Merge tag 'sound-6.10-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "This became bigger than usual, as it receives a pile of pending ASoC
  fixes. Most of changes are for device-specific issues while there are
  a few core fixes that are all rather trivial:

   - DMA-engine sync fixes

   - Continued MIDI2 conversion fixes

   - Various ASoC Intel SOF fixes

   - A series of ASoC topology fixes for memory handling

   - AMD ACP fix, curing a recent regression, too

   - Platform / codec-specific fixes for mediatek, atmel, realtek, etc"

* tag 'sound-6.10-rc6' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (40 commits)
  ASoC: rt5645: fix issue of random interrupt from push-button
  ALSA: seq: Fix missing MSB in MIDI2 SPP conversion
  ASoC: amd: yc: Fix non-functional mic on ASUS M5602RA
  ALSA: hda/realtek: fix mute/micmute LEDs don't work for EliteBook 645/665 G11.
  ALSA: hda/realtek: Fix conflicting quirk for PCI SSID 17aa:3820
  ALSA: dmaengine_pcm: terminate dmaengine before synchronize
  ALSA: hda/relatek: Enable Mute LED on HP Laptop 15-gw0xxx
  ALSA: PCM: Allow resume only for suspended streams
  ALSA: seq: Fix missing channel at encoding RPN/NRPN MIDI2 messages
  ASoC: mediatek: mt8195: Add platform entry for ETDM1_OUT_BE dai link
  ASoC: fsl-asoc-card: set priv->pdev before using it
  ASoC: amd: acp: move chip->flag variable assignment
  ASoC: amd: acp: remove i2s configuration check in acp_i2s_probe()
  ASoC: amd: acp: add a null check for chip_pdev structure
  ASoC: Intel: soc-acpi: mtl: fix speaker no sound on Dell SKU 0C64
  ASoC: q6apm-lpass-dai: close graph on prepare errors
  ASoC: cs35l56: Disconnect ASP1 TX sources when ASP1 DAI is hooked up
  ASoC: topology: Fix route memory corruption
  ASoC: rt722-sdca-sdw: add debounce time for type detection
  ASoC: SOF: sof-audio: Skip unprepare for in-use widgets on error rollback
  ...

MAINTAINERS

@@ -18209,6 +18209,7 @@ QCOM AUDIO (ASoC) DRIVERS
 M:	Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
 M:	Banajit Goswami <bgoswami@quicinc.com>
 L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+L:	linux-arm-msm@vger.kernel.org
 S:	Supported
 F:	Documentation/devicetree/bindings/soc/qcom/qcom,apr*
 F:	Documentation/devicetree/bindings/sound/qcom,*

include/sound/dmaengine_pcm.h

@@ -36,6 +36,7 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream
 int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream,
 	struct dma_chan *chan);
 int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream);
+int snd_dmaengine_pcm_sync_stop(struct snd_pcm_substream *substream);
 
 int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
 	dma_filter_fn filter_fn, void *filter_data);

sound/core/pcm_dmaengine.c

@@ -349,6 +349,16 @@ int snd_dmaengine_pcm_open_request_chan(struct snd_pcm_substream *substream,
 }
 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan);
 
+int snd_dmaengine_pcm_sync_stop(struct snd_pcm_substream *substream)
+{
+	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+
+	dmaengine_synchronize(prtd->dma_chan);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_sync_stop);
+
 /**
  * snd_dmaengine_pcm_close - Close a dmaengine based PCM substream
  * @substream: PCM substream
@@ -358,6 +368,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_open_request_chan);
 int snd_dmaengine_pcm_close(struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+	struct dma_tx_state state;
+	enum dma_status status;
+
+	status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state);
+	if (status == DMA_PAUSED)
+		dmaengine_terminate_async(prtd->dma_chan);
 
 	dmaengine_synchronize(prtd->dma_chan);
 	kfree(prtd);
@@ -378,6 +394,12 @@ EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_close);
 int snd_dmaengine_pcm_close_release_chan(struct snd_pcm_substream *substream)
 {
 	struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream);
+	struct dma_tx_state state;
+	enum dma_status status;
+
+	status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state);
+	if (status == DMA_PAUSED)
+		dmaengine_terminate_async(prtd->dma_chan);
 
 	dmaengine_synchronize(prtd->dma_chan);
 	dma_release_channel(prtd->dma_chan);

sound/core/pcm_native.c

@@ -1775,6 +1775,8 @@ static int snd_pcm_pre_resume(struct snd_pcm_substream *substream,
 			      snd_pcm_state_t state)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	if (runtime->state != SNDRV_PCM_STATE_SUSPENDED)
+		return -EBADFD;
 	if (!(runtime->info & SNDRV_PCM_INFO_RESUME))
 		return -ENOSYS;
 	runtime->trigger_master = substream;

sound/core/seq/seq_ump_convert.c

@@ -791,7 +791,8 @@ static int paf_ev_to_ump_midi2(const struct snd_seq_event *event,
 
