Merge tag 'sound-7.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Nothing too thrilling here, but we see lots of driver updates and bug
  fixes, including quirk additions and refactoring works, while there
  have been little changes in the core functionality. Here are some
  highlights:

  Core:
   - Add validation for the control API put callback
   - Fixes in compress-offload API timestamp handling
   - Continued ASoC core API cleanups

  ASoC:
   - Add support for bus keepers (for Apple devices in future)
   - Enhancements to the SDCA support, including retaskable jacks
   - Test improvements for Cirrus Logic drivers
   - Lots of fixes for the NXP, nVidia and Qualcomm
   - Support for AMD RPL DMIC, Cirrus Logic CS42L43 and CS47L47, nVidia
     machines with CPCAP and WM8962

  USB-audio:
   - Quirks for Huawei Headset, Focusrite Novation, MV-Silicon, Studio
     1824, Arturia AF16Rig, Hotone Audio, Feaulle Rainbow, PreSonus
     AudioBox, Moondrop Ju Jiu, Scarlett 18i20, etc
   - Extended mixer volume quirk handling
   - UAF and other fixes for us144mkii, 6fire and caiaq drivers

  HD-audio:
   - Add quirks or fixes for Acer, Lenovo, HP, ASUS machines
   - Fixes & cleanups of GPIO helper code

  Misc:
   - Add suspend/resume support for multiple legacy ISA and Apple
     drivers
   - Further regression fixes for ctxfi driver"

* tag 'sound-7.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (359 commits)
  ALSA: control: Validate buf_len before strnlen() in snd_ctl_elem_init_enum_names()
  ALSA: usb-audio: Fix missing error handling for get_min_max*()
  ALSA: hda/realtek - fixed speaker no sound update
  ALSA: hda/realtek: Add quirk for Acer PT316-51S headset mic
  ALSA: usb-audio: Exclude Scarlett 18i20 1st Gen from SKIP_IFACE_SETUP
  ALSA: hda/realtek: Add quirk for Legion S7 15IMH
  ALSA: hda/realtek: Add quirk for HP Spectre x360 14-ea
  ALSA: caiaq: take a reference on the USB device in create_card()
  ASoC: dt-bindings: rockchip: convert rk3399-gru-sound to DT Schema
  ALSA: sscape: Add suspend and resume support
  ALSA: sscape: Cache per-card resources for board reinitialization
  ALSA: usb-audio: Do not expose sticky mixers
  ALSA: usb-audio: Move volume control resolution check into a function
  ALSA: usb-audio: Add error checks against get_min_max*()
  ALSA: usb-audio: Add quirk for PreSonus AudioBox USB
  ALSA: interwave: guard PM-only restore helpers with CONFIG_PM
  ALSA: usb-audio: Evaluate packsize caps at the right place
  ALSA: sc6000: Restore board setup across suspend
  ALSA: sc6000: Keep the programmed board state in card-private data
  ALSA: 6fire: fix use-after-free on disconnect
  ...

21 files changed, 231 insertions, 62 deletions

diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
index 6d1ce7f5da51..45850f2d4342 100644
--- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
+++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h
@@ -140,6 +140,18 @@
 #define DISPLAY_PORT_RX_6	134
 #define DISPLAY_PORT_RX_7	135
 #define USB_RX			136
+#define LPI_MI2S_RX_0		137
+#define LPI_MI2S_TX_0		138
+#define LPI_MI2S_RX_1		139
+#define LPI_MI2S_TX_1		140
+#define LPI_MI2S_RX_2		141
+#define LPI_MI2S_TX_2		142
+#define LPI_MI2S_RX_3		143
+#define LPI_MI2S_TX_3		144
+#define LPI_MI2S_RX_4		145
+#define LPI_MI2S_TX_4		146
+#define SENARY_MI2S_RX		147
+#define SENARY_MI2S_TX		148
 
 #define LPASS_CLK_ID_PRI_MI2S_IBIT	1
 #define LPASS_CLK_ID_PRI_MI2S_EBIT	2
diff --git a/include/linux/firmware/cirrus/cs_dsp.h b/include/linux/firmware/cirrus/cs_dsp.h
index 0ec1cdc5585d..4e3baa557068 100644
--- a/include/linux/firmware/cirrus/cs_dsp.h
+++ b/include/linux/firmware/cirrus/cs_dsp.h
@@ -179,6 +179,7 @@ struct cs_dsp {
 
 	bool booted;
 	bool running;
+	bool hibernating;
 
 	struct list_head ctl_list;
 
@@ -354,4 +355,6 @@ 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);
 
