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

Author: torvalds

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

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:
 ::
 

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);

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;

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)

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),

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);

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[] = {

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,
 	},
 };