 /* set up the MIDI2 RPN/NRPN packet data from the parsed info */
 static void fill_rpn(struct snd_seq_ump_midi2_bank *cc,
-		     union snd_ump_midi2_msg *data)
+		     union snd_ump_midi2_msg *data,
+		     unsigned char channel)
 {
 	if (cc->rpn_set) {
 		data->rpn.status = UMP_MSG_STATUS_RPN;
@@ -808,6 +809,7 @@ static void fill_rpn(struct snd_seq_ump_midi2_bank *cc,
 	}
 	data->rpn.data = upscale_14_to_32bit((cc->cc_data_msb << 7) |
 					     cc->cc_data_lsb);
+	data->rpn.channel = channel;
 	cc->cc_data_msb = cc->cc_data_lsb = 0;
 }
 
@@ -855,7 +857,7 @@ static int cc_ev_to_ump_midi2(const struct snd_seq_event *event,
 		cc->cc_data_lsb = val;
 		if (!(cc->rpn_set || cc->nrpn_set))
 			return 0; // skip
-		fill_rpn(cc, data);
+		fill_rpn(cc, data, channel);
 		return 1;
 	}
 
@@ -957,7 +959,7 @@ static int ctrl14_ev_to_ump_midi2(const struct snd_seq_event *event,
 		cc->cc_data_lsb = lsb;
 		if (!(cc->rpn_set || cc->nrpn_set))
 			return 0; // skip
-		fill_rpn(cc, data);
+		fill_rpn(cc, data, channel);
 		return 1;
 	}
 
@@ -1018,7 +1020,7 @@ static int system_2p_ev_to_ump_midi2(const struct snd_seq_event *event,
 				     union snd_ump_midi2_msg *data,
 				     unsigned char status)
 {
-	return system_1p_ev_to_ump_midi1(event, dest_port,
+	return system_2p_ev_to_ump_midi1(event, dest_port,
 					 (union snd_ump_midi1_msg *)data,
 					 status);
 }

sound/pci/hda/patch_realtek.c

@@ -7525,6 +7525,7 @@ enum {
 	ALC287_FIXUP_LENOVO_THKPAD_WH_ALC1318,
 	ALC256_FIXUP_CHROME_BOOK,
 	ALC287_FIXUP_LENOVO_14ARP8_LEGION_IAH7,
+	ALC287_FIXUP_LENOVO_SSID_17AA3820,
 };
 
 /* A special fixup for Lenovo C940 and Yoga Duet 7;
@@ -7596,6 +7597,20 @@ static void alc287_fixup_lenovo_legion_7(struct hda_codec *codec,
 	__snd_hda_apply_fixup(codec, id, action, 0);
 }
 
+/* Yet more conflicting PCI SSID (17aa:3820) on two Lenovo models */
+static void alc287_fixup_lenovo_ssid_17aa3820(struct hda_codec *codec,
+					      const struct hda_fixup *fix,
+					      int action)
+{
+	int id;
+
+	if (codec->core.subsystem_id == 0x17aa3820)
+		id = ALC269_FIXUP_ASPIRE_HEADSET_MIC; /* IdeaPad 330-17IKB 81DM */
+	else /* 0x17aa3802 */
+		id =  ALC287_FIXUP_YOGA7_14ITL_SPEAKERS; /* "Yoga Duet 7 13ITL6 */
+	__snd_hda_apply_fixup(codec, id, action, 0);
+}
+
 static const struct hda_fixup alc269_fixups[] = {
 	[ALC269_FIXUP_GPIO2] = {
 		.type = HDA_FIXUP_FUNC,
@@ -9832,6 +9847,10 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC225_FIXUP_HEADSET_JACK
 	},
+	[ALC287_FIXUP_LENOVO_SSID_17AA3820] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc287_fixup_lenovo_ssid_17aa3820,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -10069,6 +10088,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x103c, 0x87b7, "HP Laptop 14-fq0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED),
+	SND_PCI_QUIRK(0x103c, 0x87d3, "HP Laptop 15-gw0xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2),
 	SND_PCI_QUIRK(0x103c, 0x87e5, "HP ProBook 440 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87e7, "HP ProBook 450 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x87f1, "HP ProBook 630 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED),
@@ -10222,6 +10242,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x8c7c, "HP ProBook 445 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8c7d, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	SND_PCI_QUIRK(0x103c, 0x8c7e, "HP ProBook 465 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8c7f, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8c80, "HP EliteBook 645 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
+	SND_PCI_QUIRK(0x103c, 0x8c81, "HP EliteBook 665 G11", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF),
 	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),
@@ -10530,7 +10553,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7),
 	SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS),
-	SND_PCI_QUIRK(0x17aa, 0x3820, "IdeaPad 330-17IKB 81DM", ALC269_FIXUP_ASPIRE_HEADSET_MIC),
+	SND_PCI_QUIRK(0x17aa, 0x3820, "IdeaPad 330 / Yoga Duet 7", ALC287_FIXUP_LENOVO_SSID_17AA3820),
 	SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
 	SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),