+void cs_dsp_hibernate(struct cs_dsp *dsp, bool hibernating);
+
 #endif
diff --git a/include/linux/mfd/arizona/pdata.h b/include/linux/mfd/arizona/pdata.h
index f72e6d4b14a7..d465dcd8c90a 100644
--- a/include/linux/mfd/arizona/pdata.h
+++ b/include/linux/mfd/arizona/pdata.h
@@ -117,11 +117,6 @@ struct arizona_pdata {
 	/** Check for line output with HPDET method */
 	bool hpdet_acc_id_line;
 
-#ifdef CONFIG_GPIOLIB_LEGACY
-	/** GPIO used for mic isolation with HPDET */
-	int hpdet_id_gpio;
-#endif
-
 	/** Channel to use for headphone detection */
 	unsigned int hpdet_channel;
 
@@ -131,11 +126,6 @@ struct arizona_pdata {
 	/** Extra debounce timeout used during initial mic detection (ms) */
 	unsigned int micd_detect_debounce;
 
-#ifdef CONFIG_GPIOLIB_LEGACY
-	/** GPIO for mic detection polarity */
-	int micd_pol_gpio;
-#endif
-
 	/** Mic detect ramp rate */
 	unsigned int micd_bias_start_time;
 
diff --git a/include/linux/mfd/cs42l43-regs.h b/include/linux/mfd/cs42l43-regs.h
index c39a49269cb7..68831f113589 100644
--- a/include/linux/mfd/cs42l43-regs.h
+++ b/include/linux/mfd/cs42l43-regs.h
@@ -1181,4 +1181,80 @@
 /* CS42L43_FW_MISSION_CTRL_MM_MCU_CFG_REG */
 #define CS42L43_FW_MISSION_CTRL_MM_MCU_CFG_DISABLE_VAL		0xF05AA50F
 
