Merge tag 'sound-6.19-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai: "This became a bit larger than wished for, often seen as a bump at the middle, but almost all changes are small device-specific fixes, so the risk must be pretty low. - SoundWire fix for missing symbol export - Fixes for device-tree bindings - A fix for OOB access in USB-audio, spotted by fuzzer - Quirks for HD-audio, SoundWire, AMD ACP - A series of ASoC tlv320 and wsa codec fixes - Other misc fixes in PCM OSS error-handling, Cirrus scodec test, ASoC ops endianess, davinci, simple-card, and tegra" * tag 'sound-6.19-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (33 commits) ALSA: hda/tas2781: Add newly-released HP laptop ASoC: rt5640: Fix duplicate clock properties in DT binding ALSA: hda/realtek: Add quirk for HP Pavilion x360 to enable mute LED ASoC: tlv320adcx140: fix word length ASoC: tlv320adcx140: Propagate error codes during probe ASoC: tlv320adcx140: fix null pointer ASoC: tlv320adcx140: invert DRE_ENABLE ASoC: sdw_utils: cs42l43: Enable Headphone pin for LINEOUT jack type ASoC: sdw_utils: Call init callbacks on the correct codec DAI soundwire: Add missing EXPORT for sdw_slave_type ALSA: usb-audio: Prevent excessive number of frames ALSA: hda/cirrus_scodec_test: Fix test suite name ALSA: hda/cirrus_scodec_test: Fix incorrect setup of gpiochip ALSA: hda/realtek: Add quirk for Asus Zephyrus G14 2025 using CS35L56, fix speakers ASoC: amd: yc: Fix microphone on ASUS M6500RE ASoC: tegra: Revert fix for uninitialized flat cache warning in tegra210_ahub ASoC: dt-bindings: rockchip-spdif: Allow "port" node ASoC: dt-bindings: realtek,rt5640: Allow 7 for realtek,jack-detect-source ASoC: dt-bindings: realtek,rt5640: Add missing properties/node ASoC: dt-bindings: realtek,rt5640: Document port node ...
2026-01-16 10:48:17 -08:00
parent c2a44a02d7 46b8d0888f
commit 711673f8dd
23 changed files with 182 additions and 41 deletions
--- a/Documentation/devicetree/bindings/sound/everest,es8316.yaml
+++ b/Documentation/devicetree/bindings/sound/everest,es8316.yaml
@@ -49,6 +49,10 @@ properties:
    items:
      - const: mclk

+  interrupts:
+    maxItems: 1
+    description: Headphone detect interrupt
+
  port:
    $ref: audio-graph-port.yaml#
    unevaluatedProperties: false
--- a/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml
+++ b/Documentation/devicetree/bindings/sound/realtek,rt5640.yaml
@@ -47,6 +47,12 @@ properties:
  reg:
    maxItems: 1

+  clocks:
+    maxItems: 1
+
+  clock-names:
+    const: mclk
+
  interrupts:
    maxItems: 1
    description: The CODEC's interrupt output.
@@ -98,6 +104,7 @@ properties:
      - 4 # Use GPIO2 for jack-detect
      - 5 # Use GPIO3 for jack-detect
      - 6 # Use GPIO4 for jack-detect
+      - 7 # Use HDA header for jack-detect

  realtek,jack-detect-not-inverted:
    description:
@@ -121,6 +128,10 @@ properties:
      - 2 # Scale current by 1.0
      - 3 # Scale current by 1.5

+  port:
+    $ref: audio-graph-port.yaml#
+    unevaluatedProperties: false
+
 required:
  - compatible
  - reg
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
@@ -70,6 +70,9 @@ properties:
  "#sound-dai-cells":
    const: 0