sound/soc/amd/acp/acp-i2s.c

@@ -588,20 +588,12 @@ static int acp_i2s_probe(struct snd_soc_dai *dai)
 {
 	struct device *dev = dai->component->dev;
 	struct acp_dev_data *adata = dev_get_drvdata(dev);
-	struct acp_resource *rsrc = adata->rsrc;
-	unsigned int val;
 
 	if (!adata->acp_base) {
 		dev_err(dev, "I2S base is NULL\n");
 		return -EINVAL;
 	}
 
-	val = readl(adata->acp_base + rsrc->i2s_pin_cfg_offset);
-	if (val != rsrc->i2s_mode) {
-		dev_err(dev, "I2S Mode not supported val %x\n", val);
-		return -EINVAL;
-	}
-
 	return 0;
 }
 

sound/soc/amd/acp/acp-pci.c

@@ -100,6 +100,7 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 		ret = -EINVAL;
 		goto release_regions;
 	}
+	chip->flag = flag;
 	dmic_dev = platform_device_register_data(dev, "dmic-codec", PLATFORM_DEVID_NONE, NULL, 0);
 	if (IS_ERR(dmic_dev)) {
 		dev_err(dev, "failed to create DMIC device\n");
@@ -139,7 +140,6 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id
 		}
 	}
 
-	chip->flag = flag;
 	memset(&pdevinfo, 0, sizeof(pdevinfo));
 
 	pdevinfo.name = chip->name;
@@ -199,10 +199,12 @@ static int __maybe_unused snd_acp_resume(struct device *dev)
 	ret = acp_init(chip);
 	if (ret)
 		dev_err(dev, "ACP init failed\n");
-	child = chip->chip_pdev->dev;
-	adata = dev_get_drvdata(&child);
-	if (adata)
-		acp_enable_interrupts(adata);
+	if (chip->chip_pdev) {
+		child = chip->chip_pdev->dev;
+		adata = dev_get_drvdata(&child);
+		if (adata)
+			acp_enable_interrupts(adata);
+	}
 	return ret;
 }
 

sound/soc/amd/yc/acp6x-mach.c

@@ -283,6 +283,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "M5402RA"),
 		}
 	},
+        {
+		.driver_data = &acp6x_card,
+		.matches = {
+			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC."),
+			DMI_MATCH(DMI_PRODUCT_NAME, "M5602RA"),
+		}
+	},
 	{
 		.driver_data = &acp6x_card,
 		.matches = {

sound/soc/atmel/atmel-classd.c

@@ -473,19 +473,22 @@ static int atmel_classd_asoc_card_init(struct device *dev,
 	if (!dai_link)
 		return -ENOMEM;
 
-	comp = devm_kzalloc(dev, sizeof(*comp), GFP_KERNEL);
+	comp = devm_kzalloc(dev, 2 * sizeof(*comp), GFP_KERNEL);
 	if (!comp)
 		return -ENOMEM;
 
-	dai_link->cpus		= comp;
+	dai_link->cpus		= &comp[0];
 	dai_link->codecs	= &snd_soc_dummy_dlc;
+	dai_link->platforms	= &comp[1];
 
 	dai_link->num_cpus	= 1;
 	dai_link->num_codecs	= 1;
+	dai_link->num_platforms = 1;
 
 	dai_link->name			= "CLASSD";
 	dai_link->stream_name		= "CLASSD PCM";
 	dai_link->cpus->dai_name	= dev_name(dev);
+	dai_link->platforms->name	= dev_name(dev);
 
 	card->dai_link	= dai_link;
 	card->num_links	= 1;

sound/soc/codecs/cs35l56-shared.c

@@ -215,6 +215,10 @@ static const struct reg_sequence cs35l56_asp1_defaults[] = {
 	REG_SEQ0(CS35L56_ASP1_FRAME_CONTROL5,	0x00020100),
 	REG_SEQ0(CS35L56_ASP1_DATA_CONTROL1,	0x00000018),
 	REG_SEQ0(CS35L56_ASP1_DATA_CONTROL5,	0x00000018),
+	REG_SEQ0(CS35L56_ASP1TX1_INPUT,		0x00000000),
+	REG_SEQ0(CS35L56_ASP1TX2_INPUT,		0x00000000),
+	REG_SEQ0(CS35L56_ASP1TX3_INPUT,		0x00000000),
+	REG_SEQ0(CS35L56_ASP1TX4_INPUT,		0x00000000),
 };
 