+/* CS42L43B VARIANT REGISTERS */
+#define CS42L43B_DEVID_VAL					0x0042A43B
+
+#define CS42L43B_DECIM_VOL_CTRL_CH1_CH2				0x00008280
+#define CS42L43B_DECIM_VOL_CTRL_CH3_CH4				0x00008284
+
+#define CS42L43B_DECIM_VOL_CTRL_CH5_CH6				0x00008290
+#define CS42L43B_DECIM_VOL_CTRL_UPDATE				0x0000829C
+
+#define CS42L43B_DECIM_HPF_WNF_CTRL5				0x000082A0
+#define CS42L43B_DECIM_HPF_WNF_CTRL6				0x000082A4
+
+#define CS42L43B_SWIRE_DP3_CH3_INPUT				0x0000C320
+#define CS42L43B_SWIRE_DP3_CH4_INPUT				0x0000C330
+#define CS42L43B_SWIRE_DP4_CH3_INPUT				0x0000C340
+#define CS42L43B_SWIRE_DP4_CH4_INPUT				0x0000C350
+
+#define CS42L43B_ISRC1DEC3_INPUT1				0x0000C780
+#define CS42L43B_ISRC1DEC4_INPUT1				0x0000C790
+#define CS42L43B_ISRC2DEC3_INPUT1				0x0000C7A0
+#define CS42L43B_ISRC2DEC4_INPUT1				0x0000C7B0
+
+#define CS42L43B_FW_MISSION_CTRL_NEED_CONFIGS			0x00117E00
+#define CS42L43B_FW_MISSION_CTRL_HAVE_CONFIGS			0x00117E04
+#define CS42L43B_FW_MISSION_CTRL_PATCH_START_ADDR_REG		0x00117E08
+#define CS42L43B_FW_MISSION_CTRL_MM_CTRL_SELECTION		0x00117E0C
+#define CS42L43B_FW_MISSION_CTRL_MM_MCU_CFG_REG			0x00117E10
+
+#define CS42L43B_MCU_SW_REV					0x00117314
+#define CS42L43B_PATCH_START_ADDR				0x00117318
+#define CS42L43B_CONFIG_SELECTION				0x0011731C
+#define CS42L43B_NEED_CONFIGS					0x00117320
+#define CS42L43B_BOOT_STATUS					0x00117330
+
+#define CS42L43B_FW_MISSION_CTRL_NEED_CONFIGS			0x00117E00
+#define CS42L43B_FW_MISSION_CTRL_HAVE_CONFIGS			0x00117E04
+#define CS42L43B_FW_MISSION_CTRL_PATCH_START_ADDR_REG		0x00117E08
+#define CS42L43B_FW_MISSION_CTRL_MM_CTRL_SELECTION		0x00117E0C
+#define CS42L43B_FW_MISSION_CTRL_MM_MCU_CFG_REG			0x00117E10
+
+#define CS42L43B_MCU_RAM_MAX					0x00117FFF
+
+/* CS42L43B_DECIM_DECIM_VOL_CTRL_CH5_CH6 */
+#define CS42L43B_DECIM6_MUTE_MASK				0x80000000
+#define CS42L43B_DECIM6_MUTE_SHIFT				31
+#define CS42L43B_DECIM6_VOL_MASK				0x3FC00000
+#define CS42L43B_DECIM6_VOL_SHIFT				22
+#define CS42L43B_DECIM6_PATH1_VOL_FALL_RATE_MASK		0x00380000
+#define CS42L43B_DECIM6_PATH1_VOL_FALL_RATE_SHIFT		19
+#define CS42L43B_DECIM6_PATH1_VOL_RISE_RATE_MASK		0x00070000
+#define CS42L43B_DECIM6_PATH1_VOL_RISE_RATE_SHIFT		16
+#define CS42L43B_DECIM5_MUTE_MASK				0x00008000
+#define CS42L43B_DECIM5_MUTE_SHIFT				15
+#define CS42L43B_DECIM5_VOL_MASK				0x00003FC0
+#define CS42L43B_DECIM5_VOL_SHIFT				6
+#define CS42L43B_DECIM5_PATH1_VOL_FALL_RATE_MASK		0x00000038
+#define CS42L43B_DECIM5_PATH1_VOL_FALL_RATE_SHIFT		3
+#define CS42L43B_DECIM5_PATH1_VOL_RISE_RATE_MASK		0x00000007
+#define CS42L43B_DECIM5_PATH1_VOL_RISE_RATE_SHIFT		0
+
+/* CS42L43B_DECIM_VOL_CTRL_UPDATE */
+#define CS42L43B_DECIM6_PATH1_VOL_TRIG_MASK			0x00000800
+#define CS42L43B_DECIM6_PATH1_VOL_TRIG_SHIFT			11
+#define CS42L43B_DECIM5_PATH1_VOL_TRIG_MASK			0x00000100
+#define CS42L43B_DECIM5_PATH1_VOL_TRIG_SHIFT			8
+#define CS42L43B_DECIM4_VOL_UPDATE_MASK				0x00000020
+#define CS42L43B_DECIM4_VOL_UPDATE_SHIFT			5
+
+/* CS42L43_ISRC1_CTRL..CS42L43_ISRC2_CTRL */
+#define CS42L43B_ISRC_DEC4_EN_MASK				0x00000008
+#define CS42L43B_ISRC_DEC4_EN_SHIFT				3
+#define CS42L43B_ISRC_DEC4_EN_WIDTH				1
+#define CS42L43B_ISRC_DEC3_EN_MASK				0x00000004
+#define CS42L43B_ISRC_DEC3_EN_SHIFT				2
+#define CS42L43B_ISRC_DEC3_EN_WIDTH				1
+
 #endif /* CS42L43_CORE_REGS_H */
diff --git a/include/linux/mfd/cs42l43.h b/include/linux/mfd/cs42l43.h
index 2239d8585e78..ff0f7e365a19 100644
--- a/include/linux/mfd/cs42l43.h
+++ b/include/linux/mfd/cs42l43.h
@@ -98,6 +98,7 @@ struct cs42l43 {
 	bool sdw_pll_active;
 	bool attached;
 	bool hw_lock;
+	long variant_id;
 };
 
 #endif /* CS42L43_CORE_EXT_H */
diff --git a/include/sound/core.h b/include/sound/core.h
index 64327e971122..4093ec82a0a1 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -133,6 +133,9 @@ struct snd_card {
 #ifdef CONFIG_SND_DEBUG
 	struct dentry *debugfs_root;    /* debugfs root for card */
 #endif
+#ifdef CONFIG_SND_CTL_DEBUG
+	struct snd_ctl_elem_value *value_buf; /* buffer for kctl->put() verification */
+#endif
 