+  port:
+    $ref: /schemas/graph.yaml#/properties/port
+
 required:
  - compatible
  - reg
--- a/drivers/soundwire/slave.c
+++ b/drivers/soundwire/slave.c
@@ -23,6 +23,7 @@ const struct device_type sdw_slave_type = {
 	.release =	sdw_slave_release,
 	.uevent =	sdw_slave_uevent,
 };
+EXPORT_SYMBOL_GPL(sdw_slave_type);

 int sdw_slave_add(struct sdw_bus *bus,
 		  struct sdw_slave_id *id, struct fwnode_handle *fwnode)
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -1402,7 +1402,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s
 #define snd_pcm_lib_mmap_iomem	NULL
 #endif

-void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime);
+int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime);

 /**
 * snd_pcm_limit_isa_dma_size - Get the max size fitting with ISA DMA transfer
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1074,7 +1074,9 @@ static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream)
 	runtime->oss.params = 0;
 	runtime->oss.prepare = 1;
 	runtime->oss.buffer_used = 0;
-	snd_pcm_runtime_buffer_set_silence(runtime);
+	err = snd_pcm_runtime_buffer_set_silence(runtime);
+	if (err < 0)
+		goto failure;

 	runtime->oss.period_frames = snd_pcm_alsa_frames(substream, oss_period_size);

--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -730,13 +730,18 @@ static void snd_pcm_buffer_access_unlock(struct snd_pcm_runtime *runtime)
 }

 /* fill the PCM buffer with the current silence format; called from pcm_oss.c */
-void snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime)
+int snd_pcm_runtime_buffer_set_silence(struct snd_pcm_runtime *runtime)
 {
-	snd_pcm_buffer_access_lock(runtime);
+	int err;
+
+	err = snd_pcm_buffer_access_lock(runtime);
+	if (err < 0)
+		return err;
 	if (runtime->dma_area)
 		snd_pcm_format_set_silence(runtime->format, runtime->dma_area,
 					   bytes_to_samples(runtime, runtime->dma_bytes));
 	snd_pcm_buffer_access_unlock(runtime);
+	return 0;
 }
 EXPORT_SYMBOL_GPL(snd_pcm_runtime_buffer_set_silence);

--- a/sound/hda/codecs/realtek/alc269.c
+++ b/sound/hda/codecs/realtek/alc269.c
@@ -6613,6 +6613,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8a2e, "HP Envy 16", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8a30, "HP Envy 17", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8a31, "HP Envy 15", ALC287_FIXUP_CS35L41_I2C_2),
+	SND_PCI_QUIRK(0x103c, 0x8a34, "HP Pavilion x360 2-in-1 Laptop 14-ek0xxx", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
 	SND_PCI_QUIRK(0x103c, 0x8a4f, "HP Victus 15-fa0xxx (MB 8A4F)", ALC245_FIXUP_HP_MUTE_LED_COEFBIT),
 	SND_PCI_QUIRK(0x103c, 0x8a6e, "HP EDNA 360", ALC287_FIXUP_CS35L41_I2C_4),
 	SND_PCI_QUIRK(0x103c, 0x8a74, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
@@ -6817,6 +6818,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8f42, "HP ZBook 8 G2a 14W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x8f57, "HP Trekker G7JC", ALC287_FIXUP_CS35L41_I2C_2),
 	SND_PCI_QUIRK(0x103c, 0x8f62, "HP ZBook 8 G2a 16W", ALC245_FIXUP_HP_TAS2781_I2C_MUTE_LED),
+	SND_PCI_QUIRK(0x1043, 0x1024, "ASUS Zephyrus G14 2025", ALC285_FIXUP_ASUS_GA403U_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1032, "ASUS VivoBook X513EA", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1043, 0x1034, "ASUS GU605C", ALC285_FIXUP_ASUS_GU605_SPI_SPEAKER2_TO_DAC1),
 	SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC),
--- a/sound/hda/codecs/side-codecs/cirrus_scodec_test.c
+++ b/sound/hda/codecs/side-codecs/cirrus_scodec_test.c
@@ -103,6 +103,7 @@ static int cirrus_scodec_test_gpio_probe(struct platform_device *pdev)