 /*

sound/soc/codecs/cs42l43-jack.c

@@ -121,7 +121,7 @@ int cs42l43_set_jack(struct snd_soc_component *component,
 		priv->buttons[3] = 735;
 	}
 
-	ret = cs42l43_find_index(priv, "cirrus,detect-us", 1000, &priv->detect_us,
+	ret = cs42l43_find_index(priv, "cirrus,detect-us", 50000, &priv->detect_us,
 				 cs42l43_accdet_us, ARRAY_SIZE(cs42l43_accdet_us));
 	if (ret < 0)
 		goto error;
@@ -433,7 +433,7 @@ irqreturn_t cs42l43_button_press(int irq, void *data)
 
 	// Wait for 2 full cycles of comb filter to ensure good reading
 	queue_delayed_work(system_wq, &priv->button_press_work,
-			   msecs_to_jiffies(10));
+			   msecs_to_jiffies(20));
 
 	return IRQ_HANDLED;
 }

sound/soc/codecs/es8326.c

@@ -857,12 +857,16 @@ static void es8326_jack_detect_handler(struct work_struct *work)
 			 * set auto-check mode, then restart jack_detect_work after 400ms.
 			 * Don't report jack status.
 			 */
-			regmap_write(es8326->regmap, ES8326_INT_SOURCE,
-					(ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
+			regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00);
 			regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01);
+			regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00);
 			es8326_enable_micbias(es8326->component);
 			usleep_range(50000, 70000);
 			regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00);
+			regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10);
+			usleep_range(50000, 70000);
+			regmap_write(es8326->regmap, ES8326_INT_SOURCE,
+					(ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON));
 			regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f);
 			regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08);
 			queue_delayed_work(system_wq, &es8326->jack_detect_work,

sound/soc/codecs/rt5645.c

@@ -81,7 +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},
+	{RT5645_IL_CMD3, 0x6728},
 };
 
 static const struct reg_default rt5645_reg[] = {
@@ -3130,20 +3130,32 @@ static void rt5645_enable_push_button_irq(struct snd_soc_component *component,
 	bool enable)
 {
 	struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component);
+	int ret;
 
 	if (enable) {
 		snd_soc_dapm_force_enable_pin(dapm, "ADC L power");
 		snd_soc_dapm_force_enable_pin(dapm, "ADC R power");
 		snd_soc_dapm_sync(dapm);
 
+		snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2,
+			RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK,
+			RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_RST);
+		usleep_range(10000, 15000);
+		snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2,
+			RT5645_EN_4BTN_IL_MASK | RT5645_RST_4BTN_IL_MASK,
+			RT5645_EN_4BTN_IL_EN | RT5645_RST_4BTN_IL_NORM);
+		msleep(50);
+		ret = snd_soc_component_read(component, RT5645_INT_IRQ_ST);
+		pr_debug("%s read %x = %x\n", __func__, RT5645_INT_IRQ_ST,
+			snd_soc_component_read(component, RT5645_INT_IRQ_ST));
+		snd_soc_component_write(component, RT5645_INT_IRQ_ST, ret);
+		ret = snd_soc_component_read(component, RT5650_4BTN_IL_CMD1);
+		pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1,
+			snd_soc_component_read(component, RT5650_4BTN_IL_CMD1));
+		snd_soc_component_write(component, RT5650_4BTN_IL_CMD1, ret);
 		snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD1, 0x3, 0x3);
 		snd_soc_component_update_bits(component,
 					RT5645_INT_IRQ_ST, 0x8, 0x8);
-		snd_soc_component_update_bits(component,
-					RT5650_4BTN_IL_CMD2, 0x8000, 0x8000);
-		snd_soc_component_read(component, RT5650_4BTN_IL_CMD1);
-		pr_debug("%s read %x = %x\n", __func__, RT5650_4BTN_IL_CMD1,
-			snd_soc_component_read(component, RT5650_4BTN_IL_CMD1));
 	} else {
 		snd_soc_component_update_bits(component, RT5650_4BTN_IL_CMD2, 0x8000, 0x0);
 		snd_soc_component_update_bits(component, RT5645_INT_IRQ_ST, 0x8, 0x0);

sound/soc/codecs/rt5645.h

@@ -2011,6 +2011,12 @@
 #define RT5645_ZCD_HP_DIS			(0x0 << 15)
 #define RT5645_ZCD_HP_EN			(0x1 << 15)
 
+/* Buttons Inline Command Function 2 (0xe0) */
+#define RT5645_EN_4BTN_IL_MASK			(0x1 << 15)
+#define RT5645_EN_4BTN_IL_EN			(0x1 << 15)
+#define RT5645_RST_4BTN_IL_MASK			(0x1 << 14)
+#define RT5645_RST_4BTN_IL_RST			(0x0 << 14)
+#define RT5645_RST_4BTN_IL_NORM			(0x1 << 14)
 
 /* Codec Private Register definition */
 /* DAC ADC Digital Volume (0x00) */

sound/soc/codecs/rt722-sdca-sdw.c

@@ -68,6 +68,7 @@ static bool rt722_sdca_mbq_readable_register(struct device *dev, unsigned int re
 	case 0x200007f:
 	case 0x2000082 ... 0x200008e:
 	case 0x2000090 ... 0x2000094:
+	case 0x3110000:
 	case 0x5300000 ... 0x5300002:
 	case 0x5400002:
 	case 0x5600000 ... 0x5600007:
@@ -125,6 +126,7 @@ static bool rt722_sdca_mbq_volatile_register(struct device *dev, unsigned int re
 	case 0x2000067:
 	case 0x2000084:
 	case 0x2000086:
+	case 0x3110000:
 		return true;
 	default:
 		return false;
@@ -350,7 +352,7 @@ static int rt722_sdca_interrupt_callback(struct sdw_slave *slave,
 
 	if (status->sdca_cascade && !rt722->disable_irq)
 		mod_delayed_work(system_power_efficient_wq,
-			&rt722->jack_detect_work, msecs_to_jiffies(30));
+			&rt722->jack_detect_work, msecs_to_jiffies(280));
 
 	mutex_unlock(&rt722->disable_irq_lock);
 

sound/soc/fsl/fsl-asoc-card.c

@@ -559,6 +559,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	if (!priv)
 		return -ENOMEM;
 
+	priv->pdev = pdev;
+
 	cpu_np = of_parse_phandle(np, "audio-cpu", 0);
 	/* Give a chance to old DT binding */
 	if (!cpu_np)
@@ -787,7 +789,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 	}
 
 	/* Initialize sound card */
-	priv->pdev = pdev;
 	priv->card.dev = &pdev->dev;
 	priv->card.owner = THIS_MODULE;
 	ret = snd_soc_of_parse_card_name(&priv->card, "model");

sound/soc/fsl/imx-pcm-dma.c

@@ -50,4 +50,5 @@ int imx_pcm_dma_init(struct platform_device *pdev)
 }
 EXPORT_SYMBOL_GPL(imx_pcm_dma_init);
 
+MODULE_DESCRIPTION("Freescale i.MX PCM DMA interface");
 MODULE_LICENSE("GPL");

sound/soc/intel/avs/topology.c

@@ -1545,8 +1545,8 @@ static int avs_route_load(struct snd_soc_component *comp, int index,
 {
 	struct snd_soc_acpi_mach *mach = dev_get_platdata(comp->card->dev);
 	size_t len = SNDRV_CTL_ELEM_ID_NAME_MAXLEN;
-	char buf[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
 	int ssp_port, tdm_slot;
+	char *buf;
 
 	/* See parse_link_formatted_string() for dynamic naming when(s). */
 	if (!avs_mach_singular_ssp(mach))
@@ -1557,13 +1557,24 @@ static int avs_route_load(struct snd_soc_component *comp, int index,
 		return 0;
 	tdm_slot = avs_mach_ssp_tdm(mach, ssp_port);
 
+	buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
 	avs_ssp_sprint(buf, len, route->source, ssp_port, tdm_slot);
-	strscpy((char *)route->source, buf, len);
+	route->source = buf;
+
+	buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+	if (!buf)
+		return -ENOMEM;
 	avs_ssp_sprint(buf, len, route->sink, ssp_port, tdm_slot);
-	strscpy((char *)route->sink, buf, len);
+	route->sink = buf;
+
 	if (route->control) {
+		buf = devm_kzalloc(comp->card->dev, len, GFP_KERNEL);
+		if (!buf)
+			return -ENOMEM;
 		avs_ssp_sprint(buf, len, route->control, ssp_port, tdm_slot);
-		strscpy((char *)route->control, buf, len);
+		route->control = buf;
 	}
 
 	return 0;

sound/soc/intel/boards/bytcr_rt5640.c

@@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 					BYT_RT5640_SSP0_AIF1 |
 					BYT_RT5640_MCLK_EN),
 	},
+	{
+		.matches = {
+			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"),
+			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"),
+		},
+		.driver_data = (void *)(BYTCR_INPUT_DEFAULTS |
+					BYT_RT5640_JD_NOT_INV |
+					BYT_RT5640_DIFF_MIC |
+					BYT_RT5640_SSP0_AIF1 |
+					BYT_RT5640_MCLK_EN),
+	},
 	{
 		.matches = {
 			DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"),

sound/soc/intel/common/soc-acpi-intel-mtl-match.c

@@ -293,7 +293,7 @@ static const struct snd_soc_acpi_adr_device rt1318_1_single_adr[] = {
 		.adr = 0x000130025D131801,
 		.num_endpoints = 1,
 		.endpoints = &single_endpoint,
-		.name_prefix = "rt1318"
+		.name_prefix = "rt1318-1"
 	}
 };
 

sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c

@@ -31,7 +31,7 @@ struct mt8183_da7219_max98357_priv {
 
 static struct snd_soc_jack_pin mt8183_da7219_max98357_jack_pins[] = {
 	{
-		.pin	= "Headphone",
+		.pin	= "Headphones",
 		.mask	= SND_JACK_HEADPHONE,
 	},
 	{
@@ -626,7 +626,7 @@ static struct snd_soc_codec_conf mt6358_codec_conf[] = {
 };
 
 static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
-	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headphones"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 	SOC_DAPM_PIN_SWITCH("Speakers"),
 	SOC_DAPM_PIN_SWITCH("Line Out"),
@@ -634,7 +634,7 @@ static const struct snd_kcontrol_new mt8183_da7219_max98357_snd_controls[] = {
 
 static const
 struct snd_soc_dapm_widget mt8183_da7219_max98357_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_HP("Headphones", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Speakers", NULL),
 	SND_SOC_DAPM_SPK("Line Out", NULL),
@@ -680,7 +680,7 @@ static struct snd_soc_codec_conf mt8183_da7219_rt1015_codec_conf[] = {
 };
 
 static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
-	SOC_DAPM_PIN_SWITCH("Headphone"),
+	SOC_DAPM_PIN_SWITCH("Headphones"),
 	SOC_DAPM_PIN_SWITCH("Headset Mic"),
 	SOC_DAPM_PIN_SWITCH("Left Spk"),
 	SOC_DAPM_PIN_SWITCH("Right Spk"),
@@ -689,7 +689,7 @@ static const struct snd_kcontrol_new mt8183_da7219_rt1015_snd_controls[] = {
 
 static const
 struct snd_soc_dapm_widget mt8183_da7219_rt1015_dapm_widgets[] = {
-	SND_SOC_DAPM_HP("Headphone", NULL),
+	SND_SOC_DAPM_HP("Headphones", NULL),
 	SND_SOC_DAPM_MIC("Headset Mic", NULL),
 	SND_SOC_DAPM_SPK("Left Spk", NULL),
 	SND_SOC_DAPM_SPK("Right Spk", NULL),

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

@@ -827,6 +827,7 @@ SND_SOC_DAILINK_DEFS(ETDM2_IN_BE,
 
 SND_SOC_DAILINK_DEFS(ETDM1_OUT_BE,
 		     DAILINK_COMP_ARRAY(COMP_CPU("ETDM1_OUT")),
+		     DAILINK_COMP_ARRAY(COMP_EMPTY()),
 		     DAILINK_COMP_ARRAY(COMP_EMPTY()));
 
 SND_SOC_DAILINK_DEFS(ETDM2_OUT_BE,

sound/soc/mxs/mxs-pcm.c

@@ -43,4 +43,5 @@ int mxs_pcm_platform_register(struct device *dev)
 }
 EXPORT_SYMBOL_GPL(mxs_pcm_platform_register);
 
+MODULE_DESCRIPTION("MXS ASoC PCM driver");
 MODULE_LICENSE("GPL");

sound/soc/qcom/qdsp6/q6apm-lpass-dais.c

@@ -141,14 +141,17 @@ static void q6apm_lpass_dai_shutdown(struct snd_pcm_substream *substream, struct
 	struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
 	int rc;
 
-	if (!dai_data->is_port_started[dai->id])
-		return;
-	rc = q6apm_graph_stop(dai_data->graph[dai->id]);
-	if (rc < 0)
-		dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+	if (dai_data->is_port_started[dai->id]) {
+		rc = q6apm_graph_stop(dai_data->graph[dai->id]);
+		dai_data->is_port_started[dai->id] = false;
+		if (rc < 0)
+			dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+	}
 
-	q6apm_graph_close(dai_data->graph[dai->id]);
-	dai_data->is_port_started[dai->id] = false;
+	if (dai_data->graph[dai->id]) {
+		q6apm_graph_close(dai_data->graph[dai->id]);
+		dai_data->graph[dai->id] = NULL;
+	}
 }
 
 static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
@@ -163,8 +166,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
 		q6apm_graph_stop(dai_data->graph[dai->id]);
 		dai_data->is_port_started[dai->id] = false;
 
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 			q6apm_graph_close(dai_data->graph[dai->id]);
+			dai_data->graph[dai->id] = NULL;
+		}
 	}
 
 	/**
@@ -183,26 +188,29 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
 
 	cfg->direction = substream->stream;
 	rc = q6apm_graph_media_format_pcm(dai_data->graph[dai->id], cfg);
-
 	if (rc) {
 		dev_err(dai->dev, "Failed to set media format %d\n", rc);
-		return rc;
+		goto err;
 	}
 
 	rc = q6apm_graph_prepare(dai_data->graph[dai->id]);
 	if (rc) {
 		dev_err(dai->dev, "Failed to prepare Graph %d\n", rc);
-		return rc;
+		goto err;
 	}
 
 	rc = q6apm_graph_start(dai_data->graph[dai->id]);
 	if (rc < 0) {
 		dev_err(dai->dev, "fail to start APM port %x\n", dai->id);
-		return rc;
+		goto err;
 	}
 	dai_data->is_port_started[dai->id] = true;
 
 	return 0;
+err:
+	q6apm_graph_close(dai_data->graph[dai->id]);
+	dai_data->graph[dai->id] = NULL;
+	return rc;
 }
 
 static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)

sound/soc/qcom/sdw.c

@@ -160,4 +160,5 @@ int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream,
 	return 0;
 }
 EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free);
+MODULE_DESCRIPTION("Qualcomm ASoC SoundWire helper functions");
 MODULE_LICENSE("GPL");

sound/soc/rockchip/rockchip_i2s_tdm.c

@@ -655,8 +655,17 @@ static int rockchip_i2s_tdm_hw_params(struct snd_pcm_substream *substream,
 	int err;
 
 	if (i2s_tdm->is_master_mode) {
-		struct clk *mclk = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
-			i2s_tdm->mclk_tx : i2s_tdm->mclk_rx;
+		struct clk *mclk;
+
+		if (i2s_tdm->clk_trcm == TRCM_TX) {
+			mclk = i2s_tdm->mclk_tx;
+		} else if (i2s_tdm->clk_trcm == TRCM_RX) {
+			mclk = i2s_tdm->mclk_rx;
+		} else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			mclk = i2s_tdm->mclk_tx;
+		} else {
+			mclk = i2s_tdm->mclk_rx;
+		}
 
 		err = clk_set_rate(mclk, DEFAULT_MCLK_FS * params_rate(params));
 		if (err)

sound/soc/soc-generic-dmaengine-pcm.c

@@ -318,6 +318,12 @@ static int dmaengine_copy(struct snd_soc_component *component,
 	return 0;
 }
 
+static int dmaengine_pcm_sync_stop(struct snd_soc_component *component,
+				   struct snd_pcm_substream *substream)
+{
+	return snd_dmaengine_pcm_sync_stop(substream);
+}
+
 static const struct snd_soc_component_driver dmaengine_pcm_component = {
 	.name		= SND_DMAENGINE_PCM_DRV_NAME,
 	.probe_order	= SND_SOC_COMP_ORDER_LATE,
@@ -327,6 +333,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = {
 	.trigger	= dmaengine_pcm_trigger,
 	.pointer	= dmaengine_pcm_pointer,
 	.pcm_construct	= dmaengine_pcm_new,
+	.sync_stop	= dmaengine_pcm_sync_stop,
 };
 
 static const struct snd_soc_component_driver dmaengine_pcm_component_process = {
@@ -339,6 +346,7 @@ static const struct snd_soc_component_driver dmaengine_pcm_component_process = {
 	.pointer	= dmaengine_pcm_pointer,
 	.copy		= dmaengine_copy,
 	.pcm_construct	= dmaengine_pcm_new,
+	.sync_stop	= dmaengine_pcm_sync_stop,
 };
 
 static const char * const dmaengine_pcm_dma_channel_names[] = {

sound/soc/soc-topology.c

@@ -1021,6 +1021,7 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
 	struct snd_soc_dapm_context *dapm = &tplg->comp->dapm;
+	const size_t maxlen = SNDRV_CTL_ELEM_ID_NAME_MAXLEN;
 	struct snd_soc_tplg_dapm_graph_elem *elem;
 	struct snd_soc_dapm_route *route;
 	int count, i;
@@ -1044,31 +1045,27 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		tplg->pos += sizeof(struct snd_soc_tplg_dapm_graph_elem);
 
 		/* validate routes */
-		if (strnlen(elem->source, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
-			    SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
+		if ((strnlen(elem->source, maxlen) == maxlen) ||
+		    (strnlen(elem->sink, maxlen) == maxlen) ||
+		    (strnlen(elem->control, maxlen) == maxlen)) {
 			ret = -EINVAL;
 			break;
 		}
-		if (strnlen(elem->sink, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
-			    SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
-			ret = -EINVAL;
-			break;
-		}
-		if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) ==
-			    SNDRV_CTL_ELEM_ID_NAME_MAXLEN) {
-			ret = -EINVAL;
+
+		route->source = devm_kstrdup(tplg->dev, elem->source, GFP_KERNEL);
+		route->sink = devm_kstrdup(tplg->dev, elem->sink, GFP_KERNEL);
+		if (!route->source || !route->sink) {
+			ret = -ENOMEM;
 			break;
 		}
 
-		route->source = elem->source;
-		route->sink = elem->sink;
-
-		/* set to NULL atm for tplg users */
-		route->connected = NULL;
-		if (strnlen(elem->control, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 0)
-			route->control = NULL;
-		else
-			route->control = elem->control;
+		if (strnlen(elem->control, maxlen) != 0) {
+			route->control = devm_kstrdup(tplg->dev, elem->control, GFP_KERNEL);
+			if (!route->control) {
+				ret = -ENOMEM;
+				break;
+			}
+		}
 
 		/* add route dobj to dobj_list */
 		route->dobj.type = SND_SOC_DOBJ_GRAPH;

sound/soc/sof/intel/hda-dai.c

@@ -379,7 +379,7 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 
 	sdev = widget_to_sdev(w);
 	if (sdev->dspless_mode_selected)
-		goto skip_tlv;
+		return 0;
 
 	/* get stream_id */
 	hext_stream = ops->get_hext_stream(sdev, cpu_dai, substream);
@@ -423,7 +423,6 @@ static int non_hda_dai_hw_params_data(struct snd_pcm_substream *substream,
 	dma_config->dma_stream_channel_map.device_count = 1;
 	dma_config->dma_priv_config_size = 0;
 
-skip_tlv:
 	return 0;
 }
 
@@ -525,6 +524,9 @@ int sdw_hda_dai_hw_params(struct snd_pcm_substream *substream,
 		return ret;
 	}
 
+	if (sdev->dspless_mode_selected)
+		return 0;
+
 	ipc4_copier = widget_to_copier(w);
 	dma_config_tlv = &ipc4_copier->dma_config_tlv[cpu_dai_id];
 	dma_config = &dma_config_tlv->dma_config;

sound/soc/sof/sof-audio.c

@@ -485,7 +485,7 @@ sof_prepare_widgets_in_path(struct snd_sof_dev *sdev, struct snd_soc_dapm_widget
 			if (ret < 0) {
 				/* unprepare the source widget */
 				if (widget_ops[widget->id].ipc_unprepare &&
-				    swidget && swidget->prepared) {
+				    swidget && swidget->prepared && swidget->use_count == 0) {
 					widget_ops[widget->id].ipc_unprepare(swidget);
 					swidget->prepared = false;
 				}

sound/soc/ti/davinci-mcasp.c

@@ -1472,10 +1472,11 @@ static int davinci_mcasp_hw_rule_min_periodsize(
 {
 	struct snd_interval *period_size = hw_param_interval(params,
 						SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+	u8 numevt = *((u8 *)rule->private);
 	struct snd_interval frames;
 
 	snd_interval_any(&frames);
-	frames.min = 64;
+	frames.min = numevt;
 	frames.integer = 1;
 
 	return snd_interval_refine(period_size, &frames);
@@ -1490,6 +1491,7 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 	u32 max_channels = 0;
 	int i, dir, ret;
 	int tdm_slots = mcasp->tdm_slots;
+	u8 *numevt;
 
 	/* Do not allow more then one stream per direction */
 	if (mcasp->substreams[substream->stream])
@@ -1589,9 +1591,12 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
 			return ret;
 	}
 
+	numevt = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
+			 &mcasp->txnumevt :
+			 &mcasp->rxnumevt;
 	snd_pcm_hw_rule_add(substream->runtime, 0,
 			    SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
-			    davinci_mcasp_hw_rule_min_periodsize, NULL,
+			    davinci_mcasp_hw_rule_min_periodsize, numevt,
 			    SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
 
 	return 0;

sound/soc/ti/omap-hdmi.c

@@ -354,11 +354,7 @@ static int omap_hdmi_audio_probe(struct platform_device *pdev)
 	if (!card)
 		return -ENOMEM;
 
-	card->name = devm_kasprintf(dev, GFP_KERNEL,
-				    "HDMI %s", dev_name(ad->dssdev));
-	if (!card->name)
-		return -ENOMEM;
-
+	card->name = "HDMI";
 	card->owner = THIS_MODULE;
 	card->dai_link =
 		devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL);