ASoC: Merge up fixes

We need some of the AMD fixes as a base for new work.

24 files changed, 142 insertions, 74 deletions

diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c
index 12aa1cef11a1..ed07fa5693d2 100644
--- a/sound/core/pcm_dmaengine.c
+++ b/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
diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c
index 0bc8617e922a..6815e751a819 100644
--- a/sound/soc/amd/acp/acp-i2s.c
+++ b/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;
 }
 
diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c
index 565623afd42e..b0304b813cad 100644
--- a/sound/soc/amd/acp/acp-pci.c
+++ b/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;
 }
 
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
index 6aed1ee443b4..ba314b279919 100644
--- a/sound/soc/atmel/atmel-classd.c
+++ b/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;
diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c
index 569bab3a2a6e..880228f89baf 100644
--- a/sound/soc/codecs/cs35l56-shared.c
+++ b/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),
 };
 
 /*
diff --git a/sound/soc/codecs/cs42l43-jack.c b/sound/soc/codecs/cs42l43-jack.c
index 901b9dbcf585..d9ab003e166b 100644
--- a/sound/soc/codecs/cs42l43-jack.c
+++ b/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;
 }
diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c
index 03b539ba540f..6a4e42e5e35b 100644
--- a/sound/soc/codecs/es8326.c
+++ b/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,
diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c
index b33da2215ade..87354bb1564e 100644
--- a/sound/soc/codecs/rt722-sdca-sdw.c
+++ b/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);
 
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5ddc0c2fe53f..eb67689dcd6e 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/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");
diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c
index 14e94270911c..4fa208d6a032 100644
--- a/sound/soc/fsl/imx-pcm-dma.c
+++ b/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");
diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c
index 35381a835c93..5cda527020c7 100644
--- a/sound/soc/intel/avs/topology.c
+++ b/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;
diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c
index b41a1147f1c3..a64d1989e28a 100644
--- a/sound/soc/intel/boards/bytcr_rt5640.c
+++ b/sound/soc/intel/boards/bytcr_rt5640.c
@@ -613,6 +613,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = {
 	{
 		.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"),
 			DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 140 CESIUM"),
 		},
 		.driver_data = (void *)(BYT_RT5640_IN1_MAP |
diff --git a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c b/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
index 48252fa9e39e..8e0ae3635a35 100644
--- a/sound/soc/intel/common/soc-acpi-intel-mtl-match.c
+++ b/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"
 	}
 };
 
diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
index acaf81fd6c9b..f848e14b091a 100644
--- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c
+++ b/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),
diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c
index df2e4be992d2..9bb08cadeb18 100644
--- a/sound/soc/mxs/mxs-pcm.c
+++ b/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");
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index ba28ec9dff86..9c98a35ad099 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -146,14 +146,17 @@ 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)
@@ -168,8 +171,10 @@ 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;
+		}
 	}
 
 	/**
@@ -188,26 +193,29 @@ 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)
diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c
index eaa8bb016e50..f2eda2ff46c0 100644
--- a/sound/soc/qcom/sdw.c
+++ b/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");
diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c
index 9fa020ef7eab..ee517d7b5b7b 100644
--- a/sound/soc/rockchip/rockchip_i2s_tdm.c
+++ b/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)
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c
index ea3bc9318412..a63e942fdc0b 100644
--- a/sound/soc/soc-generic-dmaengine-pcm.c
+++ b/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[] = {
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index b00ec01361c2..4b166294602f 100644
--- a/sound/soc/soc-topology.c
+++ b/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;
diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c
index ce675c22a5ab..c61d298ea6b3 100644
--- a/sound/soc/sof/intel/hda-dai.c
+++ b/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;
diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c
index 881eec38c2e2..9a52781bf8d8 100644
--- a/sound/soc/sof/sof-audio.c
+++ b/sound/soc/sof/sof-audio.c
@@ -485,7 +485,7 @@ sink_prepare:
 			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;
 				}
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
index 1e760c315521..2b1ed91a736c 100644
--- a/sound/soc/ti/davinci-mcasp.c
+++ b/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;
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
index 639bc83f4263..cf43ac19c4a6 100644
--- a/sound/soc/ti/omap-hdmi.c
+++ b/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);