Merge branch 'asoc-5.7' into asoc-linus

26 files changed, 407 insertions, 252 deletions

diff --git a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
index 7cd0e278ed85..a3ba2186d6a1 100644
--- a/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
+++ b/Documentation/devicetree/bindings/sound/rockchip-i2s.yaml
@@ -56,6 +56,9 @@ properties:
       - const: tx
       - const: rx
 
+  power-domains:
+    maxItems: 1
+
   rockchip,capture-channels:
     allOf:
       - $ref: /schemas/types.yaml#/definitions/uint32
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt b/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
deleted file mode 100644
index ec20c1271e92..000000000000
--- a/Documentation/devicetree/bindings/sound/rockchip-spdif.txt
+++ /dev/null
@@ -1,45 +0,0 @@
-* Rockchip SPDIF transceiver
-
-The S/PDIF audio block is a stereo transceiver that allows the
-processor to receive and transmit digital audio via an coaxial cable or
-a fibre cable.
-
-Required properties:
-
-- compatible: should be one of the following:
-   - "rockchip,rk3066-spdif"
-   - "rockchip,rk3188-spdif"
-   - "rockchip,rk3228-spdif"
-   - "rockchip,rk3288-spdif"
-   - "rockchip,rk3328-spdif"
-   - "rockchip,rk3366-spdif"
-   - "rockchip,rk3368-spdif"
-   - "rockchip,rk3399-spdif"
-- reg: physical base address of the controller and length of memory mapped
-  region.
-- interrupts: should contain the SPDIF interrupt.
-- dmas: DMA specifiers for tx dma. See the DMA client binding,
-  Documentation/devicetree/bindings/dma/dma.txt
-- dma-names: should be "tx"
-- clocks: a list of phandle + clock-specifier pairs, one for each entry
-  in clock-names.
-- clock-names: should contain following:
-   - "hclk": clock for SPDIF controller
-   - "mclk" : clock for SPDIF bus
-
-Required properties on RK3288:
-  - rockchip,grf: the phandle of the syscon node for the general register
-                   file (GRF)
-
-Example for the rk3188 SPDIF controller:
-
-spdif: spdif@1011e000 {
-	compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
-	reg = <0x1011e000 0x2000>;
-	interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
-	dmas = <&dmac1_s 8>;
-	dma-names = "tx";
-	clock-names = "hclk", "mclk";
-	clocks = <&cru HCLK_SPDIF>, <&cru SCLK_SPDIF>;
-	#sound-dai-cells = <0>;
-};
diff --git a/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
new file mode 100644
index 000000000000..c467152656f7
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/rockchip-spdif.yaml
@@ -0,0 +1,101 @@
+# SPDX-License-Identifier: GPL-2.0
+%YAML 1.2
+---
+$id: http://devicetree.org/schemas/sound/rockchip-spdif.yaml#
+$schema: http://devicetree.org/meta-schemas/core.yaml#
+
+title: Rockchip SPDIF transceiver
+
+description:
+  The S/PDIF audio block is a stereo transceiver that allows the
+  processor to receive and transmit digital audio via a coaxial or
+  fibre cable.
+
+maintainers:
+  - Heiko Stuebner <heiko@sntech.de>
+
+properties:
+  compatible:
+    oneOf:
+      - const: rockchip,rk3066-spdif
+      - const: rockchip,rk3228-spdif
+      - const: rockchip,rk3328-spdif
+      - const: rockchip,rk3366-spdif
+      - const: rockchip,rk3368-spdif
+      - const: rockchip,rk3399-spdif
+      - items:
+          - enum:
+            - rockchip,rk3188-spdif
+            - rockchip,rk3288-spdif
+          - const: rockchip,rk3066-spdif
+
+  reg:
+    maxItems: 1
+
+  interrupts:
+    maxItems: 1
+
+  clocks:
+    items:
+      - description: clock for SPDIF bus
+      - description: clock for SPDIF controller
+
+  clock-names:
+    items:
+      - const: mclk
+      - const: hclk
+
+  dmas:
+    maxItems: 1
+
+  dma-names:
+    const: tx
+
+  power-domains:
+    maxItems: 1
+
+  rockchip,grf:
+    $ref: /schemas/types.yaml#/definitions/phandle
+    description:
+      The phandle of the syscon node for the GRF register.
+      Required property on RK3288.
+
+  "#sound-dai-cells":
+    const: 0
+
+required:
+  - compatible
+  - reg
+  - interrupts
+  - clocks
+  - clock-names
+  - dmas
+  - dma-names
+  - "#sound-dai-cells"
+
+if:
+  properties:
+    compatible:
+      contains:
+        const: rockchip,rk3288-spdif
+
+then:
+  required:
+    - rockchip,grf
+
+additionalProperties: false
+
+examples:
+  - |
+    #include <dt-bindings/clock/rk3188-cru.h>
+    #include <dt-bindings/interrupt-controller/arm-gic.h>
+    spdif: spdif@1011e000 {
+      compatible = "rockchip,rk3188-spdif", "rockchip,rk3066-spdif";
+      reg = <0x1011e000 0x2000>;
+      interrupts = <GIC_SPI 32 IRQ_TYPE_LEVEL_HIGH>;
+      clocks = <&cru SCLK_SPDIF>, <&cru HCLK_SPDIF>;
+      clock-names = "mclk", "hclk";
+      dmas = <&dmac1_s 8>;
+      dma-names = "tx";
+      #sound-dai-cells = <0>;
+    };
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index d4825b82c7a3..b33abe93b905 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -351,7 +351,6 @@ struct snd_soc_dai {
 
