Merge tag 'sound-5.15-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This became a slightly large collection of changes, partly because
  I've been off in the last weeks. Most of changes are small and
  scattered while a bit big change is found in HD-audio Realtek codec
  driver; it's a very device-specific fix that has been long wanted, so
  I decided to pick up although it's in the middle RC.

  Some highlights:

   - A new guard ioctl for ALSA rawmidi API to avoid the misuse of the
     new timestamp framing mode; it's for a regression fix

   - HD-audio: a revert of the 5.15 change that might work badly, new
     quirks for Lenovo Legion & co, a follow-up fix for CS8409

   - ASoC: lots of SOF-related fixes, fsl component fixes, corrections
     of mediatek drivers

   - USB-audio: fix for the PM resume

   - FireWire: oxfw and motu fixes"

* tag 'sound-5.15-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (25 commits)
  ALSA: pcsp: Make hrtimer forwarding more robust
  ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
  ALSA: firewire-motu: fix truncated bytes in message tracepoints
  ASoC: SOF: trace: Omit error print when waking up trace sleepers
  ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX
  ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication
  ASoC: SOF: loader: release_firmware() on load failure to avoid batching
  ALSA: hda/cs8409: Setup Dolphin Headset Mic as Phantom Jack
  ALSA: pcxhr: "fix" PCXHR_REG_TO_PORT definition
  ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types
  ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types
  ASoC: SOF: Fix DSP oops stack dump output contents
  ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.
  ALSA: usb-audio: Unify mixer resume and reset_resume procedure
  Revert "ALSA: hda: Drop workaround for a hang at shutdown again"
  ALSA: oxfw: fix transmission method for Loud models based on OXFW971
  ASoC: mediatek: common: handle NULL case in suspend/resume function
  ASoC: fsl_xcvr: register platform component before registering cpu dai
  ASoC: fsl_spdif: register platform component before registering cpu dai
  ASoC: fsl_micfil: register platform component before registering cpu dai
  ...

MAINTAINERS

@@ -17883,7 +17883,8 @@ M:	Olivier Moysan <olivier.moysan@foss.st.com>
 M:	Arnaud Pouliquen <arnaud.pouliquen@foss.st.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
 S:	Maintained
-F:	Documentation/devicetree/bindings/iio/adc/st,stm32-*.yaml
+F:	Documentation/devicetree/bindings/iio/adc/st,stm32-dfsdm-adc.yaml
+F:	Documentation/devicetree/bindings/sound/st,stm32-*.yaml
 F:	sound/soc/stm/
 
 STM32 TIMER/LPTIMER DRIVERS

include/sound/rawmidi.h

@@ -98,6 +98,7 @@ struct snd_rawmidi_file {
 	struct snd_rawmidi *rmidi;
 	struct snd_rawmidi_substream *input;
 	struct snd_rawmidi_substream *output;
+	unsigned int user_pversion;	/* supported protocol version */
 };
 
 struct snd_rawmidi_str {

include/uapi/sound/asound.h

@@ -784,6 +784,7 @@ struct snd_rawmidi_status {
 
 #define SNDRV_RAWMIDI_IOCTL_PVERSION	_IOR('W', 0x00, int)
 #define SNDRV_RAWMIDI_IOCTL_INFO	_IOR('W', 0x01, struct snd_rawmidi_info)
+#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int)
 #define SNDRV_RAWMIDI_IOCTL_PARAMS	_IOWR('W', 0x10, struct snd_rawmidi_params)
 #define SNDRV_RAWMIDI_IOCTL_STATUS	_IOWR('W', 0x20, struct snd_rawmidi_status)
 #define SNDRV_RAWMIDI_IOCTL_DROP	_IOW('W', 0x30, int)

sound/core/rawmidi.c

@@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 			return -EINVAL;
 		}
 	}
+	case SNDRV_RAWMIDI_IOCTL_USER_PVERSION:
+		if (get_user(rfile->user_pversion, (unsigned int __user *)arg))
+			return -EFAULT;
+		return 0;
+
 	case SNDRV_RAWMIDI_IOCTL_PARAMS:
 	{
 		struct snd_rawmidi_params params;
 
 		if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params)))
 			return -EFAULT;
+		if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) {
+			params.mode = 0;
+			memset(params.reserved, 0, sizeof(params.reserved));
+		}
 		switch (params.stream) {
 		case SNDRV_RAWMIDI_STREAM_OUTPUT:
 			if (rfile->output == NULL)

sound/drivers/pcsp/pcsp_lib.c

@@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 	if (pointer_update)
 		pcsp_pointer_update(chip);
 