 	/* GPIO core modifies our struct gpio_chip so use a copy */
 	gpio_priv->chip = cirrus_scodec_test_gpio_chip;
+	gpio_priv->chip.parent = &pdev->dev;
 	ret = devm_gpiochip_add_data(&pdev->dev, &gpio_priv->chip, gpio_priv);
 	if (ret)
 		return dev_err_probe(&pdev->dev, ret, "Failed to add gpiochip\n");
@@ -319,7 +320,7 @@ static struct kunit_case cirrus_scodec_test_cases[] = {
 };

 static struct kunit_suite cirrus_scodec_test_suite = {
-	.name = "snd-hda-scodec-cs35l56-test",
+	.name = "snd-hda-cirrus-scodec-test",
 	.init = cirrus_scodec_test_case_init,
 	.test_cases = cirrus_scodec_test_cases,
 };
--- a/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c
+++ b/sound/hda/codecs/side-codecs/tas2781_hda_i2c.c
@@ -2,7 +2,7 @@
 //
 // TAS2781 HDA I2C driver
 //
-// Copyright 2023 - 2025 Texas Instruments, Inc.
+// Copyright 2023 - 2026 Texas Instruments, Inc.
 //
 // Author: Shenghao Ding <shenghao-ding@ti.com>
 // Current maintainer: Baojun Xu <baojun.xu@ti.com>
@@ -60,6 +60,7 @@ struct tas2781_hda_i2c_priv {
 	int (*save_calibration)(struct tas2781_hda *h);

 	int hda_chip_id;
+	bool skip_calibration;
 };

 static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data)
@@ -491,7 +492,8 @@ static void tasdevice_dspfw_init(void *context)
 	/* If calibrated data occurs error, dsp will still works with default
 	 * calibrated data inside algo.
 	 */
-	hda_priv->save_calibration(tas_hda);
+	if (!hda_priv->skip_calibration)
+		hda_priv->save_calibration(tas_hda);
 }

 static void tasdev_fw_ready(const struct firmware *fmw, void *context)
@@ -548,6 +550,7 @@ static int tas2781_hda_bind(struct device *dev, struct device *master,
 	void *master_data)
 {
 	struct tas2781_hda *tas_hda = dev_get_drvdata(dev);
+	struct tas2781_hda_i2c_priv *hda_priv = tas_hda->hda_priv;
 	struct hda_component_parent *parent = master_data;
 	struct hda_component *comp;
 	struct hda_codec *codec;
@@ -568,11 +571,22 @@ static int tas2781_hda_bind(struct device *dev, struct device *master,
 	case 0x1028:
 		tas_hda->catlog_id = DELL;
 		break;
+	case 0x103C:
+		tas_hda->catlog_id = HP;
+		break;
 	default:
 		tas_hda->catlog_id = LENOVO;
 		break;
 	}

+	/*
+	 * Using ASUS ROG Xbox Ally X (RC73XA) UEFI calibration data
+	 * causes audio dropouts during playback, use fallback data
+	 * from DSP firmware as a workaround.
+	 */
+	if (codec->core.subsystem_id == 0x10431384)
+		hda_priv->skip_calibration = true;
+
 	pm_runtime_get_sync(dev);

 	comp->dev = dev;
--- a/sound/soc/amd/yc/acp6x-mach.c
+++ b/sound/soc/amd/yc/acp6x-mach.c
@@ -416,6 +416,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M6500RC"),
 		}
 	},
+	{
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "M6500RE"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {
--- a/sound/soc/codecs/tlv320adcx140.c
+++ b/sound/soc/codecs/tlv320adcx140.c
@@ -23,7 +23,6 @@
 #include "tlv320adcx140.h"

 struct adcx140_priv {
-	struct snd_soc_component *component;
 	struct regulator *supply_areg;
 	struct gpio_desc *gpio_reset;
 	struct regmap *regmap;
@@ -338,7 +337,7 @@ static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch =
 	SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0);

 static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch =
-	SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0);
+	SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 1);