 #ifdef CONFIG_PM
 	unsigned int power_state;	/* power state */
diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h
index 28f9f5940ab6..c3b10587cb4c 100644
--- a/include/sound/cs35l56.h
+++ b/include/sound/cs35l56.h
@@ -32,9 +32,6 @@ struct snd_ctl_elem_value;
 #define CS35L56_UPDATE_REGS				0x0002A0C
 #define CS35L56_REFCLK_INPUT				0x0002C04
 #define CS35L56_GLOBAL_SAMPLE_RATE			0x0002C0C
-#define CS35L56_OTP_MEM_53				0x00300D4
-#define CS35L56_OTP_MEM_54				0x00300D8
-#define CS35L56_OTP_MEM_55				0x00300DC
 #define CS35L56_ASP1_ENABLES1				0x0004800
 #define CS35L56_ASP1_CONTROL1				0x0004804
 #define CS35L56_ASP1_CONTROL2				0x0004808
@@ -86,6 +83,9 @@ struct snd_ctl_elem_value;
 #define CS35L56_DIE_STS1				0x0017040
 #define CS35L56_DIE_STS2				0x0017044
 #define CS35L56_DSP_RESTRICT_STS1			0x00190F0
+#define CS35L56_OTP_MEM_53				0x00300D4
+#define CS35L56_OTP_MEM_54				0x00300D8
+#define CS35L56_OTP_MEM_55				0x00300DC
 #define CS35L56_DSP1_XMEM_PACKED_0			0x2000000
 #define CS35L56_DSP1_XMEM_PACKED_6143			0x2005FFC
 #define CS35L56_DSP1_XMEM_UNPACKED32_0			0x2400000
@@ -435,6 +435,7 @@ ssize_t cs35l56_cal_data_debugfs_read(struct cs35l56_base *cs35l56_base,
 ssize_t cs35l56_cal_data_debugfs_write(struct cs35l56_base *cs35l56_base,
 				       const char __user *from, size_t count,
 				       loff_t *ppos);
+int cs35l56_factory_calibrate(struct cs35l56_base *cs35l56_base);
 void cs35l56_create_cal_debugfs(struct cs35l56_base *cs35l56_base,
 				const struct cs35l56_cal_debugfs_fops *fops);
 void cs35l56_remove_cal_debugfs(struct cs35l56_base *cs35l56_base);
diff --git a/include/sound/gus.h b/include/sound/gus.h
index 321ae93625eb..3feb42627de1 100644
--- a/include/sound/gus.h
+++ b/include/sound/gus.h
@@ -536,6 +536,7 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus,
 			       struct snd_gf1_dma_block * block,
 			       int atomic,
 			       int synth);
+void snd_gf1_dma_suspend(struct snd_gus_card *gus);
 
 /* gus_volume.c */
 
@@ -552,6 +553,8 @@ struct snd_gus_voice *snd_gf1_alloc_voice(struct snd_gus_card * gus, int type, i
 void snd_gf1_free_voice(struct snd_gus_card * gus, struct snd_gus_voice *voice);
 int snd_gf1_start(struct snd_gus_card * gus);
 int snd_gf1_stop(struct snd_gus_card * gus);
+int snd_gf1_suspend(struct snd_gus_card *gus);
+int snd_gf1_resume(struct snd_gus_card *gus);
 
 /* gus_mixer.c */
 
@@ -572,6 +575,8 @@ int snd_gus_create(struct snd_card *card,
 		   int effect,
 		   struct snd_gus_card ** rgus);
 int snd_gus_initialize(struct snd_gus_card * gus);
+int snd_gus_suspend(struct snd_gus_card *gus);
+int snd_gus_resume(struct snd_gus_card *gus);
 
 /* gus_irq.c */
 
@@ -583,6 +588,8 @@ void snd_gus_irq_profile_init(struct snd_gus_card *gus);
 /* gus_uart.c */
 
 int snd_gf1_rawmidi_new(struct snd_gus_card *gus, int device);
+void snd_gf1_uart_suspend(struct snd_gus_card *gus);
+void snd_gf1_uart_resume(struct snd_gus_card *gus);
 
 /* gus_dram.c */
 int snd_gus_dram_write(struct snd_gus_card *gus, char __user *ptr,
@@ -593,5 +600,6 @@ int snd_gus_dram_read(struct snd_gus_card *gus, char __user *ptr,
 /* gus_timer.c */
 void snd_gf1_timers_init(struct snd_gus_card *gus);
 void snd_gf1_timers_done(struct snd_gus_card *gus);
+void snd_gf1_timers_resume(struct snd_gus_card *gus);
 
 #endif /* __SOUND_GUS_H */
diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h
index 5d9f0ef228af..24581080e26a 100644
--- a/include/sound/hda_codec.h
+++ b/include/sound/hda_codec.h
@@ -336,6 +336,17 @@ snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
 	return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
 }
 
+/* sync after write */
+static inline int
+snd_hda_codec_write_sync(struct hda_codec *codec, hda_nid_t nid, int flags,
+			 unsigned int verb, unsigned int parm)
+{
+	/* use snd_hda_codec_read() for writing;
+	 * the returned value is usually discarded
+	 */
+	return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
+}
+
 #define snd_hda_param_read(codec, nid, param) \
 	snd_hdac_read_parm(&(codec)->core, nid, param)
 #define snd_hda_get_sub_nodes(codec, nid, start_nid) \
@@ -470,6 +481,10 @@ void snd_hda_unlock_devices(struct hda_bus *bus);
 void snd_hda_bus_reset(struct hda_bus *bus);
 void snd_hda_bus_reset_codecs(struct hda_bus *bus);
 