-	hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
+	hrtimer_forward_now(handle, ns_to_ktime(ns));
 
 	return HRTIMER_RESTART;
 }

sound/firewire/motu/amdtp-motu.c

@@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 
 	/* This is just for v2/v3 protocol. */
 	for (i = 0; i < data_blocks; ++i) {
-		*frames = (be32_to_cpu(buffer[1]) << 16) |
-			  (be32_to_cpu(buffer[2]) >> 16);
+		*frames = be32_to_cpu(buffer[1]);
+		*frames <<= 16;
+		*frames |= be32_to_cpu(buffer[2]) >> 16;
+		++frames;
 		buffer += data_block_quadlets;
-		frames++;
 	}
 }
 

sound/firewire/oxfw/oxfw.c

@@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id
 			model = val;
 	}
 
-	/*
-	 * Mackie Onyx Satellite with base station has a quirk to report a wrong
-	 * value in 'dbs' field of CIP header against its format information.
-	 */
-	if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+	if (vendor == VENDOR_LOUD) {
+		// Mackie Onyx Satellite with base station has a quirk to report a wrong
+		// value in 'dbs' field of CIP header against its format information.
 		oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS;
 
+		// OXFW971-based models may transfer events by blocking method.
+		if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD))
+			oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION;
+	}
+
 	return 0;
 }
 

sound/pci/hda/hda_intel.c

@@ -883,10 +883,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev)
 	return azx_get_pos_posbuf(chip, azx_dev);
 }
 
-static void azx_shutdown_chip(struct azx *chip)
+static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset)
 {
 	azx_stop_chip(chip);
-	azx_enter_link_reset(chip);
+	if (!skip_link_reset)
+		azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
 	display_power(chip, false);
 }
@@ -895,6 +896,11 @@ static void azx_shutdown_chip(struct azx *chip)
 static DEFINE_MUTEX(card_list_lock);
 static LIST_HEAD(card_list);
 
+static void azx_shutdown_chip(struct azx *chip)
+{
+	__azx_shutdown_chip(chip, false);
+}
+
 static void azx_add_card_list(struct azx *chip)
 {
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
@@ -2357,7 +2363,7 @@ static void azx_shutdown(struct pci_dev *pci)
 		return;
 	chip = card->private_data;
 	if (chip && chip->running)
-		azx_shutdown_chip(chip);
+		__azx_shutdown_chip(chip, true);
 }
 
 /* PCI IDs */

sound/pci/hda/patch_cs8409.c

@@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac
 		snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true,
 				      SND_JACK_HEADPHONE, NULL);
 
+		snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true,
+				      SND_JACK_MICROPHONE, NULL);
+
 		cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID);
 		cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID);
 		cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID);

sound/pci/hda/patch_realtek.c

@@ -6429,6 +6429,20 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
 	hda_fixup_thinkpad_acpi(codec, fix, action);
 }
 
+/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */
+static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec,
+						  const struct hda_fixup *fix,
+						  int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->gen.suppress_auto_mute = 1;
+		break;
+	}
+}
+
 /* for alc295_fixup_hp_top_speakers */
 #include "hp_x360_helper.c"
 
@@ -6646,6 +6660,10 @@ enum {
 	ALC623_FIXUP_LENOVO_THINKSTATION_P340,
 	ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
 	ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST,
+	ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS,
+	ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+	ALC287_FIXUP_YOGA7_14ITL_SPEAKERS,
+	ALC287_FIXUP_13S_GEN2_SPEAKERS
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8236,6 +8254,113 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
 	},
+	[ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		//.v.verbs = legion_15imhg05_coefs,
+		.v.verbs = (const struct hda_verb[]) {
+			 // set left speaker Legion 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 // set right speaker Legion 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			 {}
+		},
+		.chained = true,
+		.chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+	},
+	[ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc287_fixup_legion_15imhg05_speakers,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
+	[ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			 // set left speaker Yoga 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 // set right speaker Yoga 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x46 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			 {}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
+	[ALC287_FIXUP_13S_GEN2_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8630,6 +8755,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
 	SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
 	SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
+	SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),