 	/* bit field */
 	unsigned int probed:1;
-	unsigned int started[SNDRV_PCM_STREAM_LAST + 1];
 };
 
 static inline struct snd_soc_pcm_stream *
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 13458e4fbb13..946f88a6c63d 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -790,6 +790,9 @@ struct snd_soc_dai_link {
 	const struct snd_soc_pcm_stream *params;
 	unsigned int num_params;
 
+	struct snd_soc_dapm_widget *playback_widget;
+	struct snd_soc_dapm_widget *capture_widget;
+
 	unsigned int dai_fmt;           /* format to set on init */
 
 	enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */
diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c
index 024a7ee54cd5..e499c00e0c66 100644
--- a/sound/soc/amd/acp3x-rt5682-max9836.c
+++ b/sound/soc/amd/acp3x-rt5682-max9836.c
@@ -89,9 +89,9 @@ static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd)
 	}
 
 	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN);
-	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOICECOMMAND);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEUP);
+	snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN);
 
 	ret = snd_soc_component_set_jack(component, &pco_jack, NULL);
 	if (ret) {
diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c
index fba9b749839d..f26b77faed59 100644
--- a/sound/soc/codecs/hdac_hdmi.c
+++ b/sound/soc/codecs/hdac_hdmi.c
@@ -142,14 +142,14 @@ static struct hdac_hdmi_pcm *
 hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi,
 			   struct hdac_hdmi_cvt *cvt)
 {
-	struct hdac_hdmi_pcm *pcm = NULL;
+	struct hdac_hdmi_pcm *pcm;
 
 	list_for_each_entry(pcm, &hdmi->pcm_list, head) {
 		if (pcm->cvt == cvt)
-			break;
+			return pcm;
 	}
 
-	return pcm;
+	return NULL;
 }
 
 static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm,
diff --git a/sound/soc/codecs/madera.c b/sound/soc/codecs/madera.c
index 40de9d7811d1..a448d2a2918a 100644
--- a/sound/soc/codecs/madera.c
+++ b/sound/soc/codecs/madera.c
@@ -1903,7 +1903,6 @@ const struct soc_enum madera_isrc_fsh[] = {
 			      MADERA_ISRC4_FSH_SHIFT, 0xf,
 			      MADERA_RATE_ENUM_SIZE,
 			      madera_rate_text, madera_rate_val),
-
 };
 EXPORT_SYMBOL_GPL(madera_isrc_fsh);
 
@@ -1924,7 +1923,6 @@ const struct soc_enum madera_isrc_fsl[] = {
 			      MADERA_ISRC4_FSL_SHIFT, 0xf,
 			      MADERA_RATE_ENUM_SIZE,
 			      madera_rate_text, madera_rate_val),
-
 };
 EXPORT_SYMBOL_GPL(madera_isrc_fsl);
 
@@ -1938,7 +1936,6 @@ const struct soc_enum madera_asrc1_rate[] = {
 			      MADERA_ASYNC_RATE_ENUM_SIZE,
 			      madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE,
 			      madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE),
-
 };
 EXPORT_SYMBOL_GPL(madera_asrc1_rate);
 
@@ -1964,7 +1961,6 @@ const struct soc_enum madera_asrc2_rate[] = {
 			      MADERA_ASYNC_RATE_ENUM_SIZE,
 			      madera_rate_text + MADERA_SYNC_RATE_ENUM_SIZE,
 			      madera_rate_val + MADERA_SYNC_RATE_ENUM_SIZE),
-
 };
 EXPORT_SYMBOL_GPL(madera_asrc2_rate);
 
diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c
index d5130193b4a2..e8a8bf7b4ffe 100644
--- a/sound/soc/codecs/sgtl5000.c
+++ b/sound/soc/codecs/sgtl5000.c
@@ -1653,6 +1653,40 @@ static int sgtl5000_i2c_probe(struct i2c_client *client,
 		dev_err(&client->dev,
 			"Error %d initializing CHIP_CLK_CTRL\n", ret);
 
+	/* Mute everything to avoid pop from the following power-up */
+	ret = regmap_write(sgtl5000->regmap, SGTL5000_CHIP_ANA_CTRL,
+			   SGTL5000_CHIP_ANA_CTRL_DEFAULT);
+	if (ret) {
+		dev_err(&client->dev,
+			"Error %d muting outputs via CHIP_ANA_CTRL\n", ret);
+		goto disable_clk;
+	}
+
+	/*
+	 * If VAG is powered-on (e.g. from previous boot), it would be disabled
+	 * by the write to ANA_POWER in later steps of the probe code. This
+	 * may create a loud pop even with all outputs muted. The proper way
+	 * to circumvent this is disabling the bit first and waiting the proper
+	 * cool-down time.
+	 */
+	ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ANA_POWER, &value);
+	if (ret) {
+		dev_err(&client->dev, "Failed to read ANA_POWER: %d\n", ret);
+		goto disable_clk;
+	}
+	if (value & SGTL5000_VAG_POWERUP) {
+		ret = regmap_update_bits(sgtl5000->regmap,
+					 SGTL5000_CHIP_ANA_POWER,
+					 SGTL5000_VAG_POWERUP,
+					 0);
+		if (ret) {
+			dev_err(&client->dev, "Error %d disabling VAG\n", ret);
+			goto disable_clk;
+		}
+
+		msleep(SGTL5000_VAG_POWERDOWN_DELAY);
+	}
+
 	/* Follow section 2.2.1.1 of AN3663 */
 	ana_pwr = SGTL5000_ANA_POWER_DEFAULT;
 	if (sgtl5000->num_supplies <= VDDD) {
diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h
index a4bf4bca95bf..56ec5863f250 100644
--- a/sound/soc/codecs/sgtl5000.h
+++ b/sound/soc/codecs/sgtl5000.h
@@ -233,6 +233,7 @@
 /*
  * SGTL5000_CHIP_ANA_CTRL
  */
+#define SGTL5000_CHIP_ANA_CTRL_DEFAULT		0x0133
 #define SGTL5000_LINE_OUT_MUTE			0x0100
 #define SGTL5000_HP_SEL_MASK			0x0040
 #define SGTL5000_HP_SEL_SHIFT			6
diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c
index 1554631cb397..5b7f9fcf6cbf 100644
--- a/sound/soc/codecs/tas571x.c
+++ b/sound/soc/codecs/tas571x.c
@@ -820,8 +820,10 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 
 	priv->regmap = devm_regmap_init(dev, NULL, client,
 					priv->chip->regmap_config);
-	if (IS_ERR(priv->regmap))
-		return PTR_ERR(priv->regmap);
+	if (IS_ERR(priv->regmap)) {
+		ret = PTR_ERR(priv->regmap);
+		goto disable_regs;
+	}
 
 	priv->pdn_gpio = devm_gpiod_get_optional(dev, "pdn", GPIOD_OUT_LOW);
 	if (IS_ERR(priv->pdn_gpio)) {
@@ -845,7 +847,7 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 
 	ret = regmap_write(priv->regmap, TAS571X_OSC_TRIM_REG, 0);
 	if (ret)
-		return ret;
+		goto disable_regs;
 
 	usleep_range(50000, 60000);
 
@@ -861,12 +863,20 @@ static int tas571x_i2c_probe(struct i2c_client *client,
 		 */
 		ret = regmap_update_bits(priv->regmap, TAS571X_MVOL_REG, 1, 0);
 		if (ret)
-			return ret;
+			goto disable_regs;
 	}
 
-	return devm_snd_soc_register_component(&client->dev,
+	ret = devm_snd_soc_register_component(&client->dev,
 				      &priv->component_driver,
 				      &tas571x_dai, 1);
+	if (ret)
+		goto disable_regs;
+
+	return ret;
+
+disable_regs:
+	regulator_bulk_disable(priv->chip->num_supply_names, priv->supplies);
+	return ret;
 }
 
 static int tas571x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c
index f2d6f2f81f14..d39d479e2378 100644
--- a/sound/soc/codecs/wsa881x.c
+++ b/sound/soc/codecs/wsa881x.c
@@ -394,6 +394,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* COMP */
 		.num = 2,
@@ -401,6 +402,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* BOOST */
 		.num = 3,
@@ -408,6 +410,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}, {
 		/* VISENSE */
 		.num = 4,
@@ -415,6 +418,7 @@ static struct sdw_dpn_prop wsa_sink_dpn_prop[WSA881X_MAX_SWR_PORTS] = {
 		.min_ch = 1,
 		.max_ch = 1,
 		.simple_ch_prep_sm = true,
+		.read_only_wordlength = true,
 	}
 };
 
diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
index bcedec6c6117..7d85bd5aff9f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c
@@ -113,14 +113,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
 	}
 };
 
-static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
-	{
-		.adr = 0x000210025D130800,
-		.num_endpoints = 1,
-		.endpoints = &single_endpoint,
-	}
-};
-
 static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
 	{
 		.adr = 0x000110025D130800,
diff --git a/sound/soc/intel/common/soc-acpi-intel-icl-match.c b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
index ef8500349f2f..16ec9f382b0f 100644
--- a/sound/soc/intel/common/soc-acpi-intel-icl-match.c
+++ b/sound/soc/intel/common/soc-acpi-intel-icl-match.c
@@ -87,14 +87,6 @@ static const struct snd_soc_acpi_adr_device rt1308_1_adr[] = {
 	}
 };
 
-static const struct snd_soc_acpi_adr_device rt1308_2_adr[] = {
-	{
-		.adr = 0x000210025D130800,
-		.num_endpoints = 1,
-		.endpoints = &single_endpoint,
-	}
-};
-
 static const struct snd_soc_acpi_adr_device rt1308_1_group1_adr[] = {
 	{
 		.adr = 0x000110025D130800,
diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c
index d55e3ad96716..287ad2aa27f3 100644
--- a/sound/soc/qcom/apq8096.c
+++ b/sound/soc/qcom/apq8096.c
@@ -116,10 +116,8 @@ static int apq8096_platform_probe(struct platform_device *pdev)
 	card->dev = dev;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
-	if (ret) {
-		dev_err(dev, "Error parsing OF data\n");
+	if (ret)
 		goto err;
-	}
 
 	apq8096_add_be_ops(card);
 	ret = snd_soc_register_card(card);
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index c1a7624eaf17..2a5302f1db98 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -902,6 +902,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -917,6 +919,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -931,6 +935,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -946,6 +952,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -960,6 +968,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -975,6 +985,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -989,6 +1001,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 			.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_8000 |
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
@@ -1004,6 +1018,8 @@ static struct snd_soc_dai_driver q6afe_dais[] = {
 				 SNDRV_PCM_RATE_16000,
 			.formats = SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE,
+			.channels_min = 1,
+			.channels_max = 8,
 			.rate_min =     8000,
 			.rate_max =     48000,
 		},
diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c
index b2de65c7f95c..68e9388ff46f 100644
--- a/sound/soc/qcom/sdm845.c
+++ b/sound/soc/qcom/sdm845.c
@@ -559,10 +559,8 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev)
 	card->dev = dev;
 	dev_set_drvdata(dev, card);
 	ret = qcom_snd_parse_of(card);
-	if (ret) {
-		dev_err(dev, "Error parsing OF data\n");
+	if (ret)
 		goto parse_dt_fail;
-	}
 
 	data->card = card;
 	snd_soc_card_set_drvdata(card, data);
diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c
index 358887848293..5e95c30fb2ba 100644
--- a/sound/soc/samsung/s3c-i2s-v2.c
+++ b/sound/soc/samsung/s3c-i2s-v2.c
@@ -656,60 +656,6 @@ void s3c_i2sv2_cleanup(struct snd_soc_dai *dai,
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_cleanup);
 
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-	u32 iismod;
-
-	if (dai->active) {
-		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
-		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
-		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
-		/* some basic suspend checks */
-
-		iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			pr_warn("%s: RXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			pr_warn("%s: TXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			pr_warn("%s: IIS active\n", __func__);
-	}
-
-	return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
-	struct s3c_i2sv2_info *i2s = to_info(dai);
-
-	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
-		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
-	if (dai->active) {
-		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
-		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
-		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
-		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
-		       i2s->regs + S3C2412_IISFIC);
-
-		ndelay(250);
-		writel(0x0, i2s->regs + S3C2412_IISFIC);
-	}
-
-	return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume  NULL
-#endif
-
 int s3c_i2sv2_register_component(struct device *dev, int id,
 			   const struct snd_soc_component_driver *cmp_drv,
 			   struct snd_soc_dai_driver *dai_drv)
@@ -727,9 +673,6 @@ int s3c_i2sv2_register_component(struct device *dev, int id,
 	if (!ops->delay)
 		ops->delay = s3c2412_i2s_delay;
 
-	dai_drv->suspend = s3c2412_i2s_suspend;
-	dai_drv->resume = s3c2412_i2s_resume;
-
 	return devm_snd_soc_register_component(dev, cmp_drv, dai_drv, 1);
 }
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_component);
diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c
index 787a3f6e9f24..b35d828c1cfe 100644
--- a/sound/soc/samsung/s3c2412-i2s.c
+++ b/sound/soc/samsung/s3c2412-i2s.c
@@ -117,6 +117,60 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
 	return 0;
 }
 
