Merge tag 'sound-6.17-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Only small fixes.

   - ASoC Cirrus codec fixes

   - A regression fix for the recent TAS2781 codec refactoring

   - A fix for user-timer error handling

   - Fixes for USB-audio descriptor validators

   - Usual HD-audio and ASoC device-specific quirks"

* tag 'sound-6.17-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: usb-audio: Use correct sub-type for UAC3 feature unit validation
  ALSA: timer: fix ida_free call while not allocated
  ASoC: cs35l56: Remove SoundWire Clock Divider workaround for CS35L63
  ASoC: cs35l56: Handle new algorithms IDs for CS35L63
  ASoC: cs35l56: Update Firmware Addresses for CS35L63 for production silicon
  ALSA: hda: tas2781: Fix wrong reference of tasdevice_priv
  ALSA: hda/realtek: Audio disappears on HP 15-fc000 after warm boot again
  ALSA: hda/realtek: Fix headset mic on ASUS Zenbook 14
  ASoC: codecs: ES9389: Modify the standby configuration
  ALSA: usb-audio: Fix size validation in convert_chmap_v3()
  ALSA: hda/tas2781: Add name prefix tas2781 for tas2781's dvc_tlv and amp_vol_tlv
  ALSA: hda/realtek: Add support for HP EliteBook x360 830 G6 and EliteBook 830 G6

Author: torvalds

include/sound/cs35l56.h

@@ -107,8 +107,8 @@
 #define CS35L56_DSP1_PMEM_5114				0x3804FE8
 
 #define CS35L63_DSP1_FW_VER				CS35L56_DSP1_FW_VER
-#define CS35L63_DSP1_HALO_STATE				0x280396C
-#define CS35L63_DSP1_PM_CUR_STATE			0x28042C8
+#define CS35L63_DSP1_HALO_STATE				0x2803C04
+#define CS35L63_DSP1_PM_CUR_STATE			0x2804518
 #define CS35L63_PROTECTION_STATUS			0x340009C
 #define CS35L63_TRANSDUCER_ACTUAL_PS			0x34000F4
 #define CS35L63_MAIN_RENDER_USER_MUTE			0x3400020
@@ -306,6 +306,7 @@ struct cs35l56_base {
 	struct gpio_desc *reset_gpio;
 	struct cs35l56_spi_payload *spi_payload_buf;
 	const struct cs35l56_fw_reg *fw_reg;
+	const struct cirrus_amp_cal_controls *calibration_controls;
 };
 
 static inline bool cs35l56_is_otp_register(unsigned int reg)

include/sound/tas2781-tlv.h

@@ -2,7 +2,7 @@
 //
 // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier
 //
-// Copyright (C) 2022 - 2024 Texas Instruments Incorporated
+// Copyright (C) 2022 - 2025 Texas Instruments Incorporated
 // https://www.ti.com
 //
 // The TAS2781 driver implements a flexible and configurable
@@ -15,7 +15,7 @@
 #ifndef __TAS2781_TLV_H__
 #define __TAS2781_TLV_H__
 
-static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 50, 0);
-static const __maybe_unused DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0);
+static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_dvc_tlv, -10000, 50, 0);
+static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2781_amp_tlv, 1100, 50, 0);
 
 #endif

sound/core/timer.c

@@ -2139,14 +2139,14 @@ static int snd_utimer_create(struct snd_timer_uinfo *utimer_info,
 		goto err_take_id;
 	}
 
+	utimer->id = utimer_id;
+
 	utimer->name = kasprintf(GFP_KERNEL, "snd-utimer%d", utimer_id);
 	if (!utimer->name) {
 		err = -ENOMEM;
 		goto err_get_name;
 	}
 