sound/pci/pcxhr/pcxhr_core.c

@@ -52,7 +52,7 @@
 #define PCXHR_DSP 2
 
 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN)
-#undef  PCXHR_REG_TO_PORT(x)
+#error  PCXHR_REG_TO_PORT(x)
 #else
 #define PCXHR_REG_TO_PORT(x)	((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP)
 #endif

sound/soc/fsl/fsl_esai.c

@@ -1073,6 +1073,16 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto err_pm_get_sync;
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
+		goto err_pm_get_sync;
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
 					      &fsl_esai_dai, 1);
 	if (ret) {
@@ -1082,12 +1092,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
 	INIT_WORK(&esai_priv->work, fsl_esai_hw_reset);
 
-	ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
-	if (ret) {
-		dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
-		goto err_pm_get_sync;
-	}
-
 	return ret;
 
 err_pm_get_sync:

sound/soc/fsl/fsl_micfil.c

@@ -737,18 +737,23 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	regcache_cache_only(micfil->regmap, true);
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to pcm register\n");
+		return ret;
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component,
 					      &fsl_micfil_dai, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register component %s\n",
 			fsl_micfil_component.name);
-		return ret;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
-	if (ret)
-		dev_err(&pdev->dev, "failed to pcm register\n");
-
 	return ret;
 }
 

sound/soc/fsl/fsl_sai.c

@@ -1152,11 +1152,10 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto err_pm_get_sync;
 
-	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
-					      &sai->cpu_dai_drv, 1);
-	if (ret)
-		goto err_pm_get_sync;
-
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
 	if (sai->soc_data->use_imx_pcm) {
 		ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
 		if (ret)
@@ -1167,6 +1166,11 @@ static int fsl_sai_probe(struct platform_device *pdev)
 			goto err_pm_get_sync;
 	}
 
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+					      &sai->cpu_dai_drv, 1);
+	if (ret)
+		goto err_pm_get_sync;
+
 	return ret;
 
 err_pm_get_sync:

sound/soc/fsl/fsl_spdif.c

@@ -1434,16 +1434,20 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	regcache_cache_only(spdif_priv->regmap, true);
 
-	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
-					      &spdif_priv->cpu_dai_drv, 1);
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
 	if (ret) {
-		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+		dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
 		goto err_pm_disable;
 	}
 
-	ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+					      &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
-		dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		goto err_pm_disable;
 	}
 

sound/soc/fsl/fsl_xcvr.c

@@ -1215,18 +1215,23 @@ static int fsl_xcvr_probe(struct platform_device *pdev)
 	pm_runtime_enable(dev);
 	regcache_cache_only(xcvr->regmap, true);
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
+	if (ret) {
+		dev_err(dev, "failed to pcm register\n");
+		return ret;
+	}
+
 	ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp,
 					      &fsl_xcvr_dai, 1);
 	if (ret) {
 		dev_err(dev, "failed to register component %s\n",
 			fsl_xcvr_comp.name);
-		return ret;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
-	if (ret)
-		dev_err(dev, "failed to pcm register\n");
-
 	return ret;
 }
 

sound/soc/intel/boards/sof_sdw.c

@@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			      cpus + *cpu_id, cpu_dai_num,
 			      codecs, codec_num,
 			      NULL, &sdw_ops);
+		/*
+		 * SoundWire DAILINKs use 'stream' functions and Bank Switch operations
+		 * based on wait_for_completion(), tag them as 'nonatomic'.
+		 */
+		dai_links[*be_index].nonatomic = true;
 
 		ret = set_codec_init_func(card, link, dai_links + (*be_index)++,
 					  playback, group_id);

sound/soc/mediatek/Kconfig

 # SPDX-License-Identifier: GPL-2.0-only
 config SND_SOC_MEDIATEK
 	tristate
+	select REGMAP_MMIO
 
 config SND_SOC_MT2701
 	tristate "ASoC support for Mediatek MT2701 chip"
@@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682
 config SND_SOC_MT8195
 	tristate "ASoC support for Mediatek MT8195 chip"
 	depends on ARCH_MEDIATEK || COMPILE_TEST
+	depends on COMMON_CLK
 	select SND_SOC_MEDIATEK
+	select MFD_SYSCON if SND_SOC_MT6359
 	help
 	  This adds ASoC platform driver support for Mediatek MT8195 chip
 	  that can be used with other codecs.

