Merge tag 'sound-6.13' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Hopefully the last PR for 6.13. This became bigger than wished due to
  the timing after holiday breaks.

  The only large LOC is the additional document for Cirrus codec which
  is nice for users (and absolutely safe). All the rest are small fixes
  in ASoC Rcar and codecs as well as HD-audio quirks (And no fix for USB
  guitar pedals seen yet :)"

* tag 'sound-6.13' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda/realtek: Fix volume adjustment issue on Lenovo ThinkBook 16P Gen5
  ALSA: hda/realtek: fixup ASUS H7606W
  ALSA: hda/realtek: fixup ASUS GA605W
  ALSA: hda/realtek: Add support for Ayaneo System using CS35L41 HDA
  ASoC: rsnd: check rsnd_adg_clk_enable() return value
  ASoC: cs42l43: Add codec force suspend/resume ops
  ALSA: doc: Add codecs/index.rst to top-level index
  ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57
  ASoC: samsung: Add missing depends on I2C
  MAINTAINERS: add missing maintainers for Simple Audio Card
  ASoC: samsung: Add missing selects for MFD_WM8994
  ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
  ASoC: wm8994: Add depends on MFD core
  ASoC: tas2781: Fix occasional calibration failture
  ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise

10 files changed, 65 insertions, 19 deletions

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b74b566f675e..ad66378d7321 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10641,6 +10641,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x1e1f, "ASUS Vivobook 15 X1504VAP", ALC2XX_FIXUP_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS),
 	SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS),
+	SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1),
+	SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1),
 	SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401),
 	SND_PCI_QUIRK(0x1043, 0x1eb3, "ASUS Ally RCLA72", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x1043, 0x1ed3, "ASUS HN7306W", ALC287_FIXUP_CS35L41_I2C_2),
@@ -10930,8 +10932,8 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x38e0, "Yoga Y990 Intel VECO Dual", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x38f8, "Yoga Book 9i", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C),
-	SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2),
-	SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
+	SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_MG_RTKC_CSAMP_CS35L41_I2C_THINKPAD),
 	SND_PCI_QUIRK(0x17aa, 0x38fd, "ThinkBook plus Gen5 Hybrid", ALC287_FIXUP_TAS2781_I2C),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3913, "Lenovo 145", ALC236_FIXUP_LENOVO_INV_DMIC),
@@ -10995,6 +10997,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1945, "Redmi G", ALC256_FIXUP_ASUS_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1947, "RedmiBook Air", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
+	SND_PCI_QUIRK(0x1f66, 0x0105, "Ayaneo Portable Game Player", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x2782, 0x0214, "VAIO VJFE-CL", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13),
 	SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO),
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0f2df7c91e18..0b9e87dc2b6c 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -2451,6 +2451,7 @@ config SND_SOC_WM8993
 
 config SND_SOC_WM8994
 	tristate
+	depends on MFD_WM8994
 
 config SND_SOC_WM8995
 	tristate
diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c
index 4236f78beec0..83c21c17fb80 100644
--- a/sound/soc/codecs/cs42l43.c
+++ b/sound/soc/codecs/cs42l43.c
@@ -2404,6 +2404,7 @@ static int cs42l43_codec_runtime_resume(struct device *dev)
 
 static const struct dev_pm_ops cs42l43_codec_pm_ops = {
 	RUNTIME_PM_OPS(NULL, cs42l43_codec_runtime_resume, NULL)
+	SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume)
 };
 
 static const struct platform_device_id cs42l43_codec_id_table[] = {
diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c
index 61729e5b50a8..f508df01145b 100644
--- a/sound/soc/codecs/es8316.c
+++ b/sound/soc/codecs/es8316.c
@@ -39,7 +39,9 @@ struct es8316_priv {
 	struct snd_soc_jack *jack;
 	int irq;
 	unsigned int sysclk;
-	unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)];
+	/* ES83xx supports halving the MCLK so it supports twice as many rates
+	 */
+	unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2];
 	struct snd_pcm_hw_constraint_list sysclk_constraints;
 	bool jd_inverted;
 };
@@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 
 		if (freq % ratio == 0)
 			es8316->allowed_rates[count++] = freq / ratio;
+
+		/* We also check if the halved MCLK produces a valid rate
+		 * since the codec supports halving the MCLK.
+		 */
+		if ((freq / ratio) % 2 == 0)
+			es8316->allowed_rates[count++] = freq / ratio / 2;
 	}
 