-	utimer->id = utimer_id;
-
 	tid.dev_sclass = SNDRV_TIMER_SCLASS_APPLICATION;
 	tid.dev_class = SNDRV_TIMER_CLASS_GLOBAL;
 	tid.card = -1;

sound/hda/codecs/realtek/alc269.c

@@ -510,6 +510,15 @@ static void alc256_shutup(struct hda_codec *codec)
 		hp_pin = 0x21;
 
 	alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */
+
+	/* 3k pull low control for Headset jack. */
+	/* NOTE: call this before clearing the pin, otherwise codec stalls */
+	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
+	 * when booting with headset plugged. So skip setting it for the codec alc257
+	 */
+	if (spec->en_3kpull_low)
+		alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
+
 	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
 
 	if (hp_pin_sense) {
@@ -520,14 +529,6 @@ static void alc256_shutup(struct hda_codec *codec)
 
 		msleep(75);
 
-	/* 3k pull low control for Headset jack. */
-	/* NOTE: call this before clearing the pin, otherwise codec stalls */
-	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
-	 * when booting with headset plugged. So skip setting it for the codec alc257
-	 */
-		if (spec->en_3kpull_low)
-			alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
-
 		if (!spec->no_shutup_pins)
 			snd_hda_codec_write(codec, hp_pin, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
@@ -3579,6 +3580,7 @@ enum {
 	ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE,
 	ALC294_FIXUP_ASUS_MIC,
 	ALC294_FIXUP_ASUS_HEADSET_MIC,
+	ALC294_FIXUP_ASUS_I2C_HEADSET_MIC,
 	ALC294_FIXUP_ASUS_SPK,
 	ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE,
 	ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE,
@@ -4889,6 +4891,15 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC269_FIXUP_HEADSET_MIC
 	},
+	[ALC294_FIXUP_ASUS_I2C_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x19, 0x03a19020 }, /* use as headset mic */
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC287_FIXUP_CS35L41_I2C_2
+	},
 	[ALC294_FIXUP_ASUS_SPK] = {
 		.type = HDA_FIXUP_VERBS,
 		.v.verbs = (const struct hda_verb[]) {
@@ -6368,6 +6379,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3),
 	SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360),
 	SND_PCI_QUIRK(0x103c, 0x8537, "HP ProBook 440 G6", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8548, "HP EliteBook x360 830 G6", ALC285_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x854a, "HP EliteBook 830 G6", ALC285_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x85c6, "HP Pavilion x360 Convertible 14-dy1xxx", ALC295_FIXUP_HP_MUTE_LED_COEFBIT11),
 	SND_PCI_QUIRK(0x103c, 0x85de, "HP Envy x360 13-ar0xxx", ALC285_FIXUP_HP_ENVY_X360),
 	SND_PCI_QUIRK(0x103c, 0x860f, "HP ZBook 15 G6", ALC285_FIXUP_HP_GPIO_AMP_INIT),
@@ -6728,7 +6741,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE),
-	SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC294_FIXUP_ASUS_I2C_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2),
 	SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2),

sound/hda/codecs/side-codecs/tas2781_hda_i2c.c

@@ -267,7 +267,7 @@ static const struct snd_kcontrol_new tas2770_snd_controls[] = {
 static const struct snd_kcontrol_new tas2781_snd_controls[] = {
 	ACARD_SINGLE_RANGE_EXT_TLV("Speaker Analog Volume", TAS2781_AMP_LEVEL,
 		1, 0, 20, 0, tas2781_amp_getvol,
-		tas2781_amp_putvol, amp_vol_tlv),
+		tas2781_amp_putvol, tas2781_amp_tlv),
 	ACARD_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0,
 		tas2781_force_fwload_get, tas2781_force_fwload_put),
 };
@@ -305,7 +305,7 @@ static int tas2563_save_calibration(struct tas2781_hda *h)
 	efi_char16_t efi_name[TAS2563_CAL_VAR_NAME_MAX];
 	unsigned long max_size = TAS2563_CAL_DATA_SIZE;
 	unsigned char var8[TAS2563_CAL_VAR_NAME_MAX];
-	struct tasdevice_priv *p = h->hda_priv;
+	struct tasdevice_priv *p = h->priv;
 	struct calidata *cd = &p->cali_data;
 	struct cali_reg *r = &cd->cali_reg_array;
 	unsigned int offset = 0;

