Merge tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "A collection of small fixes for 6.10-rc1. Most of changes are various
  device-specific fixes and quirks, while there are a few small changes
  in ALSA core timer and module / built-in fixes"

* tag 'sound-fix-6.10-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek: fix mute/micmute LEDs don't work for ProBook 440/460 G11.
  ALSA: core: Enable proc module when CONFIG_MODULES=y
  ALSA: core: Fix NULL module pointer assignment at card init
  ALSA: hda/realtek: Enable headset mic of JP-IK LEAP W502 with ALC897
  ASoC: dt-bindings: stm32: Ensure compatible pattern matches whole string
  ASoC: tas2781: Fix wrong loading calibrated data sequence
  ASoC: tas2552: Add TX path for capturing AUDIO-OUT data
  ALSA: usb-audio: Fix for sampling rates support for Mbox3
  Documentation: sound: Fix trailing whitespaces
  ALSA: timer: Set lower bound of start tick time
  ASoC: codecs: ES8326: solve hp and button detect issue
  ASoC: rt5645: mic-in detection threshold modification
  ASoC: Intel: sof_sdw_rt_sdca_jack_common: Use name_prefix for `-sdca` detection

13 files changed, 86 insertions, 106 deletions

diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
index b46a4778807d..68f97b462598 100644
--- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
+++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml
@@ -72,7 +72,7 @@ patternProperties:
     properties:
       compatible:
         description: Compatible for SAI sub-block A or B.
-        pattern: "st,stm32-sai-sub-[ab]"
+        pattern: "^st,stm32-sai-sub-[ab]$"
 
       "#sound-dai-cells":
         const: 0
diff --git a/Documentation/sound/hd-audio/notes.rst b/Documentation/sound/hd-audio/notes.rst
index a9e35b1f87bd..ef6a4513cce7 100644
--- a/Documentation/sound/hd-audio/notes.rst
+++ b/Documentation/sound/hd-audio/notes.rst
@@ -15,7 +15,7 @@ problem is broken BIOS, and the rest is the driver implementation.
 This document explains the brief trouble-shooting and debugging
 methods for the	HD-audio hardware.
 
-The HD-audio component consists of two parts: the controller chip and 
+The HD-audio component consists of two parts: the controller chip and
 the codec chips on the HD-audio bus.  Linux provides a single driver
 for all controllers, snd-hda-intel.  Although the driver name contains
 a word of a well-known hardware vendor, it's not specific to it but for
@@ -81,7 +81,7 @@ the wake-up timing.  It wakes up a few samples before actually
 processing the data on the buffer.  This caused a lot of problems, for
 example, with ALSA dmix or JACK.  Since 2.6.27 kernel, the driver puts
 an artificial delay to the wake up timing.  This delay is controlled
-via ``bdl_pos_adj`` option. 
+via ``bdl_pos_adj`` option.
 
 When ``bdl_pos_adj`` is a negative value (as default), it's assigned to
 an appropriate value depending on the controller chip.  For Intel
@@ -144,7 +144,7 @@ see a regression wrt the sound quality (stuttering, etc) or a lock-up
 in the recent kernel, try to pass ``enable_msi=0`` option to disable
 MSI.  If it works, you can add the known bad device to the blacklist
 defined in hda_intel.c.  In such a case, please report and give the
-patch back to the upstream developer. 
+patch back to the upstream developer.
 
 
 HD-Audio Codec
@@ -375,7 +375,7 @@ HD-Audio Reconfiguration
 ------------------------
 This is an experimental feature to allow you re-configure the HD-audio
 codec dynamically without reloading the driver.  The following sysfs
-files are available under each codec-hwdep device directory (e.g. 
+files are available under each codec-hwdep device directory (e.g.
 /sys/class/sound/hwC0D0):
 
 vendor_id
@@ -433,7 +433,7 @@ re-configure based on that state, run like below:
 ::
 
     # echo 0x14 0x9993013f > /sys/class/sound/hwC0D0/user_pin_configs
-    # echo 1 > /sys/class/sound/hwC0D0/reconfig  
+    # echo 1 > /sys/class/sound/hwC0D0/reconfig
 
