Merge tag 'sound-4.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound fixes from Takashi Iwai:
 "Now we hit the usual ASoC-fix-flood in the middle of release.

  Most of the changes are trivial and device-specific, while one
  significant change is the fix for unbalanced of_graph_*() refcounts.
  This involved a change in the graph API itself that had been a bit
  messy"

* tag 'sound-4.13-rc4' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
  ALSA: hda - Fix speaker output from VAIO VPCL14M1R
  device property: Fix usecount for of_graph_get_port_parent()
  ASoC: rt5665: fix wrong register for bclk ratio control
  ASoC: Intel: Use MCLK instead of BLCK as the sysclock for RT5514 codec on kabylake platform
  ASoC: Intel: Enabling ASRC for RT5663 codec on kabylake platform
  ASoC: codecs: msm8916-analog: fix DIG_CLK_CTL_RXD3_CLK_EN define
  ASoC: Intel: Skylake: Fix missing sentinels in sst_acpi_mach
  ASoC: sh: hac: add missing "int ret"
  ASoC: samsung: odroid: Fix EPLL frequency values
  ASoC: sgtl5000: Use snd_soc_kcontrol_codec()
  ASoC: rt5665: fix GPIO6 pin function define
  ASoC: ux500: Restore platform DAI assignments
  ASoC: fix pcm-creation regression
  ASoC: do not close shared backend dailink
  ASoC: pxa: SND_PXA2XX_SOC should depend on HAS_DMA
  ASoC: Intel: Skylake: Fix default dma_buffer_size
  ASoC: rt5663: Update the HW default values based on the shipping version
  ASoC: imx-ssi: add check on platform_get_irq return value

drivers/of/property.c

@@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node)
 {
 	unsigned int depth;
 
+	if (!node)
+		return NULL;
+
+	/*
+	 * Preserve usecount for passed in node as of_get_next_parent()
+	 * will do of_node_put() on it.
+	 */
+	of_node_get(node);
+
 	/* Walk 3 levels up only if there is 'ports' node. */
 	for (depth = 3; depth && node; depth--) {
 		node = of_get_next_parent(node);
@@ -728,12 +737,16 @@ EXPORT_SYMBOL(of_graph_get_port_parent);
 struct device_node *of_graph_get_remote_port_parent(
 			       const struct device_node *node)
 {
-	struct device_node *np;
+	struct device_node *np, *pp;
 
 	/* Get remote endpoint node. */
 	np = of_graph_get_remote_endpoint(node);
 
-	return of_graph_get_port_parent(np);
+	pp = of_graph_get_port_parent(np);
+
+	of_node_put(np);
+
+	return pp;
 }
 EXPORT_SYMBOL(of_graph_get_remote_port_parent);
 

include/sound/soc.h

@@ -795,10 +795,6 @@ struct snd_soc_component_driver {
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
 
-	/* pcm creation and destruction */
-	int (*pcm_new)(struct snd_soc_pcm_runtime *);
-	void (*pcm_free)(struct snd_pcm *);
-
 	/* DT */
 	int (*of_xlate_dai_name)(struct snd_soc_component *component,
 				 struct of_phandle_args *args,
@@ -874,8 +870,6 @@ struct snd_soc_component {
 	void (*remove)(struct snd_soc_component *);
 	int (*suspend)(struct snd_soc_component *);
 	int (*resume)(struct snd_soc_component *);
-	int (*pcm_new)(struct snd_soc_pcm_runtime *);
-	void (*pcm_free)(struct snd_pcm *);
 