+void snd_hda_codec_set_gpio(struct hda_codec *codec, unsigned int mask,
+			    unsigned int dir, unsigned int data,
+			    unsigned int delay);
+
 int snd_hda_codec_set_name(struct hda_codec *codec, const char *name);
 
 /*
diff --git a/include/sound/hda_verbs.h b/include/sound/hda_verbs.h
index 006d358acce2..6066954409aa 100644
--- a/include/sound/hda_verbs.h
+++ b/include/sound/hda_verbs.h
@@ -56,7 +56,12 @@ enum {
 #define AC_VERB_GET_DIGI_CONVERT_1		0x0f0d
 #define AC_VERB_GET_DIGI_CONVERT_2		0x0f0e /* unused */
 #define AC_VERB_GET_VOLUME_KNOB_CONTROL		0x0f0f
-/* f10-f1a: GPIO */
+/* f10-f1a: GPI/GPO/GPIO */
+#define AC_VERB_GET_GPI_DATA			0x0f10
+#define AC_VERB_GET_GPI_WAKE_MASK		0x0f11
+#define AC_VERB_GET_GPI_UNSOLICITED_RSP_MASK	0x0f12
+#define AC_VERB_GET_GPI_STICKY_MASK		0x0f13
+#define AC_VERB_GET_GPO_DATA			0x0f14
 #define AC_VERB_GET_GPIO_DATA			0x0f15
 #define AC_VERB_GET_GPIO_MASK			0x0f16
 #define AC_VERB_GET_GPIO_DIRECTION		0x0f17
@@ -99,6 +104,11 @@ enum {
 #define AC_VERB_SET_DIGI_CONVERT_2		0x70e
 #define AC_VERB_SET_DIGI_CONVERT_3		0x73e
 #define AC_VERB_SET_VOLUME_KNOB_CONTROL		0x70f
+#define AC_VERB_SET_GPI_DATA			0x710
+#define AC_VERB_SET_GPI_WAKE_MASK		0x711
+#define AC_VERB_SET_SPI_UNSOLICITED_RSP_MASK	0x712
+#define AC_VERB_SET_GPI_STICKY_MASK		0x713
+#define AC_VERB_SET_GPO_DATA			0x714
 #define AC_VERB_SET_GPIO_DATA			0x715
 #define AC_VERB_SET_GPIO_MASK			0x716
 #define AC_VERB_SET_GPIO_DIRECTION		0x717
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index a7860c047503..76fc33dce537 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -729,6 +729,10 @@ static inline void __snd_pcm_set_state(struct snd_pcm_runtime *runtime,
 	runtime->status->state = state; /* copy for mmap */
 }
 
+void snd_pcm_set_state(struct snd_pcm_substream *substream,
+		       snd_pcm_state_t state);
+snd_pcm_state_t snd_pcm_get_state(struct snd_pcm_substream *substream);
+
 /**
  * bytes_to_samples - Unit conversion of the size from bytes to samples
  * @runtime: PCM runtime instance
diff --git a/include/sound/sdca_asoc.h b/include/sound/sdca_asoc.h
index aa9124f93218..46a61a52decc 100644
--- a/include/sound/sdca_asoc.h
+++ b/include/sound/sdca_asoc.h
@@ -13,6 +13,8 @@
 struct device;
 struct regmap;
 struct sdca_function_data;
+struct snd_ctl_elem_value;
+struct snd_kcontrol;
 struct snd_kcontrol_new;
 struct snd_pcm_hw_params;
 struct snd_pcm_substream;
@@ -23,6 +25,42 @@ struct snd_soc_dai_ops;
 struct snd_soc_dapm_route;
 struct snd_soc_dapm_widget;
 