 
 Hint Strings
@@ -494,7 +494,7 @@ indep_hp (bool)
     mixer control, if available
 add_stereo_mix_input (bool)
     add the stereo mix (analog-loopback mix) to the input mux if
-    available 
+    available
 add_jack_modes (bool)
     add "xxx Jack Mode" enum controls to each I/O jack for allowing to
     change the headphone amp and mic bias VREF capabilities
@@ -504,7 +504,7 @@ power_save_node (bool)
     stream states
 power_down_unused (bool)
     power down the unused widgets, a subset of power_save_node, and
-    will be dropped in future 
+    will be dropped in future
 add_hp_mic (bool)
     add the headphone to capture source if possible
 hp_mic_detect (bool)
@@ -603,7 +603,7 @@ present.
 
 The patch module option is specific to each card instance, and you
 need to give one file name for each instance, separated by commas.
-For example, if you have two cards, one for an on-board analog and one 
+For example, if you have two cards, one for an on-board analog and one
 for an HDMI video board, you may pass patch option like below:
 ::
 
diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h
index ea9af2726a53..7fba7ea26a4b 100644
--- a/include/sound/tas2781-dsp.h
+++ b/include/sound/tas2781-dsp.h
@@ -2,7 +2,7 @@
 //
 // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
 //
-// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
 // https://www.ti.com
 //
 // The TAS2781 driver implements a flexible and configurable
@@ -13,8 +13,8 @@
 // Author: Kevin Lu <kevin-lu@ti.com>
 //
 
-#ifndef __TASDEVICE_DSP_H__
-#define __TASDEVICE_DSP_H__
+#ifndef __TAS2781_DSP_H__
+#define __TAS2781_DSP_H__
 
 #define MAIN_ALL_DEVICES			0x0d
 #define MAIN_DEVICE_A				0x01
@@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context);
 int tasdevice_select_tuningprm_cfg(void *context, int prm,
 	int cfg_no, int rca_conf_no);
 int tasdevice_prmg_load(void *context, int prm_no);
-int tasdevice_prmg_calibdata_load(void *context, int prm_no);
 void tasdevice_tuning_switch(void *context, int state);
 int tas2781_load_calibration(void *context, char *file_name,
 	unsigned short i);
diff --git a/sound/core/init.c b/sound/core/init.c
index 6b127864a1a3..4e52bbe32786 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -50,7 +50,7 @@ MODULE_PARM_DESC(slots, "Module names assigned to the slots.");
 static int module_slot_match(struct module *module, int idx)
 {
 	int match = 1;
-#ifdef MODULE
+#ifdef CONFIG_MODULES
 	const char *s1, *s2;
 
 	if (!module || !*module->name || !slots[idx])
@@ -77,7 +77,7 @@ static int module_slot_match(struct module *module, int idx)
 		if (!c1)
 			break;
 	}
-#endif /* MODULE */
+#endif /* CONFIG_MODULES */
 	return match;
 }
 
@@ -311,10 +311,8 @@ static int snd_card_init(struct snd_card *card, struct device *parent,
 	}
 	card->dev = parent;
 	card->number = idx;
-#ifdef MODULE
-	WARN_ON(!module);
+	WARN_ON(IS_MODULE(CONFIG_SND) && !module);
 	card->module = module;
-#endif
 	INIT_LIST_HEAD(&card->devices);
 	init_rwsem(&card->controls_rwsem);
 	rwlock_init(&card->ctl_files_rwlock);
@@ -969,7 +967,7 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer)
 
 #endif
 
-#ifdef MODULE
+#ifdef CONFIG_MODULES
 static void snd_card_module_info_read(struct snd_info_entry *entry,
 				      struct snd_info_buffer *buffer)
 {
@@ -997,7 +995,7 @@ int __init snd_card_info_init(void)
 	if (snd_info_register(entry) < 0)
 		return -ENOMEM; /* freed in error path */
 
-#ifdef MODULE
+#ifdef CONFIG_MODULES
 	entry = snd_info_create_module_entry(THIS_MODULE, "modules", NULL);
 	if (!entry)
 		return -ENOMEM;
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 4d2ee99c12a3..d104adc75a8b 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -544,6 +544,14 @@ static int snd_timer_start1(struct snd_timer_instance *timeri,
 			     SNDRV_TIMER_IFLG_START))
 		return -EBUSY;
 