 	if (count) {
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index a5603b617688..b06eead7e0f6 100644
--- a/sound/soc/codecs/es8326.c
+++ b/sound/soc/codecs/es8326.c
@@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 					0x0F, 0x0F);
 			if (es8326->version > ES8326_VERSION_B) {
 				regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
-				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10);
+				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30);
 			}
 		}
 	} else {
@@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 			regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r);
 			es8326->calibrated = true;
 		}
+		regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00);
+                regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00);
 		if (direction == SNDRV_PCM_STREAM_PLAYBACK) {
 			regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01);
 			usleep_range(1000, 5000);
@@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction)
 		} else {
 			msleep(300);
 			if (es8326->version > ES8326_VERSION_B) {
-				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50);
+				regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70);
 				regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00);
 			}
 			regmap_update_bits(es8326->regmap,  ES8326_ADC_MUTE,
@@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec,
 		regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00);
 		regmap_update_bits(es8326->regmap,  ES8326_CLK_CTL, 0x20, 0x20);
 		regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00);
+		if (es8326->version > ES8326_VERSION_B) {
+			regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
+			regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30);
+		}
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
@@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec,
 		regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b);
 		regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00);
 		regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT);
+		if (es8326->version > ES8326_VERSION_B) {
+			regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
+			regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10);
+		}
+		regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0);
+		regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80);
 		break;
 	case SND_SOC_BIAS_OFF:
 		clk_disable_unprepare(es8326->mclk);
@@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work)
 	case 0x6f:
 	case 0x4b:
 		/* button volume up */
-		cur_button = SND_JACK_BTN_1;
+		if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B))
+			cur_button = SND_JACK_BTN_0;
+		else
+			cur_button = SND_JACK_BTN_1;
 		break;
 	case 0x27:
 		/* button volume down */
@@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component)
 	regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66);
 	es8326_disable_micbias(es8326->component);
 	if (es8326->version > ES8326_VERSION_B) {
-		regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13);
+		regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10);
 		regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40);
 	}
 
diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c
index fb8cd2284fe8..728bf78ae71f 100644
--- a/sound/soc/codecs/tas2781-i2c.c
+++ b/sound/soc/codecs/tas2781-i2c.c
@@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = {
 	X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false),
-	X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false),
+	X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true),
 	X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true),
 	X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false),
 	X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false),
diff --git a/sound/soc/renesas/rcar/adg.c b/sound/soc/renesas/rcar/adg.c
index 0f190abf00e7..191f212d338c 100644
--- a/sound/soc/renesas/rcar/adg.c
+++ b/sound/soc/renesas/rcar/adg.c
@@ -374,12 +374,12 @@ int rsnd_adg_ssi_clk_try_start(struct rsnd_mod *ssi_mod, unsigned int rate)
 	return 0;
 }
 
-void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
+int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 {
 	struct rsnd_adg *adg = rsnd_priv_to_adg(priv);
 	struct rsnd_mod *adg_mod = rsnd_mod_get(adg);
 	struct clk *clk;
-	int i;
+	int ret = 0, i;
 
 	if (enable) {
 		rsnd_mod_bset(adg_mod, BRGCKR, 0x80770000, adg->ckr);
@@ -389,18 +389,33 @@ void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable)
 
 	for_each_rsnd_clkin(clk, adg, i) {
 		if (enable) {
-			clk_prepare_enable(clk);
+			ret = clk_prepare_enable(clk);
 