sound/soc/mediatek/common/mtk-afe-fe-dai.c

@@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component)
 			devm_kcalloc(dev, afe->reg_back_up_list_num,
 				     sizeof(unsigned int), GFP_KERNEL);
 
-	for (i = 0; i < afe->reg_back_up_list_num; i++)
-		regmap_read(regmap, afe->reg_back_up_list[i],
-			    &afe->reg_back_up[i]);
+	if (afe->reg_back_up) {
+		for (i = 0; i < afe->reg_back_up_list_num; i++)
+			regmap_read(regmap, afe->reg_back_up_list[i],
+				    &afe->reg_back_up[i]);
+	}
 
 	afe->suspended = true;
 	afe->runtime_suspend(dev);
@@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component)
 
 	afe->runtime_resume(dev);
 
-	if (!afe->reg_back_up)
+	if (!afe->reg_back_up) {
 		dev_dbg(dev, "%s no reg_backup\n", __func__);
-
-	for (i = 0; i < afe->reg_back_up_list_num; i++)
-		mtk_regmap_write(regmap, afe->reg_back_up_list[i],
-				 afe->reg_back_up[i]);
+	} else {
+		for (i = 0; i < afe->reg_back_up_list_num; i++)
+			mtk_regmap_write(regmap, afe->reg_back_up_list[i],
+					 afe->reg_back_up[i]);
+	}
 
 	afe->suspended = false;
 	return 0;

sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c

@@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
 	return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
 }
 
-static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
-					      struct snd_pcm_hw_params *params)
+static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+				       struct snd_pcm_hw_params *params)
 
 {
 	/* fix BE i2s format to 32bit, clean param mask first */
@@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
 		.no_pcm = 1,
 		.dpcm_playback = 1,
 		.ops = &mt8195_dptx_ops,
-		.be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
+		.be_hw_params_fixup = mt8195_dptx_hw_params_fixup,
 		SND_SOC_DAILINK_REG(DPTX_BE),
 	},
 	[DAI_LINK_ETDM1_IN_BE] = {
@@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
 			SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 		.dpcm_playback = 1,
-		.be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
 		SND_SOC_DAILINK_REG(ETDM3_OUT_BE),
 	},
 	[DAI_LINK_PCM1_BE] = {

sound/soc/sof/core.c

@@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev)
 			dev_warn(dev, "error: %d failed to prepare DSP for device removal",
 				 ret);
 
-		snd_sof_fw_unload(sdev);
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
 		snd_sof_free_trace(sdev);
@@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev)
 		snd_sof_remove(sdev);
 
 	/* release firmware */
-	release_firmware(pdata->fw);
-	pdata->fw = NULL;
+	snd_sof_fw_unload(sdev);
 
 	return 0;
 }

sound/soc/sof/imx/imx8.c

@@ -365,7 +365,14 @@ static int imx8_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-	return type;
+	/* Only IRAM and SRAM bars are valid */
+	switch (type) {
+	case SOF_FW_BLK_TYPE_IRAM:
+	case SOF_FW_BLK_TYPE_SRAM:
+		return type;
+	default:
+		return -EINVAL;
+	}
 }
 
 static void imx8_ipc_msg_data(struct snd_sof_dev *sdev,

sound/soc/sof/imx/imx8m.c

@@ -228,7 +228,14 @@ static int imx8m_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-	return type;
+	/* Only IRAM and SRAM bars are valid */
+	switch (type) {
+	case SOF_FW_BLK_TYPE_IRAM:
+	case SOF_FW_BLK_TYPE_SRAM:
+		return type;
+	default:
+		return -EINVAL;
+	}
 }
 
 static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev,

sound/soc/sof/loader.c

@@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev)
 	ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev);
 
 	if (ret < 0) {
-		dev_err(sdev->dev, "error: request firmware %s failed err: %d\n",
-			fw_filename, ret);
 		dev_err(sdev->dev,
-			"you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n");
+			"error: sof firmware file is missing, you might need to\n");
+		dev_err(sdev->dev,
+			"       download it from https://github.com/thesofproject/sof-bin/\n");
 		goto err;
 	} else {
 		dev_dbg(sdev->dev, "request_firmware %s successful\n",
@@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware);
 void snd_sof_fw_unload(struct snd_sof_dev *sdev)
 {
 	/* TODO: support module unloading at runtime */
+	release_firmware(sdev->pdata->fw);
+	sdev->pdata->fw = NULL;
 }
 EXPORT_SYMBOL(snd_sof_fw_unload);

sound/soc/sof/trace.c

@@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev)
 		return;
 
 	if (sdev->dtrace_is_enabled) {
-		dev_err(sdev->dev, "error: waking up any trace sleepers\n");
 		sdev->dtrace_error = true;
 		wake_up(&sdev->trace_sleep);
 	}

sound/soc/sof/xtensa/core.c

@@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack,
 	 * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63
 	 */
 	for (i = 0; i < stack_words; i += 4) {
-		hex_dump_to_buffer(stack + i * 4, 16, 16, 4,
+		hex_dump_to_buffer(stack + i, 16, 16, 4,
 				   buf, sizeof(buf), false);
-		dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf);
+		dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf);
 	}
 }
 

sound/usb/card.c

@@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 	return 0;
 }
 
-static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
+static int usb_audio_resume(struct usb_interface *intf)
 {
 	struct snd_usb_audio *chip = usb_get_intfdata(intf);
 	struct snd_usb_stream *as;
@@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 	 * we just notify and restart the mixers
 	 */
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
-		err = snd_usb_mixer_resume(mixer, reset_resume);
+		err = snd_usb_mixer_resume(mixer);
 		if (err < 0)
 			goto err_out;
 	}
@@ -1100,20 +1100,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 	atomic_dec(&chip->active); /* allow autopm after this point */
 	return err;
 }
-
-static int usb_audio_resume(struct usb_interface *intf)
-{
-	return __usb_audio_resume(intf, false);
-}
-
-static int usb_audio_reset_resume(struct usb_interface *intf)
-{
-	return __usb_audio_resume(intf, true);
-}
 #else
 #define usb_audio_suspend	NULL
 #define usb_audio_resume	NULL
-#define usb_audio_reset_resume	NULL
+#define usb_audio_resume	NULL
 #endif		/* CONFIG_PM */
 
 static const struct usb_device_id usb_audio_ids [] = {
@@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = {
 	.disconnect =	usb_audio_disconnect,
 	.suspend =	usb_audio_suspend,
 	.resume =	usb_audio_resume,
-	.reset_resume =	usb_audio_reset_resume,
+	.reset_resume =	usb_audio_resume,
 	.id_table =	usb_audio_ids,
 	.supports_autosuspend = 1,
 };

sound/usb/mixer.c

@@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
 	return 0;
 }
 
-static int default_mixer_reset_resume(struct usb_mixer_elem_list *list)
-{
-	int err;
-
-	if (list->resume) {
-		err = list->resume(list);
-		if (err < 0)
-			return err;
-	}
-	return restore_mixer_value(list);
-}
-
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer)
 {
 	struct usb_mixer_elem_list *list;
-	usb_mixer_elem_resume_func_t f;
 	int id, err;
 
 	/* restore cached mixer values */
 	for (id = 0; id < MAX_ID_ELEMS; id++) {
 		for_each_mixer_elem(list, mixer, id) {
-			if (reset_resume)
-				f = list->reset_resume;
-			else
-				f = list->resume;
-			if (f) {
-				err = f(list);
+			if (list->resume) {
+				err = list->resume(list);
 				if (err < 0)
 					return err;
 			}
@@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 	list->id = unitid;
 	list->dump = snd_usb_mixer_dump_cval;
 #ifdef CONFIG_PM
-	list->resume = NULL;
-	list->reset_resume = default_mixer_reset_resume;
+	list->resume = restore_mixer_value;
 #endif
 }

sound/usb/mixer.h

@@ -70,7 +70,6 @@ struct usb_mixer_elem_list {
 	bool is_std_info;
 	usb_mixer_elem_dump_func_t dump;
 	usb_mixer_elem_resume_func_t resume;
-	usb_mixer_elem_resume_func_t reset_resume;
 };
 
 /* iterate over mixer element list of the given unit id */
@@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 
 #ifdef CONFIG_PM
 int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer);
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume);
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer);
 #endif
 
 int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,

sound/usb/mixer_quirks.c

@@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
 		*listp = list;
 	list->mixer = mixer;
 	list->id = id;
-	list->reset_resume = resume;
+	list->resume = resume;
 	kctl = snd_ctl_new1(knew, list);
 	if (!kctl) {
 		kfree(list);