sound/hda/codecs/side-codecs/tas2781_hda_spi.c

@@ -494,9 +494,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol,
 
 static struct snd_kcontrol_new tas2781_snd_ctls[] = {
 	ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_AMP_LEVEL, 1, 0, 20, 0,
-		tas2781_amp_getvol, tas2781_amp_putvol, amp_vol_tlv),
+		tas2781_amp_getvol, tas2781_amp_putvol,
+		tas2781_amp_tlv),
 	ACARD_SINGLE_RANGE_EXT_TLV(NULL, TAS2781_DVC_LVL, 0, 0, 200, 1,
-		tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv),
+		tas2781_digital_getvol, tas2781_digital_putvol,
+		tas2781_dvc_tlv),
 	ACARD_SINGLE_BOOL_EXT(NULL, 0, tas2781_force_fwload_get,
 		tas2781_force_fwload_put),
 };

sound/soc/codecs/cs35l56-sdw.c

@@ -393,74 +393,6 @@ static int cs35l56_sdw_update_status(struct sdw_slave *peripheral,
 	return 0;
 }
 
-static int cs35l63_sdw_kick_divider(struct cs35l56_private *cs35l56,
-				    struct sdw_slave *peripheral)
-{
-	unsigned int curr_scale_reg, next_scale_reg;
-	int curr_scale, next_scale, ret;
-
-	if (!cs35l56->base.init_done)
-		return 0;
-
-	if (peripheral->bus->params.curr_bank) {
-		curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1;
-		next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0;
-	} else {
-		curr_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B0;
-		next_scale_reg = SDW_SCP_BUSCLOCK_SCALE_B1;
-	}
-
-	/*
-	 * Current clock scale value must be different to new value.
-	 * Modify current to guarantee this. If next still has the dummy
-	 * value we wrote when it was current, the core code has not set
-	 * a new scale so restore its original good value
-	 */
-	curr_scale = sdw_read_no_pm(peripheral, curr_scale_reg);
-	if (curr_scale < 0) {
-		dev_err(cs35l56->base.dev, "Failed to read current clock scale: %d\n", curr_scale);
-		return curr_scale;
-	}
-
-	next_scale = sdw_read_no_pm(peripheral, next_scale_reg);
-	if (next_scale < 0) {
-		dev_err(cs35l56->base.dev, "Failed to read next clock scale: %d\n", next_scale);
-		return next_scale;
-	}
-
-	if (next_scale == CS35L56_SDW_INVALID_BUS_SCALE) {
-		next_scale = cs35l56->old_sdw_clock_scale;
-		ret = sdw_write_no_pm(peripheral, next_scale_reg, next_scale);
-		if (ret < 0) {
-			dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n",
-				ret);
-			return ret;
-		}
-	}
-
-	cs35l56->old_sdw_clock_scale = curr_scale;
-	ret = sdw_write_no_pm(peripheral, curr_scale_reg, CS35L56_SDW_INVALID_BUS_SCALE);
-	if (ret < 0) {
-		dev_err(cs35l56->base.dev, "Failed to modify current clock scale: %d\n", ret);
-		return ret;
-	}
-
-	dev_dbg(cs35l56->base.dev, "Next bus scale: %#x\n", next_scale);
-
-	return 0;
-}
-
-static int cs35l56_sdw_bus_config(struct sdw_slave *peripheral,
-				  struct sdw_bus_params *params)
-{
-	struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev);
-
-	if ((cs35l56->base.type == 0x63) && (cs35l56->base.rev < 0xa1))
-		return cs35l63_sdw_kick_divider(cs35l56, peripheral);
-
-	return 0;
-}
-
 static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral,
 					       enum sdw_clk_stop_mode mode,
 					       enum sdw_clk_stop_type type)
@@ -476,7 +408,6 @@ static const struct sdw_slave_ops cs35l56_sdw_ops = {
 	.read_prop = cs35l56_sdw_read_prop,
 	.interrupt_callback = cs35l56_sdw_interrupt,
 	.update_status = cs35l56_sdw_update_status,
-	.bus_config = cs35l56_sdw_bus_config,
 #ifdef DEBUG
 	.clk_stop = cs35l56_sdw_clk_stop,
 #endif

sound/soc/codecs/cs35l56-shared.c

@@ -838,6 +838,15 @@ const struct cirrus_amp_cal_controls cs35l56_calibration_controls = {
 };
 EXPORT_SYMBOL_NS_GPL(cs35l56_calibration_controls, "SND_SOC_CS35L56_SHARED");
 