+	/* check the actual time for the start tick;
+	 * bail out as error if it's way too low (< 100us)
+	 */
+	if (start) {
+		if ((u64)snd_timer_hw_resolution(timer) * ticks < 100000)
+			return -EINVAL;
+	}
+
 	if (start)
 		timeri->ticks = timeri->cticks = ticks;
 	else if (!timeri->cticks)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index a696943aec0d..e3c0b9d5552d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10194,8 +10194,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8c70, "HP EliteBook 835 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c71, "HP EliteBook 845 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c72, "HP EliteBook 865 G11", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c89, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c8a, "HP EliteBook 630", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c8c, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c8d, "HP ProBook 440 G11", ALC236_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x8c8e, "HP ProBook 460 G11", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c90, "HP EliteBook 640", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c91, "HP EliteBook 660", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x8c96, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
@@ -12028,6 +12031,7 @@ enum {
 	ALC897_FIXUP_LENOVO_HEADSET_MODE,
 	ALC897_FIXUP_HEADSET_MIC_PIN2,
 	ALC897_FIXUP_UNIS_H3C_X500S,
+	ALC897_FIXUP_HEADSET_MIC_PIN3,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -12474,10 +12478,18 @@ static const struct hda_fixup alc662_fixups[] = {
 			{}
 		},
 	},
+	[ALC897_FIXUP_HEADSET_MIC_PIN3] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a11050 }, /* use as headset mic */
+			{ }
+		},
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_FIXUP_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1019, 0x9859, "JP-IK LEAP W502", ALC897_FIXUP_HEADSET_MIC_PIN3),
 	SND_PCI_QUIRK(0x1025, 0x022f, "Acer Aspire One", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0241, "Packard Bell DOTS", ALC662_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE),
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 833ea52638ab..03b539ba540f 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -829,8 +829,8 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			/* mute adc when mic path switch */
 			regmap_write(es8326->regmap, ES8326_ADC1_SRC, 0x44);
 			regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
-			es8326->hp = 0;
 		}
+		es8326->hp = 0;
 		regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
 		regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x0a);
 		regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x03);
@@ -981,7 +981,7 @@ static int es8326_resume(struct snd_soc_component *component)
 	regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0);
 	usleep_range(10000, 15000);
 	regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9);
-	regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xcb);
+	regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xd8);
 	/* set headphone default type and detect pin */
 	regmap_write(es8326->regmap, ES8326_HPDET_TYPE, 0x83);
 	regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05);
@@ -1018,7 +1018,7 @@ static int es8326_resume(struct snd_soc_component *component)
 
 	regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F);
 	/* select vdda as micbias source */
-	regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x23);
+	regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x03);
 	/* set dac dsmclip = 1 */
 	regmap_write(es8326->regmap, ES8326_DAC_DSM, 0x08);
 	regmap_write(es8326->regmap, ES8326_DAC_VPPSCALE, 0x15);
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 05f574bf8b8f..cdb7ff7020e9 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -81,6 +81,7 @@ static const struct reg_sequence init_list[] = {
 static const struct reg_sequence rt5650_init_list[] = {
 	{0xf6,	0x0100},
 	{RT5645_PWR_ANLG1, 0x02},
+	{RT5645_IL_CMD3, 0x0018},
 };
 
 static const struct reg_default rt5645_reg[] = {
diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c
index 40f5f27e74c0..a7ed59ec49a6 100644
--- a/sound/soc/codecs/tas2552.c
+++ b/sound/soc/codecs/tas2552.c
@@ -2,7 +2,8 @@
 /*
  * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier
  *
- * Copyright (C) 2014 Texas Instruments Incorporated -  https://www.ti.com
+ * Copyright (C) 2014 - 2024 Texas Instruments Incorporated -
+ *	https://www.ti.com
  *
  * Author: Dan Murphy <dmurphy@ti.com>
  */
@@ -119,12 +120,14 @@ static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] =
 			 &tas2552_input_mux_control),
 