+/* convenient macro to handle the mono volume in 7.8 fixed format representation */
+#define SDCA_SINGLE_Q78_TLV(xname, xreg, xmin, xmax, xstep, 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, \
+	.get = sdca_asoc_q78_get_volsw, \
+	.put = sdca_asoc_q78_put_volsw, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) { \
+		.reg = (xreg), .rreg = (xreg), \
+		.min = (xmin), .max = (xmax), \
+		.shift = (xstep), .rshift = (xstep), \
+		.sign_bit = 15 \
+	} \
+}
+
+/* convenient macro for stereo volume in 7.8 fixed format with separate registers for L/R */
+#define SDCA_DOUBLE_Q78_TLV(xname, xreg_l, xreg_r, xmin, xmax, xstep, 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, \
+	.get = sdca_asoc_q78_get_volsw, \
+	.put = sdca_asoc_q78_put_volsw, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) { \
+		.reg = (xreg_l), .rreg = (xreg_r), \
+		.min = (xmin), .max = (xmax), \
+		.shift = (xstep), .rshift = (xstep), \
+		.sign_bit = 15 \
+	} \
+}
+
 int sdca_asoc_count_component(struct device *dev, struct sdca_function_data *function,
 			      int *num_widgets, int *num_routes, int *num_controls,
 			      int *num_dais);
@@ -57,5 +95,8 @@ int sdca_asoc_hw_params(struct device *dev, struct regmap *regmap,
 			struct snd_pcm_substream *substream,
 			struct snd_pcm_hw_params *params,
 			struct snd_soc_dai *dai);
-
+int sdca_asoc_q78_put_volsw(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol);
+int sdca_asoc_q78_get_volsw(struct snd_kcontrol *kcontrol,
+			    struct snd_ctl_elem_value *ucontrol);
 #endif // __SDCA_ASOC_H__
diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h
index 69a9c9c4d0e9..915e6ae5f68d 100644
--- a/include/sound/simple_card_utils.h
+++ b/include/sound/simple_card_utils.h
@@ -54,6 +54,11 @@ struct prop_nums {
 	int platforms;
 };
 
+enum simple_util_sysclk_order {
+	SIMPLE_SYSCLK_ORDER_CODEC_FIRST = 0,
+	SIMPLE_SYSCLK_ORDER_CPU_FIRST,
+};
+
 struct simple_util_priv {
 	struct snd_soc_card snd_card;
 	struct simple_dai_props {
@@ -63,6 +68,7 @@ struct simple_util_priv {
 		struct snd_soc_codec_conf *codec_conf;
 		struct prop_nums num;
 		unsigned int mclk_fs;
+		enum simple_util_sysclk_order sysclk_order;
 	} *dai_props;
 	struct simple_util_jack hp_jack;
 	struct simple_util_jack mic_jack;
diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h
index 2a2b74b24a60..7e158d27ae8d 100644
--- a/include/sound/soc-component.h
+++ b/include/sound/soc-component.h
@@ -86,10 +86,10 @@ struct snd_soc_component_driver {
 		     unsigned int reg, unsigned int val);
 
 	/* pcm creation and destruction */
-	int (*pcm_construct)(struct snd_soc_component *component,
-			     struct snd_soc_pcm_runtime *rtd);
-	void (*pcm_destruct)(struct snd_soc_component *component,
-			     struct snd_pcm *pcm);
+	int (*pcm_new)(struct snd_soc_component *component,
+		       struct snd_soc_pcm_runtime *rtd);
+	void (*pcm_free)(struct snd_soc_component *component,
+			 struct snd_pcm *pcm);
 
 	/* component wide operations */
 	int (*set_sysclk)(struct snd_soc_component *component,
@@ -224,7 +224,6 @@ struct snd_soc_component {
 	int num_dai;
 
 	struct regmap *regmap;
-	int val_bytes;
 
 	struct mutex io_mutex;
 
@@ -327,7 +326,7 @@ int snd_soc_component_stream_event(struct snd_soc_component *component,
 int snd_soc_component_set_bias_level(struct snd_soc_component *component,
 				     enum snd_soc_bias_level level);
 
-void snd_soc_component_setup_regmap(struct snd_soc_component *component);
+int snd_soc_component_regmap_val_bytes(struct snd_soc_component *component);
 #ifdef CONFIG_REGMAP
 void snd_soc_component_init_regmap(struct snd_soc_component *component,
 				   struct regmap *regmap);
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 224396927aef..6a42812bba8c 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -53,6 +53,21 @@ struct snd_compr_stream;
 #define SND_SOC_POSSIBLE_DAIFMT_PDM		(1 << SND_SOC_DAI_FORMAT_PDM)
 
 /*
+ * DAI TDM slot idle modes
+ *
+ * Describes a CODEC/CPU's behaviour when not actively receiving or
+ * transmitting on a given TDM slot. NONE is undefined behaviour.
+ * Add new modes to the end.
+ */
+#define SND_SOC_DAI_TDM_IDLE_NONE	0
+#define SND_SOC_DAI_TDM_IDLE_OFF	1
+#define SND_SOC_DAI_TDM_IDLE_ZERO	2
+#define SND_SOC_DAI_TDM_IDLE_PULLDOWN	3
+#define SND_SOC_DAI_TDM_IDLE_HIZ	4
+#define SND_SOC_DAI_TDM_IDLE_PULLUP	5
+#define SND_SOC_DAI_TDM_IDLE_DRIVE_HIGH	6
+
+/*
  * DAI Clock gating.
  *
  * DAI bit clocks can be gated (disabled) when the DAI is not
@@ -181,6 +196,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
 
