Merge tag 'asoc-fix-v4.3-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus

ASoC: Fixes for v4.3

A disappointingly large set of fixes, though none of them very big and
very widely spread over many different drivers.  Nothing especially
stands out, it's mostly all device specific and relatively minor.

18 files changed, 119 insertions, 69 deletions

diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index 885683a..e040621 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -9,6 +9,14 @@ menuconfig SND_ARM
 	  Drivers that are implemented on ASoC can be found in
 	  "ALSA for SoC audio support" section.
 
+config SND_PXA2XX_LIB
+	tristate
+	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
+	select SND_DMAENGINE_PCM
+
+config SND_PXA2XX_LIB_AC97
+	bool
+
 if SND_ARM
 
 config SND_ARMAACI
@@ -21,13 +29,6 @@ config SND_PXA2XX_PCM
 	tristate
 	select SND_PCM
 
-config SND_PXA2XX_LIB
-	tristate
-	select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97
-
-config SND_PXA2XX_LIB_AC97
-	bool
-
 config SND_PXA2XX_AC97
 	tristate "AC97 driver for the Intel PXA2xx chip"
 	depends on ARCH_PXA
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 38e853a..0bf9d62 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 {
 	struct resource *iores, *dmares;
 	unsigned long sel;
-	int ret;
 	struct au1xpsc_audio_data *wd;
 
 	wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data),
diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c
index 4972bf3..268a28b 100644
--- a/sound/soc/codecs/rt5645.c
+++ b/sound/soc/codecs/rt5645.c
@@ -732,14 +732,14 @@ static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = {
 static const struct snd_kcontrol_new rt5645_dac_l_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_L_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_L_SFT, 1, 1),
 };
 
 static const struct snd_kcontrol_new rt5645_dac_r_mix[] = {
 	SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_ADCMIX_R_SFT, 1, 1),
-	SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER,
+	SOC_DAPM_SINGLE_AUTODISABLE("DAC1 Switch", RT5645_AD_DA_MIXER,
 			RT5645_M_DAC1_R_SFT, 1, 1),
 };
 
@@ -1381,7 +1381,7 @@ static void hp_amp_power(struct snd_soc_codec *codec, int on)
 				regmap_write(rt5645->regmap, RT5645_PR_BASE +
 					RT5645_MAMP_INT_REG2, 0xfc00);
 				snd_soc_write(codec, RT5645_DEPOP_M2, 0x1140);
-				mdelay(5);
+				msleep(40);
 				rt5645->hp_on = true;
 			} else {
 				/* depop parameters */
@@ -2829,13 +2829,12 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert)
 			snd_soc_dapm_sync(dapm);
 			rt5645->jack_type = SND_JACK_HEADPHONE;
 		}
-
-		snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d);
-		snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001);
 	} else { /* jack out */
 		rt5645->jack_type = 0;
 
+		regmap_update_bits(rt5645->regmap, RT5645_HP_VOL,
+			RT5645_L_MUTE | RT5645_R_MUTE,
+			RT5645_L_MUTE | RT5645_R_MUTE);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL2,
 			RT5645_CBJ_MN_JD, RT5645_CBJ_MN_JD);
 		regmap_update_bits(rt5645->regmap, RT5645_IN1_CTRL1,
@@ -2880,8 +2879,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec,
 		rt5645->en_button_func = true;
 		regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1,
 				RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ);
-		regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1,
-				RT5645_HP_CB_MASK, RT5645_HP_CB_PU);
 		regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1,
 				RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL);
 	}
@@ -3205,6 +3202,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = {
 			DMI_MATCH(DMI_PRODUCT_NAME, "Celes"),
 		},
 	},
+	{
+		.ident = "Google Ultima",
+		.callback = strago_quirk_cb,
+		.matches = {
+			DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"),
+		},
+	},
 	{ }
 };
 
diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c
index f2c6ad4..581ec15 100644
--- a/sound/soc/codecs/wm0010.c
+++ b/sound/soc/codecs/wm0010.c
@@ -577,7 +577,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec);
 	unsigned long flags;
 	int ret;
-	const struct firmware *fw;
 	struct spi_message m;
 	struct spi_transfer t;
 	struct dfw_pllrec pll_rec;
@@ -623,14 +622,6 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 	wm0010->state = WM0010_OUT_OF_RESET;
 	spin_unlock_irqrestore(&wm0010->irq_lock, flags);
 
-	/* First the bootloader */
-	ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev);
-	if (ret != 0) {
-		dev_err(codec->dev, "Failed to request stage2 loader: %d\n",
-			ret);
-		goto abort;
-	}
-
 	if (!wait_for_completion_timeout(&wm0010->boot_completion,
 					 msecs_to_jiffies(20)))
 		dev_err(codec->dev, "Failed to get interrupt from DSP\n");
@@ -673,7 +664,7 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 		img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA);
 		if (!img_swap)
-			goto abort;
+			goto abort_out;
 
 		/* We need to re-order for 0010 */
 		byte_swap_64((u64 *)&pll_rec, img_swap, len);
@@ -688,16 +679,16 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 		spi_message_add_tail(&t, &m);
 
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "First PLL write failed: %d\n", ret);
-			goto abort;
+			goto abort_swap;
 		}
 
 		/* Use a second send of the message to get the return status */
 		ret = spi_sync(spi, &m);
-		if (ret != 0) {
+		if (ret) {
 			dev_err(codec->dev, "Second PLL write failed: %d\n", ret);
-			goto abort;
+			goto abort_swap;
 		}
 
 		p = (u32 *)out;
@@ -730,6 +721,10 @@ static int wm0010_boot(struct snd_soc_codec *codec)
 
 	return 0;
 
+abort_swap:
+	kfree(img_swap);
+abort_out:
+	kfree(out);
 abort:
 	/* Put the chip back into reset */
 	wm0010_halt(codec);
diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c
index e3b7d0c..dbd8840 100644
--- a/sound/soc/codecs/wm8960.c
+++ b/sound/soc/codecs/wm8960.c
@@ -211,28 +211,38 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol,
 	return wm8960_set_deemph(codec);
 }
 
-static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0);
-static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1);
+static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1);
+static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1725, 75, 0);
+static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0);
 static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
-static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1);
+static const DECLARE_TLV_DB_SCALE(lineinboost_tlv, -1500, 300, 1);
+static const unsigned int micboost_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 1, TLV_DB_SCALE_ITEM(0, 1300, 0),
+	2, 3, TLV_DB_SCALE_ITEM(2000, 900, 0),
+};
 
 static const struct snd_kcontrol_new wm8960_snd_controls[] = {
 SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL,
-		 0, 63, 0, adc_tlv),
+		 0, 63, 0, inpga_tlv),
 SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL,
 	6, 1, 0),
 SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL,
 	7, 1, 0),
 
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume",
-	       WM8960_INBMIX1, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume",
-	       WM8960_INBMIX1, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX1, 1, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume",
-	       WM8960_INBMIX2, 4, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 4, 7, 0, lineinboost_tlv),
 SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume",
-	       WM8960_INBMIX2, 1, 7, 0, boost_tlv),
+	       WM8960_INBMIX2, 1, 7, 0, lineinboost_tlv),
+SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT1 Volume",
+		WM8960_RINPATH, 4, 3, 0, micboost_tlv),
+SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT1 Volume",
+		WM8960_LINPATH, 4, 3, 0, micboost_tlv),
 
 SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC,
 		 0, 255, 0, dac_tlv),
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index b4eb975..293e47a 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -2944,7 +2944,8 @@ static int wm8962_mute(struct snd_soc_dai *dai, int mute)
 				   WM8962_DAC_MUTE, val);
 }
 
-#define WM8962_RATES SNDRV_PCM_RATE_8000_96000
+#define WM8962_RATES (SNDRV_PCM_RATE_8000_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define WM8962_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index add6bb9..7d45d98 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	u8 rx_ser = 0;
 	u8 slots = mcasp->tdm_slots;
 	u8 max_active_serializers = (channels + slots - 1) / slots;
-	int active_serializers, numevt, n;
+	int active_serializers, numevt;
 	u32 reg;
 	/* Default configuration */
 	if (mcasp->version < MCASP_VERSION_3)
@@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
 	 * The number of words for numevt need to be in steps of active
 	 * serializers.
 	 */
-	n = numevt % active_serializers;
-	if (n)
-		numevt += (active_serializers - n);
+	numevt = (numevt / active_serializers) * active_serializers;
+
 	while (period_words % numevt && numevt > 0)
 		numevt -= active_serializers;
 	if (numevt <= 0)
@@ -1299,6 +1298,7 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
 		.ops 		= &davinci_mcasp_dai_ops,
 
 		.symmetric_samplebits	= 1,
+		.symmetric_rates	= 1,
 	},
 	{
 		.name		= "davinci-mcasp.1",
@@ -1685,7 +1685,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "common");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_common_irq_handler,
@@ -1702,7 +1702,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "rx");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_rx_irq_handler,
@@ -1717,7 +1717,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev)
 
 	irq = platform_get_irq_byname(pdev, "tx");
 	if (irq >= 0) {
-		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n",
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
 					  dev_name(&pdev->dev));
 		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
 						davinci_mcasp_tx_irq_handler,
diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c
index 5aeb6ed..96f55ae 100644
--- a/sound/soc/fsl/fsl-asoc-card.c
+++ b/sound/soc/fsl/fsl-asoc-card.c
@@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev)
 		priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM;
 	} else {
 		dev_err(&pdev->dev, "unknown Device Tree compatible\n");
-		return -EINVAL;
+		ret = -EINVAL;
+		goto asrc_fail;
 	}
 
 	/* Common settings for corresponding Freescale CPU DAI driver */
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 8ec6fb2..37c5cd4 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids);
 
 static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private)
 {
-	return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97);
+	return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+		SND_SOC_DAIFMT_AC97;
 }
 
 static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private)
@@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev,
 				CCSR_SSI_SCR_TCH_EN);
 	}
 
-	if (fmt & SND_SOC_DAIFMT_AC97)
+	if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97)
 		fsl_ssi_setup_ac97(ssi_private);
 
 	return 0;
diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c
index f6efa9d..b27f25f 100644
--- a/sound/soc/intel/haswell/sst-haswell-ipc.c
+++ b/sound/soc/intel/haswell/sst-haswell-ipc.c
@@ -302,6 +302,10 @@ struct sst_hsw {
 	struct sst_hsw_ipc_dx_reply dx;
 	void *dx_context;
 	dma_addr_t dx_context_paddr;
+	enum sst_hsw_device_id dx_dev;
+	enum sst_hsw_device_mclk dx_mclk;
+	enum sst_hsw_device_mode dx_mode;
+	u32 dx_clock_divider;
 
 	/* boot */
 	wait_queue_head_t boot_wait;
@@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw,
 
 	trace_ipc_request("set device config", dev);
 
-	config.ssp_interface = dev;
-	config.clock_frequency = mclk;
-	config.mode = mode;
-	config.clock_divider = clock_divider;
+	hsw->dx_dev = config.ssp_interface = dev;
+	hsw->dx_mclk = config.clock_frequency = mclk;
+	hsw->dx_mode = config.mode = mode;
+	hsw->dx_clock_divider = config.clock_divider = clock_divider;
 	if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER)
 		config.channels = 4;
 	else
@@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw)
 		return -EIO;
 	}
 
-	/* Set ADSP SSP port settings */
-	ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0,
-					SST_HSW_DEVICE_MCLK_FREQ_24_MHZ,
-					SST_HSW_DEVICE_CLOCK_MASTER, 9);
+	/* Set ADSP SSP port settings - sadly the FW does not store SSP port
+	   settings as part of the PM context. */
+	ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk,
+					hsw->dx_mode, hsw->dx_clock_divider);
 	if (ret < 0)
 		dev_err(dev, "error: SSP re-initialization failed\n");
 
diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c
index d190fe0..f5baf3c 100644
--- a/sound/soc/mediatek/mtk-afe-pcm.c
+++ b/sound/soc/mediatek/mtk-afe-pcm.c
@@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream,
 	memif->substream = substream;
 
 	snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware);
+
+	/*
+	 * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be
+	 * smaller than period_size due to AFE's internal buffer.
+	 * This easily leads to overrun when avail_min is period_size.
+	 * One more period can hold the possible unread buffer.
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		ret = snd_pcm_hw_constraint_minmax(runtime,
+						   SNDRV_PCM_HW_PARAM_PERIODS,
+						   3,
+						   mtk_afe_hardware.periods_max);
+		if (ret < 0) {
+			dev_err(afe->dev, "hw_constraint_minmax failed\n");
+			return ret;
+		}
+	}
 	ret = snd_pcm_hw_constraint_integer(runtime,
 					    SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index 39cea80..f2bf866 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -1,7 +1,6 @@
 config SND_PXA2XX_SOC
 	tristate "SoC Audio for the Intel PXA2xx chip"
 	depends on ARCH_PXA
-	select SND_ARM
 	select SND_PXA2XX_LIB
 	help
 	  Say Y or M if you want to add support for codecs attached to
@@ -25,7 +24,6 @@ config SND_PXA2XX_AC97
 config SND_PXA2XX_SOC_AC97
 	tristate
 	select AC97_BUS
-	select SND_ARM
 	select SND_PXA2XX_LIB_AC97
 	select SND_SOC_AC97_BUS
 
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 1f60546..9e4b04e 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
 	.reset	= pxa2xx_ac97_cold_reset,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12;
+static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = {
 	.filter_data	= &pxa2xx_ac97_pcm_stereo_in_req,
 };
 
-static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11;
+static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12;
 static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = {
 	.addr		= __PREG(PCDR),
 	.addr_width	= DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index f4bf21a..ff8bda4 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w,
 
 	default:
 		WARN(1, "Unknown event %d\n", event);
-		return -EINVAL;
+		ret = -EINVAL;
 	}
 
 out:
diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c
index 362c69a..53dd085 100644
--- a/sound/soc/soc-utils.c
+++ b/sound/soc/soc-utils.c
@@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec;
 			SNDRV_PCM_FMTBIT_S32_LE | \
 			SNDRV_PCM_FMTBIT_U32_LE | \
 			SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
+/*
+ * The dummy CODEC is only meant to be used in situations where there is no
+ * actual hardware.
+ *
+ * If there is actual hardware even if it does not have a control bus
+ * the hardware will still have constraints like supported samplerates, etc.
+ * which should be modelled. And the data flow graph also should be modelled
+ * using DAPM.
+ */
 static struct snd_soc_dai_driver dummy_dai = {
 	.name = "snd-soc-dummy-dai",
 	.playback = {
diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig
index 0a53053..4fb9141 100644
--- a/sound/soc/spear/Kconfig
+++ b/sound/soc/spear/Kconfig
@@ -1,6 +1,6 @@
 config SND_SPEAR_SOC
 	tristate
-	select SND_DMAENGINE_PCM
+	select SND_SOC_GENERIC_DMAENGINE_PCM
 
 config SND_SPEAR_SPDIF_OUT
 	tristate
diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c
index f6eefe1..843f037 100644
--- a/sound/soc/sti/uniperif_player.c
+++ b/sound/soc/sti/uniperif_player.c
@@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	of_property_read_u32(pnode, "version", &player->ver);
-	if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+	if (of_property_read_u32(pnode, "version", &player->ver) ||
+	    player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
 		dev_err(dev, "Unknown uniperipheral version ");
 		return -EINVAL;
 	}
@@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev,
 	if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0)
 		info->underflow_enabled = 1;
 
-	of_property_read_u32(pnode, "uniperiph-id", &info->id);
+	if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) {
+		dev_err(dev, "uniperipheral id not defined");
+		return -EINVAL;
+	}
 
 	/* Read the device mode property */
-	of_property_read_string(pnode, "mode", &mode);
+	if (of_property_read_string(pnode, "mode", &mode)) {
+		dev_err(dev, "uniperipheral mode not defined");
+		return -EINVAL;
+	}
 
 	if (strcasecmp(mode, "hdmi") == 0)
 		info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI;
diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c
index c502626..f791239 100644
--- a/sound/soc/sti/uniperif_reader.c
+++ b/sound/soc/sti/uniperif_reader.c
@@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev,
 	if (!info)
 		return -ENOMEM;
 
-	of_property_read_u32(node, "version", &reader->ver);
+	if (of_property_read_u32(node, "version", &reader->ver) ||
+	    reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) {
+		dev_err(&pdev->dev, "Unknown uniperipheral version ");
+		return -EINVAL;
+	}
 
 	/* Save the info structure */
 	reader->info = info;