 /* Output Mixer */
 static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = {
@@ -699,7 +698,6 @@ static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state)
 {
 	int pwr_ctrl = 0;
 	int ret = 0;
-	struct snd_soc_component *component = adcx140->component;

 	if (power_state)
 		pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ;
@@ -711,7 +709,7 @@ static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state)
 		ret = regmap_write(adcx140->regmap, ADCX140_PHASE_CALIB,
 			adcx140->phase_calib_on ? 0x00 : 0x40);
 		if (ret)
-			dev_err(component->dev, "%s: register write error %d\n",
+			dev_err(adcx140->dev, "%s: register write error %d\n",
 				__func__, ret);
 	}

@@ -727,7 +725,7 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream,
 	struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component);
 	u8 data = 0;

-	switch (params_width(params)) {
+	switch (params_physical_width(params)) {
 	case 16:
 		data = ADCX140_16_BIT_WORD;
 		break;
@@ -742,7 +740,7 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream,
 		break;
 	default:
 		dev_err(component->dev, "%s: Unsupported width %d\n",
-			__func__, params_width(params));
+			__func__, params_physical_width(params));
 		return -EINVAL;
 	}

@@ -1156,6 +1154,9 @@ static int adcx140_i2c_probe(struct i2c_client *i2c)
 	adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev,
 						      "reset", GPIOD_OUT_LOW);
 	if (IS_ERR(adcx140->gpio_reset))
+		return dev_err_probe(&i2c->dev, PTR_ERR(adcx140->gpio_reset),
+				     "Failed to get Reset GPIO\n");
+	if (!adcx140->gpio_reset)
 		dev_info(&i2c->dev, "Reset GPIO not defined\n");

 	adcx140->supply_areg = devm_regulator_get_optional(adcx140->dev,
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -678,6 +678,7 @@ struct wsa881x_priv {
 	 */
 	unsigned int sd_n_val;
 	int active_ports;
+	bool hw_init;
 	bool port_prepared[WSA881X_MAX_SWR_PORTS];
 	bool port_enable[WSA881X_MAX_SWR_PORTS];
 };
@@ -687,6 +688,9 @@ static void wsa881x_init(struct wsa881x_priv *wsa881x)
 	struct regmap *rm = wsa881x->regmap;
 	unsigned int val = 0;

+	if (wsa881x->hw_init)
+		return;
+
 	regmap_register_patch(wsa881x->regmap, wsa881x_rev_2_0,
 			      ARRAY_SIZE(wsa881x_rev_2_0));

@@ -724,6 +728,8 @@ static void wsa881x_init(struct wsa881x_priv *wsa881x)
 	regmap_update_bits(rm, WSA881X_OTP_REG_28, 0x3F, 0x3A);
 	regmap_update_bits(rm, WSA881X_BONGO_RESRV_REG1, 0xFF, 0xB2);
 	regmap_update_bits(rm, WSA881X_BONGO_RESRV_REG2, 0xFF, 0x05);
+
+	wsa881x->hw_init = true;
 }

 static int wsa881x_component_probe(struct snd_soc_component *comp)
@@ -1067,6 +1073,9 @@ static int wsa881x_update_status(struct sdw_slave *slave,
 {
 	struct wsa881x_priv *wsa881x = dev_get_drvdata(&slave->dev);

+	if (status == SDW_SLAVE_UNATTACHED)
+		wsa881x->hw_init = false;
+
 	if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0)
 		wsa881x_init(wsa881x);

--- a/sound/soc/codecs/wsa883x.c
+++ b/sound/soc/codecs/wsa883x.c
@@ -475,6 +475,7 @@ struct wsa883x_priv {
 	int active_ports;
 	int dev_mode;
 	int comp_offset;
+	bool hw_init;
 	/*
 	 * Protects temperature reading code (related to speaker protection) and
 	 * fields: temperature and pa_on.
@@ -1043,6 +1044,9 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x)
 	struct regmap *regmap = wsa883x->regmap;
 	int variant, version, ret;