 	SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_AIF_OUT("ASI OUT", "DAC Capture", 0, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0),
 	SND_SOC_DAPM_POST("Post Event", tas2552_post_event),
 
-	SND_SOC_DAPM_OUTPUT("OUT")
+	SND_SOC_DAPM_OUTPUT("OUT"),
+	SND_SOC_DAPM_INPUT("DMIC")
 };
 
 static const struct snd_soc_dapm_route tas2552_audio_map[] = {
@@ -134,6 +137,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = {
 	{"ClassD", NULL, "Input selection"},
 	{"OUT", NULL, "ClassD"},
 	{"ClassD", NULL, "PLL"},
+	{"ASI OUT", NULL, "DMIC"}
 };
 
 #ifdef CONFIG_PM
@@ -538,6 +542,13 @@ static struct snd_soc_dai_driver tas2552_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = TAS2552_FORMATS,
 		},
+		.capture = {
+			.stream_name = "Capture",
+			.channels_min = 2,
+			.channels_max = 2,
+			.rates = SNDRV_PCM_RATE_8000_192000,
+			.formats = TAS2552_FORMATS,
+		},
 		.ops = &tas2552_speaker_dai_ops,
 	},
 };
diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c
index a6be81adcb83..265a8ca25cbb 100644
--- a/sound/soc/codecs/tas2781-fmwlib.c
+++ b/sound/soc/codecs/tas2781-fmwlib.c
@@ -2151,6 +2151,24 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv,
 	return ret;
 }
 
+static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i)
+{
+	struct tasdevice_calibration *cal;
+	struct tasdevice_fw *cal_fmw;
+
+	cal_fmw = priv->tasdevice[i].cali_data_fmw;
+
+	/* No calibrated data for current devices, playback will go ahead. */
+	if (!cal_fmw)
+		return;
+
+	cal = cal_fmw->calibrations;
+	if (cal)
+		return;
+
+	load_calib_data(priv, &cal->dev_data);
+}
+
 int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
 	int cfg_no, int rca_conf_no)
 {
@@ -2210,21 +2228,9 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
 		for (i = 0; i < tas_priv->ndev; i++) {
 			if (tas_priv->tasdevice[i].is_loaderr == true)
 				continue;
-			else if (tas_priv->tasdevice[i].is_loaderr == false
-				&& tas_priv->tasdevice[i].is_loading == true) {
-				struct tasdevice_fw *cal_fmw =
-					tas_priv->tasdevice[i].cali_data_fmw;
-
-				if (cal_fmw) {
-					struct tasdevice_calibration
-						*cal = cal_fmw->calibrations;
-
-					if (cal)
-						load_calib_data(tas_priv,
-							&(cal->dev_data));
-				}
+			if (tas_priv->tasdevice[i].is_loaderr == false &&
+				tas_priv->tasdevice[i].is_loading == true)
 				tas_priv->tasdevice[i].cur_prog = prm_no;
-			}
 		}
 	}
 
@@ -2245,11 +2251,15 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no,
 		tasdevice_load_data(tas_priv, &(conf->dev_data));
 		for (i = 0; i < tas_priv->ndev; i++) {
 			if (tas_priv->tasdevice[i].is_loaderr == true) {
-				status |= 1 << (i + 4);
+				status |= BIT(i + 4);
 				continue;
-			} else if (tas_priv->tasdevice[i].is_loaderr == false
-				&& tas_priv->tasdevice[i].is_loading == true)
+			}
+
+			if (tas_priv->tasdevice[i].is_loaderr == false &&
+				tas_priv->tasdevice[i].is_loading == true) {
+				tasdev_load_calibrated_data(tas_priv, i);
 				tas_priv->tasdevice[i].cur_conf = cfg_no;
+			}
 		}
 	} else
 		dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n",
@@ -2308,65 +2318,6 @@ out:
 }
 EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_load, SND_SOC_TAS2781_FMWLIB);
 
-int tasdevice_prmg_calibdata_load(void *context, int prm_no)
-{
-	struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
-	struct tasdevice_fw *tas_fmw = tas_priv->fmw;
-	struct tasdevice_prog *program;
-	int prog_status = 0;
-	int i;
-
-	if (!tas_fmw) {
-		dev_err(tas_priv->dev, "%s: Firmware is NULL\n", __func__);
-		goto out;
-	}
-
-	if (prm_no >= tas_fmw->nr_programs) {
-		dev_err(tas_priv->dev,
-			"%s: prm(%d) is not in range of Programs %u\n",
-			__func__, prm_no, tas_fmw->nr_programs);
-		goto out;
-	}
-
-	for (i = 0, prog_status = 0; i < tas_priv->ndev; i++) {
-		if (prm_no >= 0 && tas_priv->tasdevice[i].cur_prog != prm_no) {
-			tas_priv->tasdevice[i].cur_conf = -1;
-			tas_priv->tasdevice[i].is_loading = true;
-			prog_status++;
-		}
-		tas_priv->tasdevice[i].is_loaderr = false;
-	}
-
-	if (prog_status) {
-		program = &(tas_fmw->programs[prm_no]);
-		tasdevice_load_data(tas_priv, &(program->dev_data));
-		for (i = 0; i < tas_priv->ndev; i++) {
-			if (tas_priv->tasdevice[i].is_loaderr == true)
-				continue;
-			else if (tas_priv->tasdevice[i].is_loaderr == false
-				&& tas_priv->tasdevice[i].is_loading == true) {
-				struct tasdevice_fw *cal_fmw =
-					tas_priv->tasdevice[i].cali_data_fmw;
-
-				if (cal_fmw) {
-					struct tasdevice_calibration *cal =
-						cal_fmw->calibrations;
-
-					if (cal)
-						load_calib_data(tas_priv,
-							&(cal->dev_data));
-				}
-				tas_priv->tasdevice[i].cur_prog = prm_no;
-			}
-		}
-	}
-
-out:
-	return prog_status;
-}
-EXPORT_SYMBOL_NS_GPL(tasdevice_prmg_calibdata_load,
-	SND_SOC_TAS2781_FMWLIB);
-
 void tasdevice_tuning_switch(void *context, int state)
 {
 	struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context;
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index b5abff230e43..9350972dfefe 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -2,7 +2,7 @@
 //
 // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier
 //
-// Copyright (C) 2022 - 2023 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
 // https://www.ti.com
 //
 // The TAS2563/TAS2781 driver implements a flexible and configurable
@@ -414,7 +414,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw,
 				__func__, tas_priv->cal_binaryname[i]);
 	}
 
-	tasdevice_prmg_calibdata_load(tas_priv, 0);
+	tasdevice_prmg_load(tas_priv, 0);
 	tas_priv->cur_prog = 0;
 out:
 	if (tas_priv->fw_state == TASDEVICE_DSP_FW_FAIL) {
diff --git a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
index 701b0372f59e..012195c50519 100644
--- a/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
+++ b/sound/soc/intel/boards/sof_sdw_rt_sdca_jack_common.c
@@ -109,7 +109,7 @@ int rt_sdca_jack_rtd_init(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *d
 		return -ENOMEM;
 
 	for (i = 0; i < ARRAY_SIZE(need_sdca_suffix); i++) {
-		if (strstr(codec_dai->name, need_sdca_suffix[i])) {
+		if (strstr(component->name_prefix, need_sdca_suffix[i])) {
 			/* Add -sdca suffix for existing UCMs */
 			card->components = devm_kasprintf(card->dev, GFP_KERNEL,
 							  "%s-sdca", card->components);
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 2f961f0e9378..58156fbca02c 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1740,7 +1740,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
 	u32 current_rate;
 
 	// Get current rate from card and check if changing it is needed
-	snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+	snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
 					0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
 	current_rate = le32_to_cpu(buff4);
 	dev_dbg(&subs->dev->dev,
@@ -1765,7 +1765,7 @@ static void mbox3_set_format_quirk(struct snd_usb_substream *subs,
 
 	// Check whether the change was successful
 	buff4 = 0;
-	snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0),
+	snd_usb_ctl_msg(subs->dev, usb_rcvctrlpipe(subs->dev, 0),
 					0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4);
 	if (new_rate != le32_to_cpu(buff4))
 		dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate");