Merge v7.2.6linux-rolling-stable

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

20 files changed, 131 insertions, 100 deletions

diff --git a/sound/soc/amd/acp-config.c b/sound/soc/amd/acp-config.c
index 88e4230d66c3..8e1563824a97 100644
--- a/sound/soc/amd/acp-config.c
+++ b/sound/soc/amd/acp-config.c
@@ -54,7 +54,7 @@ static const struct dmi_system_id acp70_acpi_flag_override_table[] = {
 	{
 		.matches = {
 			DMI_MATCH(DMI_BOARD_VENDOR, "ASUSTeK COMPUTER INC"),
-			DMI_MATCH(DMI_PRODUCT_NAME, "TUF Gaming A14 FA401EA"),
+			DMI_MATCH(DMI_PRODUCT_NAME, "FA401EA"),
 		},
 	},
 	{
diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c
index 492165c0e1ea..ebe116f32661 100644
--- a/sound/soc/apple/mca.c
+++ b/sound/soc/apple/mca.c
@@ -210,10 +210,10 @@ static void mca_fe_early_trigger(struct snd_pcm_substream *substream, int cmd,
 			   SERDES_STATUS_EN | SERDES_STATUS_RST,
 			   SERDES_STATUS_RST);
 		/*
-		 * Experiments suggest that it takes at most ~1 us
-		 * for the bit to clear, so wait 2 us for good measure.
+		 * The SERDES cluster needs a bit of time to reset itself
+		 * and settle before we start poking it. This is... slow...
 		 */
-		udelay(2);
+		udelay(25);
 		WARN_ON(readl_relaxed(cl->base + serdes_unit + REG_SERDES_STATUS) &
 			SERDES_STATUS_RST);
 		mca_modify(cl, serdes_conf, SERDES_CONF_SYNC_SEL,
diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c
index a451d5d1f8ab..bb449f08e30c 100644
--- a/sound/soc/codecs/rt700-sdw.c
+++ b/sound/soc/codecs/rt700-sdw.c
@@ -458,10 +458,8 @@ static void rt700_sdw_remove(struct sdw_slave *slave)
 {
 	struct rt700_priv *rt700 = dev_get_drvdata(&slave->dev);
 
-	if (rt700->hw_init) {
-		cancel_delayed_work_sync(&rt700->jack_detect_work);
-		cancel_delayed_work_sync(&rt700->jack_btn_check_work);
-	}
+	cancel_delayed_work_sync(&rt700->jack_detect_work);
+	cancel_delayed_work_sync(&rt700->jack_btn_check_work);
 
 	pm_runtime_disable(&slave->dev);
 }
diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c
index e5d27266370e..4fbbf2ca5bf9 100644
--- a/sound/soc/codecs/tas2783-sdw.c
+++ b/sound/soc/codecs/tas2783-sdw.c
@@ -1098,7 +1098,13 @@ static s32 tas2783_sdca_dev_resume(struct device *dev)
 	}
 
 	regcache_cache_only(tas_dev->regmap, false);
-	regcache_sync(tas_dev->regmap);
+	ret = regcache_sync(tas_dev->regmap);
+	if (ret) {
+		regcache_cache_only(tas_dev->regmap, true);
+		regcache_mark_dirty(tas_dev->regmap);
+		return ret;
+	}
+
 	return 0;
 }
 
@@ -1224,9 +1230,23 @@ static s32 tas_update_status(struct sdw_slave *slave,
 	if (tas_dev->hw_init || tas_dev->status != SDW_SLAVE_ATTACHED)
 		return 0;
 
-	/* updated the cache data to device */
 	regcache_cache_only(tas_dev->regmap, false);
-	regcache_sync(tas_dev->regmap);
+
+	/*
+	 * The device is attaching uninitialized: either this is the first
+	 * attach, or it lost power (and with it all register and DSP state)
+	 * while the controller was power-gated during system suspend. The
+	 * cache still holds the pre-suspend values, and tas_io_init() below
+	 * resets the device via TAS2783_SW_RESET anyway, so syncing it back
+	 * is both useless and harmful: later read-modify-write updates would
+	 * compare against stale data and skip the hardware write.
+	 *
+	 * Drop the cache instead, so that subsequent accesses see the real
+	 * hardware state. Syncing after the reset is not an option either:
+	 * the cache accepts registers for which tas2783_sdca_mbq_size()
+	 * returns 0, and writing those back fails with -EINVAL.
+	 */
+	regcache_drop_region(tas_dev->regmap, 0, UINT_MAX);
 
 	/* perform I/O transfers required for Slave initialization */
 	return tas_io_init(&slave->dev, slave);
diff --git a/sound/soc/codecs/tas675x.c b/sound/soc/codecs/tas675x.c
index 82526362de7b..404706b62156 100644
--- a/sound/soc/codecs/tas675x.c
+++ b/sound/soc/codecs/tas675x.c
@@ -1924,15 +1924,6 @@ static const struct reg_default tas675x_reg_defaults[] = {
 	{ TAS675X_PWM_PHASE_M_CTRL_CH2_REG, 0x00 },
 	{ TAS675X_PWM_PHASE_M_CTRL_CH3_REG, 0x00 },
 	{ TAS675X_PWM_PHASE_M_CTRL_CH4_REG, 0x00 },
-	{ TAS675X_DC_LDG_CTRL_REG,         0x00 },
-	{ TAS675X_DC_LDG_LO_CTRL_REG,      0x00 },
-	{ TAS675X_DC_LDG_TIME_CTRL_REG,    0x00 },
-	{ TAS675X_DC_LDG_SL_CH1_CH2_CTRL_REG, 0x11 },
-	{ TAS675X_DC_LDG_SL_CH3_CH4_CTRL_REG, 0x11 },
-	{ TAS675X_AC_LDG_CTRL_REG,         0x10 },
-	{ TAS675X_TWEETER_DETECT_CTRL_REG, 0x08 },
-	{ TAS675X_TWEETER_DETECT_THRESH_REG, 0x00 },
-	{ TAS675X_AC_LDG_FREQ_CTRL_REG,    0xC8 },
 	{ TAS675X_REPORT_ROUTING_1_REG,    0x00 },
 	{ TAS675X_OTSD_RECOVERY_EN_REG,    0x00 },
 	{ TAS675X_REPORT_ROUTING_2_REG,    0xA2 },
@@ -1943,6 +1934,15 @@ static const struct reg_default tas675x_reg_defaults[] = {
 	{ TAS675X_GPIO1_OUTPUT_SEL_REG,    0x00 },
 	{ TAS675X_GPIO2_OUTPUT_SEL_REG,    0x00 },
 	{ TAS675X_GPIO_CTRL_REG,           TAS675X_GPIO_CTRL_RSTVAL },
+	{ TAS675X_DC_LDG_CTRL_REG,         0x00 },
+	{ TAS675X_DC_LDG_LO_CTRL_REG,      0x00 },
+	{ TAS675X_DC_LDG_TIME_CTRL_REG,    0x00 },
+	{ TAS675X_DC_LDG_SL_CH1_CH2_CTRL_REG, 0x11 },
+	{ TAS675X_DC_LDG_SL_CH3_CH4_CTRL_REG, 0x11 },
+	{ TAS675X_AC_LDG_CTRL_REG,         0x10 },
+	{ TAS675X_TWEETER_DETECT_CTRL_REG, 0x08 },
+	{ TAS675X_TWEETER_DETECT_THRESH_REG, 0x00 },
+	{ TAS675X_AC_LDG_FREQ_CTRL_REG,    0xC8 },
 	{ TAS675X_OTW_CTRL_CH1_CH2_REG,    0x11 },
 	{ TAS675X_OTW_CTRL_CH3_CH4_REG,    0x11 },
 };
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 8ae59c094878..04940879dfd8 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -244,7 +244,7 @@ config SND_SOC_MPC5200_AC97
 config SND_MPC52xx_SOC_PCM030
 	tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712"
 	depends on PPC_MPC5200_SIMPLE
-	select SND_SOC_MPC5200_AC97
+	depends on SND_SOC_MPC5200_AC97
 	select SND_SOC_WM9712
 	help
 	  Say Y if you want to add support for sound on the Phytec pcm030
@@ -253,7 +253,7 @@ config SND_MPC52xx_SOC_PCM030
 config SND_MPC52xx_SOC_EFIKA
 	tristate "SoC AC97 Audio support for bbplan Efika and STAC9766"
 	depends on PPC_EFIKA
-	select SND_SOC_MPC5200_AC97
+	depends on SND_SOC_MPC5200_AC97
 	select SND_SOC_STAC9766
 	help
 	  Say Y if you want to add support for sound on the Efika.
diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c
index 4d3266d9cbb7..e9c728bae8b9 100644
--- a/sound/soc/fsl/efika-audio-fabric.c
+++ b/sound/soc/fsl/efika-audio-fabric.c
@@ -22,8 +22,6 @@
 #include <sound/initval.h>
 #include <sound/soc.h>
 
-#include "mpc5200_dma.h"
-
 #define DRV_NAME "efika-audio-fabric"
 
 SND_SOC_DAILINK_DEFS(analog,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 709543308fe9..eff46666c8ad 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -728,8 +728,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 
 	cpu_pdev = of_find_device_by_node(cpu_np);
 	if (!cpu_pdev) {
-		dev_err(&pdev->dev, "failed to find CPU DAI device\n");
-		ret = -EINVAL;
+		ret = dev_err_probe(&pdev->dev, -EPROBE_DEFER,
+				    "failed to find CPU DAI device\n");
 		goto fail;
 	}
 
diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c
index f819f33ec46b..2885cc10b02d 100644
--- a/sound/soc/fsl/fsl_audmix.c
+++ b/sound/soc/fsl/fsl_audmix.c
@@ -457,6 +457,9 @@ static const struct of_device_id fsl_audmix_ids[] = {
 };
 MODULE_DEVICE_TABLE(of, fsl_audmix_ids);
 
+static int fsl_audmix_runtime_resume(struct device *dev);
+static int fsl_audmix_runtime_suspend(struct device *dev);
+
 static int fsl_audmix_probe(struct platform_device *pdev)
 {
 	struct device *dev = &pdev->dev;
@@ -488,13 +491,25 @@ static int fsl_audmix_probe(struct platform_device *pdev)
 	spin_lock_init(&priv->lock);
 	platform_set_drvdata(pdev, priv);
 	pm_runtime_enable(dev);
+	if (!pm_runtime_enabled(dev)) {
+		ret = fsl_audmix_runtime_resume(dev);
+		if (ret)
+			goto err_disable_pm;
+	}
+
+	ret = pm_runtime_resume_and_get(dev);
+	if (ret < 0)
+		goto err_pm_get_sync;
+
+	/* To enable regmap cache only when runtime PM enabled */
+	pm_runtime_put(dev);
 
 	ret = devm_snd_soc_register_component(dev, &fsl_audmix_component,
 					      fsl_audmix_dai,
 					      ARRAY_SIZE(fsl_audmix_dai));
 	if (ret) {
 		dev_err(dev, "failed to register ASoC DAI\n");
-		goto err_disable_pm;
+		goto err_pm_get_sync;
 	}
 
 	/*
@@ -506,12 +521,15 @@ static int fsl_audmix_probe(struct platform_device *pdev)
 		if (IS_ERR(priv->pdev)) {
 			ret = PTR_ERR(priv->pdev);
 			dev_err(dev, "failed to register platform: %d\n", ret);
-			goto err_disable_pm;
+			goto err_pm_get_sync;
 		}
 	}
 
 	return 0;
 
+err_pm_get_sync:
+	if (!pm_runtime_status_suspended(dev))
+		fsl_audmix_runtime_suspend(dev);
 err_disable_pm:
 	pm_runtime_disable(dev);
 	return ret;
@@ -522,6 +540,8 @@ static void fsl_audmix_remove(struct platform_device *pdev)
 	struct fsl_audmix *priv = dev_get_drvdata(&pdev->dev);
 
 	pm_runtime_disable(&pdev->dev);
+	if (!pm_runtime_status_suspended(&pdev->dev))
+		fsl_audmix_runtime_suspend(&pdev->dev);
 
 	if (priv->pdev)
 		platform_device_unregister(priv->pdev);
diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c
index 5542c4ee6d12..a7a5dba83a2f 100644
--- a/sound/soc/fsl/pcm030-audio-fabric.c
+++ b/sound/soc/fsl/pcm030-audio-fabric.c
@@ -13,8 +13,6 @@
 
 #include <sound/soc.h>
 
-#include "mpc5200_dma.h"
-
 #define DRV_NAME "pcm030-audio-fabric"
 
 struct pcm030_audio_data {
diff --git a/sound/soc/meson/meson-card-utils.c b/sound/soc/meson/meson-card-utils.c
index cdb759b466ad..8617a4661a33 100644
--- a/sound/soc/meson/meson-card-utils.c
+++ b/sound/soc/meson/meson-card-utils.c
@@ -50,25 +50,20 @@ int meson_card_reallocate_links(struct snd_soc_card *card,
 			 num_links * sizeof(*priv->card.dai_link),
 			 GFP_KERNEL | __GFP_ZERO);
 	if (!links)
-		goto err_links;
+		return -ENOMEM;
+
+	priv->card.dai_link = links;
+	priv->card.num_links = num_links;
 
 	ldata = krealloc(priv->link_data,
 			 num_links * sizeof(*priv->link_data),
 			 GFP_KERNEL | __GFP_ZERO);
+	/* meson_card_clean_references() will free the links on this error path */
 	if (!ldata)
-		goto err_ldata;
+		return -ENOMEM;
 
-	priv->card.dai_link = links;
 	priv->link_data = ldata;
-	priv->card.num_links = num_links;
 	return 0;
-
-err_ldata:
-	kfree(links);
-err_links:
-	dev_err(priv->card.dev, "failed to allocate links\n");
-	return -ENOMEM;
-
 }
 EXPORT_SYMBOL_GPL(meson_card_reallocate_links);
 
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index f8054c1c59fa..3a0abcb0bfbd 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -769,13 +769,10 @@ static int pxa_ssp_probe(struct snd_soc_dai *dai)
 			goto err_priv;
 		}
 
-		priv->extclk = devm_clk_get(dev, "extclk");
+		priv->extclk = devm_clk_get_optional(dev, "extclk");
 		if (IS_ERR(priv->extclk)) {
 			ret = PTR_ERR(priv->extclk);
-			if (ret == -EPROBE_DEFER)
-				goto err_priv;
-
-			priv->extclk = NULL;
+			goto err_priv;
 		}
 	} else {
 		priv->ssp = pxa_ssp_request(dai->id + 1, "SoC audio");
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index 641d6d243229..f167b9dae3fa 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -802,14 +802,17 @@ EXPORT_SYMBOL_GPL(q6apm_graph_prepare);
 int q6apm_graph_start(struct q6apm_graph *graph)
 {
 	struct audioreach_graph *ar_graph = graph->ar_graph;
-	int ret = 0;
+	int ret;
 
-	if (ar_graph->start_count == 0)
+	if (ar_graph->start_count == 0) {
 		ret = audioreach_graph_mgmt_cmd(ar_graph, APM_CMD_GRAPH_START);
+		if (ret)
+			return ret;
+	}
 
 	ar_graph->start_count++;
 
-	return ret;
+	return 0;
 }
 EXPORT_SYMBOL_GPL(q6apm_graph_start);
 
@@ -817,6 +820,9 @@ int q6apm_graph_stop(struct q6apm_graph *graph)
 {
 	struct audioreach_graph *ar_graph = graph->ar_graph;
 
+	if (ar_graph->start_count == 0)
+		return 0;
+
 	if (--ar_graph->start_count > 0)
 		return 0;
 
diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c
index f80f697a5d55..f80e8d498156 100644
--- a/sound/soc/samsung/i2s.c
+++ b/sound/soc/samsung/i2s.c
@@ -8,6 +8,7 @@
 #include <dt-bindings/sound/samsung-i2s.h>
 #include <linux/delay.h>
 #include <linux/slab.h>
+#include <linux/cleanup.h>
 #include <linux/clk.h>
 #include <linux/clk-provider.h>
 #include <linux/io.h>
@@ -512,7 +513,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 	u32 mod, mask, val = 0;
 	int ret = 0;
 
-	pm_runtime_get_sync(dai->dev);
+	guard(pm_runtime_active)(dai->dev);
 
 	scoped_guard(spinlock_irqsave, &priv->lock)
 		mod = readl(priv->addr + I2SMOD);
@@ -537,8 +538,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 					&& (mod & cdcon_mask))))) {
 			dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
-			ret = -EAGAIN;
-			goto err;
+			return -EAGAIN;
 		}
 
 		if (dir == SND_SOC_CLOCK_IN)
@@ -566,7 +566,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 				} else {
 					priv->rclk_srcrate =
 						clk_get_rate(priv->op_clk);
-					goto done;
+					return 0;
 				}
 			}
 
@@ -580,14 +580,14 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 			if (WARN_ON(IS_ERR(priv->op_clk))) {
 				ret = PTR_ERR(priv->op_clk);
 				priv->op_clk = NULL;
-				goto err;
+				return ret;
 			}
 
 			ret = clk_prepare_enable(priv->op_clk);
 			if (ret) {
 				clk_put(priv->op_clk);
 				priv->op_clk = NULL;
-				goto err;
+				return ret;
 			}
 			priv->rclk_srcrate = clk_get_rate(priv->op_clk);
 
@@ -595,11 +595,10 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 				|| (clk_id && !(mod & rsrc_mask))) {
 			dev_err(&i2s->pdev->dev,
 				"%s:%d Other DAI busy\n", __func__, __LINE__);
-			ret = -EAGAIN;
-			goto err;
+			return -EAGAIN;
 		} else {
 			/* Call can't be on the active DAI */
-			goto done;
+			return 0;
 		}
 
 		if (clk_id == 1)
@@ -607,8 +606,7 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 		break;
 	default:
 		dev_err(&i2s->pdev->dev, "We don't serve that!\n");
-		ret = -EINVAL;
-		goto err;
+		return -EINVAL;
 	}
 
 	scoped_guard(spinlock_irqsave, &priv->lock) {
@@ -616,13 +614,8 @@ static int i2s_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int rfs,
 		mod = (mod & ~mask) | val;
 		writel(mod, priv->addr + I2SMOD);
 	}
-done:
-	pm_runtime_put(dai->dev);
 
 	return 0;
-err:
-	pm_runtime_put(dai->dev);
-	return ret;
 }
 
 static int i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 4ad126bd4f70..646f26b7b415 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3608,7 +3608,7 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol,
 	val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l;
 	mask = e->mask << e->shift_l;
 	if (e->shift_l != e->shift_r) {
-		if (item[1] > e->items)
+		if (item[1] >= e->items)
 			return -EINVAL;
 		val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r;
 		mask |= e->mask << e->shift_r;
diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c
index 6fdfb667cce8..8f37fbdd3f7f 100644
--- a/sound/soc/sof/ipc4-topology.c
+++ b/sound/soc/sof/ipc4-topology.c
@@ -1127,6 +1127,7 @@ static int sof_ipc4_widget_setup_comp_src(struct snd_sof_widget *swidget)
 			"Invalid number of formats: input: %d, output: %d\n",
 			src->available_fmt.num_input_formats,
 			src->available_fmt.num_output_formats);
+		ret = -EINVAL;
 		goto err;
 	}
 
@@ -1179,6 +1180,7 @@ static int sof_ipc4_widget_setup_comp_asrc(struct snd_sof_widget *swidget)
 			"Invalid number of formats: input: %d, output: %d\n",
 			asrc->available_fmt.num_input_formats,
 			asrc->available_fmt.num_output_formats);
+		ret = -EINVAL;
 		goto err;
 	}
 
diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c
index 42a2d90bb705..31dd7a66a9cf 100644
--- a/sound/soc/sof/topology.c
+++ b/sound/soc/sof/topology.c
@@ -846,6 +846,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp,
 	struct snd_soc_tplg_mixer_control *mc =
 		container_of(hdr, struct snd_soc_tplg_mixer_control, hdr);
 	int tlv[SOF_TLV_ITEMS];
+	u32 min, max;
 	unsigned int mask;
 	int ret;
 
@@ -853,6 +854,11 @@ static int sof_control_load_volume(struct snd_soc_component *scomp,
 	if (le32_to_cpu(mc->num_channels) > SND_SOC_TPLG_MAX_CHAN)
 		return -EINVAL;
 
+	min = le32_to_cpu(mc->min);
+	max = le32_to_cpu(mc->max);
+	if (min > max || max >= INT_MAX)
+		return -EINVAL;
+
 	/*
 	 * If control has more than 2 channels we need to override the info. This is because even if
 	 * ASoC layer has defined topology's max channel count to SND_SOC_TPLG_MAX_CHAN = 8, the
@@ -863,12 +869,12 @@ static int sof_control_load_volume(struct snd_soc_component *scomp,
 		kc->info = snd_sof_volume_info;
 
 	scontrol->comp_id = sdev->next_comp_id;
-	scontrol->min_volume_step = le32_to_cpu(mc->min);
-	scontrol->max_volume_step = le32_to_cpu(mc->max);
+	scontrol->min_volume_step = min;
+	scontrol->max_volume_step = max;
 	scontrol->num_channels = le32_to_cpu(mc->num_channels);
 
-	scontrol->max = le32_to_cpu(mc->max);
-	if (le32_to_cpu(mc->max) == 1)
+	scontrol->max = max;
+	if (max == 1)
 		goto skip;
 
 	/* extract tlv data */
@@ -878,7 +884,7 @@ static int sof_control_load_volume(struct snd_soc_component *scomp,
 	}
 
 	/* set up volume table */
-	ret = set_up_volume_table(scontrol, tlv, le32_to_cpu(mc->max) + 1);
+	ret = set_up_volume_table(scontrol, tlv, max + 1);
 	if (ret < 0) {
 		dev_err(scomp->dev, "error: setting up volume table\n");
 		return ret;
@@ -911,7 +917,7 @@ skip:
 	return 0;
 
 err:
-	if (le32_to_cpu(mc->max) > 1)
+	if (max > 1)
 		kfree(scontrol->volume_table);
 
 	return ret;
diff --git a/sound/soc/spacemit/k1_i2s.c b/sound/soc/spacemit/k1_i2s.c
index 8871fc15b29c..67c03b8a5e0c 100644
--- a/sound/soc/spacemit/k1_i2s.c
+++ b/sound/soc/spacemit/k1_i2s.c
@@ -52,7 +52,7 @@ struct spacemit_i2s_dev {
 
 	struct clk *sysclk;
 	struct clk *bclk;
-	struct clk *sspa_clk;
+	struct clk *func_clk;
 	struct clk *sysclk_div;
 	struct clk *c_sysclk;
 	struct clk *c_bclk;
@@ -221,7 +221,7 @@ static int spacemit_i2s_hw_params(struct snd_pcm_substream *substream,
 	if (ret)
 		return ret;
 
-	return clk_set_rate(i2s->sspa_clk, bclk_rate);
+	return clk_set_rate(i2s->func_clk, bclk_rate);
 }
 
 static int spacemit_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id,
@@ -347,22 +347,6 @@ static const struct snd_soc_dai_ops spacemit_i2s_dai_ops = {
 
 static struct snd_soc_dai_driver spacemit_i2s_dai = {
 	.ops = &spacemit_i2s_dai_ops,
-	.playback = {
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = SPACEMIT_PCM_RATES,
-		.rate_min = SNDRV_PCM_RATE_8000,
-		.rate_max = SNDRV_PCM_RATE_48000,
-		.formats = SPACEMIT_PCM_FORMATS,
-	},
-	.capture = {
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = SPACEMIT_PCM_RATES,
-		.rate_min = SNDRV_PCM_RATE_8000,
-		.rate_max = SNDRV_PCM_RATE_48000,
-		.formats = SPACEMIT_PCM_FORMATS,
-	},
 	.symmetric_rate = 1,
 };
 
@@ -392,6 +376,8 @@ static int spacemit_i2s_init_dai(struct spacemit_i2s_dev *i2s,
 		dai->playback.channels_min = 1;
 		dai->playback.channels_max = 2;
 		dai->playback.rates = SPACEMIT_PCM_RATES;
+		dai->playback.rate_min = SNDRV_PCM_RATE_8000;
+		dai->playback.rate_max = SNDRV_PCM_RATE_48000;
 		dai->playback.formats = SPACEMIT_PCM_FORMATS;
 
 		i2s->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -404,6 +390,8 @@ static int spacemit_i2s_init_dai(struct spacemit_i2s_dev *i2s,
 		dai->capture.channels_min = 1;
 		dai->capture.channels_max = 2;
 		dai->capture.rates = SPACEMIT_PCM_RATES;
+		dai->capture.rate_min = SNDRV_PCM_RATE_8000;
+		dai->capture.rate_max = SNDRV_PCM_RATE_48000;
 		dai->capture.formats = SPACEMIT_PCM_FORMATS;
 
 		i2s->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES;
@@ -445,14 +433,14 @@ static int spacemit_i2s_probe(struct platform_device *pdev)
 	if (IS_ERR(i2s->bclk))
 		return dev_err_probe(i2s->dev, PTR_ERR(i2s->bclk), "failed to enable bit clock\n");
 
-	clk = devm_clk_get_enabled(i2s->dev, "sspa_bus");
+	clk = devm_clk_get_enabled(i2s->dev, "bus");
 	if (IS_ERR(clk))
-		return dev_err_probe(i2s->dev, PTR_ERR(clk), "failed to enable sspa_bus clock\n");
+		return dev_err_probe(i2s->dev, PTR_ERR(clk), "failed to enable bus clock\n");
 
-	i2s->sspa_clk = devm_clk_get_enabled(i2s->dev, "sspa");
-	if (IS_ERR(i2s->sspa_clk))
-		return dev_err_probe(i2s->dev, PTR_ERR(i2s->sspa_clk),
-				     "failed to enable sspa clock\n");
+	i2s->func_clk = devm_clk_get_enabled(i2s->dev, "func");
+	if (IS_ERR(i2s->func_clk))
+		return dev_err_probe(i2s->dev, PTR_ERR(i2s->func_clk),
+				     "failed to enable func clock\n");
 
 	i2s->sysclk_div = devm_clk_get_optional_enabled(i2s->dev, "sysclk_div");
 	if (IS_ERR(i2s->sysclk_div))
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
index c214ae0d7b95..312298e0b004 100644
--- a/sound/soc/ti/j721e-evm.c
+++ b/sound/soc/ti/j721e-evm.c
@@ -4,6 +4,7 @@
  *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
  */
 
+#include <linux/cleanup.h>
 #include <linux/clk.h>
 #include <linux/module.h>
 #include <linux/of.h>
@@ -263,7 +264,7 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
 	int ret = 0;
 	int i;
 
-	guard(mutex)(&priv->mutex);
+	mutex_lock(&priv->mutex);
 
 	domain->active++;
 
@@ -303,6 +304,7 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream)
 out:
 	if (ret)
 		domain->active--;
+	mutex_unlock(&priv->mutex);
 
 	return ret;
 }
diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c
index f15336197ed3..798f17a6960a 100644
--- a/sound/soc/xilinx/xlnx_formatter_pcm.c
+++ b/sound/soc/xilinx/xlnx_formatter_pcm.c
@@ -384,7 +384,7 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	if (err) {
 		dev_err(component->dev,
 			"Unable to set constraint on period bytes\n");
-		return err;
+		goto error;
 	}
 
 	/* Resize the buffer bytes as divisible by 64 */
@@ -394,7 +394,7 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	if (err) {
 		dev_err(component->dev,
 			"Unable to set constraint on buffer bytes\n");
-		return err;
+		goto error;
 	}
 
 	/* Set periods as integer multiple */
@@ -403,7 +403,7 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	if (err < 0) {
 		dev_err(component->dev,
 			"Unable to set constraint on periods to be integer\n");
-		return err;
+		goto error;
 	}
 
 	/* enable DMA IOC irq */
@@ -412,6 +412,14 @@ static int xlnx_formatter_pcm_open(struct snd_soc_component *component,
 	writel(val, stream_data->mmio + XLNX_AUD_CTRL);
 
 	return 0;
+
+error:
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		adata->play_stream = NULL;
+	else
+		adata->capture_stream = NULL;
+	kfree(stream_data);
+	return err;
 }
 
 static int xlnx_formatter_pcm_close(struct snd_soc_component *component,