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+	u32 iismod;
+
+	if (component->active) {
+		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+		/* some basic suspend checks */
+
+		iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+			pr_warn("%s: RXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+			pr_warn("%s: TXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+			pr_warn("%s: IIS active\n", __func__);
+	}
+
+	return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_component *component)
+{
+	struct s3c_i2sv2_info *i2s = snd_soc_component_get_drvdata(component);
+
+	pr_info("component_active %d, IISMOD %08x, IISCON %08x\n",
+		component->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+	if (component->active) {
+		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		ndelay(250);
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+	}
+
+	return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume  NULL
+#endif
+
 #define S3C2412_I2S_RATES \
 	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
@@ -146,6 +200,8 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = {
 
 static const struct snd_soc_component_driver s3c2412_i2s_component = {
 	.name		= "s3c2412-i2s",
+	.suspend	= s3c2412_i2s_suspend,
+	.resume		= s3c2412_i2s_resume,
 };
 
 static int s3c2412_iis_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c
index fc5d089868df..d51fb3a39448 100644
--- a/sound/soc/sh/rcar/ssi.c
+++ b/sound/soc/sh/rcar/ssi.c
@@ -407,7 +407,7 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod,
 	 * We shouldn't exchange SWSP after running.
 	 * This means, parent needs to care it.
 	 */
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		goto init_end;
 
 	if (rsnd_io_is_play(io))
@@ -559,7 +559,7 @@ static int rsnd_ssi_start(struct rsnd_mod *mod,
 	 * EN is for data output.
 	 * SSI parent EN is not needed.
 	 */
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	ssi->cr_en = EN;
@@ -582,7 +582,7 @@ static int rsnd_ssi_stop(struct rsnd_mod *mod,
 	if (!rsnd_ssi_is_run_mods(mod, io))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	cr  =	ssi->cr_own	|
@@ -620,7 +620,7 @@ static int rsnd_ssi_irq(struct rsnd_mod *mod,
 	if (rsnd_is_gen1(priv))
 		return 0;
 
-	if (rsnd_ssi_is_parent(mod, io))
+	if (rsnd_ssi_is_parent(mod, io) && !rsnd_ssi_multi_slaves(io))
 		return 0;
 
 	if (!rsnd_ssi_is_run_mods(mod, io))
diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c
index f35d88211887..9c7c3e7539c9 100644
--- a/sound/soc/sh/rcar/ssiu.c
+++ b/sound/soc/sh/rcar/ssiu.c
@@ -221,7 +221,7 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod,
 			i;
 
 		for_each_rsnd_mod_array(i, pos, io, rsnd_ssi_array) {
-			shift	= (i * 4) + 16;
+			shift	= (i * 4) + 20;
 			val	= (val & ~(0xF << shift)) |
 				rsnd_mod_id(pos) << shift;
 		}
diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c
index 8f3cad8db89a..31c41559034b 100644
--- a/sound/soc/soc-dai.c
+++ b/sound/soc/soc-dai.c
@@ -295,24 +295,17 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai,
 {
 	int ret = 0;
 
-	if (!dai->started[substream->stream] &&
-	    dai->driver->ops->startup)
+	if (dai->driver->ops->startup)
 		ret = dai->driver->ops->startup(substream, dai);
 
-	if (ret == 0)
-		dai->started[substream->stream] = 1;
-
 	return ret;
 }
 
 void snd_soc_dai_shutdown(struct snd_soc_dai *dai,
 			 struct snd_pcm_substream *substream)
 {
-	if (dai->started[substream->stream] &&
-	    dai->driver->ops->shutdown)
+	if (dai->driver->ops->shutdown)
 		dai->driver->ops->shutdown(substream, dai);
-
-	dai->started[substream->stream] = 0;
 }
 
 int snd_soc_dai_prepare(struct snd_soc_dai *dai,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index 679ed60d850e..a4143ca190d0 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -4165,6 +4165,8 @@ snd_soc_dapm_new_dai(struct snd_soc_card *card,
 	w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template);
 	if (IS_ERR(w)) {
 		ret = PTR_ERR(w);
+		dev_err(rtd->dev, "ASoC: Failed to create %s widget: %d\n",
+			link_name, ret);
 		goto outfree_kcontrol_news;
 	}
 
@@ -4283,52 +4285,58 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card)
 	return 0;
 }
 
-static void dapm_add_valid_dai_widget(struct snd_soc_card *card,
-				      struct snd_soc_pcm_runtime *rtd,
-				      struct snd_soc_dai *codec_dai,
-				      struct snd_soc_dai *cpu_dai)
+static void dapm_connect_dai_routes(struct snd_soc_dapm_context *dapm,
+				    struct snd_soc_dai *src_dai,
+				    struct snd_soc_dapm_widget *src,
+				    struct snd_soc_dapm_widget *dai,
+				    struct snd_soc_dai *sink_dai,
+				    struct snd_soc_dapm_widget *sink)
 {
-	struct snd_soc_dapm_widget *playback = NULL, *capture = NULL;
-	struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu;
+	dev_dbg(dapm->dev, "connected DAI link %s:%s -> %s:%s\n",
+		src_dai->component->name, src->name,
+		sink_dai->component->name, sink->name);
+
+	if (dai) {
+		snd_soc_dapm_add_path(dapm, src, dai, NULL, NULL);
+		src = dai;
+	}
+
+	snd_soc_dapm_add_path(dapm, src, sink, NULL, NULL);
+}
+
+static void dapm_connect_dai_pair(struct snd_soc_card *card,
+				  struct snd_soc_pcm_runtime *rtd,
+				  struct snd_soc_dai *codec_dai,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct snd_soc_dai_link *dai_link = rtd->dai_link;
+	struct snd_soc_dapm_widget *dai, *codec, *playback_cpu, *capture_cpu;
 	struct snd_pcm_substream *substream;
 	struct snd_pcm_str *streams = rtd->pcm->streams;
 
-	if (rtd->dai_link->params) {
+	if (dai_link->params) {
 		playback_cpu = cpu_dai->capture_widget;
 		capture_cpu = cpu_dai->playback_widget;
 	} else {
-		playback = cpu_dai->playback_widget;
-		capture = cpu_dai->capture_widget;
-		playback_cpu = playback;
-		capture_cpu = capture;
+		playback_cpu = cpu_dai->playback_widget;
+		capture_cpu = cpu_dai->capture_widget;
 	}
 
 	/* connect BE DAI playback if widgets are valid */
 	codec = codec_dai->playback_widget;
 
 	if (playback_cpu && codec) {
-		if (!playback) {
+		if (dai_link->params && !dai_link->playback_widget) {
 			substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-			playback = snd_soc_dapm_new_dai(card, substream,
-							"playback");
-			if (IS_ERR(playback)) {
-				dev_err(rtd->dev,
-					"ASoC: Failed to create DAI %s: %ld\n",
-					codec_dai->name,
-					PTR_ERR(playback));
+			dai = snd_soc_dapm_new_dai(card, substream, "playback");
+			if (IS_ERR(dai))
 				goto capture;
-			}
-
-			snd_soc_dapm_add_path(&card->dapm, playback_cpu,
-					      playback, NULL, NULL);
+			dai_link->playback_widget = dai;
 		}
 
-		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-			cpu_dai->component->name, playback_cpu->name,
-			codec_dai->component->name, codec->name);
-
-		snd_soc_dapm_add_path(&card->dapm, playback, codec,
-				      NULL, NULL);
+		dapm_connect_dai_routes(&card->dapm, cpu_dai, playback_cpu,
+					dai_link->playback_widget,
+					codec_dai, codec);
 	}
 
 capture:
@@ -4336,52 +4344,20 @@ capture:
 	codec = codec_dai->capture_widget;
 
 	if (codec && capture_cpu) {
-		if (!capture) {
+		if (dai_link->params && !dai_link->capture_widget) {
 			substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream;
-			capture = snd_soc_dapm_new_dai(card, substream,
-						       "capture");
-			if (IS_ERR(capture)) {
-				dev_err(rtd->dev,
-					"ASoC: Failed to create DAI %s: %ld\n",
-					codec_dai->name,
-					PTR_ERR(capture));
+			dai = snd_soc_dapm_new_dai(card, substream, "capture");
+			if (IS_ERR(dai))
 				return;
-			}
-
-			snd_soc_dapm_add_path(&card->dapm, capture,
-					      capture_cpu, NULL, NULL);
+			dai_link->capture_widget = dai;
 		}
 
-		dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n",
-			codec_dai->component->name, codec->name,
-			cpu_dai->component->name, capture_cpu->name);
-
-		snd_soc_dapm_add_path(&card->dapm, codec, capture,
-				      NULL, NULL);
+		dapm_connect_dai_routes(&card->dapm, codec_dai, codec,
+					dai_link->capture_widget,
+					cpu_dai, capture_cpu);
 	}
 }
 
-static void dapm_connect_dai_link_widgets(struct snd_soc_card *card,
-					  struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_dai *codec_dai;
-	int i;
-
-	if (rtd->num_cpus == 1) {
-		for_each_rtd_codec_dais(rtd, i, codec_dai)
-			dapm_add_valid_dai_widget(card, rtd, codec_dai,
-						  rtd->cpu_dais[0]);
-	} else if (rtd->num_codecs == rtd->num_cpus) {
-		for_each_rtd_codec_dais(rtd, i, codec_dai)
-			dapm_add_valid_dai_widget(card, rtd, codec_dai,
-						  rtd->cpu_dais[i]);
-	} else {
-		dev_err(card->dev,
-			"N cpus to M codecs link is not supported yet\n");
-	}
-
-}
-
 static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
 	int event)
 {
@@ -4422,6 +4398,8 @@ static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream,
 void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 {
 	struct snd_soc_pcm_runtime *rtd;
+	struct snd_soc_dai *codec_dai;
+	int i;
 
 	/* for each BE DAI link... */
 	for_each_card_rtds(card, rtd)  {
@@ -4432,7 +4410,18 @@ void snd_soc_dapm_connect_dai_link_widgets(struct snd_soc_card *card)
 		if (rtd->dai_link->dynamic)
 			continue;
 
-		dapm_connect_dai_link_widgets(card, rtd);
+		if (rtd->num_cpus == 1) {
+			for_each_rtd_codec_dais(rtd, i, codec_dai)
+				dapm_connect_dai_pair(card, rtd, codec_dai,
+						      rtd->cpu_dais[0]);
+		} else if (rtd->num_codecs == rtd->num_cpus) {
+			for_each_rtd_codec_dais(rtd, i, codec_dai)
+				dapm_connect_dai_pair(card, rtd, codec_dai,
+						      rtd->cpu_dais[i]);
+		} else {
+			dev_err(card->dev,
+				"N cpus to M codecs link is not supported yet\n");
+		}
 	}
 }
 
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 289aebc15529..1f302de44052 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2911,8 +2911,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	int i;
 
 	if (rtd->dai_link->dynamic || rtd->dai_link->no_pcm) {
-		playback = rtd->dai_link->dpcm_playback;
-		capture = rtd->dai_link->dpcm_capture;
+		cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+		if (rtd->num_cpus > 1) {
+			dev_err(rtd->dev,
+				"DPCM doesn't support Multi CPU yet\n");
+			return -EINVAL;
+		}
+
+		playback = rtd->dai_link->dpcm_playback &&
+			   snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK);
+		capture = rtd->dai_link->dpcm_capture &&
+			  snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE);
 	} else {
 		/* Adapt stream for codec2codec links */
 		int cpu_capture = rtd->dai_link->params ?
diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c
index 87f75edba3dc..6df3b0d12d87 100644
--- a/sound/soc/soc-topology.c
+++ b/sound/soc/soc-topology.c
@@ -894,7 +894,13 @@ static int soc_tplg_dmixer_create(struct soc_tplg *tplg, unsigned int count,
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc, &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc,
@@ -1118,6 +1124,7 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 	struct snd_soc_tplg_hdr *hdr)
 {
 	struct snd_soc_tplg_ctl_hdr *control_hdr;
+	int ret;
 	int i;
 
 	if (tplg->pass != SOC_TPLG_PASS_MIXER) {
@@ -1146,25 +1153,30 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg,
 		case SND_SOC_TPLG_CTL_RANGE:
 		case SND_SOC_TPLG_DAPM_CTL_VOLSW:
 		case SND_SOC_TPLG_DAPM_CTL_PIN:
-			soc_tplg_dmixer_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dmixer_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_ENUM:
 		case SND_SOC_TPLG_CTL_ENUM_VALUE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT:
 		case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE:
-			soc_tplg_denum_create(tplg, 1,
-					      le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_denum_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		case SND_SOC_TPLG_CTL_BYTES:
-			soc_tplg_dbytes_create(tplg, 1,
-					       le32_to_cpu(hdr->payload_size));
+			ret = soc_tplg_dbytes_create(tplg, 1,
+					le32_to_cpu(hdr->payload_size));
 			break;
 		default:
 			soc_bind_err(tplg, control_hdr, i);
 			return -EINVAL;
 		}
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: invalid control\n");
+			return ret;
+		}
+
 	}
 
 	return 0;
@@ -1272,7 +1284,9 @@ static int soc_tplg_dapm_graph_elems_load(struct soc_tplg *tplg,
 		routes[i]->dobj.index = tplg->index;
 		list_add(&routes[i]->dobj.list, &tplg->comp->dobj_list);
 
-		soc_tplg_add_route(tplg, routes[i]);
+		ret = soc_tplg_add_route(tplg, routes[i]);
+		if (ret < 0)
+			break;
 
 		/* add route, but keep going if some fail */
 		snd_soc_dapm_add_routes(dapm, routes[i], 1);
@@ -1355,7 +1369,13 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create(
 		}
 
 		/* create any TLV data */
-		soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		err = soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr);
+		if (err < 0) {
+			dev_err(tplg->dev, "ASoC: failed to create TLV %s\n",
+				mc->hdr.name);
+			kfree(sm);
+			continue;
+		}
 
 		/* pass control to driver for optional further init */
 		err = soc_tplg_init_kcontrol(tplg, &kc[i],
@@ -1766,10 +1786,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg)
 	return 0;
 }
 
-static void set_stream_info(struct snd_soc_pcm_stream *stream,
+static int set_stream_info(struct snd_soc_pcm_stream *stream,
 	struct snd_soc_tplg_stream_caps *caps)
 {
 	stream->stream_name = kstrdup(caps->name, GFP_KERNEL);
+	if (!stream->stream_name)
+		return -ENOMEM;
+
 	stream->channels_min = le32_to_cpu(caps->channels_min);
 	stream->channels_max = le32_to_cpu(caps->channels_max);
 	stream->rates = le32_to_cpu(caps->rates);
@@ -1777,6 +1800,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream,
 	stream->rate_max = le32_to_cpu(caps->rate_max);
 	stream->formats = le64_to_cpu(caps->formats);
 	stream->sig_bits = le32_to_cpu(caps->sig_bits);
+
+	return 0;
 }
 
 static void set_dai_flags(struct snd_soc_dai_driver *dai_drv,
@@ -1812,20 +1837,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	if (dai_drv == NULL)
 		return -ENOMEM;
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!dai_drv->name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 	dai_drv->id = le32_to_cpu(pcm->dai_id);
 
 	if (pcm->playback) {
 		stream = &dai_drv->playback;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->capture) {
 		stream = &dai_drv->capture;
 		caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (pcm->compress)
@@ -1835,11 +1869,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		kfree(dai_drv->playback.stream_name);
-		kfree(dai_drv->capture.stream_name);
-		kfree(dai_drv->name);
-		kfree(dai_drv);
-		return ret;
+		goto err;
 	}
 
 	dai_drv->dobj.index = tplg->index;
@@ -1860,6 +1890,14 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg,
 		return ret;
 	}
 
+	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	kfree(dai_drv->name);
+	kfree(dai_drv);
+
 	return ret;
 }
 
@@ -1916,11 +1954,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg,
 	if (strlen(pcm->pcm_name)) {
 		link->name = kstrdup(pcm->pcm_name, GFP_KERNEL);
 		link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL);
+		if (!link->name || !link->stream_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
 	}
 	link->id = le32_to_cpu(pcm->pcm_id);
 
-	if (strlen(pcm->dai_name))
+	if (strlen(pcm->dai_name)) {
 		link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL);
+		if (!link->cpus->dai_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
 
 	link->codecs->name = "snd-soc-dummy";
 	link->codecs->dai_name = "snd-soc-dummy-dai";
@@ -2088,7 +2135,9 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg,
 			_pcm = pcm;
 		} else {
 			abi_match = false;
-			pcm_new_ver(tplg, pcm, &_pcm);
+			ret = pcm_new_ver(tplg, pcm, &_pcm);
+			if (ret < 0)
+				return ret;
 		}
 
 		/* create the FE DAIs and DAI links */
@@ -2436,13 +2485,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	if (d->playback) {
 		stream = &dai_drv->playback;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->capture) {
 		stream = &dai_drv->capture;
 		caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE];
-		set_stream_info(stream, caps);
+		ret = set_stream_info(stream, caps);
+		if (ret < 0)
+			goto err;
 	}
 
 	if (d->flag_mask)
@@ -2454,10 +2507,15 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg,
 	ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai);
 	if (ret < 0) {
 		dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n");
-		return ret;
+		goto err;
 	}
 
 	return 0;
+
+err:
+	kfree(dai_drv->playback.stream_name);
+	kfree(dai_drv->capture.stream_name);
+	return ret;
 }
 
 /* load physical DAI elements */
@@ -2466,7 +2524,7 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 {
 	struct snd_soc_tplg_dai *dai;
 	int count;
-	int i;
+	int i, ret;
 
 	count = le32_to_cpu(hdr->count);
 
@@ -2481,7 +2539,12 @@ static int soc_tplg_dai_elems_load(struct soc_tplg *tplg,
 			return -EINVAL;
 		}
 
-		soc_tplg_dai_config(tplg, dai);
+		ret = soc_tplg_dai_config(tplg, dai);
+		if (ret < 0) {
+			dev_err(tplg->dev, "ASoC: failed to configure DAI\n");
+			return ret;
+		}
+
 		tplg->pos += (sizeof(*dai) + le32_to_cpu(dai->priv.size));
 	}
 
@@ -2589,7 +2652,7 @@ static int soc_valid_header(struct soc_tplg *tplg,
 	}
 
 	/* big endian firmware objects not supported atm */
-	if (hdr->magic == SOC_TPLG_MAGIC_BIG_ENDIAN) {
+	if (le32_to_cpu(hdr->magic) == SOC_TPLG_MAGIC_BIG_ENDIAN) {
 		dev_err(tplg->dev,
 			"ASoC: pass %d big endian not supported header got %x at offset 0x%lx size 0x%zx.\n",
 			tplg->pass, hdr->magic,
diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c
index 0d0c9afd8791..34a7c3d6fb91 100644
--- a/sound/soc/stm/stm32_sai_sub.c
+++ b/sound/soc/stm/stm32_sai_sub.c
@@ -837,7 +837,7 @@ static int stm32_sai_set_config(struct snd_soc_dai *cpu_dai,
 		cr1 = SAI_XCR1_DS_SET(SAI_DATASIZE_32);
 		break;
 	default:
-		dev_err(cpu_dai->dev, "Data format not supported");
+		dev_err(cpu_dai->dev, "Data format not supported\n");
 		return -EINVAL;
 	}