 	/* machine specific init */
 	int (*init)(struct snd_soc_component *component);

sound/pci/hda/patch_realtek.c

@@ -2296,6 +2296,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3),
 	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT),
 	SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP),
+	SND_PCI_QUIRK(0x104d, 0x9060, "Sony Vaio VPCL14M1R", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP),
 	SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP),
 

sound/soc/codecs/msm8916-wcd-analog.c

@@ -36,7 +36,7 @@
 #define CDC_D_CDC_DIG_CLK_CTL		(0xf04A)
 #define DIG_CLK_CTL_RXD1_CLK_EN		BIT(0)
 #define DIG_CLK_CTL_RXD2_CLK_EN		BIT(1)
-#define DIG_CLK_CTL_RXD3_CLK_EN		BIT(3)
+#define DIG_CLK_CTL_RXD3_CLK_EN		BIT(2)
 #define DIG_CLK_CTL_TXD_CLK_EN		BIT(4)
 #define DIG_CLK_CTL_NCP_CLK_EN_MASK	BIT(6)
 #define DIG_CLK_CTL_NCP_CLK_EN		BIT(6)

sound/soc/codecs/rt5663.c

@@ -466,7 +466,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0006, 0x1000 },
 	{ 0x000a, 0x0000 },
 	{ 0x0010, 0x000f },
-	{ 0x0015, 0x42c1 },
+	{ 0x0015, 0x42f1 },
 	{ 0x0016, 0x0000 },
 	{ 0x0018, 0x000b },
 	{ 0x0019, 0xafaf },
@@ -509,7 +509,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x008a, 0x0000 },
 	{ 0x008b, 0x0000 },
 	{ 0x008c, 0x0003 },
-	{ 0x008e, 0x0004 },
+	{ 0x008e, 0x0008 },
 	{ 0x008f, 0x1000 },
 	{ 0x0090, 0x0646 },
 	{ 0x0091, 0x0e3e },
@@ -520,7 +520,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0098, 0x0000 },
 	{ 0x009a, 0x0000 },
 	{ 0x009f, 0x0000 },
-	{ 0x00ae, 0x2000 },
+	{ 0x00ae, 0x6000 },
 	{ 0x00af, 0x0000 },
 	{ 0x00b6, 0x0000 },
 	{ 0x00b7, 0x0000 },
@@ -538,7 +538,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x00d9, 0x08f9 },
 	{ 0x00db, 0x0008 },
 	{ 0x00dc, 0x00c0 },
-	{ 0x00dd, 0x6724 },
+	{ 0x00dd, 0x6729 },
 	{ 0x00de, 0x3131 },
 	{ 0x00df, 0x0008 },
 	{ 0x00e0, 0x4000 },
@@ -578,7 +578,7 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0116, 0x0000 },
 	{ 0x0117, 0x0f00 },
 	{ 0x0118, 0x0006 },
-	{ 0x0125, 0x2224 },
+	{ 0x0125, 0x2424 },
 	{ 0x0126, 0x5550 },
 	{ 0x0127, 0x0400 },
 	{ 0x0128, 0x7711 },
@@ -596,8 +596,8 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0145, 0x0002 },
 	{ 0x0146, 0x0000 },
 	{ 0x0160, 0x0e80 },
-	{ 0x0161, 0x0020 },
-	{ 0x0162, 0x0080 },
+	{ 0x0161, 0x0080 },
+	{ 0x0162, 0x0200 },
 	{ 0x0163, 0x0800 },
 	{ 0x0164, 0x0000 },
 	{ 0x0165, 0x0000 },
@@ -676,8 +676,8 @@ static const struct reg_default rt5663_reg[] = {
 	{ 0x0251, 0x0000 },
 	{ 0x0252, 0x028a },
 	{ 0x02fa, 0x0000 },
-	{ 0x02fb, 0x0000 },
-	{ 0x02fc, 0x0000 },
+	{ 0x02fb, 0x00a4 },
+	{ 0x02fc, 0x0300 },
 	{ 0x0300, 0x0000 },
 	{ 0x03d0, 0x0000 },
 	{ 0x03d1, 0x0000 },