+	if (wsa883x->hw_init)
+		return 0;
+
 	ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant);
 	if (ret)
 		return ret;
@@ -1054,22 +1058,23 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x)

 	switch (variant) {
 	case WSA8830:
-		dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n",
-			 version);
+		dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8830\n",
+			version);
 		break;
 	case WSA8835:
-		dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n",
-			 version);
+		dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835\n",
+			version);
 		break;
 	case WSA8832:
-		dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n",
-			 version);
+		dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8832\n",
+			version);
 		break;
 	case WSA8835_V2:
-		dev_info(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n",
-			 version);
+		dev_dbg(wsa883x->dev, "WSA883X Version 1_%d, Variant: WSA8835_V2\n",
+			version);
 		break;
 	default:
+		dev_warn(wsa883x->dev, "unknown variant: %d\n", variant);
 		break;
 	}

@@ -1085,6 +1090,8 @@ static int wsa883x_init(struct wsa883x_priv *wsa883x)
 				   wsa883x->comp_offset);
 	}

+	wsa883x->hw_init = true;
+
 	return 0;
 }

@@ -1093,6 +1100,9 @@ static int wsa883x_update_status(struct sdw_slave *slave,
 {
 	struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev);

+	if (status == SDW_SLAVE_UNATTACHED)
+		wsa883x->hw_init = false;
+
 	if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0)
 		return wsa883x_init(wsa883x);

--- a/sound/soc/codecs/wsa884x.c
+++ b/sound/soc/codecs/wsa884x.c
@@ -1534,7 +1534,7 @@ static void wsa884x_init(struct wsa884x_priv *wsa884x)

 	wsa884x_set_gain_parameters(wsa884x);

-	wsa884x->hw_init = false;
+	wsa884x->hw_init = true;
 }

 static int wsa884x_update_status(struct sdw_slave *slave,
@@ -2109,7 +2109,6 @@ static int wsa884x_probe(struct sdw_slave *pdev,

 	/* Start in cache-only until device is enumerated */
 	regcache_cache_only(wsa884x->regmap, true);
-	wsa884x->hw_init = true;

 	if (IS_REACHABLE(CONFIG_HWMON)) {
 		struct device *hwmon;
--- a/sound/soc/generic/simple-card-utils.c
+++ b/sound/soc/generic/simple-card-utils.c
@@ -1179,9 +1179,9 @@ void graph_util_parse_link_direction(struct device_node *np,
 	bool is_playback_only = of_property_read_bool(np, "playback-only");
 	bool is_capture_only  = of_property_read_bool(np, "capture-only");

-	if (playback_only)
+	if (np && playback_only)
 		*playback_only = is_playback_only;
-	if (capture_only)
+	if (np && capture_only)
 		*capture_only = is_capture_only;
 }
 EXPORT_SYMBOL_GPL(graph_util_parse_link_direction);
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -764,6 +764,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = {
 		.driver_data = (void *)(SOC_SDW_CODEC_SPKR),
 	},
 	/* Pantherlake devices*/
+	{
+		.callback = sof_sdw_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0DD6")
+		},
+		.driver_data = (void *)(SOC_SDW_SIDECAR_AMPS),
+	},
 	{
 		.callback = sof_sdw_quirk_cb,
 		.matches = {
--- a/sound/soc/sdw_utils/soc_sdw_cs42l43.c
+++ b/sound/soc/sdw_utils/soc_sdw_cs42l43.c
@@ -44,7 +44,7 @@ static const struct snd_soc_dapm_route cs42l43_dmic_map[] = {
 static struct snd_soc_jack_pin soc_jack_pins[] = {
 	{
 		.pin    = "Headphone",
-		.mask   = SND_JACK_HEADPHONE,
+		.mask   = SND_JACK_HEADPHONE | SND_JACK_LINEOUT,
 	},
 	{
 		.pin    = "Headset Mic",
--- a/sound/soc/sdw_utils/soc_sdw_utils.c
+++ b/sound/soc/sdw_utils/soc_sdw_utils.c
@@ -841,6 +841,19 @@ struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_part(const u64 adr)
 }
 EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_part, "SND_SOC_SDW_UTILS");