 			/*
 			 * We shouldn't use clk_get_rate() under
 			 * atomic context. Let's keep it when
 			 * rsnd_adg_clk_enable() was called
 			 */
+			if (ret < 0)
+				break;
+
 			adg->clkin_rate[i] = clk_get_rate(clk);
 		} else {
-			clk_disable_unprepare(clk);
+			if (adg->clkin_rate[i])
+				clk_disable_unprepare(clk);
+
+			adg->clkin_rate[i] = 0;
 		}
 	}
+
+	/*
+	 * rsnd_adg_clk_enable() might return error (_disable() will not).
+	 * We need to rollback in such case
+	 */
+	if (ret < 0)
+		rsnd_adg_clk_disable(priv);
+
+	return ret;
 }
 
 static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv,
@@ -753,7 +768,10 @@ int rsnd_adg_probe(struct rsnd_priv *priv)
 	if (ret)
 		return ret;
 
-	rsnd_adg_clk_enable(priv);
+	ret = rsnd_adg_clk_enable(priv);
+	if (ret)
+		return ret;
+
 	rsnd_adg_clk_dbg_info(priv, NULL);
 
 	return 0;
diff --git a/sound/soc/renesas/rcar/core.c b/sound/soc/renesas/rcar/core.c
index e2234928c9e8..d3709fd0409e 100644
--- a/sound/soc/renesas/rcar/core.c
+++ b/sound/soc/renesas/rcar/core.c
@@ -2086,9 +2086,7 @@ static int __maybe_unused rsnd_resume(struct device *dev)
 {
 	struct rsnd_priv *priv = dev_get_drvdata(dev);
 
-	rsnd_adg_clk_enable(priv);
-
-	return 0;
+	return rsnd_adg_clk_enable(priv);
 }
 
 static const struct dev_pm_ops rsnd_pm_ops = {
diff --git a/sound/soc/renesas/rcar/rsnd.h b/sound/soc/renesas/rcar/rsnd.h
index 3c164d8e3b16..a5f54b65313c 100644
--- a/sound/soc/renesas/rcar/rsnd.h
+++ b/sound/soc/renesas/rcar/rsnd.h
@@ -608,7 +608,7 @@ int rsnd_adg_set_cmd_timsel_gen2(struct rsnd_mod *cmd_mod,
 				 struct rsnd_dai_stream *io);
 #define rsnd_adg_clk_enable(priv)	rsnd_adg_clk_control(priv, 1)
 #define rsnd_adg_clk_disable(priv)	rsnd_adg_clk_control(priv, 0)
-void rsnd_adg_clk_control(struct rsnd_priv *priv, int enable);
+int rsnd_adg_clk_control(struct rsnd_priv *priv, int enable);
 void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m);
 
 /*
diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig
index 4b1ea7b2c796..60b4b7b75215 100644
--- a/sound/soc/samsung/Kconfig
+++ b/sound/soc/samsung/Kconfig
@@ -127,8 +127,9 @@ config SND_SOC_SAMSUNG_TM2_WM5110
 
 config SND_SOC_SAMSUNG_ARIES_WM8994
 	tristate "SoC I2S Audio support for WM8994 on Aries"
-	depends on SND_SOC_SAMSUNG && MFD_WM8994 && IIO && EXTCON
+	depends on SND_SOC_SAMSUNG && I2C && IIO && EXTCON
 	select SND_SOC_BT_SCO
+	select MFD_WM8994
 	select SND_SOC_WM8994
 	select SND_SAMSUNG_I2S
 	help
@@ -140,8 +141,9 @@ config SND_SOC_SAMSUNG_ARIES_WM8994
 
 config SND_SOC_SAMSUNG_MIDAS_WM1811
 	tristate "SoC I2S Audio support for Midas boards"
-	depends on SND_SOC_SAMSUNG && IIO
+	depends on SND_SOC_SAMSUNG && I2C && IIO
 	select SND_SAMSUNG_I2S
+	select MFD_WM8994
 	select SND_SOC_WM8994
 	help
 	  Say Y if you want to add support for SoC audio on the Midas boards.