, drivers, etc)
   - TAS5825 HD-audio side-codec support"

* tag 'sound-6.18-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (454 commits)
  ALSA: usb-audio: don't hardcode gain for output channel of Presonus Studio
  ALSA: usb-audio: add the initial mix for Presonus Studio 1824c
  ALSA: doc: improved docs about quirk_flags in snd-usb-audio
  ALSA: usb-audio: make param quirk_flags change-able in runtime
  ALSA: usb-audio: improve module param quirk_flags
  ALSA: usb-audio: add two-way convert between name and bit for QUIRK_FLAG_*
  ALSA: usb-audio: fix race condition to UAF in snd_usbmidi_free
  ALSA: usb-audio: add mono main switch to Presonus S1824c
  ALSA: compress: document 'chan_map' member in snd_dec_opus
  ASoC: cs35l56: Add support for CS35L56 B2 silicon
  ASoC: cs35l56: Set fw_regs table after getting REVID
  ALSA: hda/realtek: Add quirk for HP Spectre 14t-ea100
  ASoc: tas2783A: Fix an error code in probe()
  ASoC: tlv320aic3x: Fix class-D initialization for tlv320aic3007
  ASoC: qcom: sc8280xp: use sa8775p/ subdir for QCS9100 / QCS9075
  ASoC: stm32: sai: manage context in set_sysclk callback
  ASoC: renesas: msiof: ignore 1st FSERR
  ASoC: renesas: msiof: Add note for The possibility of R/L opposite Capture
  ASoC: renesas: msiof: setup both (Playback/Capture) in the same time
  ASoC: renesas: msiof: tidyup DMAC stop timing
  ...

26 files changed, 543 insertions, 154 deletions

diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h
index 0832776262ac..e6a3476bcef1 100644
--- a/include/linux/soundwire/sdw.h
+++ b/include/linux/soundwire/sdw.h
@@ -19,6 +19,7 @@
 
 struct dentry;
 struct fwnode_handle;
+struct device_node;
 
 struct sdw_bus;
 struct sdw_slave;
@@ -1086,6 +1087,10 @@ int sdw_stream_add_slave(struct sdw_slave *slave,
 int sdw_stream_remove_slave(struct sdw_slave *slave,
 			    struct sdw_stream_runtime *stream);
 
+struct device *of_sdw_find_device_by_node(struct device_node *np);
+
+int sdw_slave_get_current_bank(struct sdw_slave *sdev);
+
 int sdw_slave_get_scale_index(struct sdw_slave *slave, u8 *base);
 
 /* messaging and data APIs */
@@ -1119,6 +1124,18 @@ static inline int sdw_stream_remove_slave(struct sdw_slave *slave,
 	return -EINVAL;
 }
 
+static inline struct device *of_sdw_find_device_by_node(struct device_node *np)
+{
+	WARN_ONCE(1, "SoundWire API is disabled");
+	return NULL;
+}
+
+static inline int sdw_slave_get_current_bank(struct sdw_slave *sdev)
+{
+	WARN_ONCE(1, "SoundWire API is disabled");
+	return -EINVAL;
+}
+
 /* messaging and data APIs */
 static inline int sdw_read(struct sdw_slave *slave, u32 addr)
 {
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index b55c9eeb2b54..9e3d801e45ec 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -161,7 +161,7 @@ struct snd_compr_ops {
 			struct snd_compr_metadata *metadata);
 	int (*trigger)(struct snd_compr_stream *stream, int cmd);
 	int (*pointer)(struct snd_compr_stream *stream,
-			struct snd_compr_tstamp *tstamp);
+		       struct snd_compr_tstamp64 *tstamp);
 	int (*copy)(struct snd_compr_stream *stream, char __user *buf,
 		       size_t count);
 	int (*mmap)(struct snd_compr_stream *stream,
diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h
index 5459c221badf..43a87a39110c 100644
--- a/include/sound/cs-amp-lib.h
+++ b/include/sound/cs-amp-lib.h
@@ -49,6 +49,7 @@ int cs_amp_write_cal_coeffs(struct cs_dsp *dsp,
 			    const struct cirrus_amp_cal_data *data);
 int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index,
 				    struct cirrus_amp_cal_data *out_data);
+int cs_amp_get_vendor_spkid(struct device *dev);
 
 struct cs_amp_test_hooks {
 	efi_status_t (*get_efi_variable)(efi_char16_t *name,
diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h
index 7c8bbe8ad1e2..ab044ce2aa8b 100644
--- a/include/sound/cs35l56.h
+++ b/include/sound/cs35l56.h
@@ -85,7 +85,9 @@
 #define CS35L56_DSP1_XMEM_UNPACKED24_0			0x2800000
 #define CS35L56_DSP1_FW_VER				0x2800010
 #define CS35L56_DSP1_HALO_STATE				0x28021E0
+#define CS35L56_B2_DSP1_HALO_STATE			0x2803D20
 #define CS35L56_DSP1_PM_CUR_STATE			0x2804308
+#define CS35L56_B2_DSP1_PM_CUR_STATE			0x2804678
 #define CS35L56_DSP1_XMEM_UNPACKED24_8191		0x2807FFC
 #define CS35L56_DSP1_CORE_BASE				0x2B80000
 #define CS35L56_DSP1_SCRATCH1				0x2B805C0
@@ -337,9 +339,6 @@ extern const struct regmap_config cs35l56_regmap_sdw;
 extern const struct regmap_config cs35l63_regmap_i2c;
 extern const struct regmap_config cs35l63_regmap_sdw;
 
-extern const struct cs35l56_fw_reg cs35l56_fw_reg;
-extern const struct cs35l56_fw_reg cs35l63_fw_reg;
-
 extern const struct cirrus_amp_cal_controls cs35l56_calibration_controls;
 
 extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC];
diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h
index 1ef13bcdc43f..9472f0a966a2 100644
--- a/include/sound/dmaengine_pcm.h
+++ b/include/sound/dmaengine_pcm.h
@@ -69,6 +69,10 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream)
  * @peripheral_config: peripheral configuration for programming peripheral
  * for dmaengine transfer
  * @peripheral_size: peripheral configuration buffer size
+ * @port_window_size: The length of the register area in words the data need
+ * to be accessed on the device side. It is only used for devices which is using
+ * an area instead of a single register to send/receive the data. Typically the
+ * DMA loops in this area in order to transfer the data.
  */
 struct snd_dmaengine_dai_dma_data {
 	dma_addr_t addr;
@@ -80,6 +84,7 @@ struct snd_dmaengine_dai_dma_data {
 	unsigned int flags;
 	void *peripheral_config;
 	size_t peripheral_size;
+	u32 port_window_size;
 };
 
 void snd_dmaengine_pcm_set_config_from_dai_data(
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 38db50b280eb..4f94565c9d15 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1842,8 +1842,7 @@ unsigned int snd_emu10k1_ptr20_read(struct snd_emu10k1 * emu, unsigned int reg,
 void snd_emu10k1_ptr20_write(struct snd_emu10k1 *emu, unsigned int reg, unsigned int chn, unsigned int data);
 int snd_emu10k1_spi_write(struct snd_emu10k1 * emu, unsigned int data);
 int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, u32 reg, u32 value);
-static inline void snd_emu1010_fpga_lock(struct snd_emu10k1 *emu) { mutex_lock(&emu->emu1010.lock); };
-static inline void snd_emu1010_fpga_unlock(struct snd_emu10k1 *emu) { mutex_unlock(&emu->emu1010.lock); };
+DEFINE_GUARD(snd_emu1010_fpga_lock, struct snd_emu10k1 *, mutex_lock(&(_T)->emu1010.lock), mutex_unlock(&(_T)->emu1010.lock))
 void snd_emu1010_fpga_write_lock(struct snd_emu10k1 *emu, u32 reg, u32 value);
 void snd_emu1010_fpga_write(struct snd_emu10k1 *emu, u32 reg, u32 value);
 void snd_emu1010_fpga_read(struct snd_emu10k1 *emu, u32 reg, u32 *value);
diff --git a/include/sound/gus.h b/include/sound/gus.h
index 1c8fb6c93e50..321ae93625eb 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -515,7 +515,6 @@ struct _SND_IW_LFO_PROGRAM {
 