+int snd_soc_dai_set_tdm_idle(struct snd_soc_dai *dai,
+			     unsigned int tx_mask, unsigned int rx_mask,
+			     int tx_mode, int rx_mode);
+
 int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai,
 	unsigned int tx_num, const unsigned int *tx_slot,
 	unsigned int rx_num, const unsigned int *rx_slot);
@@ -297,6 +316,9 @@ struct snd_soc_dai_ops {
 	int (*set_tdm_slot)(struct snd_soc_dai *dai,
 		unsigned int tx_mask, unsigned int rx_mask,
 		int slots, int slot_width);
+	int (*set_tdm_idle)(struct snd_soc_dai *dai,
+			    unsigned int tx_mask, unsigned int rx_mask,
+			    int tx_mode, int rx_mode);
 	int (*set_channel_map)(struct snd_soc_dai *dai,
 		unsigned int tx_num, const unsigned int *tx_slot,
 		unsigned int rx_num, const unsigned int *rx_slot);
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index 49f0fe05db01..4f8fb7622a13 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -424,6 +424,7 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_input = 0,		/* input pin */
 	snd_soc_dapm_output,		/* output pin */
 	snd_soc_dapm_mux,		/* selects 1 analog signal from many inputs */
+	snd_soc_dapm_mux_named_ctl,	/* mux with named controls */
 	snd_soc_dapm_demux,		/* connects the input to one of multiple outputs */
 	snd_soc_dapm_mixer,		/* mixes several analog signals together */
 	snd_soc_dapm_mixer_named_ctl,	/* mixer with named controls */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 7d8376c8e1be..5e3eb617d832 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -311,6 +311,12 @@ struct platform_device;
 	.info = snd_soc_info_bool_ext, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = xdata }
+#define SOC_SINGLE_BOOL_EXT_ACC(xname, xdata, xhandler_get, xhandler_put, xaccess) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.access = xaccess, \
+	.info = snd_soc_info_bool_ext, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = xdata }
 #define SOC_ENUM_EXT(xname, xenum, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_enum_double, \
@@ -422,11 +428,6 @@ struct snd_soc_jack_pin;
 #include <sound/soc-dpcm.h>
 #include <sound/soc-topology.h>
 
-enum snd_soc_pcm_subclass {
-	SND_SOC_PCM_CLASS_PCM	= 0,
-	SND_SOC_PCM_CLASS_BE	= 1,
-};
-
 int snd_soc_register_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);
@@ -465,6 +466,7 @@ struct snd_soc_component *snd_soc_lookup_component_nolocked(struct device *dev,
 							    const char *driver_name);
 struct snd_soc_component *snd_soc_lookup_component(struct device *dev,
 						   const char *driver_name);
+struct snd_soc_component *snd_soc_lookup_component_by_name(const char *component_name);
 
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd);
 #ifdef CONFIG_SND_SOC_COMPRESS
@@ -999,7 +1001,6 @@ struct snd_soc_card {
 
 	/* Mutex for PCM operations */
 	struct mutex pcm_mutex;
-	enum snd_soc_pcm_subclass pcm_subclass;
 
 	int (*probe)(struct snd_soc_card *card);
 	int (*late_probe)(struct snd_soc_card *card);
@@ -1026,8 +1027,6 @@ struct snd_soc_card {
 	void (*remove_dai_link)(struct snd_soc_card *,
 			    struct snd_soc_dai_link *link);
 
-	long pmdown_time;
-
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;  /* predefined links only */
 	int num_links;  /* predefined links only */
@@ -1072,9 +1071,6 @@ struct snd_soc_card {
 	struct list_head dapm_list;
 	struct list_head dapm_dirty;
 