+static const struct cirrus_amp_cal_controls cs35l63_calibration_controls = {
+	.alg_id =	0xbf210,
+	.mem_region =	WMFW_ADSP2_YM,
+	.ambient =	"CAL_AMBIENT",
+	.calr =		"CAL_R",
+	.status =	"CAL_STATUS",
+	.checksum =	"CAL_CHECKSUM",
+};
+
 int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base)
 {
 	u64 silicon_uid = 0;
@@ -912,19 +921,31 @@ EXPORT_SYMBOL_NS_GPL(cs35l56_read_prot_status, "SND_SOC_CS35L56_SHARED");
 void cs35l56_log_tuning(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_dsp)
 {
 	__be32 pid, sid, tid;
+	unsigned int alg_id;
 	int ret;
 
+	switch (cs35l56_base->type) {
+	case 0x54:
+	case 0x56:
+	case 0x57:
+		alg_id = 0x9f212;
+		break;
+	default:
+		alg_id = 0xbf212;
+		break;
+	}
+
 	scoped_guard(mutex, &cs_dsp->pwr_lock) {
 		ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_PRJCT_ID",
-							    WMFW_ADSP2_XM, 0x9f212),
+							    WMFW_ADSP2_XM, alg_id),
 					     0, &pid, sizeof(pid));
 		if (!ret)
 			ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_CHNNL_ID",
-								    WMFW_ADSP2_XM, 0x9f212),
+								    WMFW_ADSP2_XM, alg_id),
 						     0, &sid, sizeof(sid));
 		if (!ret)
 			ret = cs_dsp_coeff_read_ctrl(cs_dsp_get_ctl(cs_dsp, "AS_SNPSHT_ID",
-								    WMFW_ADSP2_XM, 0x9f212),
+								    WMFW_ADSP2_XM, alg_id),
 						     0, &tid, sizeof(tid));
 	}
 
@@ -974,8 +995,10 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base)
 	case 0x35A54:
 	case 0x35A56:
 	case 0x35A57:
+		cs35l56_base->calibration_controls = &cs35l56_calibration_controls;
 		break;
 	case 0x35A630:
+		cs35l56_base->calibration_controls = &cs35l63_calibration_controls;
 		devid = devid >> 4;
 		break;
 	default:

sound/soc/codecs/cs35l56.c

@@ -695,7 +695,7 @@ static int cs35l56_write_cal(struct cs35l56_private *cs35l56)
 		return ret;
 
 	ret = cs_amp_write_cal_coeffs(&cs35l56->dsp.cs_dsp,
-				      &cs35l56_calibration_controls,
+				      cs35l56->base.calibration_controls,
 				      &cs35l56->base.cal_data);
 
 	wm_adsp_stop(&cs35l56->dsp);

sound/soc/codecs/cs35l56.h

@@ -20,8 +20,6 @@
 #define CS35L56_SDW_GEN_INT_MASK_1	0xc1
 #define CS35L56_SDW_INT_MASK_CODEC_IRQ	BIT(0)
 
-#define CS35L56_SDW_INVALID_BUS_SCALE	0xf
-
 #define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE)
 #define CS35L56_TX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE \
 			    | SNDRV_PCM_FMTBIT_S32_LE)
@@ -52,7 +50,6 @@ struct cs35l56_private {
 	u8 asp_slot_count;
 	bool tdm_mode;
 	bool sysclk_set;
-	u8 old_sdw_clock_scale;
 	u8 sdw_link_num;
 	u8 sdw_unique_id;
 };