sound/soc/codecs/rt5665.c

@@ -4368,12 +4368,12 @@ static int rt5665_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio)
 		switch (dai->id) {
 		case RT5665_AIF2_1:
 		case RT5665_AIF2_2:
-			snd_soc_update_bits(codec, RT5665_ADDA_CLK_1,
+			snd_soc_update_bits(codec, RT5665_ADDA_CLK_2,
 				RT5665_I2S_BCLK_MS2_MASK,
 				RT5665_I2S_BCLK_MS2_64);
 			break;
 		case RT5665_AIF3:
-			snd_soc_update_bits(codec, RT5665_ADDA_CLK_1,
+			snd_soc_update_bits(codec, RT5665_ADDA_CLK_2,
 				RT5665_I2S_BCLK_MS3_MASK,
 				RT5665_I2S_BCLK_MS3_64);
 			break;

sound/soc/codecs/rt5665.h

@@ -1692,8 +1692,8 @@
 #define RT5665_GP6_PIN_MASK			(0x3 << 5)
 #define RT5665_GP6_PIN_SFT			5
 #define RT5665_GP6_PIN_GPIO6			(0x0 << 5)
-#define RT5665_GP6_PIN_BCLK3			(0x0 << 5)
-#define RT5665_GP6_PIN_PDM_SCL			(0x1 << 5)
+#define RT5665_GP6_PIN_BCLK3			(0x1 << 5)
+#define RT5665_GP6_PIN_PDM_SCL			(0x2 << 5)
 #define RT5665_GP7_PIN_MASK			(0x3 << 3)
 #define RT5665_GP7_PIN_SFT			3
 #define RT5665_GP7_PIN_GPIO7			(0x0 << 3)

sound/soc/codecs/sgtl5000.c

@@ -409,7 +409,7 @@ static int dac_put_volsw(struct snd_kcontrol *kcontrol,
 static int avc_get_threshold(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	int db, i;
 	u16 reg = snd_soc_read(codec, SGTL5000_DAP_AVC_THRESHOLD);
 
@@ -442,7 +442,7 @@ static int avc_get_threshold(struct snd_kcontrol *kcontrol,
 static int avc_put_threshold(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
 {
-	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
 	int db;
 	u16 reg;
 

sound/soc/fsl/imx-ssi.c

@@ -527,6 +527,10 @@ static int imx_ssi_probe(struct platform_device *pdev)
 	}
 
 	ssi->irq = platform_get_irq(pdev, 0);
+	if (ssi->irq < 0) {
+		dev_err(&pdev->dev, "Failed to get IRQ: %d\n", ssi->irq);
+		return ssi->irq;
+	}
 
 	ssi->clk = devm_clk_get(&pdev->dev, NULL);
 	if (IS_ERR(ssi->clk)) {

sound/soc/generic/audio-graph-card.c

@@ -224,9 +224,11 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 
 	of_for_each_phandle(&it, rc, node, "dais", NULL, 0) {
 		ret = asoc_graph_card_dai_link_of(it.node, priv, idx++);
-		of_node_put(it.node);
-		if (ret < 0)
+		if (ret < 0) {
+			of_node_put(it.node);
+
 			return ret;
+		}
 	}
 
 	return asoc_simple_card_parse_card_name(card, NULL);
@@ -239,10 +241,8 @@ static int asoc_graph_get_dais_count(struct device *dev)
 	int count = 0;
 	int rc;
 
-	of_for_each_phandle(&it, rc, node, "dais", NULL, 0) {
+	of_for_each_phandle(&it, rc, node, "dais", NULL, 0)
 		count++;
-		of_node_put(it.node);
-	}
 
 	return count;
 }

sound/soc/generic/audio-graph-scu-card.c

@@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 		codec_ep = of_graph_get_remote_endpoint(cpu_ep);
 		rcpu_ep  = of_graph_get_remote_endpoint(codec_ep);
 
-		of_node_put(cpu_port);
 		of_node_put(cpu_ep);
 		of_node_put(codec_ep);
 		of_node_put(rcpu_ep);
@@ -232,8 +231,10 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 
 		ret = asoc_simple_card_parse_daifmt(dev, cpu_ep, codec_ep,
 							    NULL, &daifmt);
-		if (ret < 0)
+		if (ret < 0) {
+			of_node_put(cpu_port);
 			goto parse_of_err;
+		}
 	}
 
 	dai_idx = 0;
@@ -250,7 +251,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 			codec_ep = of_graph_get_remote_endpoint(cpu_ep);
 			codec_port = of_graph_get_port_parent(codec_ep);
 
-			of_node_put(cpu_port);
 			of_node_put(cpu_ep);
 			of_node_put(codec_ep);
 			of_node_put(codec_port);
@@ -266,13 +266,17 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv)
 
 				/* Back-End (= Codec) */
 				ret = asoc_graph_card_dai_link_of(codec_ep, priv, daifmt, dai_idx++, 0);
-				if (ret < 0)
+				if (ret < 0) {
+					of_node_put(cpu_port);
 					goto parse_of_err;
+				}
 			} else {
 				/* Front-End (= CPU) */
 				ret = asoc_graph_card_dai_link_of(cpu_ep, priv, daifmt, dai_idx++, 1);
-				if (ret < 0)
+				if (ret < 0) {
+					of_node_put(cpu_port);
 					goto parse_of_err;
+				}
 			}
 		}
 	}
@@ -306,7 +310,6 @@ static int asoc_graph_get_dais_count(struct device *dev)
 		codec_ep = of_graph_get_remote_endpoint(cpu_ep);
 		codec_port = of_graph_get_port_parent(codec_ep);
 
-		of_node_put(cpu_port);
 		of_node_put(cpu_ep);
 		of_node_put(codec_ep);
 		of_node_put(codec_port);

sound/soc/generic/simple-card-utils.c

@@ -263,6 +263,9 @@ static int asoc_simple_card_get_dai_id(struct device_node *ep)
 			id = i;
 		i++;
 	}
+
+	of_node_put(node);
+
 	if (id < 0)
 		return -ENODEV;
 
@@ -282,11 +285,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep,
 	if (!dai_name)
 		return 0;
 
-	/*
-	 * of_graph_get_port_parent() will call
-	 * of_node_put(). So, call of_node_get() here
-	 */
-	of_node_get(ep);
 	node = of_graph_get_port_parent(ep);
 
 	/* Get dai->name */

sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c

@@ -319,7 +319,9 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream,
 	int ret;
 
 	/* use ASRC for internal clocks, as PLL rate isn't multiple of BCLK */
-	rt5663_sel_asrc_clk_src(codec_dai->codec, RT5663_DA_STEREO_FILTER, 1);
+	rt5663_sel_asrc_clk_src(codec_dai->codec,
+			RT5663_DA_STEREO_FILTER | RT5663_AD_STEREO_FILTER,
+			RT5663_CLK_SEL_I2S1_ASRC);
 
 	ret = snd_soc_dai_set_sysclk(codec_dai,
 			RT5663_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN);
@@ -349,19 +351,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 				return ret;
 			}
 
-			ret = snd_soc_dai_set_pll(codec_dai, 0,
-				RT5514_PLL1_S_BCLK, RT5514_AIF1_BCLK_FREQ,
-						RT5514_AIF1_SYSCLK_FREQ);
-			if (ret < 0) {
-				dev_err(rtd->dev, "set bclk err: %d\n", ret);
-				return ret;
-			}
-
 			ret = snd_soc_dai_set_sysclk(codec_dai,
-				RT5514_SCLK_S_PLL1, RT5514_AIF1_SYSCLK_FREQ,
-							SND_SOC_CLOCK_IN);
+				RT5514_SCLK_S_MCLK, 24576000, SND_SOC_CLOCK_IN);
 			if (ret < 0) {
-				dev_err(rtd->dev, "set sclk err: %d\n", ret);
+				dev_err(rtd->dev, "set sysclk err: %d\n", ret);
 				return ret;
 			}
 		}

sound/soc/intel/skylake/skl-messages.c

@@ -540,6 +540,14 @@ static void skl_setup_cpr_gateway_cfg(struct skl_sst *ctx,
 	cpr_mconfig->gtw_cfg.dma_buffer_size =
 				mconfig->dma_buffer_size * dma_io_buf;
 
+	/* fallback to 2ms default value */
+	if (!cpr_mconfig->gtw_cfg.dma_buffer_size) {
+		if (mconfig->hw_conn_type == SKL_CONN_SOURCE)
+			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->obs;
+		else
+			cpr_mconfig->gtw_cfg.dma_buffer_size = 2 * mconfig->ibs;
+	}
+
 	cpr_mconfig->cpr_feature_mask = 0;
 	cpr_mconfig->gtw_cfg.config_length  = 0;
 

sound/soc/intel/skylake/skl.c

@@ -941,6 +941,7 @@ static struct sst_acpi_mach sst_bxtp_devdata[] = {
 		.machine_quirk = sst_acpi_codec_list,
 		.quirk_data = &bxt_codecs,
 	},
+	{}
 };
 
 static struct sst_acpi_mach sst_kbl_devdata[] = {
@@ -991,6 +992,7 @@ static struct sst_acpi_mach sst_glk_devdata[] = {
 		.drv_name = "glk_alc298s_i2s",
 		.fw_filename = "intel/dsp_fw_glk.bin",
 	},
+	{}
 };
 
 /* PCI IDs */

sound/soc/pxa/Kconfig

 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA || COMPILE_TEST
+	depends on HAS_DMA
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to

sound/soc/samsung/odroid.c

@@ -42,17 +42,17 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream,
 	switch (params_rate(params)) {
 	case 32000:
 	case 64000:
-		pll_freq = 131072000U;
+		pll_freq = 131072006U;
 		break;
 	case 44100:
 	case 88200:
 	case 176400:
-		pll_freq = 180633600U;
+		pll_freq = 180633609U;
 		break;
 	case 48000:
 	case 96000:
 	case 192000:
-		pll_freq = 196608000U;
+		pll_freq = 196608001U;
 		break;
 	default:
 		return -EINVAL;

sound/soc/sh/hac.c

@@ -315,6 +315,8 @@ static const struct snd_soc_component_driver sh4_hac_component = {
 
 static int hac_soc_platform_probe(struct platform_device *pdev)
 {
+	int ret;
+
 	ret = snd_soc_set_ac97_ops(&hac_ac97_ops);
 	if (ret != 0)
 		return ret;

sound/soc/soc-core.c

@@ -3171,8 +3171,6 @@ static int snd_soc_component_initialize(struct snd_soc_component *component,
 	component->remove = component->driver->remove;
 	component->suspend = component->driver->suspend;
 	component->resume = component->driver->resume;
-	component->pcm_new = component->driver->pcm_new;
-	component->pcm_free = component->driver->pcm_free;
 
 	dapm = &component->dapm;
 	dapm->dev = dev;
@@ -3360,25 +3358,6 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component)
 	platform->driver->remove(platform);
 }
 
-static int snd_soc_platform_drv_pcm_new(struct snd_soc_pcm_runtime *rtd)
-{
-	struct snd_soc_platform *platform = rtd->platform;
-
-	if (platform->driver->pcm_new)
-		return platform->driver->pcm_new(rtd);
-	else
-		return 0;
-}
-
-static void snd_soc_platform_drv_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	struct snd_soc_platform *platform = rtd->platform;
-
-	if (platform->driver->pcm_free)
-		platform->driver->pcm_free(pcm);
-}
-
 /**
  * snd_soc_add_platform - Add a platform to the ASoC core
  * @dev: The parent device for the platform
@@ -3402,10 +3381,6 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform,
 		platform->component.probe = snd_soc_platform_drv_probe;
 	if (platform_drv->remove)
 		platform->component.remove = snd_soc_platform_drv_remove;
-	if (platform_drv->pcm_new)
-		platform->component.pcm_new = snd_soc_platform_drv_pcm_new;
-	if (platform_drv->pcm_free)
-		platform->component.pcm_free = snd_soc_platform_drv_pcm_free;
 
 #ifdef CONFIG_DEBUG_FS
 	platform->component.debugfs_prefix = "platform";
@@ -4113,6 +4088,8 @@ int snd_soc_get_dai_id(struct device_node *ep)
 	}
 	mutex_unlock(&client_mutex);
 
+	of_node_put(node);
+
 	return ret;
 }
 EXPORT_SYMBOL_GPL(snd_soc_get_dai_id);

sound/soc/soc-pcm.c

@@ -181,6 +181,10 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir,
 		dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n",
 				be->dai_link->name, event, dir);
 
+		if ((event == SND_SOC_DAPM_STREAM_STOP) &&
+		    (be->dpcm[dir].users >= 1))
+			continue;
+
 		snd_soc_dapm_stream_event(be, dir, event);
 	}
 
@@ -2628,25 +2632,12 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream)
 	return ret;
 }
 
-static void soc_pcm_free(struct snd_pcm *pcm)
-{
-	struct snd_soc_pcm_runtime *rtd = pcm->private_data;
-	struct snd_soc_component *component;
-
-	list_for_each_entry(component, &rtd->card->component_dev_list,
-			    card_list) {
-		if (component->pcm_free)
-			component->pcm_free(pcm);
-	}
-}
-
 /* create a new pcm */
 int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 {
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
-	struct snd_soc_component *component;
 	struct snd_pcm *pcm;
 	char new_name[64];
 	int ret = 0, playback = 0, capture = 0;
@@ -2756,18 +2747,17 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 	if (capture)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &rtd->ops);
 
-	list_for_each_entry(component, &rtd->card->component_dev_list, card_list) {
-		if (component->pcm_new) {
-			ret = component->pcm_new(rtd);
-			if (ret < 0) {
-				dev_err(component->dev,
-					"ASoC: pcm constructor failed: %d\n",
-					ret);
-				return ret;
-			}
+	if (platform->driver->pcm_new) {
+		ret = platform->driver->pcm_new(rtd);
+		if (ret < 0) {
+			dev_err(platform->dev,
+				"ASoC: pcm constructor failed: %d\n",
+				ret);
+			return ret;
 		}
 	}
-	pcm->private_free = soc_pcm_free;
+
+	pcm->private_free = platform->driver->pcm_free;
 out:
 	dev_info(rtd->card->dev, "%s <-> %s mapping ok\n",
 		 (rtd->num_codecs > 1) ? "multicodec" : rtd->codec_dai->name,

sound/soc/ux500/mop500.c

@@ -33,6 +33,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = {
 		.stream_name = "ab8500_0",
 		.cpu_dai_name = "ux500-msp-i2s.1",
 		.codec_dai_name = "ab8500-codec-dai.0",
+		.platform_name = "ux500-msp-i2s.1",
 		.codec_name = "ab8500-codec.0",
 		.init = mop500_ab8500_machine_init,
 		.ops = mop500_ab8500_ops,
@@ -42,6 +43,7 @@ static struct snd_soc_dai_link mop500_dai_links[] = {
 		.stream_name = "ab8500_1",
 		.cpu_dai_name = "ux500-msp-i2s.3",
 		.codec_dai_name = "ab8500-codec-dai.1",
+		.platform_name = "ux500-msp-i2s.3",
 		.codec_name = "ab8500-codec.0",
 		.init = NULL,
 		.ops = mop500_ab8500_ops,
@@ -85,6 +87,8 @@ static int mop500_of_probe(struct platform_device *pdev,
 	for (i = 0; i < 2; i++) {
 		mop500_dai_links[i].cpu_of_node = msp_np[i];
 		mop500_dai_links[i].cpu_dai_name = NULL;
+		mop500_dai_links[i].platform_of_node = msp_np[i];
+		mop500_dai_links[i].platform_name = NULL;
 		mop500_dai_links[i].codec_of_node = codec_np;
 		mop500_dai_links[i].codec_name = NULL;
 	}