+static struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_sdw_id(const struct sdw_slave_id *id)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(codec_info_list); i++)
+		if (id->part_id == codec_info_list[i].part_id &&
+		    (!codec_info_list[i].version_id ||
+		     id->sdw_version == codec_info_list[i].version_id))
+			return &codec_info_list[i];
+
+	return NULL;
+}
+
 struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_acpi(const u8 *acpi_id)
 {
 	int i;
@@ -873,22 +886,46 @@ struct asoc_sdw_codec_info *asoc_sdw_find_codec_info_dai(const char *dai_name, i
 }
 EXPORT_SYMBOL_NS(asoc_sdw_find_codec_info_dai, "SND_SOC_SDW_UTILS");

+static int asoc_sdw_find_codec_info_dai_index(const struct asoc_sdw_codec_info *codec_info,
+					      const char *dai_name)
+{
+	int i;
+
+	for (i = 0; i < codec_info->dai_num; i++) {
+		if (!strcmp(codec_info->dais[i].dai_name, dai_name))
+			return i;
+	}
+
+	return -ENOENT;
+}
+
 int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
 {
 	struct snd_soc_card *card = rtd->card;
 	struct snd_soc_dapm_context *dapm = snd_soc_card_to_dapm(card);
 	struct asoc_sdw_codec_info *codec_info;
 	struct snd_soc_dai *dai;
+	struct sdw_slave *sdw_peripheral;
 	const char *spk_components="";
 	int dai_index;
 	int ret;
 	int i;

 	for_each_rtd_codec_dais(rtd, i, dai) {
-		codec_info = asoc_sdw_find_codec_info_dai(dai->name, &dai_index);
+		if (is_sdw_slave(dai->component->dev))
+			sdw_peripheral = dev_to_sdw_dev(dai->component->dev);
+		else if (dai->component->dev->parent && is_sdw_slave(dai->component->dev->parent))
+			sdw_peripheral = dev_to_sdw_dev(dai->component->dev->parent);
+		else
+			continue;
+
+		codec_info = asoc_sdw_find_codec_info_sdw_id(&sdw_peripheral->id);
 		if (!codec_info)
 			return -EINVAL;

+		dai_index = asoc_sdw_find_codec_info_dai_index(codec_info, dai->name);
+		WARN_ON(dai_index < 0);
+
 		/*
 		 * A codec dai can be connected to different dai links for capture and playback,
 		 * but we only need to call the rtd_init function once.
@@ -898,6 +935,10 @@ int asoc_sdw_rtd_init(struct snd_soc_pcm_runtime *rtd)
 		if (codec_info->dais[dai_index].rtd_init_done)
 			continue;

+		dev_dbg(card->dev, "%#x/%s initializing for %s/%s\n",
+			codec_info->part_id, codec_info->dais[dai_index].dai_name,
+			dai->component->name, dai->name);
+
 		/*
 		 * Add card controls and dapm widgets for the first codec dai.
 		 * The controls and widgets will be used for all codec dais.
--- a/sound/soc/soc-ops.c
+++ b/sound/soc/soc-ops.c
@@ -543,11 +543,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol,
 			ucontrol->value.bytes.data[0] &= ~params->mask;
 			break;
 		case 2:
-			((u16 *)(&ucontrol->value.bytes.data))[0]
+			((__be16 *)(&ucontrol->value.bytes.data))[0]
 				&= cpu_to_be16(~params->mask);
 			break;
 		case 4:
-			((u32 *)(&ucontrol->value.bytes.data))[0]
+			((__be32 *)(&ucontrol->value.bytes.data))[0]
 				&= cpu_to_be32(~params->mask);
 			break;
 		default:
--- a/sound/soc/tegra/tegra210_ahub.c
+++ b/sound/soc/tegra/tegra210_ahub.c
@@ -2077,7 +2077,7 @@ static const struct regmap_config tegra210_ahub_regmap_config = {
 	.val_bits		= 32,
 	.reg_stride		= 4,
 	.max_register		= TEGRA210_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT_S,
+	.cache_type		= REGCACHE_FLAT,
 };