-	/* attached dynamic objects */
-	struct list_head dobj_list;
-
 	/* Generic DAPM context for the card */
 	struct snd_soc_dapm_context *dapm;
 	struct snd_soc_dapm_stats dapm_stats;
@@ -1239,7 +1235,6 @@ struct soc_mixer_control {
 	unsigned int sign_bit;
 	unsigned int invert:1;
 	unsigned int autodisable:1;
-	unsigned int sdca_q78:1;
 #ifdef CONFIG_SND_SOC_TOPOLOGY
 	struct snd_soc_dobj dobj;
 #endif
@@ -1340,15 +1335,6 @@ void snd_soc_of_parse_node_prefix(struct device_node *np,
 				   struct snd_soc_codec_conf *codec_conf,
 				   struct device_node *of_node,
 				   const char *propname);
-static inline
-void snd_soc_of_parse_audio_prefix(struct snd_soc_card *card,
-				   struct snd_soc_codec_conf *codec_conf,
-				   struct device_node *of_node,
-				   const char *propname)
-{
-	snd_soc_of_parse_node_prefix(card->dev->of_node,
-				     codec_conf, of_node, propname);
-}
 
 int snd_soc_of_parse_audio_routing(struct snd_soc_card *card,
 				   const char *propname);
@@ -1412,6 +1398,9 @@ struct snd_soc_dai *snd_soc_find_dai(
 struct snd_soc_dai *snd_soc_find_dai_with_mutex(
 	const struct snd_soc_dai_link_component *dlc);
 
+void soc_pcm_set_dai_params(struct snd_soc_dai *dai,
+			    struct snd_pcm_hw_params *params);
+
 #include <sound/soc-dai.h>
 
 static inline
@@ -1517,7 +1506,7 @@ static inline void _snd_soc_dapm_mutex_assert_held_d(struct snd_soc_dapm_context
  */
 static inline void _snd_soc_dpcm_mutex_lock_c(struct snd_soc_card *card)
 {
-	mutex_lock_nested(&card->pcm_mutex, card->pcm_subclass);
+	mutex_lock(&card->pcm_mutex);
 }
 
 static inline void _snd_soc_dpcm_mutex_unlock_c(struct snd_soc_card *card)
diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h
index 227347c8f0b3..489083183673 100644
--- a/include/sound/soc_sdw_utils.h
+++ b/include/sound/soc_sdw_utils.h
@@ -71,6 +71,7 @@ struct asoc_sdw_aux_info {
 };
 
 struct asoc_sdw_codec_info {
+	const int vendor_id;
 	const int part_id;
 	const int version_id;
 	const char *name_prefix;
@@ -82,6 +83,8 @@ struct asoc_sdw_codec_info {
 	const int dai_num;
 	struct asoc_sdw_aux_info auxs[SOC_SDW_MAX_AUX_NUM];
 	const int aux_num;
+	/* Force AMP-style name_prefix handling (append AMP index) even if MIC/Jack DAIs exist */
+	const bool is_amp;
 
 	int (*codec_card_late_probe)(struct snd_soc_card *card);
 
@@ -259,6 +262,8 @@ 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_cs42l45_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
 int asoc_sdw_cs42l45_dmic_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int asoc_sdw_cs47l47_hs_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai);
+int asoc_sdw_cs47l47_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 */
diff --git a/include/sound/tea6330t.h b/include/sound/tea6330t.h
index 1c77b78f6533..3a34033d2aa3 100644
--- a/include/sound/tea6330t.h
+++ b/include/sound/tea6330t.h
@@ -12,5 +12,6 @@
 int snd_tea6330t_detect(struct snd_i2c_bus *bus, int equalizer);
 int snd_tea6330t_update_mixer(struct snd_card *card, struct snd_i2c_bus *bus,
 			      int equalizer, int fader);
+int snd_tea6330t_restore_mixer(struct snd_i2c_bus *bus);
 
 #endif /* __SOUND_TEA6330T_H */
diff --git a/include/sound/timer.h b/include/sound/timer.h
index 760e132cc0cd..83bafe70cf33 100644
--- a/include/sound/timer.h
+++ b/include/sound/timer.h
@@ -102,6 +102,7 @@ struct snd_timer_instance {
 	unsigned int slave_id;
 	struct list_head open_list;
 	struct list_head active_list;
+	struct list_head master_list;
 	struct list_head ack_list;
 	struct list_head slave_list_head;
 	struct list_head slave_active_head;
diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h
deleted file mode 100644
index 2e42ea2d0cfd..000000000000
--- a/include/sound/uda1380.h
+++ /dev/null
@@ -1,19 +0,0 @@
-/* SPDX-License-Identifier: GPL-2.0-only */
-/*
- * UDA1380 ALSA SoC Codec driver
- *
- * Copyright 2009 Philipp Zabel
- */
-
-#ifndef __UDA1380_H
-#define __UDA1380_H
-
-struct uda1380_platform_data {
-	int gpio_power;
-	int gpio_reset;
-	int dac_clk;
-#define UDA1380_DAC_CLK_SYSCLK 0
-#define UDA1380_DAC_CLK_WSPLL  1
-};
-
-#endif /* __UDA1380_H */