 /* gus_mem.c */
 
-void snd_gf1_mem_lock(struct snd_gf1_mem * alloc, int xup);
 int snd_gf1_mem_xfree(struct snd_gf1_mem * alloc, struct snd_gf1_mem_block * block);
 struct snd_gf1_mem_block *snd_gf1_mem_alloc(struct snd_gf1_mem * alloc, int owner,
 				       char *name, int size, int w_16,
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index ddc9c392f93f..5d9f0ef228af 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -360,8 +360,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 			   hda_nid_t nid, int recursive);
 unsigned int snd_hda_get_num_devices(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
-			u8 *dev_list, int max_devices);
+unsigned int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+				u8 *dev_list, unsigned int max_devices);
 int snd_hda_get_dev_select(struct hda_codec *codec, hda_nid_t nid);
 int snd_hda_set_dev_select(struct hda_codec *codec, hda_nid_t nid, int dev_id);
 
@@ -503,6 +503,36 @@ static inline bool hda_codec_need_resume(struct hda_codec *codec)
 	return !codec->relaxed_resume && codec->jacktbl.used;
 }
 
+/*
+ * PM with auto-cleanup: call like CLASS(snd_hda_power, pm)(codec)
+ * If the error handling is needed, refer pm.err.
+ */
+struct __hda_power_obj {
+	struct hda_codec *codec;
+	int err;
+};
+
+static inline struct __hda_power_obj __snd_hda_power_up(struct hda_codec *codec)
+{
+	struct __hda_power_obj T = { .codec = codec };
+	T.err = snd_hda_power_up(codec);
+	return T;
+}
+
+static inline struct __hda_power_obj __snd_hda_power_up_pm(struct hda_codec *codec)
+{
+	struct __hda_power_obj T = { .codec = codec };
+	T.err = snd_hda_power_up_pm(codec);
+	return T;
+}
+
+DEFINE_CLASS(snd_hda_power, struct __hda_power_obj,
+	     snd_hda_power_down((_T).codec), __snd_hda_power_up(codec),
+	     struct hda_codec *codec)
+DEFINE_CLASS(snd_hda_power_pm, struct __hda_power_obj,
+	     snd_hda_power_down_pm((_T).codec), __snd_hda_power_up_pm(codec),
+	     struct hda_codec *codec)
+
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
 /*
  * patch firmware
diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h
index d38234f8fe44..4e0c1d8af09f 100644
--- a/include/sound/hdaudio.h
+++ b/include/sound/hdaudio.h
@@ -651,6 +651,7 @@ int snd_hdac_stream_set_lpib(struct hdac_stream *azx_dev, u32 value);
 #define snd_hdac_dsp_lock(dev)		mutex_lock(&(dev)->dsp_mutex)
 #define snd_hdac_dsp_unlock(dev)	mutex_unlock(&(dev)->dsp_mutex)
 #define snd_hdac_stream_is_locked(dev)	((dev)->locked)
+DEFINE_GUARD(snd_hdac_dsp_lock, struct hdac_stream *, snd_hdac_dsp_lock(_T), snd_hdac_dsp_unlock(_T))
 /* DSP loader helpers */
 int snd_hdac_dsp_prepare(struct hdac_stream *azx_dev, unsigned int format,
 			 unsigned int byte_size, struct snd_dma_buffer *bufp);
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 2caa807c6249..d78cda866888 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -47,7 +47,7 @@ struct snd_compress_ops {
 		       struct snd_compr_stream *stream, int cmd);
 	int (*pointer)(struct snd_soc_component *component,
 		       struct snd_compr_stream *stream,
-		       struct snd_compr_tstamp *tstamp);
+		       struct snd_compr_tstamp64 *tstamp);
 	int (*copy)(struct snd_soc_component *component,
 		    struct snd_compr_stream *stream, char __user *buf,
 		    size_t count);
@@ -261,89 +261,18 @@ struct snd_soc_component {
 	list_for_each_entry_safe(dai, _dai, &(component)->dai_list, list)
 
 /**
- * snd_soc_dapm_to_component() - Casts a DAPM context to the component it is
- *  embedded in
- * @dapm: The DAPM context to cast to the component
- *
- * This function must only be used on DAPM contexts that are known to be part of
- * a component (e.g. in a component driver). Otherwise the behavior is
- * undefined.
- */
-static inline struct snd_soc_component *snd_soc_dapm_to_component(
-	struct snd_soc_dapm_context *dapm)
-{
-	return container_of(dapm, struct snd_soc_component, dapm);
-}
-
-/**
- * snd_soc_component_get_dapm() - Returns the DAPM context associated with a
+ * snd_soc_component_to_dapm() - Returns the DAPM context associated with a
  *  component
  * @component: The component for which to get the DAPM context
  */
-static inline struct snd_soc_dapm_context *snd_soc_component_get_dapm(
+static inline struct snd_soc_dapm_context *snd_soc_component_to_dapm(
 	struct snd_soc_component *component)
 {
 	return &component->dapm;
 }
 
-/**
- * snd_soc_component_init_bias_level() - Initialize COMPONENT DAPM bias level
- * @component: The COMPONENT for which to initialize the DAPM bias level
- * @level: The DAPM level to initialize to
- *
- * Initializes the COMPONENT DAPM bias level. See snd_soc_dapm_init_bias_level()
- */
-static inline void
-snd_soc_component_init_bias_level(struct snd_soc_component *component,
-				  enum snd_soc_bias_level level)
-{
-	snd_soc_dapm_init_bias_level(
-		snd_soc_component_get_dapm(component), level);
-}
-
-/**
- * snd_soc_component_get_bias_level() - Get current COMPONENT DAPM bias level
- * @component: The COMPONENT for which to get the DAPM bias level
- *
- * Returns: The current DAPM bias level of the COMPONENT.
- */
-static inline enum snd_soc_bias_level
-snd_soc_component_get_bias_level(struct snd_soc_component *component)
-{
-	return snd_soc_dapm_get_bias_level(
-		snd_soc_component_get_dapm(component));
-}
-
-/**
- * snd_soc_component_force_bias_level() - Set the COMPONENT DAPM bias level
- * @component: The COMPONENT for which to set the level
- * @level: The level to set to
- *
- * Forces the COMPONENT bias level to a specific state. See
- * snd_soc_dapm_force_bias_level().
- */
-static inline int
-snd_soc_component_force_bias_level(struct snd_soc_component *component,
-				   enum snd_soc_bias_level level)
-{
-	return snd_soc_dapm_force_bias_level(
-		snd_soc_component_get_dapm(component),
-		level);
-}
-
-/**
- * snd_soc_dapm_kcontrol_component() - Returns the component associated to a
- * kcontrol
- * @kcontrol: The kcontrol
- *
- * This function must only be used on DAPM contexts that are known to be part of
- * a COMPONENT (e.g. in a COMPONENT driver). Otherwise the behavior is undefined
- */
-static inline struct snd_soc_component *snd_soc_dapm_kcontrol_component(
-	struct snd_kcontrol *kcontrol)
-{
-	return snd_soc_dapm_to_component(snd_soc_dapm_kcontrol_dapm(kcontrol));
-}
+// FIXME
+#define snd_soc_component_get_dapm	snd_soc_component_to_dapm
 
 /**
  * snd_soc_component_cache_sync() - Sync the register cache with the hardware
@@ -498,7 +427,7 @@ int snd_soc_component_compr_get_codec_caps(struct snd_compr_stream *cstream,
 					   struct snd_compr_codec_caps *codec);
 int snd_soc_component_compr_ack(struct snd_compr_stream *cstream, size_t bytes);
 int snd_soc_component_compr_pointer(struct snd_compr_stream *cstream,
-				    struct snd_compr_tstamp *tstamp);
+				    struct snd_compr_tstamp64 *tstamp);
 int snd_soc_component_compr_copy(struct snd_compr_stream *cstream,
 				 char __user *buf, size_t count);
 int snd_soc_component_compr_set_metadata(struct snd_compr_stream *cstream,
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 166c29557e9d..224396927aef 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -256,7 +256,7 @@ int snd_soc_dai_compr_ack(struct snd_soc_dai *dai,
 			  size_t bytes);
 int snd_soc_dai_compr_pointer(struct snd_soc_dai *dai,
 			      struct snd_compr_stream *cstream,
-			      struct snd_compr_tstamp *tstamp);
+			      struct snd_compr_tstamp64 *tstamp);
 int snd_soc_dai_compr_set_metadata(struct snd_soc_dai *dai,
 				   struct snd_compr_stream *cstream,
 				   struct snd_compr_metadata *metadata);
@@ -383,8 +383,9 @@ struct snd_soc_cdai_ops {
 			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 (*pointer)(struct snd_compr_stream *stream,
+		       struct snd_compr_tstamp64 *tstamp,
+		       struct snd_soc_dai *dai);
 	int (*ack)(struct snd_compr_stream *, size_t,
 			struct snd_soc_dai *);
 };
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 0b5c7e6a90c8..75941324886b 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -583,11 +583,9 @@ struct snd_soc_dapm_update {
 struct snd_soc_dapm_context {
 	enum snd_soc_bias_level bias_level;
 
-	/* bit field */
-	unsigned int idle_bias_off:1;		/* Use BIAS_OFF instead of STANDBY */
-	unsigned int suspend_bias_off:1;	/* Use BIAS_OFF in suspend if the DAPM is idle */
+	bool idle_bias;				/* Use BIAS_OFF instead of STANDBY when false */
 
-	struct device *dev;			/* from parent - for debug */
+	struct device *dev;			/* from parent - for debug */ /* REMOVE ME */
 	struct snd_soc_component *component;	/* parent component */
 	struct snd_soc_card *card;		/* parent card */
 
@@ -660,6 +658,12 @@ 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);
 int snd_soc_dapm_widget_name_cmp(struct snd_soc_dapm_widget *widget, const char *s);
+struct device *snd_soc_dapm_to_dev(struct snd_soc_dapm_context *dapm);
+struct snd_soc_card *snd_soc_dapm_to_card(struct snd_soc_dapm_context *dapm);
+struct snd_soc_component *snd_soc_dapm_to_component(struct snd_soc_dapm_context *dapm);
+
+bool snd_soc_dapm_get_idle_bias(struct snd_soc_dapm_context *dapm);
+void snd_soc_dapm_set_idle_bias(struct snd_soc_dapm_context *dapm, bool on);
 
 /* dapm path setup */
 int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
@@ -699,7 +703,6 @@ int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin);
 int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, const char *pin);
 int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin);
-unsigned int dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol);
 void snd_soc_dapm_mark_endpoints_dirty(struct snd_soc_card *card);
 
 /*
@@ -718,10 +721,23 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
 	bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, enum snd_soc_dapm_direction));
 void snd_soc_dapm_dai_free_widgets(struct snd_soc_dapm_widget_list **list);
 
-struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm(struct snd_kcontrol *kcontrol);
-struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_widget(struct snd_kcontrol *kcontrol);
+struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_to_dapm(struct snd_kcontrol *kcontrol);
+struct snd_soc_dapm_widget *snd_soc_dapm_kcontrol_to_widget(struct snd_kcontrol *kcontrol);
+struct snd_soc_component *snd_soc_dapm_kcontrol_to_component(struct snd_kcontrol *kcontrol);
+unsigned int snd_soc_dapm_kcontrol_get_value(const struct snd_kcontrol *kcontrol);
 
 int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level);
+enum snd_soc_bias_level snd_soc_dapm_get_bias_level(struct snd_soc_dapm_context *dapm);
+void snd_soc_dapm_init_bias_level(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level);
+
+// REMOVE ME !!
+#define snd_soc_component_force_bias_level(c, l)	snd_soc_dapm_force_bias_level(&(c)->dapm, l)
+#define snd_soc_component_get_bias_level(c)		snd_soc_dapm_get_bias_level(&(c)->dapm)
+#define snd_soc_component_init_bias_level(c, l)		snd_soc_dapm_init_bias_level(&(c)->dapm, l)
+#define snd_soc_dapm_kcontrol_widget			snd_soc_dapm_kcontrol_to_widget
+#define snd_soc_dapm_kcontrol_dapm			snd_soc_dapm_kcontrol_to_dapm
+#define dapm_kcontrol_get_value				snd_soc_dapm_kcontrol_get_value
+#define snd_soc_dapm_kcontrol_component			snd_soc_dapm_kcontrol_to_component
 
 #define for_each_dapm_widgets(list, i, widget)				\
 	for ((i) = 0;							\
@@ -729,37 +745,6 @@ int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm, enum snd_so
 	     (i)++)
 
 /**
- * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level
- * @dapm: The DAPM context to initialize
- * @level: The DAPM level to initialize to
- *
- * This function only sets the driver internal state of the DAPM level and will
- * not modify the state of the device. Hence it should not be used during normal
- * operation, but only to synchronize the internal state to the device state.
- * E.g. during driver probe to set the DAPM level to the one corresponding with
- * the power-on reset state of the device.
- *
- * To change the DAPM state of the device use snd_soc_dapm_set_bias_level().
- */
-static inline void snd_soc_dapm_init_bias_level(
-	struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level)
-{
-	dapm->bias_level = level;
-}
-
-/**
- * snd_soc_dapm_get_bias_level() - Get current DAPM bias level
- * @dapm: The context for which to get the bias level
- *
- * Returns: The current bias level of the passed DAPM context.
- */
-static inline enum snd_soc_bias_level snd_soc_dapm_get_bias_level(
-	struct snd_soc_dapm_context *dapm)
-{
-	return dapm->bias_level;
-}
-
-/**
  * snd_soc_dapm_widget_for_each_path - Iterates over all paths in the
  *   specified direction of a widget
  * @w: The widget
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 1fffef311c41..ddc508ff7b9b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -1120,6 +1120,11 @@ static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card)
 	return card && card->instantiated;
 }
 
+static inline struct snd_soc_dapm_context *snd_soc_card_to_dapm(struct snd_soc_card *card)
+{
+	return &card->dapm;
+}
+
 /* SoC machine DAI configuration, glues a codec and cpu DAI together */
 struct snd_soc_pcm_runtime {
 	struct device *dev;
diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 6049a5d0cfcd..3c5e9b2af7f1 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -248,5 +248,13 @@ int asoc_sdw_cs42l43_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_so
 int asoc_sdw_cs42l43_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
 int asoc_sdw_cs_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
 int asoc_sdw_maxim_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+/* TI */
+int asoc_sdw_ti_amp_init(struct snd_soc_card *card,
+			 struct snd_soc_dai_link *dai_links,
+			 struct asoc_sdw_codec_info *info,
+			 bool playback);
+int asoc_sdw_ti_spk_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int asoc_sdw_ti_amp_initial_settings(struct snd_soc_card *card,
+				     const char *name_prefix);
 
 #endif
diff --git a/include/sound/sof/ipc4/header.h b/include/sound/sof/ipc4/header.h
index e85c7afd85a4..15fac532688e 100644
--- a/include/sound/sof/ipc4/header.h
+++ b/include/sound/sof/ipc4/header.h
@@ -326,10 +326,14 @@ struct sof_ipc4_base_module_cfg {
 #define SOF_IPC4_MOD_INSTANCE_SHIFT		16
 #define SOF_IPC4_MOD_INSTANCE_MASK		GENMASK(23, 16)
 #define SOF_IPC4_MOD_INSTANCE(x)		((x) << SOF_IPC4_MOD_INSTANCE_SHIFT)
+#define SOF_IPC4_MOD_INSTANCE_GET(x)		(((x) & SOF_IPC4_MOD_INSTANCE_MASK) \
+						 >> SOF_IPC4_MOD_INSTANCE_SHIFT)
 
 #define SOF_IPC4_MOD_ID_SHIFT			0
 #define SOF_IPC4_MOD_ID_MASK			GENMASK(15, 0)
 #define SOF_IPC4_MOD_ID(x)			((x) << SOF_IPC4_MOD_ID_SHIFT)
+#define SOF_IPC4_MOD_ID_GET(x)			(((x) & SOF_IPC4_MOD_ID_MASK) \
+						 >> SOF_IPC4_MOD_ID_SHIFT)
 
 /* init module ipc msg */
 #define SOF_IPC4_MOD_EXT_PARAM_SIZE_SHIFT	0
diff --git a/include/sound/soundfont.h b/include/sound/soundfont.h
index 8a40cc15f66d..48f8cf6de3ac 100644
--- a/include/sound/soundfont.h
+++ b/include/sound/soundfont.h
@@ -114,5 +114,23 @@ int snd_sf_calc_parm_decay(int msec);
 extern int snd_sf_vol_table[128];
 int snd_sf_linear_to_log(unsigned int amount, int offset, int ratio);
 
+/* lock access to sflist */
+static inline void snd_soundfont_lock_preset(struct snd_sf_list *sflist)
+{
+	mutex_lock(&sflist->presets_mutex);
+	guard(spinlock_irqsave)(&sflist->lock);
+	sflist->presets_locked = 1;
+}
+
+/* remove lock */
+static inline void snd_soundfont_unlock_preset(struct snd_sf_list *sflist)
+{
+	guard(spinlock_irqsave)(&sflist->lock);
+	sflist->presets_locked = 0;
+	mutex_unlock(&sflist->presets_mutex);
+}
+
+DEFINE_GUARD(snd_soundfont_lock_preset, struct snd_sf_list *,
+	     snd_soundfont_lock_preset(_T), snd_soundfont_unlock_preset(_T))
 
 #endif /* __SOUND_SOUNDFONT_H */
diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h
index c3a9efa73d5d..dd6ee45ad096 100644
--- a/include/sound/tas2781-dsp.h
+++ b/include/sound/tas2781-dsp.h
@@ -34,6 +34,7 @@
 #define PPC3_VERSION_TAS2781_BASIC_MIN		0x14600
 #define PPC3_VERSION_TAS2781_ALPHA_MIN		0x4a00
 #define PPC3_VERSION_TAS2781_BETA_MIN		0x19400
+#define PPC3_VERSION_TAS5825_BASE		0x114200
 #define TASDEVICE_DEVICE_SUM			8
 #define TASDEVICE_CONFIG_SUM			64
 
@@ -53,6 +54,8 @@ enum tasdevice_dsp_dev_idx {
 	TASDEVICE_DSP_TAS_2781_DUAL_MONO,
 	TASDEVICE_DSP_TAS_2781_21,
 	TASDEVICE_DSP_TAS_2781_QUAD,
+	TASDEVICE_DSP_TAS_5825_MONO,
+	TASDEVICE_DSP_TAS_5825_DUAL,
 	TASDEVICE_DSP_TAS_MAX_DEVICE
 };
 
@@ -198,6 +201,14 @@ struct tasdevice_rca {
 	int ncfgs;
 	struct tasdevice_config_info **cfg_info;
 	int profile_cfg_id;
+	/*
+	 * Since version 0x105, the keyword 'init' was introduced into the
+	 * profile, which is used for chip initialization, particularly to
+	 * store common settings for other non-initialization profiles.
+	 * if (init_profile_id < 0)
+	 *         No init profile inside the RCA firmware.
+	 */
+	int init_profile_id;
 };
 
 void tasdevice_select_cfg_blk(void *context, int conf_no,
diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h
index 3875e92f1ec5..ddd997ac3216 100644
--- a/include/sound/tas2781.h
+++ b/include/sound/tas2781.h
@@ -49,10 +49,12 @@
 #define TASDEVICE_REG(book, page, reg)	(((book * 256 * 128) + \
 					(page * 128)) + reg)
 
-/* Software Reset */
+/* Software Reset, compatble with new device (TAS5825). */
 #define TASDEVICE_REG_SWRESET		TASDEVICE_REG(0x0, 0x0, 0x01)
 #define TASDEVICE_REG_SWRESET_RESET	BIT(0)
 
+#define TAS5825_REG_SWRESET_RESET	(BIT(0) | BIT(4))
+
 /* Checksum */
 #define TASDEVICE_CHECKSUM_REG		TASDEVICE_REG(0x0, 0x0, 0x7e)
 
@@ -110,8 +112,17 @@
 #define TAS2781_RUNTIME_RE_REG		TASDEVICE_REG(0x64, 0x63, 0x44)
 
 enum audio_device {
+	TAS2020,
+	TAS2118,
+	TAS2120,
+	TAS2320,
 	TAS2563,
+	TAS2570,
+	TAS2572,
 	TAS2781,
+	TAS5825,
+	TAS5827,
+	TAS_OTHERS,
 };
 
 enum dspbin_type {
@@ -194,6 +205,7 @@ struct tasdevice_priv {
 	unsigned char coef_binaryname[64];
 	unsigned char rca_binaryname[64];
 	unsigned char dev_name[32];
+	const unsigned char (*dvc_tlv_table)[4];
 	const char *name_prefix;
 	unsigned char ndev;
 	unsigned int dspbin_typ;
diff --git a/include/sound/tas2x20-tlv.h b/include/sound/tas2x20-tlv.h
new file mode 100644
index 000000000000..6e6bcec4a0a1
--- /dev/null
+++ b/include/sound/tas2x20-tlv.h
@@ -0,0 +1,259 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+//
+// ALSA SoC Texas Instruments TAS2x20/TAS2118 Audio Smart Amplifier
+//
+// Copyright (C) 2025 Texas Instruments Incorporated
+// https://www.ti.com
+//
+// The TAS2x20/TAS2118 hda driver implements for one, two, or even multiple
+// TAS2x20/TAS2118 chips.
+//
+// Author: Baojun Xu <baojun.xu@ti.com>
+//
+
+#ifndef __TAS2X20_TLV_H__
+#define __TAS2X20_TLV_H__
+
+#define TAS2X20_DVC_LEVEL		TASDEVICE_REG(0x0, 0x2, 0x0c)
+#define TAS2X20_AMP_LEVEL		TASDEVICE_REG(0x0, 0x0, 0x07)
+
+static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_dvc_tlv, 1650, 50, 0);
+static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2x20_amp_tlv, 2100, 50, 0);
+
+/* pow(10, db/20) * pow(2,22) */
+static const __maybe_unused unsigned char tas2x20_dvc_table[][4] = {
+	{ 0X00, 0X00, 0X0D, 0X00 }, /* -110.0db */
+	{ 0X00, 0X00, 0X0E, 0X00 }, /* -109.5db */
+	{ 0X00, 0X00, 0X0E, 0X00 }, /* -109.0db */
+	{ 0X00, 0X00, 0X0F, 0X00 }, /* -108.5db */
+	{ 0X00, 0X00, 0X10, 0X00 }, /* -108.0db */
+	{ 0X00, 0X00, 0X11, 0X00 }, /* -107.5db */
+	{ 0X00, 0X00, 0X12, 0X00 }, /* -107.0db */
+	{ 0X00, 0X00, 0X13, 0X00 }, /* -106.5db */
+	{ 0X00, 0X00, 0X15, 0X00 }, /* -106.0db */
+	{ 0X00, 0X00, 0X16, 0X00 }, /* -105.5db */
+	{ 0X00, 0X00, 0X17, 0X00 }, /* -105.0db */
+	{ 0X00, 0X00, 0X18, 0X00 }, /* -104.5db */
+	{ 0X00, 0X00, 0X1A, 0X00 }, /* -104.0db */
+	{ 0X00, 0X00, 0X1C, 0X00 }, /* -103.5db */
+	{ 0X00, 0X00, 0X1D, 0X00 }, /* -103.0db */
+	{ 0X00, 0X00, 0X1F, 0X00 }, /* -102.5db */
+	{ 0X00, 0X00, 0X21, 0X00 }, /* -102.0db */
+	{ 0X00, 0X00, 0X23, 0X00 }, /* -101.5db */
+	{ 0X00, 0X00, 0X25, 0X00 }, /* -101.0db */
+	{ 0X00, 0X00, 0X27, 0X00 }, /* -100.5db */
+	{ 0X00, 0X00, 0X29, 0X00 }, /* -100.0db */
+	{ 0X00, 0X00, 0X2C, 0X00 }, /* -99.5db */
+	{ 0X00, 0X00, 0X2F, 0X00 }, /* -99.0db */
+	{ 0X00, 0X00, 0X31, 0X00 }, /* -98.5db */
+	{ 0X00, 0X00, 0X34, 0X00 }, /* -98.0db */
+	{ 0X00, 0X00, 0X37, 0X00 }, /* -97.5db */
+	{ 0X00, 0X00, 0X3B, 0X00 }, /* -97.0db */
+	{ 0X00, 0X00, 0X3E, 0X00 }, /* -96.5db */
+	{ 0X00, 0X00, 0X42, 0X00 }, /* -96.0db */
+	{ 0X00, 0X00, 0X46, 0X00 }, /* -95.5db */
+	{ 0X00, 0X00, 0X4A, 0X00 }, /* -95.0db */
+	{ 0X00, 0X00, 0X4F, 0X00 }, /* -94.5db */
+	{ 0X00, 0X00, 0X53, 0X00 }, /* -94.0db */
+	{ 0X00, 0X00, 0X58, 0X00 }, /* -93.5db */
+	{ 0X00, 0X00, 0X5D, 0X00 }, /* -93.0db */
+	{ 0X00, 0X00, 0X63, 0X00 }, /* -92.5db */
+	{ 0X00, 0X00, 0X69, 0X00 }, /* -92.0db */
+	{ 0X00, 0X00, 0X6F, 0X00 }, /* -91.5db */
+	{ 0X00, 0X00, 0X76, 0X00 }, /* -91.0db */
+	{ 0X00, 0X00, 0X7D, 0X00 }, /* -90.5db */
+	{ 0X00, 0X00, 0X84, 0X00 }, /* -90.0db */
+	{ 0X00, 0X00, 0X8C, 0X00 }, /* -89.5db */
+	{ 0X00, 0X00, 0X94, 0X00 }, /* -89.0db */
+	{ 0X00, 0X00, 0X9D, 0X00 }, /* -88.5db */
+	{ 0X00, 0X00, 0XA6, 0X00 }, /* -88.0db */
+	{ 0X00, 0X00, 0XB0, 0X00 }, /* -87.5db */
+	{ 0X00, 0X00, 0XBB, 0X00 }, /* -87.0db */
+	{ 0X00, 0X00, 0XC6, 0X00 }, /* -86.5db */
+	{ 0X00, 0X00, 0XD2, 0X00 }, /* -86.0db */
+	{ 0X00, 0X00, 0XDE, 0X00 }, /* -85.5db */
+	{ 0X00, 0X00, 0XEB, 0X00 }, /* -85.0db */
+	{ 0X00, 0X00, 0XF9, 0X00 }, /* -84.5db */
+	{ 0X00, 0X01, 0X08, 0X00 }, /* -84.0db */
+	{ 0X00, 0X01, 0X18, 0X00 }, /* -83.5db */
+	{ 0X00, 0X01, 0X28, 0X00 }, /* -83.0db */
+	{ 0X00, 0X01, 0X3A, 0X00 }, /* -82.5db */
+	{ 0X00, 0X01, 0X4D, 0X00 }, /* -82.0db */
+	{ 0X00, 0X01, 0X60, 0X00 }, /* -81.5db */
+	{ 0X00, 0X01, 0X75, 0X00 }, /* -81.0db */
+	{ 0X00, 0X01, 0X8B, 0X00 }, /* -80.5db */
+	{ 0X00, 0X01, 0XA3, 0X00 }, /* -80.0db */
+	{ 0X00, 0X01, 0XBC, 0X00 }, /* -79.5db */
+	{ 0X00, 0X01, 0XD6, 0X00 }, /* -79.0db */
+	{ 0X00, 0X01, 0XF2, 0X00 }, /* -78.5db */
+	{ 0X00, 0X02, 0X10, 0X00 }, /* -78.0db */
+	{ 0X00, 0X02, 0X2F, 0X00 }, /* -77.5db */
+	{ 0X00, 0X02, 0X50, 0X00 }, /* -77.0db */
+	{ 0X00, 0X02, 0X73, 0X00 }, /* -76.5db */
+	{ 0X00, 0X02, 0X98, 0X00 }, /* -76.0db */
+	{ 0X00, 0X02, 0XC0, 0X00 }, /* -75.5db */
+	{ 0X00, 0X02, 0XE9, 0X00 }, /* -75.0db */
+	{ 0X00, 0X03, 0X16, 0X00 }, /* -74.5db */
+	{ 0X00, 0X03, 0X44, 0X00 }, /* -74.0db */
+	{ 0X00, 0X03, 0X76, 0X00 }, /* -73.5db */
+	{ 0X00, 0X03, 0XAA, 0X00 }, /* -73.0db */
+	{ 0X00, 0X03, 0XE2, 0X00 }, /* -72.5db */
+	{ 0X00, 0X04, 0X1D, 0X00 }, /* -72.0db */
+	{ 0X00, 0X04, 0X5B, 0X00 }, /* -71.5db */
+	{ 0X00, 0X04, 0X9E, 0X00 }, /* -71.0db */
+	{ 0X00, 0X04, 0XE4, 0X00 }, /* -70.5db */
+	{ 0X00, 0X05, 0X2E, 0X00 }, /* -70.0db */
+	{ 0X00, 0X05, 0X7C, 0X00 }, /* -69.5db */
+	{ 0X00, 0X05, 0XD0, 0X00 }, /* -69.0db */
+	{ 0X00, 0X06, 0X28, 0X00 }, /* -68.5db */
+	{ 0X00, 0X06, 0X85, 0X00 }, /* -68.0db */
+	{ 0X00, 0X06, 0XE8, 0X00 }, /* -67.5db */
+	{ 0X00, 0X07, 0X51, 0X00 }, /* -67.0db */
+	{ 0X00, 0X07, 0XC0, 0X00 }, /* -66.5db */
+	{ 0X00, 0X08, 0X36, 0X00 }, /* -66.0db */
+	{ 0X00, 0X08, 0XB2, 0X00 }, /* -65.5db */
+	{ 0X00, 0X09, 0X36, 0X00 }, /* -65.0db */
+	{ 0X00, 0X09, 0XC2, 0X00 }, /* -64.5db */
+	{ 0X00, 0X0A, 0X56, 0X00 }, /* -64.0db */
+	{ 0X00, 0X0A, 0XF3, 0X00 }, /* -63.5db */
+	{ 0X00, 0X0B, 0X99, 0X00 }, /* -63.0db */
+	{ 0X00, 0X0C, 0X49, 0X00 }, /* -62.5db */
+	{ 0X00, 0X0D, 0X03, 0X00 }, /* -62.0db */
+	{ 0X00, 0X0D, 0XC9, 0X00 }, /* -61.5db */
+	{ 0X00, 0X0E, 0X9A, 0X00 }, /* -61.0db */
+	{ 0X00, 0X0F, 0X77, 0X00 }, /* -60.5db */
+	{ 0X00, 0X10, 0X62, 0X00 }, /* -60.0db */
+	{ 0X00, 0X11, 0X5A, 0X00 }, /* -59.5db */
+	{ 0X00, 0X12, 0X62, 0X00 }, /* -59.0db */
+	{ 0X00, 0X13, 0X78, 0X00 }, /* -58.5db */
+	{ 0X00, 0X14, 0XA0, 0X00 }, /* -58.0db */
+	{ 0X00, 0X15, 0XD9, 0X00 }, /* -57.5db */
+	{ 0X00, 0X17, 0X24, 0X00 }, /* -57.0db */
+	{ 0X00, 0X18, 0X83, 0X00 }, /* -56.5db */
+	{ 0X00, 0X19, 0XF7, 0X00 }, /* -56.0db */
+	{ 0X00, 0X1B, 0X81, 0X00 }, /* -55.5db */
+	{ 0X00, 0X1D, 0X22, 0X00 }, /* -55.0db */
+	{ 0X00, 0X1E, 0XDC, 0X00 }, /* -54.5db */
+	{ 0X00, 0X20, 0XB0, 0X00 }, /* -54.0db */
+	{ 0X00, 0X22, 0XA0, 0X00 }, /* -53.5db */
+	{ 0X00, 0X24, 0XAD, 0X00 }, /* -53.0db */
+	{ 0X00, 0X26, 0XDA, 0X00 }, /* -52.5db */
+	{ 0X00, 0X29, 0X27, 0X00 }, /* -52.0db */
+	{ 0X00, 0X2B, 0X97, 0X00 }, /* -51.5db */
+	{ 0X00, 0X2E, 0X2D, 0X00 }, /* -51.0db */
+	{ 0X00, 0X30, 0XE9, 0X00 }, /* -50.5db */
+	{ 0X00, 0X33, 0XCF, 0X00 }, /* -50.0db */
+	{ 0X00, 0X36, 0XE1, 0X00 }, /* -49.5db */
+	{ 0X00, 0X3A, 0X21, 0X00 }, /* -49.0db */
+	{ 0X00, 0X3D, 0X93, 0X00 }, /* -48.5db */
+	{ 0X00, 0X41, 0X39, 0X00 }, /* -48.0db */
+	{ 0X00, 0X45, 0X17, 0X00 }, /* -47.5db */
+	{ 0X00, 0X49, 0X2F, 0X00 }, /* -47.0db */
+	{ 0X00, 0X4D, 0X85, 0X00 }, /* -46.5db */
+	{ 0X00, 0X52, 0X1D, 0X00 }, /* -46.0db */
+	{ 0X00, 0X56, 0XFA, 0X00 }, /* -45.5db */
+	{ 0X00, 0X5C, 0X22, 0X00 }, /* -45.0db */
+	{ 0X00, 0X61, 0X97, 0X00 }, /* -44.5db */
+	{ 0X00, 0X67, 0X60, 0X00 }, /* -44.0db */
+	{ 0X00, 0X6D, 0X80, 0X00 }, /* -43.5db */
+	{ 0X00, 0X73, 0XFD, 0X00 }, /* -43.0db */
+	{ 0X00, 0X7A, 0XDC, 0X00 }, /* -42.5db */
+	{ 0X00, 0X82, 0X24, 0X00 }, /* -42.0db */
+	{ 0X00, 0X89, 0XDA, 0X00 }, /* -41.5db */
+	{ 0X00, 0X92, 0X05, 0X00 }, /* -41.0db */
+	{ 0X00, 0X9A, 0XAC, 0X00 }, /* -40.5db */
+	{ 0X00, 0XA3, 0XD7, 0X00 }, /* -40.0db */
+	{ 0X00, 0XAD, 0X8C, 0X00 }, /* -39.5db */
+	{ 0X00, 0XB7, 0XD4, 0X00 }, /* -39.0db */
+	{ 0X00, 0XC2, 0XB9, 0X00 }, /* -38.5db */
+	{ 0X00, 0XCE, 0X43, 0X00 }, /* -38.0db */
+	{ 0X00, 0XDA, 0X7B, 0X00 }, /* -37.5db */
+	{ 0X00, 0XE7, 0X6E, 0X00 }, /* -37.0db */
+	{ 0X00, 0XF5, 0X24, 0X00 }, /* -36.5db */
+	{ 0X01, 0X03, 0XAB, 0X00 }, /* -36.0db */
+	{ 0X01, 0X13, 0X0E, 0X00 }, /* -35.5db */
+	{ 0X01, 0X23, 0X5A, 0X00 }, /* -35.0db */
+	{ 0X01, 0X34, 0X9D, 0X00 }, /* -34.5db */
+	{ 0X01, 0X46, 0XE7, 0X00 }, /* -34.0db */
+	{ 0X01, 0X5A, 0X46, 0X00 }, /* -33.5db */
+	{ 0X01, 0X6E, 0XCA, 0X00 }, /* -33.0db */
+	{ 0X01, 0X84, 0X86, 0X00 }, /* -32.5db */
+	{ 0X01, 0X9B, 0X8C, 0X00 }, /* -32.0db */
+	{ 0X01, 0XB3, 0XEE, 0X00 }, /* -31.5db */
+	{ 0X01, 0XCD, 0XC3, 0X00 }, /* -31.0db */
+	{ 0X01, 0XE9, 0X20, 0X00 }, /* -30.5db */
+	{ 0X02, 0X06, 0X1B, 0X00 }, /* -30.0db */
+	{ 0X02, 0X24, 0XCE, 0X00 }, /* -29.5db */
+	{ 0X02, 0X45, 0X53, 0X00 }, /* -29.0db */
+	{ 0X02, 0X67, 0XC5, 0X00 }, /* -28.5db */
+	{ 0X02, 0X8C, 0X42, 0X00 }, /* -28.0db */
+	{ 0X02, 0XB2, 0XE8, 0X00 }, /* -27.5db */
+	{ 0X02, 0XDB, 0XD8, 0X00 }, /* -27.0db */
+	{ 0X03, 0X07, 0X36, 0X00 }, /* -26.5db */
+	{ 0X03, 0X35, 0X25, 0X00 }, /* -26.0db */
+	{ 0X03, 0X65, 0XCD, 0X00 }, /* -25.5db */
+	{ 0X03, 0X99, 0X57, 0X00 }, /* -25.0db */
+	{ 0X03, 0XCF, 0XEE, 0X00 }, /* -24.5db */
+	{ 0X04, 0X09, 0XC2, 0X00 }, /* -24.0db */
+	{ 0X04, 0X47, 0X03, 0X00 }, /* -23.5db */
+	{ 0X04, 0X87, 0XE5, 0X00 }, /* -23.0db */
+	{ 0X04, 0XCC, 0XA0, 0X00 }, /* -22.5db */
+	{ 0X05, 0X15, 0X6D, 0X00 }, /* -22.0db */
+	{ 0X05, 0X62, 0X8A, 0X00 }, /* -21.5db */
+	{ 0X05, 0XB4, 0X39, 0X00 }, /* -21.0db */
+	{ 0X06, 0X0A, 0XBF, 0X00 }, /* -20.5db */
+	{ 0X06, 0X66, 0X66, 0X00 }, /* -20.0db */
+	{ 0X06, 0XC7, 0X7B, 0X00 }, /* -19.5db */
+	{ 0X07, 0X2E, 0X50, 0X00 }, /* -19.0db */
+	{ 0X07, 0X9B, 0X3D, 0X00 }, /* -18.5db */
+	{ 0X08, 0X0E, 0X9F, 0X00 }, /* -18.0db */
+	{ 0X08, 0X88, 0XD7, 0X00 }, /* -17.5db */
+	{ 0X09, 0X0A, 0X4D, 0X00 }, /* -17.0db */
+	{ 0X09, 0X93, 0X6E, 0X00 }, /* -16.5db */
+	{ 0X0A, 0X24, 0XB0, 0X00 }, /* -16.0db */
+	{ 0X0A, 0XBE, 0X8D, 0X00 }, /* -15.5db */
+	{ 0X0B, 0X61, 0X88, 0X00 }, /* -15.0db */
+	{ 0X0C, 0X0E, 0X2B, 0X00 }, /* -14.5db */
+	{ 0X0C, 0XC5, 0X09, 0X00 }, /* -14.0db */
+	{ 0X0D, 0X86, 0XBD, 0X00 }, /* -13.5db */
+	{ 0X0E, 0X53, 0XEB, 0X00 }, /* -13.0db */
+	{ 0X0F, 0X2D, 0X42, 0X00 }, /* -12.5db */
+	{ 0X10, 0X13, 0X79, 0X00 }, /* -12.0db */
+	{ 0X11, 0X07, 0X54, 0X00 }, /* -11.5db */
+	{ 0X12, 0X09, 0XA3, 0X00 }, /* -11.0db */
+	{ 0X13, 0X1B, 0X40, 0X00 }, /* -10.5db */
+	{ 0X14, 0X3D, 0X13, 0X00 }, /* -10.0db */
+	{ 0X15, 0X70, 0X12, 0X00 }, /* -9.5db */
+	{ 0X16, 0XB5, 0X43, 0X00 }, /* -9.0db */
+	{ 0X18, 0X0D, 0XB8, 0X00 }, /* -8.5db */
+	{ 0X19, 0X7A, 0X96, 0X00 }, /* -8.0db */
+	{ 0X1A, 0XFD, 0X13, 0X00 }, /* -7.5db */
+	{ 0X1C, 0X96, 0X76, 0X00 }, /* -7.0db */
+	{ 0X1E, 0X48, 0X1C, 0X00 }, /* -6.5db */
+	{ 0X20, 0X13, 0X73, 0X00 }, /* -6.0db */
+	{ 0X21, 0XFA, 0X02, 0X00 }, /* -5.5db */
+	{ 0X23, 0XFD, 0X66, 0X00 }, /* -5.0db */
+	{ 0X26, 0X1F, 0X54, 0X00 }, /* -4.5db */
+	{ 0X28, 0X61, 0X9A, 0X00 }, /* -4.0db */
+	{ 0X2A, 0XC6, 0X25, 0X00 }, /* -3.5db */
+	{ 0X2D, 0X4E, 0XFB, 0X00 }, /* -3.0db */
+	{ 0X2F, 0XFE, 0X44, 0X00 }, /* -2.5db */
+	{ 0X32, 0XD6, 0X46, 0X00 }, /* -2.0db */
+	{ 0X35, 0XD9, 0X6B, 0X00 }, /* -1.5db */
+	{ 0X39, 0X0A, 0X41, 0X00 }, /* -1.0db */
+	{ 0X3C, 0X6B, 0X7E, 0X00 }, /* -0.5db */
+	{ 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */
+	{ 0X43, 0XCA, 0XD0, 0X00 }, /* 0.5db */
+	{ 0X47, 0XCF, 0X26, 0X00 }, /* 1.0db */
+	{ 0X4C, 0X10, 0X6B, 0X00 }, /* 1.5db */
+	{ 0X50, 0X92, 0X3B, 0X00 }, /* 2.0db */
+	{ 0X55, 0X58, 0X6A, 0X00 }, /* 2.5db */
+	{ 0X5A, 0X67, 0X03, 0X00 }, /* 3.0db */
+	{ 0X5F, 0XC2, 0X53, 0X00 }, /* 3.5db */
+	{ 0X65, 0X6E, 0XE3, 0X00 }, /* 4.0db */
+	{ 0X6B, 0X71, 0X86, 0X00 }, /* 4.5db */
+	{ 0X71, 0XCF, 0X54, 0X00 }, /* 5.0db */
+	{ 0X78, 0X8D, 0XB4, 0X00 }, /* 5.5db */
+	{ 0X7F, 0XB2, 0X61, 0X00 }, /* 6.0db */
+};
+#endif
diff --git a/include/sound/tas5825-tlv.h b/include/sound/tas5825-tlv.h
new file mode 100644
index 000000000000..95f2d3fad120
--- /dev/null
+++ b/include/sound/tas5825-tlv.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+//
+// ALSA SoC Texas Instruments TAS5825 Audio Smart Amplifier
+//
+// Copyright (C) 2025 Texas Instruments Incorporated
+// https://www.ti.com
+//
+// The TAS5825 hda driver implements for one or two TAS5825 chips.
+//
+// Author: Baojun Xu <baojun.xu@ti.com>
+//
+
+#ifndef __TAS5825_TLV_H__
+#define __TAS5825_TLV_H__
+
+#define TAS5825_DVC_LEVEL		TASDEVICE_REG(0x0, 0x0, 0x4c)
+#define TAS5825_AMP_LEVEL		TASDEVICE_REG(0x0, 0x0, 0x54)
+
+static const __maybe_unused DECLARE_TLV_DB_SCALE(
+		tas5825_dvc_tlv, -10300, 50, 0);
+static const __maybe_unused DECLARE_TLV_DB_SCALE(
+		tas5825_amp_tlv, -1550, 50, 0);
+
+#endif
diff --git a/include/sound/tlv320dac33-plat.h b/include/sound/tlv320dac33-plat.h
deleted file mode 100644
index 7a7249a896e3..000000000000
--- a/include/sound/tlv320dac33-plat.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * Platform header for Texas Instruments TLV320DAC33 codec driver
- *
- * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
- *
- * Copyright:   (C) 2009 Nokia Corporation
- */
-
-#ifndef __TLV320DAC33_PLAT_H
-#define __TLV320DAC33_PLAT_H
-
-struct tlv320dac33_platform_data {
-	int power_gpio;
-	int mode1_latency; /* latency caused by the i2c writes in us */
-	int auto_fifo_config; /* FIFO config based on the period size */
-	int keep_bclk;	/* Keep the BCLK running in FIFO modes */
-	u8 burst_bclkdiv;
-};
-
-#endif /* __TLV320DAC33_PLAT_H */
diff --git a/include/uapi/sound/compress_offload.h b/include/uapi/sound/compress_offload.h
index d62eb93af0ed..b610683fd8db 100644
--- a/include/uapi/sound/compress_offload.h
+++ b/include/uapi/sound/compress_offload.h
@@ -13,8 +13,7 @@
 #include <sound/asound.h>
 #include <sound/compress_params.h>
 
-
-#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 3, 0)
+#define SNDRV_COMPRESS_VERSION SNDRV_PROTOCOL_VERSION(0, 4, 1)
 /**
  * struct snd_compressed_buffer - compressed buffer
  * @fragment_size: size of buffer fragment in bytes
@@ -57,6 +56,25 @@ struct snd_compr_tstamp {
 } __attribute__((packed, aligned(4)));
 
 /**
+ * struct snd_compr_tstamp64 - timestamp descriptor with fields in 64 bit
+ * @byte_offset: Byte offset in ring buffer to DSP
+ * @copied_total: Total number of bytes copied from/to ring buffer to/by DSP
+ * @pcm_frames: Frames decoded or encoded by DSP. This field will evolve by
+ *	large steps and should only be used to monitor encoding/decoding
+ *	progress. It shall not be used for timing estimates.
+ * @pcm_io_frames: Frames rendered or received by DSP into a mixer or an audio
+ * output/input. This field should be used for A/V sync or time estimates.
+ * @sampling_rate: sampling rate of audio
+ */
+struct snd_compr_tstamp64 {
+	__u32 byte_offset;
+	__u64 copied_total;
+	__u64 pcm_frames;
+	__u64 pcm_io_frames;
+	__u32 sampling_rate;
+} __attribute__((packed, aligned(4)));
+
+/**
  * struct snd_compr_avail - avail descriptor
  * @avail: Number of bytes available in ring buffer for writing/reading
  * @tstamp: timestamp information
@@ -66,6 +84,16 @@ struct snd_compr_avail {
 	struct snd_compr_tstamp tstamp;
 } __attribute__((packed, aligned(4)));
 
+/**
+ * struct snd_compr_avail64 - avail descriptor with tstamp in 64 bit format
+ * @avail: Number of bytes available in ring buffer for writing/reading
+ * @tstamp: timestamp information
+ */
+struct snd_compr_avail64 {
+	__u64 avail;
+	struct snd_compr_tstamp64 tstamp;
+} __attribute__((packed, aligned(4)));
+
 enum snd_compr_direction {
 	SND_COMPRESS_PLAYBACK = 0,
 	SND_COMPRESS_CAPTURE,
@@ -189,6 +217,7 @@ struct snd_compr_task_status {
  * Note: only codec params can be changed runtime and stream params cant be
  * SNDRV_COMPRESS_GET_PARAMS: Query codec params
  * SNDRV_COMPRESS_TSTAMP: get the current timestamp value
+ * SNDRV_COMPRESS_TSTAMP64: get the current timestamp value in 64 bit format
  * SNDRV_COMPRESS_AVAIL: get the current buffer avail value.
  * This also queries the tstamp properties
  * SNDRV_COMPRESS_PAUSE: Pause the running stream
@@ -211,6 +240,8 @@ struct snd_compr_task_status {
 						 struct snd_compr_metadata)
 #define SNDRV_COMPRESS_TSTAMP		_IOR('C', 0x20, struct snd_compr_tstamp)
 #define SNDRV_COMPRESS_AVAIL		_IOR('C', 0x21, struct snd_compr_avail)
+#define SNDRV_COMPRESS_TSTAMP64		_IOR('C', 0x22, struct snd_compr_tstamp64)
+#define SNDRV_COMPRESS_AVAIL64		_IOR('C', 0x23, struct snd_compr_avail64)
 #define SNDRV_COMPRESS_PAUSE		_IO('C', 0x30)
 #define SNDRV_COMPRESS_RESUME		_IO('C', 0x31)
 #define SNDRV_COMPRESS_START		_IO('C', 0x32)
diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h
index bc7648a30746..d7db6b4e1166 100644
--- a/include/uapi/sound/compress_params.h
+++ b/include/uapi/sound/compress_params.h
@@ -43,7 +43,8 @@
 #define SND_AUDIOCODEC_BESPOKE               ((__u32) 0x0000000E)
 #define SND_AUDIOCODEC_ALAC                  ((__u32) 0x0000000F)
 #define SND_AUDIOCODEC_APE                   ((__u32) 0x00000010)
-#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_APE
+#define SND_AUDIOCODEC_OPUS_RAW              ((__u32) 0x00000011)
+#define SND_AUDIOCODEC_MAX                   SND_AUDIOCODEC_OPUS_RAW
 
 /*
  * Profile and modes are listed with bit masks. This allows for a
@@ -324,6 +325,43 @@ struct snd_dec_ape {
 	__u32 seek_table_present;
 } __attribute__((packed, aligned(4)));
 
+/**
+ * struct snd_dec_opus - Opus decoder parameters (raw opus packets)
+ * @version: Usually should be '1' but can be split into major (4 upper bits)
+ * and minor (4 lower bits) sub-fields.
+ * @num_channels: Number of output channels.
+ * @pre_skip: Number of samples to discard at 48 kHz.
+ * @sample_rate: Sample rate of original input.
+ * @output_gain: Gain to apply when decoding (in Q7.8 format).
+ * @mapping_family: Order and meaning of output channels. Only values 0 and 1
+ * are expected; values 2..255 are not recommended for playback.
+ *
+ * @chan_map: Optional channel mapping table. Describes mapping of opus streams
+ *            to decoded channels. Fields:
+ * @chan_map.stream_count: Number of streams encoded in each Ogg packet.
+ * @chan_map.coupled_count: Number of streams whose decoders are used
+ *                          for two channels.
+ * @chan_map.channel_map: Which decoded channel to be used for each one.
+ *                        Supports only mapping families 0 and 1,
+ *                        max number of channels is 8.
+ *
+ * These options were extracted from RFC7845 Section 5.
+ */
+
+struct snd_dec_opus {
+	__u8 version;
+	__u8 num_channels;
+	__u16 pre_skip;
+	__u32 sample_rate;
+	__u16 output_gain;
+	__u8 mapping_family;
+	struct snd_dec_opus_ch_map {
+		__u8 stream_count;
+		__u8 coupled_count;
+		__u8 channel_map[8];
+	} chan_map;
+} __attribute__((packed, aligned(4)));
+
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -334,6 +372,7 @@ union snd_codec_options {
 	struct snd_dec_wma wma_d;
 	struct snd_dec_alac alac_d;
 	struct snd_dec_ape ape_d;
+	struct snd_dec_opus opus_d;
 	struct {
 		__u32 out_sample_rate;
 	} src_d;
diff --git a/include/uapi/sound/intel/avs/tokens.h b/include/uapi/sound/intel/avs/tokens.h
index c9f845b3c523..f3ff6aae09a9 100644
--- a/include/uapi/sound/intel/avs/tokens.h
+++ b/include/uapi/sound/intel/avs/tokens.h
@@ -133,6 +133,21 @@ enum avs_tplg_token {
 	AVS_TKN_PATH_FE_FMT_ID_U32			= 1902,
 	AVS_TKN_PATH_BE_FMT_ID_U32			= 1903,
 
+	/* struct avs_tplg_path_template (conditional) */
+	AVS_TKN_CONDPATH_TMPL_ID_U32			= 1801,
+	AVS_TKN_CONDPATH_TMPL_SOURCE_TPLG_NAME_STRING	= 2002,
+	AVS_TKN_CONDPATH_TMPL_SOURCE_PATH_TMPL_ID_U32	= 2003,
+	AVS_TKN_CONDPATH_TMPL_SINK_TPLG_NAME_STRING	= 2004,
+	AVS_TKN_CONDPATH_TMPL_SINK_PATH_TMPL_ID_U32	= 2005,
+	AVS_TKN_CONDPATH_TMPL_COND_TYPE_U32		= 2006,
+	AVS_TKN_CONDPATH_TMPL_OVERRIDABLE_BOOL		= 2007,
+	AVS_TKN_CONDPATH_TMPL_PRIORITY_U8		= 2008,
+
+	/* struct avs_tplg_path (conditional) */
+	AVS_TKN_CONDPATH_ID_U32				= 1901,
+	AVS_TKN_CONDPATH_SOURCE_PATH_ID_U32		= 2102,
+	AVS_TKN_CONDPATH_SINK_PATH_ID_U32		= 2103,
+
 	/* struct avs_tplg_pin_format */
 	AVS_TKN_PIN_FMT_INDEX_U32			= 2201,
 	AVS_TKN_PIN_FMT_IOBS_U32			= 2202,
diff --git a/include/uapi/sound/snd_ar_tokens.h b/include/uapi/sound/snd_ar_tokens.h
index b9b9093b4396..6b8102eaa121 100644
--- a/include/uapi/sound/snd_ar_tokens.h
+++ b/include/uapi/sound/snd_ar_tokens.h
@@ -3,6 +3,8 @@
 #ifndef __SND_AR_TOKENS_H__
 #define __SND_AR_TOKENS_H__
 
+#include <linux/types.h>
+
 #define APM_SUB_GRAPH_PERF_MODE_LOW_POWER	0x1
 #define APM_SUB_GRAPH_PERF_MODE_LOW_LATENCY	0x2
 
@@ -118,6 +120,12 @@ enum ar_event_types {
  *						LPAIF_WSA = 2,
  *						LPAIF_VA = 3,
  *						LPAIF_AXI = 4
+ * Possible values for MI2S
+ *						I2S_INTF_TYPE_PRIMARY = 0,
+ *						I2S_INTF_TYPE_SECONDARY = 1,
+ *						I2S_INTF_TYPE_TERTIARY = 2,
+ *						I2S_INTF_TYPE_QUATERNARY = 3,
+ *						I2S_INTF_TYPE_QUINARY = 4,
  *
  * %AR_TKN_U32_MODULE_FMT_INTERLEAVE:		PCM Interleaving
  *						PCM_INTERLEAVED = 1,
@@ -184,8 +192,8 @@ enum ar_event_types {
 #define AR_TKN_U32_MODULE_INSTANCE_ID		201
 #define AR_TKN_U32_MODULE_MAX_IP_PORTS		202
 #define AR_TKN_U32_MODULE_MAX_OP_PORTS		203
-#define AR_TKN_U32_MODULE_IN_PORTS		204
-#define AR_TKN_U32_MODULE_OUT_PORTS		205
+#define AR_TKN_U32_MODULE_IN_PORTS		204 /* deprecated */
+#define AR_TKN_U32_MODULE_OUT_PORTS		205 /* deprecated */
 #define AR_TKN_U32_MODULE_SRC_OP_PORT_ID	206
 #define AR_TKN_U32_MODULE_DST_IN_PORT_ID	207
 #define AR_TKN_U32_MODULE_SRC_INSTANCE_ID	208
@@ -232,4 +240,12 @@ enum ar_event_types {
 #define AR_TKN_U32_MODULE_LOG_TAP_POINT_ID	260
 #define AR_TKN_U32_MODULE_LOG_MODE		261
 
+#define SND_SOC_AR_TPLG_MODULE_CFG_TYPE 0x01001006
+struct audioreach_module_priv_data {
+	__le32 size;	/* size in bytes of the array, including all elements */
+	__le32 type;	/* SND_SOC_AR_TPLG_MODULE_CFG_TYPE */
+	__le32 priv[2];	/* Private data for future expansion */
+	__le32 data[0];	/* config data */
+};
+
 #endif /* __SND_AR_TOKENS_H__ */
diff --git a/include/uapi/sound/sof/tokens.h b/include/uapi/sound/sof/tokens.h
index c28c766270de..9ce72fbd6f11 100644
--- a/include/uapi/sound/sof/tokens.h
+++ b/include/uapi/sound/sof/tokens.h
@@ -106,6 +106,8 @@
  */
 #define SOF_TKN_COMP_NO_WNAME_IN_KCONTROL_NAME	417
 
+#define SOF_TKN_COMP_SCHED_DOMAIN		418
+
 /* SSP */
 #define SOF_TKN_INTEL_SSP_CLKS_CONTROL		500
 #define SOF_TKN_INTEL_SSP_MCLK_ID		501