 static const struct regmap_config tegra186_ahub_regmap_config = {
@@ -2085,7 +2085,7 @@ static const struct regmap_config tegra186_ahub_regmap_config = {
 	.val_bits		= 32,
 	.reg_stride		= 4,
 	.max_register		= TEGRA186_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT_S,
+	.cache_type		= REGCACHE_FLAT,
 };

 static const struct regmap_config tegra264_ahub_regmap_config = {
@@ -2094,7 +2094,7 @@ static const struct regmap_config tegra264_ahub_regmap_config = {
 	.reg_stride		= 4,
 	.writeable_reg		= tegra264_ahub_wr_reg,
 	.max_register		= TEGRA264_MAX_REGISTER_ADDR,
-	.cache_type		= REGCACHE_FLAT_S,
+	.cache_type		= REGCACHE_FLAT,
 };

 static const struct tegra_ahub_soc_data soc_data_tegra210 = {
--- a/sound/soc/ti/davinci-evm.c
+++ b/sound/soc/ti/davinci-evm.c
@@ -194,27 +194,32 @@ static int davinci_evm_probe(struct platform_device *pdev)
 		return -EINVAL;

 	dai->cpus->of_node = of_parse_phandle(np, "ti,mcasp-controller", 0);
-	if (!dai->cpus->of_node)
-		return -EINVAL;
+	if (!dai->cpus->of_node) {
+		ret = -EINVAL;
+		goto err_put;
+	}

 	dai->platforms->of_node = dai->cpus->of_node;

 	evm_soc_card.dev = &pdev->dev;
 	ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model");
 	if (ret)
-		return ret;
+		goto err_put;

 	mclk = devm_clk_get(&pdev->dev, "mclk");
 	if (PTR_ERR(mclk) == -EPROBE_DEFER) {
-		return -EPROBE_DEFER;
+		ret = -EPROBE_DEFER;
+		goto err_put;
 	} else if (IS_ERR(mclk)) {
 		dev_dbg(&pdev->dev, "mclk not found.\n");
 		mclk = NULL;
 	}

 	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
-	if (!drvdata)
-		return -ENOMEM;
+	if (!drvdata) {
+		ret = -ENOMEM;
+		goto err_put;
+	}

 	drvdata->mclk = mclk;

@@ -224,7 +229,8 @@ static int davinci_evm_probe(struct platform_device *pdev)
 		if (!drvdata->mclk) {
 			dev_err(&pdev->dev,
 				"No clock or clock rate defined.\n");
-			return -EINVAL;
+			ret = -EINVAL;
+			goto err_put;
 		}
 		drvdata->sysclk = clk_get_rate(drvdata->mclk);
 	} else if (drvdata->mclk) {
@@ -240,8 +246,25 @@ static int davinci_evm_probe(struct platform_device *pdev)
 	snd_soc_card_set_drvdata(&evm_soc_card, drvdata);
 	ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card);

-	if (ret)
+	if (ret) {
 		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		goto err_put;
+	}
+
+	return ret;
+
+err_put:
+	dai->platforms->of_node = NULL;
+
+	if (dai->cpus->of_node) {
+		of_node_put(dai->cpus->of_node);
+		dai->cpus->of_node = NULL;
+	}
+
+	if (dai->codecs->of_node) {
+		of_node_put(dai->codecs->of_node);
+		dai->codecs->of_node = NULL;
+	}

 	return ret;
 }
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -1553,7 +1553,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs,

 		for (i = 0; i < ctx->packets; i++) {
 			counts = snd_usb_endpoint_next_packet_size(ep, ctx, i, avail);
-			if (counts < 0)
+			if (counts < 0 || frames + counts >= ep->max_urb_frames)
 				break;
 			/* set up descriptor */
 			urb->iso_frame_desc[i].offset = frames * stride;